From 2e02aab9a70fc3cfea254ab7390f691096cc679e Mon Sep 17 00:00:00 2001 From: roytam1 Date: Thu, 6 Apr 2023 15:48:27 +0800 Subject: [PATCH] import changes from `dev' branch of rmottola/Arctic-Fox: - Bug 1198458: Webrtc updated to branch 43; pull made 2015-09-29 09:00AM PDT rs=jesup (c45a789c99) - Bug 1159489: WebRTC bitrate limits for video depend on input resolution and framerate r=pkerr (e3691a247c) - Bug 1132318: merge SelectSendFrameRate with SelectSendResolution r=bwc (bcc232994c) - Bug 1182289: Clean up dispatches in WebrtcGmpVideoEncoder/Decoder. r=jesup, a=abillings (b5e2030b07) - Bug 1167306: Fix preprocessor goof that disabled the load manager and some preference handling. r=jesup (0cc0dee688) - Bug 1198458: Rollup of changes previously applied to media/webrtc/trunk/webrtc and fixes to those rs=jesup r=froyd,jib,bwc,jesup,gcp,sotaro,pkerr,pehrsons (fe384d5e63) - Bug 1198458 Unbreak build on BSDs by fixing non-POSIX thread includes/usage. r=jesup (4ad8e88c17) - Bug 1198458: Fix typo in merges in OMX rs=bustage,kwierso on a CLOSED TREE (211b3b1a3f) - Bug 1226146: fix sndio audio_device backend after webrtc 43 landing in bug 1198458 r=jesup NPOTB (2b10ba3e86) - Bug 1231106 - Make BSDs fall-through to ASSERT as well. r=jesup (775d4fedae) - Bug 1231109 - Drop FreeBSD checks for unsupported versions. r=jld r=jesup (3cd55166fd) - Bug 1161079: Fix VideoCodecStats to allow for collecting encoder and decoder stats r=jib (1f98af8939) - add limits to fix gcc12 compile (d78098980f) - Bug 1193495 - Part 1: Test case. r=mt (add2ded009) - Bug 1193495 - Part 2: Maintain clones of supported codecs for each level, and do necessary checking to prevent payload-type clashes. r=mt (d45a24dba7) - Bug 1191301 - Re-enable the use of media.navigator.video.use_tmmbr pref. r=bwc (d2723821d4) - Bug 1094447 - Use UDP/TLS/RTP/SAVPF for audio/video m-lines. r=drno (e8e1dafd0a) - Bug 1173599 - a=imageattr support. r=mt (612fc343d7) - Bug 1173601 - Add a=simulcast support. r=mt (7c5303bacb) - Bug 1203246 - Factor track negotiation stuff out of JsepSessionImpl, and other simplification. r=mt (b71c809b78) - Bug 1212907 - a=rid support. r=mt (7479ab6984) - Bug 1212908 - Update a=simulcast to match new grammar in 03 draft. r=mt (e205d3b0dd) - Bug 1192390 - Part 1: Lay architectural groundwork for simulcast negotiation. r=mt r=jesup (e75dda3139) - Bug 1223160 - added SDP parser file reader. r=bwc (6752195791) - Bug 1192390 - Part 2: Simulcast and RID negotiation. r=mt (51b151ef52) - fix some warnings stuff (fed8f513a8) - Bug 1161317: Fix bug where sendonly video RTCP would be treated as outgoing RTP r=jesup (e24371fbe0) - Bug 1226347: Import cherry-pick of AEC changes from 43->48 (delay-agnostic AEC). r=pkerr (d3a074f4d8) - Bug 1226347 - Part 2: Allow control of AEC via prefs. r=rjesup (58f142005b) - Bug 1228788 - Force QT device release to happen on the main thread. r=jesup (885e9d1236) - Bug 1162218 - Make worker idle thread timeouts more strict, r=baku. (a7d2106987) - No bug. Remove a stray debugging printf of mine. r=me. (77b84cda62) - minor (22564a666e) - Bug 1224237 - Remove the !baseURL check from ServiceWorkerContainer::Register;r=bkelly (8b13c4dc49) - Bug 1196157 - Marks left by performance marks should print the domain of the application and not the complete URL. r=baku (d133708d8d) - Bug 1211970 - "Muted errors in workers are not correctly reported to the console". r=bz (42c15275f4) - Bug 1208559 - Tests. r=bholley (e16a30caa5) - Bug 1045891 - Tests for child-src r=ckerschb (90aa832cd0) - Bug 1223647: CSP erroneously inherited into dedicated workers. r=ckerschb (6fd8d9bfc9) - Bug 1218433 - Use AsyncOpen2 in dom/workers/ScriptLoader.cpp - part 1, r=sicking (a7e9187e52) - Bug 1218433 - Use AsyncOpen2 in dom/workers/ScriptLoader.cpp - part 3 - WPT, r=sicking, r=Ms2ger (e5e3c69f6a) - Bug 1218433 - Use AsyncOpen2 in dom/workers/ScriptLoader.cpp - part 2 - WPT, r=sicking, r=Ms2ger (1dd2d871ec) - Bug 1211967 - Fix how we report errors when loading a worker from a data url, r=bz (8517368daa) - nsRefPtr - RefPtr (34bb404530) - Bug 1231055 - Fix tags usage in PluginProvider. r=dtownsend (870b0e71eb) - bug 1228792 - remove use of array comprehensions r=mossop (3e31f18e83) - bug 1228792 - use standard version of catch r=mossop (f306557ca2) - Bug 1228009. Geolocation code needs to handle failures on its ErrorResults. r=smaug (8778a9e264) - Bug 1228707. Add a away to call Web IDL callbacks while ignoring any errors from them, and use it in a few places. r=smaug (a414e0d711) - Bug 1201692. Add a fast path to ExplicitChildIterator::Seek for the common case of seeking an actual DOM child of the parent node. r=wchen (01234ad43a) - Bug 1202186 - use nsISensitiveInfoHidden for console methods, r=baku (888b4506ad) - Bug 1223774 - Console API should check if the outer window exists, r=smaug (e0d7f408dc) - Bug 1200551 - Handle multiple %c formatters without a string between them by using only the last one for styling;r=baku,r=past (463550117a) - Bug 1213719 - Back out bug 1170314 for duplicate functionality. r=smaug (ae74e0ad52) - Bug 1154076 followup: Mark ConsoleRunnable::Run() as override. rs=ehsan (337181faab) - Bug 1127703 - "Support iteration on FormData" r=bz (4ddd461e99) - Bug 1230509 - BlobImplFile should return false in IsDateUnknown and IsSizeUnknown, r=bz (1079bfe2ab) - Bug 1198095 - FileReader should dispatch an error if the blob changes size in the meantime the read is executed, r=bz (263993a172) - Bug 1231094 - patch 1 - nsDOMFileReader to mozilla::dom::FileReader, r=sicking (aa8c3ff373) - Bug 1231094 - patch 2 - Get rid of FileIOObject, r=sicking (44af1e17dd) - Bug 1231100 - Get rid of nsIDOMFileReader - patch 1, r=sicking (c909d9d793) - Bug 1231100 - Get rid of nsIDOMFileReader - patch 2, r=sicking (4d3da1c566) - Bug 1161183: Don't show the add-on version in the list view. r=dao (002d8f6fb1) - Bug 1229519: Fix toolkit/modules to pass eslint checks. r=mak (8bbd9c8fe0) --- b2g/components/LogCapture.jsm | 5 +- build/clang-plugin/clang-plugin.cpp | 1 + build/gyp.mozbuild | 12 +- dom/apps/ImportExport.jsm | 5 +- dom/base/BlobSet.h | 3 +- dom/base/ChildIterator.cpp | 24 + dom/base/ChildIterator.h | 12 +- dom/base/Console.cpp | 58 +- dom/base/Console.h | 4 - dom/base/File.cpp | 33 +- dom/base/File.h | 7 +- dom/base/FileIOObject.cpp | 246 - dom/base/FileIOObject.h | 103 - .../{nsDOMFileReader.cpp => FileReader.cpp} | 509 +- dom/base/{nsDOMFileReader.h => FileReader.h} | 145 +- dom/base/MultipartBlobImpl.cpp | 67 +- 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layout/media/symbols.def.in | 1 + media/mtransport/nr_socket_prsock.cpp | 8 +- .../third_party/nICEr/src/util/mbslen.c | 9 +- media/webrtc/moz.build | 14 +- media/webrtc/signaling/signaling.gyp | 5 +- .../src/common/EncodingConstraints.h | 46 + .../signaling/src/common/NullTransport.h | 4 +- .../signaling/src/jsep/JsepCodecDescription.h | 308 +- media/webrtc/signaling/src/jsep/JsepSession.h | 2 +- .../signaling/src/jsep/JsepSessionImpl.cpp | 482 +- .../signaling/src/jsep/JsepSessionImpl.h | 34 +- media/webrtc/signaling/src/jsep/JsepTrack.cpp | 469 + media/webrtc/signaling/src/jsep/JsepTrack.h | 133 +- .../signaling/src/jsep/JsepTrackEncoding.h | 50 + .../webrtc/signaling/src/jsep/JsepTrackImpl.h | 92 - .../src/media-conduit/AudioConduit.cpp | 64 +- .../src/media-conduit/AudioConduit.h | 12 +- .../signaling/src/media-conduit/CodecConfig.h | 26 +- .../src/media-conduit/CodecStatistics.cpp | 27 +- .../src/media-conduit/CodecStatistics.h | 6 +- .../src/media-conduit/GmpVideoCodec.cpp | 4 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| 5 +- media/webrtc/trunk/webrtc/base/httpclient.cc | 8 +- media/webrtc/trunk/webrtc/base/httpclient.h | 10 +- media/webrtc/trunk/webrtc/base/httpcommon.cc | 22 +- media/webrtc/trunk/webrtc/base/httpcommon.h | 12 +- media/webrtc/trunk/webrtc/base/httprequest.cc | 2 + media/webrtc/trunk/webrtc/base/httprequest.h | 1 + media/webrtc/trunk/webrtc/base/httpserver.h | 12 +- media/webrtc/trunk/webrtc/base/ipaddress.cc | 21 +- media/webrtc/trunk/webrtc/base/ipaddress.h | 2 + .../trunk/webrtc/base/ipaddress_unittest.cc | 35 + media/webrtc/trunk/webrtc/base/json.cc | 4 + media/webrtc/trunk/webrtc/base/json.h | 4 +- .../webrtc/trunk/webrtc/base/json_unittest.cc | 10 +- media/webrtc/trunk/webrtc/base/logging.cc | 13 +- .../webrtc/trunk/webrtc/base/macasyncsocket.h | 39 +- .../trunk/webrtc/base/maccocoasocketserver.h | 6 +- .../trunk/webrtc/base/macsocketserver.cc | 8 + .../trunk/webrtc/base/macsocketserver.h | 22 +- .../trunk/webrtc/base/macwindowpicker.h | 17 +- 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.../modules/audio_coding/main/acm2/acm_red.h | 42 - .../audio_coding/main/acm2/acm_send_test.cc | 6 +- .../audio_coding/main/acm2/acm_send_test.h | 19 +- .../main/acm2/acm_send_test_oldapi.cc | 13 +- .../main/acm2/acm_send_test_oldapi.h | 20 +- .../audio_coding/main/acm2/acm_speex.cc | 324 - .../audio_coding/main/acm2/acm_speex.h | 67 - .../main/acm2/audio_coding_module.gypi | 63 +- .../main/acm2/audio_coding_module_impl.cc | 1550 +- .../main/acm2/audio_coding_module_impl.h | 324 +- .../main/acm2/audio_coding_module_unittest.cc | 186 +- .../audio_coding_module_unittest_oldapi.cc | 407 +- .../audio_coding/main/acm2/codec_manager.cc | 612 + .../audio_coding/main/acm2/codec_manager.h | 101 + .../main/acm2/initial_delay_manager.h | 2 +- .../acm2/initial_delay_manager_unittest.cc | 3 +- .../modules/audio_coding/main/acm2/nack.cc | 4 +- .../modules/audio_coding/main/acm2/nack.h | 10 +- .../audio_coding/main/acm2/nack_unittest.cc | 26 +- .../main/interface/audio_coding_module.h | 164 +- 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.../source/time_scheduler.cc | 13 +- .../source/time_scheduler.h | 8 +- .../webrtc/modules/audio_device/BUILD.gn | 5 + .../modules/audio_device/android/OWNERS | 1 - .../audio_device/android/audio_common.h | 3 + .../android/audio_device_template.h | 482 +- .../android/audio_device_unittest.cc | 884 + .../android/audio_device_utility_android.cc | 36 +- .../android/audio_device_utility_android.h | 20 +- .../audio_device/android/audio_manager.cc | 204 + .../audio_device/android/audio_manager.h | 141 + .../audio_device/android/audio_manager_jni.cc | 30 +- .../audio_device/android/audio_manager_jni.h | 2 - .../audio_device/android/audio_record_jni.cc | 1389 +- .../audio_device/android/audio_record_jni.h | 233 +- .../audio_device/android/audio_track_jni.cc | 1607 +- .../audio_device/android/audio_track_jni.h | 229 +- .../android/ensure_initialized.cc | 51 + .../android/ensure_initialized.h} | 12 +- .../audio_device/android/fine_audio_buffer.h | 4 +- .../android/fine_audio_buffer_unittest.cc | 4 +- .../voiceengine/WebRtcAudioManager.java | 152 + .../webrtc/voiceengine/WebRtcAudioRecord.java | 380 +- .../webrtc/voiceengine/WebRtcAudioTrack.java | 482 +- .../webrtc/voiceengine/WebRtcAudioUtils.java | 53 + .../android/low_latency_event_unittest.cc | 14 +- .../audio_device/android/opensles_input.cc | 53 +- .../audio_device/android/opensles_input.h | 26 +- .../audio_device/android/opensles_output.cc | 27 +- .../audio_device/android/opensles_output.h | 23 +- .../audio_device/android/single_rw_fifo.cc | 6 + .../audio_device/android/single_rw_fifo.h | 4 +- .../android/single_rw_fifo_unittest.cc | 4 +- .../modules/audio_device/audio_device.gypi | 24 +- .../audio_device/audio_device_buffer.h | 1 - .../audio_device/audio_device_generic.cc | 8 +- .../audio_device/audio_device_generic.h | 7 +- .../modules/audio_device/audio_device_impl.cc | 60 +- .../modules/audio_device/audio_device_impl.h | 206 +- .../audio_device/audio_device_tests.isolate | 8 - 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66 +- .../mac/portaudio/pa_ringbuffer.h | 48 +- .../audio_device/sndio/audio_device_sndio.cc | 38 +- .../audio_device/sndio/audio_device_sndio.h | 150 +- .../sndio/audio_device_utility_sndio.h | 2 +- .../test/audio_device_test_api.cc | 9 +- .../audio_device/test/func_test_manager.cc | 19 +- .../audio_device/test/func_test_manager.h | 2 +- .../audio_device/win/audio_device_wave_win.cc | 52 +- .../audio_device/win/audio_device_wave_win.h | 5 +- .../webrtc/modules/audio_processing/BUILD.gn | 125 +- .../modules/audio_processing/aec/aec_core.c | 633 +- .../modules/audio_processing/aec/aec_core.h | 48 +- .../audio_processing/aec/aec_core_internal.h | 89 +- .../audio_processing/aec/aec_core_mips.c | 2 +- .../audio_processing/aec/aec_core_neon.c | 3 + .../modules/audio_processing/aec/aec_rdft.c | 20 +- .../modules/audio_processing/aec/aec_rdft.h | 16 +- .../audio_processing/aec/aec_resampler.c | 34 +- .../audio_processing/aec/aec_resampler.h | 10 +- .../audio_processing/aec/echo_cancellation.c | 403 +- .../aec/echo_cancellation_internal.h | 6 +- .../aec/include/echo_cancellation.h | 109 +- .../aec/system_delay_unittest.cc | 29 +- .../modules/audio_processing/aecm/aecm_core.c | 133 +- .../modules/audio_processing/aecm/aecm_core.h | 125 +- .../audio_processing/aecm/aecm_core_c.c | 101 +- .../audio_processing/aecm/aecm_core_mips.c | 43 +- .../audio_processing/aecm/aecm_core_neon.S | 171 - .../audio_processing/aecm/aecm_core_neon.c | 421 +- .../aecm/aecm_core_neon_offsets.c | 26 - .../aecm/echo_control_mobile.c | 242 +- .../aecm/include/echo_control_mobile.h | 89 +- .../modules/audio_processing/agc/agc.cc | 158 + .../webrtc/modules/audio_processing/agc/agc.h | 69 + .../audio_processing/agc/agc_audio_proc.cc | 269 + .../audio_processing/agc/agc_audio_proc.h | 83 + .../agc/agc_audio_proc_internal.h | 81 + .../agc/agc_audio_proc_unittest.cc | 61 + .../agc/agc_manager_direct.cc | 435 + .../audio_processing/agc/agc_manager_direct.h | 98 + .../audio_processing/agc/agc_unittest.cc | 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mode 100644 testing/web-platform/meta/mixed-content/blockable/no-opt-in/same-host-http/worker-request/top-level/swap-scheme-redirect/no-opt-in-blocks.https.html.ini create mode 100644 testing/web-platform/meta/workers/Worker_cross_origin_security_err.htm.ini delete mode 100644 testing/web-platform/meta/workers/interfaces/WorkerUtils/importScripts/006.html.ini diff --git a/b2g/components/LogCapture.jsm b/b2g/components/LogCapture.jsm index 2170d1f597..803028d574 100644 --- a/b2g/components/LogCapture.jsm +++ b/b2g/components/LogCapture.jsm @@ -10,6 +10,7 @@ const Cu = Components.utils; const Ci = Components.interfaces; const Cc = Components.classes; +Cu.importGlobalProperties(['FileReader']); Cu.import("resource://gre/modules/XPCOMUtils.jsm"); XPCOMUtils.defineLazyModuleGetter(this, "Promise", "resource://gre/modules/Promise.jsm"); @@ -198,9 +199,7 @@ var LogCapture = { try { this.ensureLoaded(); - let fr = Cc["@mozilla.org/files/filereader;1"] - .createInstance(Ci.nsIDOMFileReader); - + let fr = new FileReader(); fr.onload = function(evt) { deferred.resolve(new Uint8Array(evt.target.result)); }; diff --git a/build/clang-plugin/clang-plugin.cpp b/build/clang-plugin/clang-plugin.cpp index 8cab4b4458..bc2ce02e0e 100644 --- a/build/clang-plugin/clang-plugin.cpp +++ b/build/clang-plugin/clang-plugin.cpp @@ -164,6 +164,7 @@ bool isInIgnoredNamespaceForImplicitCtor(const Decl *decl) { name == "__gnu_cxx" || // gnu C++ lib name == "boost" || // boost name == "webrtc" || // upstream webrtc + name == "rtc" || // upstream webrtc 'base' package name.substr(0, 4) == "icu_" || // icu name == "google" || // protobuf name == "google_breakpad" || // breakpad diff --git a/build/gyp.mozbuild b/build/gyp.mozbuild index be53d0defc..55be1cea12 100644 --- a/build/gyp.mozbuild +++ b/build/gyp.mozbuild @@ -5,6 +5,8 @@ # file, You can obtain one at http://mozilla.org/MPL/2.0/. gyp_vars = { + 'lsan': 0, + 'asan': 0, 'build_with_mozilla': 1, 'build_with_chromium': 0, 'use_official_google_api_keys': 0, @@ -27,14 +29,14 @@ gyp_vars = { 'build_libyuv': 0, 'build_libvpx': 0, 'build_ssl': 0, + 'build_json': 0, + 'build_icu': 0, + 'build_opus': 0, 'libyuv_dir': '/media/libyuv', 'yuv_disable_avx2': 0 if CONFIG['HAVE_X86_AVX2'] else 1, # don't use openssl 'use_openssl': 0, - # saves 4MB when webrtc_trace is off - 'enable_lazy_trace_alloc': 1 if CONFIG['RELEASE_BUILD'] else 0, - 'use_x11': 1 if CONFIG['MOZ_X11'] else 0, 'use_glib': 1 if CONFIG['GLIB_LIBS'] else 0, @@ -60,7 +62,9 @@ gyp_vars = { 'include_opus': 1, 'include_g722': 1, 'include_ilbc': 0, - 'include_isac': 0, + # We turn on ISAC because the AGC uses parts of it, and depend on the + # linker to throw away uneeded bits. + 'include_isac': 1, 'include_pcm16b': 1, } diff --git a/dom/apps/ImportExport.jsm b/dom/apps/ImportExport.jsm index 8cfbd160ae..e2797084eb 100644 --- a/dom/apps/ImportExport.jsm +++ b/dom/apps/ImportExport.jsm @@ -15,7 +15,7 @@ Cu.import("resource://gre/modules/Promise.jsm"); Cu.import("resource://gre/modules/Webapps.jsm"); Cu.import("resource://gre/modules/MessageBroadcaster.jsm"); -Cu.importGlobalProperties(['File']); +Cu.importGlobalProperties(['File', 'FileReader']); XPCOMUtils.defineLazyModuleGetter(this, "FileUtils", "resource://gre/modules/FileUtils.jsm"); @@ -255,8 +255,7 @@ this.ImportExport = { debug("_writeBlobToTempFile"); let path; return new Promise((aResolve, aReject) => { - let reader = Cc['@mozilla.org/files/filereader;1'] - .createInstance(Ci.nsIDOMFileReader); + let reader = new FileReader(); reader.onloadend = () => { path = OS.Path.join(OS.Constants.Path.tmpDir, "app-blob.zip"); debug("onloadend path=" + path); diff --git a/dom/base/BlobSet.h b/dom/base/BlobSet.h index 3a0596282d..8ca8f4aa43 100644 --- a/dom/base/BlobSet.h +++ b/dom/base/BlobSet.h @@ -32,7 +32,8 @@ public: nsTArray>& GetBlobImpls() { Flush(); return mBlobImpls; } already_AddRefed GetBlobInternal(nsISupports* aParent, - const nsACString& aContentType); + const nsACString& aContentType, + ErrorResult& aRv); protected: bool ExpandBufferSize(uint64_t aSize) diff --git a/dom/base/ChildIterator.cpp b/dom/base/ChildIterator.cpp index fef7f582d4..13dd9fc66f 100644 --- a/dom/base/ChildIterator.cpp +++ b/dom/base/ChildIterator.cpp @@ -190,6 +190,30 @@ FlattenedChildIterator::Init(bool aIgnoreXBL) } } +void +ExplicitChildIterator::Seek(nsIContent* aChildToFind) +{ + if (aChildToFind->GetParent() == mParent && + !aChildToFind->IsRootOfAnonymousSubtree()) { + // Fast path: just point ourselves to aChildToFind, which is a + // normal DOM child of ours. + MOZ_ASSERT(!ShadowRoot::IsShadowInsertionPoint(aChildToFind)); + MOZ_ASSERT(!nsContentUtils::IsContentInsertionPoint(aChildToFind)); + mChild = aChildToFind; + mIndexInInserted = 0; + mShadowIterator = nullptr; + mDefaultChild = nullptr; + mIsFirst = false; + return; + } + + // Can we add more fast paths here based on whether the parent of aChildToFind + // is a shadow insertion point or content insertion point? + + // Slow path: just walk all our kids. + Seek(aChildToFind, nullptr); +} + nsIContent* ExplicitChildIterator::Get() { diff --git a/dom/base/ChildIterator.h b/dom/base/ChildIterator.h index def339e038..dbe6def3cf 100644 --- a/dom/base/ChildIterator.h +++ b/dom/base/ChildIterator.h @@ -60,16 +60,24 @@ public: nsIContent* GetNextChild(); - // Looks for aChildToFind respecting insertion points until aChildToFind + // Looks for aChildToFind respecting insertion points until aChildToFind is + // found. This version can take shortcuts that the two-argument version + // can't, so can be faster (and in fact can be O(1) instead of O(N) in many + // cases). + void Seek(nsIContent* aChildToFind); + + // Looks for aChildToFind respecting insertion points until aChildToFind is found. // or aBound is found. If aBound is nullptr then the seek is unbounded. Returns // whether aChildToFind was found as an explicit child prior to encountering // aBound. - bool Seek(nsIContent* aChildToFind, nsIContent* aBound = nullptr) + bool Seek(nsIContent* aChildToFind, nsIContent* aBound) { // It would be nice to assert that we find aChildToFind, but bz thinks that // we might not find aChildToFind when called from ContentInserted // if first-letter frames are about. + // We can't easily take shortcuts here because we'd have to have a way to + // compare aChildToFind to aBound. nsIContent* child; do { child = GetNextChild(); diff --git a/dom/base/Console.cpp b/dom/base/Console.cpp index 38e580459f..e1b35a9728 100644 --- a/dom/base/Console.cpp +++ b/dom/base/Console.cpp @@ -18,6 +18,7 @@ #include "nsDOMNavigationTiming.h" #include "nsGlobalWindow.h" #include "nsJSUtils.h" +#include "nsNetUtil.h" #include "nsPerformance.h" #include "ScriptSettings.h" #include "WorkerPrivate.h" @@ -34,15 +35,12 @@ #include "nsIInterfaceRequestorUtils.h" #include "nsILoadContext.h" #include "nsIProgrammingLanguage.h" +#include "nsISensitiveInfoHiddenURI.h" #include "nsIServiceManager.h" #include "nsISupportsPrimitives.h" #include "nsIWebNavigation.h" #include "nsIXPConnect.h" -#ifdef MOZ_ENABLE_PROFILER_SPS -#include "nsIProfiler.h" -#endif - // The maximum allowed number of concurrent timers per page. #define MAX_PAGE_TIMERS 10000 @@ -240,7 +238,7 @@ public: } private: - NS_IMETHOD Run() + NS_IMETHOD Run() override { AssertIsOnMainThread(); @@ -311,7 +309,9 @@ private: MOZ_ASSERT(aWindow->IsInnerWindow()); nsPIDOMWindow* outerWindow = aWindow->GetOuterWindow(); - MOZ_ASSERT(outerWindow); + if (NS_WARN_IF(!outerWindow)) { + return; + } RunConsole(jsapi.cx(), outerWindow, aWindow); } @@ -697,9 +697,13 @@ Console::Console(nsPIDOMWindow* aWindow) MOZ_ASSERT(mWindow->IsInnerWindow()); mInnerID = mWindow->WindowID(); + // Without outerwindow any console message coming from this object will not + // shown in the devtools webconsole. But this should be fine because + // probably we are shutting down, or the window is CCed/GCed. nsPIDOMWindow* outerWindow = mWindow->GetOuterWindow(); - MOZ_ASSERT(outerWindow); - mOuterID = outerWindow->WindowID(); + if (outerWindow) { + mOuterID = outerWindow->WindowID(); + } } if (NS_IsMainThread()) { @@ -829,23 +833,6 @@ Console::TimeStamp(JSContext* aCx, const JS::Handle aData) return; } -#ifdef MOZ_ENABLE_PROFILER_SPS - if (aData.isString() && NS_IsMainThread()) { - if (!mProfiler) { - mProfiler = do_GetService("@mozilla.org/tools/profiler;1"); - } - if (mProfiler) { - bool active = false; - if (NS_SUCCEEDED(mProfiler->IsActive(&active)) && active) { - nsAutoJSString stringValue; - if (stringValue.init(aCx, aData)) { - mProfiler->AddMarker(NS_ConvertUTF16toUTF8(stringValue).get()); - } - } - } - } -#endif - Method(aCx, MethodTimeStamp, NS_LITERAL_STRING("timeStamp"), data); } @@ -1222,6 +1209,19 @@ Console::ProcessCallData(ConsoleCallData* aData) event.mLevel = aData->mMethodString; event.mFilename = frame.mFilename; + + nsCOMPtr filenameURI; + nsAutoCString pass; + if (NS_SUCCEEDED(NS_NewURI(getter_AddRefs(filenameURI), frame.mFilename)) && + NS_SUCCEEDED(filenameURI->GetPassword(pass)) && !pass.IsEmpty()) { + nsCOMPtr safeURI = do_QueryInterface(filenameURI); + nsAutoCString spec; + if (safeURI && + NS_SUCCEEDED(safeURI->GetSensitiveInfoHiddenSpec(spec))) { + CopyUTF8toUTF16(spec, event.mFilename); + } + } + event.mLineNumber = frame.mLineNumber; event.mColumnNumber = frame.mColumnNumber; event.mFunctionName = frame.mFunctionName; @@ -1389,7 +1389,7 @@ FlushOutput(JSContext* aCx, Sequence& aSequence, nsString &aOutput) return true; } -} // anonymous namespace +} // namespace bool Console::ProcessArguments(JSContext* aCx, @@ -1519,6 +1519,12 @@ Console::ProcessArguments(JSContext* aCx, case 'c': { + // If there isn't any output but there's already a style, then + // discard the previous style and use the next one instead. + if (output.IsEmpty() && !aStyles.IsEmpty()) { + aStyles.TruncateLength(aStyles.Length() - 1); + } + if (!FlushOutput(aCx, aSequence, output)) { return false; } diff --git a/dom/base/Console.h b/dom/base/Console.h index 506a2c3dac..394278b063 100644 --- a/dom/base/Console.h +++ b/dom/base/Console.h @@ -20,7 +20,6 @@ class nsIConsoleAPIStorage; class nsIPrincipal; -class nsIProfiler; namespace mozilla { namespace dom { @@ -205,9 +204,6 @@ private: nsCOMPtr mWindow; nsCOMPtr mStorage; RefPtr mSandbox; -#ifdef MOZ_ENABLE_PROFILER_SPS - nsCOMPtr mProfiler; -#endif nsDataHashtable mTimerRegistry; nsDataHashtable mCounterRegistry; diff --git a/dom/base/File.cpp b/dom/base/File.cpp index e3df761998..5cc20a76e4 100644 --- a/dom/base/File.cpp +++ b/dom/base/File.cpp @@ -257,7 +257,7 @@ Blob::ToFile() } already_AddRefed -Blob::ToFile(const nsAString& aName) const +Blob::ToFile(const nsAString& aName, ErrorResult& aRv) const { nsAutoTArray, 1> blobImpls; blobImpls.AppendElement(mImpl); @@ -266,7 +266,10 @@ Blob::ToFile(const nsAString& aName) const mImpl->GetType(contentType); RefPtr impl = - new MultipartBlobImpl(blobImpls, aName, contentType); + MultipartBlobImpl::Create(blobImpls, aName, contentType, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } RefPtr file = new File(mParent, impl); return file.forget(); @@ -347,7 +350,11 @@ Blob::Constructor(const GlobalObject& aGlobal, ErrorResult& aRv) { RefPtr impl = new MultipartBlobImpl(); - impl->InitializeBlob(); + impl->InitializeBlob(aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + MOZ_ASSERT(!impl->IsFile()); RefPtr blob = Blob::Create(aGlobal.GetAsSupports(), impl); @@ -851,7 +858,7 @@ BlobImplFile::GetMozFullPathInternal(nsAString& aFilename, ErrorResult& aRv) con uint64_t BlobImplFile::GetSize(ErrorResult& aRv) { - if (IsSizeUnknown()) { + if (BlobImplBase::IsSizeUnknown()) { NS_ASSERTION(mWholeFile, "Should only use lazy size when using the whole file"); int64_t fileSize; @@ -902,7 +909,7 @@ int64_t BlobImplFile::GetLastModified(ErrorResult& aRv) { NS_ASSERTION(mIsFile, "Should only be called on files"); - if (IsDateUnknown()) { + if (BlobImplBase::IsDateUnknown()) { PRTime msecs; aRv = mFile->GetLastModifiedTime(&msecs); if (NS_WARN_IF(aRv.Failed())) { @@ -1121,11 +1128,19 @@ BlobImplTemporaryBlob::GetInternalStream(nsIInputStream** aStream, // BlobSet implementation already_AddRefed -BlobSet::GetBlobInternal(nsISupports* aParent, const nsACString& aContentType) +BlobSet::GetBlobInternal(nsISupports* aParent, + const nsACString& aContentType, + ErrorResult& aRv) { - RefPtr blob = Blob::Create(aParent, - new MultipartBlobImpl(GetBlobImpls(), - NS_ConvertASCIItoUTF16(aContentType))); + RefPtr blobImpl = + MultipartBlobImpl::Create(GetBlobImpls(), + NS_ConvertASCIItoUTF16(aContentType), + aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + + RefPtr blob = Blob::Create(aParent, blobImpl); return blob.forget(); } diff --git a/dom/base/File.h b/dom/base/File.h index bb1a2cb5c9..25550faab4 100644 --- a/dom/base/File.h +++ b/dom/base/File.h @@ -118,7 +118,8 @@ public: // This method creates a new File object with the given name and the same // BlobImpl. - already_AddRefed ToFile(const nsAString& aName) const; + already_AddRefed ToFile(const nsAString& aName, + ErrorResult& aRv) const; already_AddRefed CreateSlice(uint64_t aStart, uint64_t aLength, const nsAString& aContentType, @@ -798,6 +799,10 @@ public: virtual void LookupAndCacheIsDirectory() override; + // We always have size and date for this kind of blob. + virtual bool IsSizeUnknown() const override { return false; } + virtual bool IsDateUnknown() const override { return false; } + protected: virtual ~BlobImplFile() { if (mFile && mIsTemporary) { diff --git a/dom/base/FileIOObject.cpp b/dom/base/FileIOObject.cpp deleted file mode 100644 index 8c0c097b55..0000000000 --- a/dom/base/FileIOObject.cpp +++ /dev/null @@ -1,246 +0,0 @@ -/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ -/* vim: set ts=8 sts=2 et sw=2 tw=80: */ -/* This Source Code Form is subject to the terms of the Mozilla Public - * License, v. 2.0. If a copy of the MPL was not distributed with this - * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ - -#include "FileIOObject.h" -#include "mozilla/dom/File.h" -#include "mozilla/dom/ProgressEvent.h" -#include "mozilla/EventDispatcher.h" -#include "nsComponentManagerUtils.h" -#include "nsError.h" -#include "nsIDOMEvent.h" - -#define ERROR_STR "error" -#define ABORT_STR "abort" -#define PROGRESS_STR "progress" - -namespace mozilla { -namespace dom { - -const uint64_t kUnknownSize = uint64_t(-1); - -NS_IMPL_ADDREF_INHERITED(FileIOObject, DOMEventTargetHelper) -NS_IMPL_RELEASE_INHERITED(FileIOObject, DOMEventTargetHelper) - -NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(FileIOObject) - NS_INTERFACE_MAP_ENTRY(nsITimerCallback) - NS_INTERFACE_MAP_ENTRY(nsIInputStreamCallback) -NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper) - -NS_IMPL_CYCLE_COLLECTION_CLASS(FileIOObject) - -NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(FileIOObject, - DOMEventTargetHelper) - NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mProgressNotifier) - NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mError) -NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END - -NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(FileIOObject, - DOMEventTargetHelper) - NS_IMPL_CYCLE_COLLECTION_UNLINK(mProgressNotifier) - NS_IMPL_CYCLE_COLLECTION_UNLINK(mError) -NS_IMPL_CYCLE_COLLECTION_UNLINK_END - -NS_IMPL_EVENT_HANDLER(FileIOObject, abort) -NS_IMPL_EVENT_HANDLER(FileIOObject, error) -NS_IMPL_EVENT_HANDLER(FileIOObject, progress) - -FileIOObject::FileIOObject() - : mProgressEventWasDelayed(false), - mTimerIsActive(false), - mReadyState(0), - mTotal(0), mTransferred(0) -{} - -FileIOObject::~FileIOObject() -{} - -void -FileIOObject::StartProgressEventTimer() -{ - if (!mProgressNotifier) { - mProgressNotifier = do_CreateInstance(NS_TIMER_CONTRACTID); - } - if (mProgressNotifier) { - mProgressEventWasDelayed = false; - mTimerIsActive = true; - mProgressNotifier->Cancel(); - mProgressNotifier->InitWithCallback(this, NS_PROGRESS_EVENT_INTERVAL, - nsITimer::TYPE_ONE_SHOT); - } -} - -void -FileIOObject::ClearProgressEventTimer() -{ - mProgressEventWasDelayed = false; - mTimerIsActive = false; - if (mProgressNotifier) { - mProgressNotifier->Cancel(); - } -} - -void -FileIOObject::DispatchError(nsresult rv, nsAString& finalEvent) -{ - // Set the status attribute, and dispatch the error event - switch (rv) { - case NS_ERROR_FILE_NOT_FOUND: - mError = new DOMError(GetOwner(), NS_LITERAL_STRING("NotFoundError")); - break; - case NS_ERROR_FILE_ACCESS_DENIED: - mError = new DOMError(GetOwner(), NS_LITERAL_STRING("SecurityError")); - break; - default: - mError = new DOMError(GetOwner(), NS_LITERAL_STRING("NotReadableError")); - break; - } - - // Dispatch error event to signify load failure - DispatchProgressEvent(NS_LITERAL_STRING(ERROR_STR)); - DispatchProgressEvent(finalEvent); -} - -nsresult -FileIOObject::DispatchProgressEvent(const nsAString& aType) -{ - ProgressEventInit init; - init.mBubbles = false; - init.mCancelable = false; - init.mLoaded = mTransferred; - - if (mTotal != kUnknownSize) { - init.mLengthComputable = true; - init.mTotal = mTotal; - } else { - init.mLengthComputable = false; - init.mTotal = 0; - } - RefPtr event = - ProgressEvent::Constructor(this, aType, init); - event->SetTrusted(true); - - return DispatchDOMEvent(nullptr, event, nullptr, nullptr); -} - -// nsITimerCallback -NS_IMETHODIMP -FileIOObject::Notify(nsITimer* aTimer) -{ - nsresult rv; - mTimerIsActive = false; - - if (mProgressEventWasDelayed) { - rv = DispatchProgressEvent(NS_LITERAL_STRING("progress")); - NS_ENSURE_SUCCESS(rv, rv); - - StartProgressEventTimer(); - } - - return NS_OK; -} - -// InputStreamCallback -NS_IMETHODIMP -FileIOObject::OnInputStreamReady(nsIAsyncInputStream* aStream) -{ - if (mReadyState != 1 || aStream != mAsyncStream) { - return NS_OK; - } - - uint64_t aCount; - nsresult rv = aStream->Available(&aCount); - - if (NS_SUCCEEDED(rv) && aCount) { - rv = DoReadData(aStream, aCount); - } - - if (NS_SUCCEEDED(rv)) { - rv = DoAsyncWait(aStream); - } - - if (NS_FAILED(rv) || !aCount) { - if (rv == NS_BASE_STREAM_CLOSED) { - rv = NS_OK; - } - return OnLoadEnd(rv); - } - - mTransferred += aCount; - - //Notify the timer is the appropriate timeframe has passed - if (mTimerIsActive) { - mProgressEventWasDelayed = true; - } else { - rv = DispatchProgressEvent(NS_LITERAL_STRING(PROGRESS_STR)); - NS_ENSURE_SUCCESS(rv, rv); - - StartProgressEventTimer(); - } - - return NS_OK; -} - -nsresult -FileIOObject::OnLoadEnd(nsresult aStatus) -{ - // Cancel the progress event timer - ClearProgressEventTimer(); - - // FileIOObject must be in DONE stage after an operation - mReadyState = 2; - - nsString successEvent, termEvent; - nsresult rv = DoOnLoadEnd(aStatus, successEvent, termEvent); - NS_ENSURE_SUCCESS(rv, rv); - - // Set the status field as appropriate - if (NS_FAILED(aStatus)) { - DispatchError(aStatus, termEvent); - return NS_OK; - } - - // Dispatch event to signify end of a successful operation - DispatchProgressEvent(successEvent); - DispatchProgressEvent(termEvent); - - return NS_OK; -} - -nsresult -FileIOObject::DoAsyncWait(nsIAsyncInputStream* aStream) -{ - return aStream->AsyncWait(this, - /* aFlags*/ 0, - /* aRequestedCount */ 0, - NS_GetCurrentThread()); -} - -void -FileIOObject::Abort(ErrorResult& aRv) -{ - if (mReadyState != 1) { - // XXX The spec doesn't say this - aRv.Throw(NS_ERROR_DOM_FILE_ABORT_ERR); - return; - } - - ClearProgressEventTimer(); - - mReadyState = 2; // There are DONE constants on multiple interfaces, - // but they all have value 2. - // XXX The spec doesn't say this - mError = new DOMError(GetOwner(), NS_LITERAL_STRING("AbortError")); - - nsString finalEvent; - DoAbort(finalEvent); - - // Dispatch the events - DispatchProgressEvent(NS_LITERAL_STRING(ABORT_STR)); - DispatchProgressEvent(finalEvent); -} - -} // namespace dom -} // namespace mozilla diff --git a/dom/base/FileIOObject.h b/dom/base/FileIOObject.h deleted file mode 100644 index 769f09b041..0000000000 --- a/dom/base/FileIOObject.h +++ /dev/null @@ -1,103 +0,0 @@ -/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ -/* vim: set ts=8 sts=2 et sw=2 tw=80: */ -/* This Source Code Form is subject to the terms of the Mozilla Public - * License, v. 2.0. If a copy of the MPL was not distributed with this - * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ - -#ifndef FileIOObject_h__ -#define FileIOObject_h__ - -#include "mozilla/DOMEventTargetHelper.h" -#include "nsIFile.h" -#include "nsITimer.h" -#include "nsCOMPtr.h" -#include "nsIAsyncInputStream.h" - -#include "mozilla/dom/DOMError.h" - -#define NS_PROGRESS_EVENT_INTERVAL 50 - -namespace mozilla { - -class ErrorResult; - -namespace dom { - -extern const uint64_t kUnknownSize; - -// A common base class for FileReader and FileSaver - -class FileIOObject : public DOMEventTargetHelper, - public nsIInputStreamCallback, - public nsITimerCallback -{ -public: - FileIOObject(); - - NS_DECL_ISUPPORTS_INHERITED - - // Common methods - void Abort(ErrorResult& aRv); - uint16_t ReadyState() const - { - return mReadyState; - } - DOMError* GetError() const - { - return mError; - } - - NS_METHOD GetOnabort(JSContext* aCx, JS::MutableHandle aValue); - NS_METHOD SetOnabort(JSContext* aCx, JS::Handle aValue); - NS_METHOD GetOnerror(JSContext* aCx, JS::MutableHandle aValue); - NS_METHOD SetOnerror(JSContext* aCx, JS::Handle aValue); - NS_METHOD GetOnprogress(JSContext* aCx, JS::MutableHandle aValue); - NS_METHOD SetOnprogress(JSContext* aCx, JS::Handle aValue); - - IMPL_EVENT_HANDLER(abort) - IMPL_EVENT_HANDLER(error) - IMPL_EVENT_HANDLER(progress) - - NS_DECL_NSITIMERCALLBACK - - NS_DECL_NSIINPUTSTREAMCALLBACK - - NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(FileIOObject, DOMEventTargetHelper) - -protected: - virtual ~FileIOObject(); - - // Implemented by the derived class to do whatever it needs to do for abort - virtual void DoAbort(nsAString& aEvent) = 0; - - virtual nsresult DoReadData(nsIAsyncInputStream* aStream, uint64_t aCount) = 0; - - virtual nsresult DoOnLoadEnd(nsresult aStatus, nsAString& aSuccessEvent, - nsAString& aTerminationEvent) = 0; - - nsresult OnLoadEnd(nsresult aStatus); - nsresult DoAsyncWait(nsIAsyncInputStream* aStream); - - void StartProgressEventTimer(); - void ClearProgressEventTimer(); - void DispatchError(nsresult rv, nsAString& finalEvent); - nsresult DispatchProgressEvent(const nsAString& aType); - - nsCOMPtr mProgressNotifier; - bool mProgressEventWasDelayed; - bool mTimerIsActive; - - nsCOMPtr mAsyncStream; - - RefPtr mError; - - uint16_t mReadyState; - - uint64_t mTotal; - uint64_t mTransferred; -}; - -} // namespace dom -} // namespace mozilla - -#endif diff --git a/dom/base/nsDOMFileReader.cpp b/dom/base/FileReader.cpp similarity index 50% rename from dom/base/nsDOMFileReader.cpp rename to dom/base/FileReader.cpp index a5b1b82776..0ad85b8e70 100644 --- a/dom/base/nsDOMFileReader.cpp +++ b/dom/base/FileReader.cpp @@ -4,7 +4,7 @@ * License, v. 2.0. If a copy of the MPL was not distributed with this * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ -#include "nsDOMFileReader.h" +#include "FileReader.h" #include "nsContentCID.h" #include "nsContentUtils.h" @@ -22,6 +22,7 @@ #include "mozilla/dom/EncodingUtils.h" #include "mozilla/dom/File.h" #include "mozilla/dom/FileReaderBinding.h" +#include "mozilla/dom/ProgressEvent.h" #include "xpcpublic.h" #include "nsDOMJSUtils.h" @@ -30,226 +31,137 @@ #include "nsITransport.h" #include "nsIStreamTransportService.h" -using namespace mozilla; -using namespace mozilla::dom; +namespace mozilla { +namespace dom { +#define ABORT_STR "abort" #define LOAD_STR "load" #define LOADSTART_STR "loadstart" #define LOADEND_STR "loadend" +#define ERROR_STR "error" +#define PROGRESS_STR "progress" + +const uint64_t kUnknownSize = uint64_t(-1); static NS_DEFINE_CID(kStreamTransportServiceCID, NS_STREAMTRANSPORTSERVICE_CID); -NS_IMPL_CYCLE_COLLECTION_CLASS(nsDOMFileReader) +NS_IMPL_CYCLE_COLLECTION_CLASS(FileReader) -NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(nsDOMFileReader, - FileIOObject) +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(FileReader, + DOMEventTargetHelper) NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mBlob) + NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mProgressNotifier) + NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mError) NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END -NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(nsDOMFileReader, - FileIOObject) +NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(FileReader, + DOMEventTargetHelper) tmp->mResultArrayBuffer = nullptr; NS_IMPL_CYCLE_COLLECTION_UNLINK(mBlob) + NS_IMPL_CYCLE_COLLECTION_UNLINK(mProgressNotifier) + NS_IMPL_CYCLE_COLLECTION_UNLINK(mError) NS_IMPL_CYCLE_COLLECTION_UNLINK_END - -NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN_INHERITED(nsDOMFileReader, +NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN_INHERITED(FileReader, DOMEventTargetHelper) NS_IMPL_CYCLE_COLLECTION_TRACE_JS_MEMBER_CALLBACK(mResultArrayBuffer) NS_IMPL_CYCLE_COLLECTION_TRACE_END -NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(nsDOMFileReader) - NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY - NS_INTERFACE_MAP_ENTRY(nsIDOMFileReader) +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(FileReader) + NS_INTERFACE_MAP_ENTRY(nsITimerCallback) + NS_INTERFACE_MAP_ENTRY(nsIInputStreamCallback) NS_INTERFACE_MAP_ENTRY(nsISupportsWeakReference) -NS_INTERFACE_MAP_END_INHERITING(FileIOObject) +NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper) -NS_IMPL_ADDREF_INHERITED(nsDOMFileReader, FileIOObject) -NS_IMPL_RELEASE_INHERITED(nsDOMFileReader, FileIOObject) - -NS_IMPL_EVENT_HANDLER(nsDOMFileReader, load) -NS_IMPL_EVENT_HANDLER(nsDOMFileReader, loadend) -NS_IMPL_EVENT_HANDLER(nsDOMFileReader, loadstart) -NS_IMPL_FORWARD_EVENT_HANDLER(nsDOMFileReader, abort, FileIOObject) -NS_IMPL_FORWARD_EVENT_HANDLER(nsDOMFileReader, progress, FileIOObject) -NS_IMPL_FORWARD_EVENT_HANDLER(nsDOMFileReader, error, FileIOObject) +NS_IMPL_ADDREF_INHERITED(FileReader, DOMEventTargetHelper) +NS_IMPL_RELEASE_INHERITED(FileReader, DOMEventTargetHelper) void -nsDOMFileReader::RootResultArrayBuffer() +FileReader::RootResultArrayBuffer() { mozilla::HoldJSObjects(this); } -//nsDOMFileReader constructors/initializers +//FileReader constructors/initializers -nsDOMFileReader::nsDOMFileReader() - : mFileData(nullptr), - mDataLen(0), mDataFormat(FILE_AS_BINARY), - mResultArrayBuffer(nullptr) +FileReader::FileReader(nsPIDOMWindow* aWindow) + : DOMEventTargetHelper(aWindow) + , mFileData(nullptr) + , mDataLen(0) + , mDataFormat(FILE_AS_BINARY) + , mResultArrayBuffer(nullptr) + , mProgressEventWasDelayed(false) + , mTimerIsActive(false) + , mReadyState(EMPTY) + , mTotal(0) + , mTransferred(0) { SetDOMStringToNull(mResult); } -nsDOMFileReader::~nsDOMFileReader() +FileReader::~FileReader() { FreeFileData(); mResultArrayBuffer = nullptr; - mozilla::DropJSObjects(this); + DropJSObjects(this); } - -/** - * This Init method is called from the factory constructor. - */ -nsresult -nsDOMFileReader::Init() +/* static */ already_AddRefed +FileReader::Constructor(const GlobalObject& aGlobal, ErrorResult& aRv) { - // Instead of grabbing some random global from the context stack, - // let's use the default one (junk scope) for now. - // We should move away from this Init... - BindToOwner(xpc::NativeGlobal(xpc::PrivilegedJunkScope())); - return NS_OK; -} - -/* static */ already_AddRefed -nsDOMFileReader::Constructor(const GlobalObject& aGlobal, ErrorResult& aRv) -{ - RefPtr fileReader = new nsDOMFileReader(); - + // The owner can be null when this object is used by chrome code. nsCOMPtr owner = do_QueryInterface(aGlobal.GetAsSupports()); - if (!owner) { - NS_WARNING("Unexpected owner"); - aRv.Throw(NS_ERROR_FAILURE); - return nullptr; + RefPtr fileReader = new FileReader(owner); + + if (!owner && nsContentUtils::IsCallerChrome()) { + // Instead of grabbing some random global from the context stack, + // let's use the default one (junk scope) for now. + // We should move away from this Init... + fileReader->BindToOwner(xpc::NativeGlobal(xpc::PrivilegedJunkScope())); } - fileReader->BindToOwner(owner); return fileReader.forget(); } // nsIInterfaceRequestor NS_IMETHODIMP -nsDOMFileReader::GetInterface(const nsIID & aIID, void **aResult) +FileReader::GetInterface(const nsIID & aIID, void **aResult) { return QueryInterface(aIID, aResult); } -// nsIDOMFileReader - -NS_IMETHODIMP -nsDOMFileReader::GetReadyState(uint16_t *aReadyState) -{ - *aReadyState = ReadyState(); - return NS_OK; -} - void -nsDOMFileReader::GetResult(JSContext* aCx, JS::MutableHandle aResult, - ErrorResult& aRv) -{ - aRv = GetResult(aCx, aResult); -} - -NS_IMETHODIMP -nsDOMFileReader::GetResult(JSContext* aCx, JS::MutableHandle aResult) +FileReader::GetResult(JSContext* aCx, + JS::MutableHandle aResult, + ErrorResult& aRv) { JS::Rooted result(aCx); + if (mDataFormat == FILE_AS_ARRAYBUFFER) { - if (mReadyState == nsIDOMFileReader::DONE && mResultArrayBuffer) { + if (mReadyState == DONE && mResultArrayBuffer) { result.setObject(*mResultArrayBuffer); } else { result.setNull(); } + if (!JS_WrapValue(aCx, &result)) { - return NS_ERROR_FAILURE; + aRv.Throw(NS_ERROR_FAILURE); + return; } + aResult.set(result); - return NS_OK; + return; } nsString tmpResult = mResult; if (!xpc::StringToJsval(aCx, tmpResult, aResult)) { - return NS_ERROR_FAILURE; + aRv.Throw(NS_ERROR_FAILURE); + return; } - return NS_OK; } -NS_IMETHODIMP -nsDOMFileReader::GetError(nsISupports** aError) -{ - NS_IF_ADDREF(*aError = GetError()); - return NS_OK; -} - -NS_IMETHODIMP -nsDOMFileReader::ReadAsArrayBuffer(nsIDOMBlob* aBlob, JSContext* aCx) -{ - NS_ENSURE_TRUE(aBlob, NS_ERROR_NULL_POINTER); - ErrorResult rv; - RefPtr blob = static_cast(aBlob); - ReadAsArrayBuffer(aCx, *blob, rv); - return rv.StealNSResult(); -} - -NS_IMETHODIMP -nsDOMFileReader::ReadAsBinaryString(nsIDOMBlob* aBlob) -{ - NS_ENSURE_TRUE(aBlob, NS_ERROR_NULL_POINTER); - ErrorResult rv; - RefPtr blob = static_cast(aBlob); - ReadAsBinaryString(*blob, rv); - return rv.StealNSResult(); -} - -NS_IMETHODIMP -nsDOMFileReader::ReadAsText(nsIDOMBlob* aBlob, - const nsAString &aCharset) -{ - NS_ENSURE_TRUE(aBlob, NS_ERROR_NULL_POINTER); - ErrorResult rv; - RefPtr blob = static_cast(aBlob); - ReadAsText(*blob, aCharset, rv); - return rv.StealNSResult(); -} - -NS_IMETHODIMP -nsDOMFileReader::ReadAsDataURL(nsIDOMBlob* aBlob) -{ - NS_ENSURE_TRUE(aBlob, NS_ERROR_NULL_POINTER); - ErrorResult rv; - RefPtr blob = static_cast(aBlob); - ReadAsDataURL(*blob, rv); - return rv.StealNSResult(); -} - -NS_IMETHODIMP -nsDOMFileReader::Abort() -{ - ErrorResult rv; - FileIOObject::Abort(rv); - return rv.StealNSResult(); -} - -/* virtual */ void -nsDOMFileReader::DoAbort(nsAString& aEvent) -{ - // Revert status and result attributes - SetDOMStringToNull(mResult); - mResultArrayBuffer = nullptr; - - mAsyncStream = nullptr; - mBlob = nullptr; - - //Clean up memory buffer - FreeFileData(); - - // Tell the base class which event to dispatch - aEvent = NS_LITERAL_STRING(LOADEND_STR); -} - -static -NS_METHOD +static NS_IMETHODIMP ReadFuncBinaryString(nsIInputStream* in, void* closure, const char* fromRawSegment, @@ -271,11 +183,10 @@ ReadFuncBinaryString(nsIInputStream* in, } nsresult -nsDOMFileReader::DoOnLoadEnd(nsresult aStatus, - nsAString& aSuccessEvent, - nsAString& aTerminationEvent) +FileReader::DoOnLoadEnd(nsresult aStatus, + nsAString& aSuccessEvent, + nsAString& aTerminationEvent) { - // Make sure we drop all the objects that could hold files open now. nsCOMPtr stream; mAsyncStream.swap(stream); @@ -283,25 +194,34 @@ nsDOMFileReader::DoOnLoadEnd(nsresult aStatus, RefPtr blob; mBlob.swap(blob); - aSuccessEvent = NS_LITERAL_STRING(LOAD_STR); - aTerminationEvent = NS_LITERAL_STRING(LOADEND_STR); - // Clear out the data if necessary if (NS_FAILED(aStatus)) { FreeFileData(); return NS_OK; } + // In case we read a different number of bytes, we can assume that the + // underlying storage has changed. We should not continue. + if (mDataLen != mTotal) { + DispatchError(NS_ERROR_FAILURE, aTerminationEvent); + FreeFileData(); + return NS_ERROR_FAILURE; + } + + aSuccessEvent = NS_LITERAL_STRING(LOAD_STR); + aTerminationEvent = NS_LITERAL_STRING(LOADEND_STR); + nsresult rv = NS_OK; switch (mDataFormat) { case FILE_AS_ARRAYBUFFER: { AutoJSAPI jsapi; - if (NS_WARN_IF(!jsapi.Init(mozilla::DOMEventTargetHelper::GetParentObject()))) { + if (NS_WARN_IF(!jsapi.Init(DOMEventTargetHelper::GetParentObject()))) { + FreeFileData(); return NS_ERROR_FAILURE; } RootResultArrayBuffer(); - mResultArrayBuffer = JS_NewArrayBufferWithContents(jsapi.cx(), mTotal, mFileData); + mResultArrayBuffer = JS_NewArrayBufferWithContents(jsapi.cx(), mDataLen, mFileData); if (!mResultArrayBuffer) { JS_ClearPendingException(jsapi.cx()); rv = NS_ERROR_OUT_OF_MEMORY; @@ -336,15 +256,14 @@ nsDOMFileReader::DoOnLoadEnd(nsresult aStatus, } nsresult -nsDOMFileReader::DoReadData(nsIAsyncInputStream* aStream, uint64_t aCount) +FileReader::DoReadData(nsIAsyncInputStream* aStream, uint64_t aCount) { MOZ_ASSERT(aStream); if (mDataFormat == FILE_AS_BINARY) { //Continuously update our binary string as data comes in uint32_t oldLen = mResult.Length(); - NS_ASSERTION(mResult.Length() == mDataLen, - "unexpected mResult length"); + NS_ASSERTION(mResult.Length() == mDataLen, "unexpected mResult length"); if (uint64_t(oldLen) + aCount > UINT32_MAX) return NS_ERROR_OUT_OF_MEMORY; char16_t *buf = nullptr; @@ -379,18 +298,21 @@ nsDOMFileReader::DoReadData(nsIAsyncInputStream* aStream, uint64_t aCount) // Helper methods void -nsDOMFileReader::ReadFileContent(Blob& aBlob, - const nsAString &aCharset, - eDataFormat aDataFormat, - ErrorResult& aRv) +FileReader::ReadFileContent(Blob& aBlob, + const nsAString &aCharset, + eDataFormat aDataFormat, + ErrorResult& aRv) { //Implicit abort to clear any other activity going on - Abort(); + ErrorResult error; + Abort(error); + error.SuppressException(); + mError = nullptr; SetDOMStringToNull(mResult); mTransferred = 0; mTotal = 0; - mReadyState = nsIDOMFileReader::EMPTY; + mReadyState = EMPTY; FreeFileData(); mBlob = &aBlob; @@ -398,7 +320,6 @@ nsDOMFileReader::ReadFileContent(Blob& aBlob, CopyUTF16toUTF8(aCharset, mCharset); nsresult rv; - nsCOMPtr sts = do_GetService(kStreamTransportServiceCID, &rv); if (NS_WARN_IF(NS_FAILED(rv))) { @@ -413,23 +334,21 @@ nsDOMFileReader::ReadFileContent(Blob& aBlob, } nsCOMPtr transport; - rv = sts->CreateInputTransport(stream, - /* aStartOffset */ 0, - /* aReadLimit */ -1, - /* aCloseWhenDone */ true, - getter_AddRefs(transport)); - if (NS_WARN_IF(NS_FAILED(rv))) { - aRv.Throw(rv); + aRv = sts->CreateInputTransport(stream, + /* aStartOffset */ 0, + /* aReadLimit */ -1, + /* aCloseWhenDone */ true, + getter_AddRefs(transport)); + if (NS_WARN_IF(aRv.Failed())) { return; } nsCOMPtr wrapper; - rv = transport->OpenInputStream(/* aFlags */ 0, - /* aSegmentSize */ 0, - /* aSegmentCount */ 0, - getter_AddRefs(wrapper)); - if (NS_WARN_IF(NS_FAILED(rv))) { - aRv.Throw(rv); + aRv = transport->OpenInputStream(/* aFlags */ 0, + /* aSegmentSize */ 0, + /* aSegmentCount */ 0, + getter_AddRefs(wrapper)); + if (NS_WARN_IF(aRv.Failed())) { return; } @@ -442,14 +361,16 @@ nsDOMFileReader::ReadFileContent(Blob& aBlob, return; } - rv = DoAsyncWait(mAsyncStream); - if (NS_WARN_IF(NS_FAILED(rv))) { - aRv.Throw(rv); + aRv = mAsyncStream->AsyncWait(this, + /* aFlags*/ 0, + /* aRequestedCount */ 0, + NS_GetCurrentThread()); + if (NS_WARN_IF(aRv.Failed())) { return; } //FileReader should be in loading state here - mReadyState = nsIDOMFileReader::LOADING; + mReadyState = LOADING; DispatchProgressEvent(NS_LITERAL_STRING(LOADSTART_STR)); if (mDataFormat == FILE_AS_ARRAYBUFFER) { @@ -462,11 +383,11 @@ nsDOMFileReader::ReadFileContent(Blob& aBlob, } nsresult -nsDOMFileReader::GetAsText(Blob *aBlob, - const nsACString &aCharset, - const char *aFileData, - uint32_t aDataLen, - nsAString& aResult) +FileReader::GetAsText(Blob *aBlob, + const nsACString &aCharset, + const char *aFileData, + uint32_t aDataLen, + nsAString& aResult) { // The BOM sniffing is baked into the "decode" part of the Encoding // Standard, which the File API references. @@ -502,14 +423,14 @@ nsDOMFileReader::GetAsText(Blob *aBlob, } nsresult -nsDOMFileReader::GetAsDataURL(Blob *aBlob, - const char *aFileData, - uint32_t aDataLen, - nsAString& aResult) +FileReader::GetAsDataURL(Blob *aBlob, + const char *aFileData, + uint32_t aDataLen, + nsAString& aResult) { aResult.AssignLiteral("data:"); - nsString contentType; + nsAutoString contentType; aBlob->GetType(contentType); if (!contentType.IsEmpty()) { aResult.Append(contentType); @@ -530,7 +451,197 @@ nsDOMFileReader::GetAsDataURL(Blob *aBlob, } /* virtual */ JSObject* -nsDOMFileReader::WrapObject(JSContext* aCx, JS::Handle aGivenProto) +FileReader::WrapObject(JSContext* aCx, JS::Handle aGivenProto) { return FileReaderBinding::Wrap(aCx, this, aGivenProto); } + +void +FileReader::StartProgressEventTimer() +{ + if (!mProgressNotifier) { + mProgressNotifier = do_CreateInstance(NS_TIMER_CONTRACTID); + } + + if (mProgressNotifier) { + mProgressEventWasDelayed = false; + mTimerIsActive = true; + mProgressNotifier->Cancel(); + mProgressNotifier->InitWithCallback(this, NS_PROGRESS_EVENT_INTERVAL, + nsITimer::TYPE_ONE_SHOT); + } +} + +void +FileReader::ClearProgressEventTimer() +{ + mProgressEventWasDelayed = false; + mTimerIsActive = false; + if (mProgressNotifier) { + mProgressNotifier->Cancel(); + } +} + +void +FileReader::DispatchError(nsresult rv, nsAString& finalEvent) +{ + // Set the status attribute, and dispatch the error event + switch (rv) { + case NS_ERROR_FILE_NOT_FOUND: + mError = new DOMError(GetOwner(), NS_LITERAL_STRING("NotFoundError")); + break; + case NS_ERROR_FILE_ACCESS_DENIED: + mError = new DOMError(GetOwner(), NS_LITERAL_STRING("SecurityError")); + break; + default: + mError = new DOMError(GetOwner(), NS_LITERAL_STRING("NotReadableError")); + break; + } + + // Dispatch error event to signify load failure + DispatchProgressEvent(NS_LITERAL_STRING(ERROR_STR)); + DispatchProgressEvent(finalEvent); +} + +nsresult +FileReader::DispatchProgressEvent(const nsAString& aType) +{ + ProgressEventInit init; + init.mBubbles = false; + init.mCancelable = false; + init.mLoaded = mTransferred; + + if (mTotal != kUnknownSize) { + init.mLengthComputable = true; + init.mTotal = mTotal; + } else { + init.mLengthComputable = false; + init.mTotal = 0; + } + RefPtr event = + ProgressEvent::Constructor(this, aType, init); + event->SetTrusted(true); + + return DispatchDOMEvent(nullptr, event, nullptr, nullptr); +} + +// nsITimerCallback +NS_IMETHODIMP +FileReader::Notify(nsITimer* aTimer) +{ + nsresult rv; + mTimerIsActive = false; + + if (mProgressEventWasDelayed) { + rv = DispatchProgressEvent(NS_LITERAL_STRING("progress")); + NS_ENSURE_SUCCESS(rv, rv); + + StartProgressEventTimer(); + } + + return NS_OK; +} + +// InputStreamCallback +NS_IMETHODIMP +FileReader::OnInputStreamReady(nsIAsyncInputStream* aStream) +{ + if (mReadyState != LOADING || aStream != mAsyncStream) { + return NS_OK; + } + + uint64_t aCount; + nsresult rv = aStream->Available(&aCount); + + if (NS_SUCCEEDED(rv) && aCount) { + rv = DoReadData(aStream, aCount); + } + + if (NS_SUCCEEDED(rv)) { + rv = aStream->AsyncWait(this, + /* aFlags*/ 0, + /* aRequestedCount */ 0, + NS_GetCurrentThread()); + } + + if (NS_FAILED(rv) || !aCount) { + if (rv == NS_BASE_STREAM_CLOSED) { + rv = NS_OK; + } + return OnLoadEnd(rv); + } + + mTransferred += aCount; + + //Notify the timer is the appropriate timeframe has passed + if (mTimerIsActive) { + mProgressEventWasDelayed = true; + } else { + rv = DispatchProgressEvent(NS_LITERAL_STRING(PROGRESS_STR)); + NS_ENSURE_SUCCESS(rv, rv); + + StartProgressEventTimer(); + } + + return NS_OK; +} + +nsresult +FileReader::OnLoadEnd(nsresult aStatus) +{ + // Cancel the progress event timer + ClearProgressEventTimer(); + + // FileReader must be in DONE stage after an operation + mReadyState = DONE; + + nsAutoString successEvent, termEvent; + nsresult rv = DoOnLoadEnd(aStatus, successEvent, termEvent); + NS_ENSURE_SUCCESS(rv, rv); + + // Set the status field as appropriate + if (NS_FAILED(aStatus)) { + DispatchError(aStatus, termEvent); + return NS_OK; + } + + // Dispatch event to signify end of a successful operation + DispatchProgressEvent(successEvent); + DispatchProgressEvent(termEvent); + + return NS_OK; +} + +void +FileReader::Abort(ErrorResult& aRv) +{ + if (mReadyState != LOADING) { + // XXX The spec doesn't say this + aRv.Throw(NS_ERROR_DOM_FILE_ABORT_ERR); + return; + } + + ClearProgressEventTimer(); + + mReadyState = DONE; + + // XXX The spec doesn't say this + mError = new DOMError(GetOwner(), NS_LITERAL_STRING("AbortError")); + + // Revert status and result attributes + SetDOMStringToNull(mResult); + mResultArrayBuffer = nullptr; + + mAsyncStream = nullptr; + mBlob = nullptr; + + //Clean up memory buffer + FreeFileData(); + + // Dispatch the events + DispatchProgressEvent(NS_LITERAL_STRING(ABORT_STR)); + DispatchProgressEvent(NS_LITERAL_STRING(LOADEND_STR)); +} + +} // dom namespace +} // mozilla namespace diff --git a/dom/base/nsDOMFileReader.h b/dom/base/FileReader.h similarity index 57% rename from dom/base/nsDOMFileReader.h rename to dom/base/FileReader.h index 0fa7080a67..418ef5c833 100644 --- a/dom/base/nsDOMFileReader.h +++ b/dom/base/FileReader.h @@ -4,70 +4,55 @@ * License, v. 2.0. If a copy of the MPL was not distributed with this * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ -#ifndef nsDOMFileReader_h__ -#define nsDOMFileReader_h__ +#ifndef mozilla_dom_FileReader_h +#define mozilla_dom_FileReader_h #include "mozilla/Attributes.h" -#include "nsISupportsUtils.h" -#include "nsString.h" -#include "nsWeakReference.h" -#include "nsIStreamListener.h" -#include "nsIInterfaceRequestor.h" -#include "nsJSUtils.h" -#include "nsTArray.h" -#include "prtime.h" -#include "nsITimer.h" -#include "nsIAsyncInputStream.h" +#include "mozilla/DOMEventTargetHelper.h" +#include "mozilla/dom/DOMError.h" -#include "nsIDOMFileReader.h" -#include "nsIDOMFileList.h" #include "nsCOMPtr.h" +#include "nsIAsyncInputStream.h" +#include "nsIStreamListener.h" +#include "nsISupportsUtils.h" +#include "nsIInterfaceRequestor.h" +#include "nsITimer.h" +#include "nsJSUtils.h" +#include "nsString.h" +#include "nsTArray.h" +#include "nsWeakReference.h" +#include "prtime.h" -#include "FileIOObject.h" +#define NS_PROGRESS_EVENT_INTERVAL 50 namespace mozilla { namespace dom { + class Blob; -} // namespace dom -} // namespace mozilla -class nsDOMFileReader final : public mozilla::dom::FileIOObject, - public nsIDOMFileReader, - public nsIInterfaceRequestor, - public nsSupportsWeakReference +extern const uint64_t kUnknownSize; + +class FileReader final : public DOMEventTargetHelper, + public nsIInterfaceRequestor, + public nsSupportsWeakReference, + public nsIInputStreamCallback, + public nsITimerCallback { - typedef mozilla::ErrorResult ErrorResult; - typedef mozilla::dom::GlobalObject GlobalObject; - typedef mozilla::dom::Blob Blob; - public: - nsDOMFileReader(); + explicit FileReader(nsPIDOMWindow* aWindow); NS_DECL_ISUPPORTS_INHERITED - NS_DECL_NSIDOMFILEREADER - - NS_REALLY_FORWARD_NSIDOMEVENTTARGET(mozilla::DOMEventTargetHelper) - - // nsIInterfaceRequestor + NS_DECL_NSITIMERCALLBACK + NS_DECL_NSIINPUTSTREAMCALLBACK NS_DECL_NSIINTERFACEREQUESTOR - // FileIOObject overrides - virtual void DoAbort(nsAString& aEvent) override; + NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_CLASS_INHERITED(FileReader, DOMEventTargetHelper) - virtual nsresult DoReadData(nsIAsyncInputStream* aStream, uint64_t aCount) override; - - virtual nsresult DoOnLoadEnd(nsresult aStatus, nsAString& aSuccessEvent, - nsAString& aTerminationEvent) override; - - nsPIDOMWindow* GetParentObject() const - { - return GetOwner(); - } virtual JSObject* WrapObject(JSContext* aCx, JS::Handle aGivenProto) override; // WebIDL - static already_AddRefed + static already_AddRefed Constructor(const GlobalObject& aGlobal, ErrorResult& aRv); void ReadAsArrayBuffer(JSContext* aCx, Blob& aBlob, ErrorResult& aRv) { @@ -84,23 +69,26 @@ public: ReadFileContent(aBlob, EmptyString(), FILE_AS_DATAURL, aRv); } - using FileIOObject::Abort; + void Abort(ErrorResult& aRv); - // Inherited ReadyState(). + uint16_t ReadyState() const + { + return static_cast(mReadyState); + } + + DOMError* GetError() const + { + return mError; + } void GetResult(JSContext* aCx, JS::MutableHandle aResult, ErrorResult& aRv); - using FileIOObject::GetError; - IMPL_EVENT_HANDLER(loadstart) - using FileIOObject::GetOnprogress; - using FileIOObject::SetOnprogress; + IMPL_EVENT_HANDLER(progress) IMPL_EVENT_HANDLER(load) - using FileIOObject::GetOnabort; - using FileIOObject::SetOnabort; - using FileIOObject::GetOnerror; - using FileIOObject::SetOnerror; + IMPL_EVENT_HANDLER(abort) + IMPL_EVENT_HANDLER(error) IMPL_EVENT_HANDLER(loadend) void ReadAsBinaryString(Blob& aBlob, ErrorResult& aRv) @@ -108,15 +96,15 @@ public: ReadFileContent(aBlob, EmptyString(), FILE_AS_BINARY, aRv); } +private: + virtual ~FileReader(); - nsresult Init(); - - NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_CLASS_INHERITED(nsDOMFileReader, - FileIOObject) - void RootResultArrayBuffer(); - -protected: - virtual ~nsDOMFileReader(); + // This must be in sync with dom/webidl/FileReader.webidl + enum eReadyState { + EMPTY = 0, + LOADING = 1, + DONE = 2 + }; enum eDataFormat { FILE_AS_ARRAYBUFFER, @@ -125,6 +113,8 @@ protected: FILE_AS_DATAURL }; + void RootResultArrayBuffer(); + void ReadFileContent(Blob& aBlob, const nsAString &aCharset, eDataFormat aDataFormat, ErrorResult& aRv); @@ -134,7 +124,20 @@ protected: nsresult GetAsDataURL(Blob *aBlob, const char *aFileData, uint32_t aDataLen, nsAString &aResult); - void FreeFileData() { + nsresult OnLoadEnd(nsresult aStatus); + + void StartProgressEventTimer(); + void ClearProgressEventTimer(); + void DispatchError(nsresult rv, nsAString& finalEvent); + nsresult DispatchProgressEvent(const nsAString& aType); + + nsresult DoReadData(nsIAsyncInputStream* aStream, uint64_t aCount); + + nsresult DoOnLoadEnd(nsresult aStatus, nsAString& aSuccessEvent, + nsAString& aTerminationEvent); + + void FreeFileData() + { moz_free(mFileData); mFileData = nullptr; mDataLen = 0; @@ -150,6 +153,22 @@ protected: nsString mResult; JS::Heap mResultArrayBuffer; + + nsCOMPtr mProgressNotifier; + bool mProgressEventWasDelayed; + bool mTimerIsActive; + + nsCOMPtr mAsyncStream; + + RefPtr mError; + + eReadyState mReadyState; + + uint64_t mTotal; + uint64_t mTransferred; }; -#endif +} // dom namespace +} // mozilla namespace + +#endif // mozilla_dom_FileReader_h diff --git a/dom/base/MultipartBlobImpl.cpp b/dom/base/MultipartBlobImpl.cpp index 4d2c309c2f..98671429ed 100644 --- a/dom/base/MultipartBlobImpl.cpp +++ b/dom/base/MultipartBlobImpl.cpp @@ -25,6 +25,37 @@ using namespace mozilla::dom; NS_IMPL_ISUPPORTS_INHERITED0(MultipartBlobImpl, BlobImpl) +/* static */ already_AddRefed +MultipartBlobImpl::Create(const nsTArray>& aBlobImpls, + const nsAString& aName, + const nsAString& aContentType, + ErrorResult& aRv) +{ + RefPtr blobImpl = + new MultipartBlobImpl(aBlobImpls, aName, aContentType); + blobImpl->SetLengthAndModifiedDate(aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + + return blobImpl.forget(); +} + +/* static */ already_AddRefed +MultipartBlobImpl::Create(const nsTArray>& aBlobImpls, + const nsAString& aContentType, + ErrorResult& aRv) +{ + RefPtr blobImpl = + new MultipartBlobImpl(aBlobImpls, aContentType); + blobImpl->SetLengthAndModifiedDate(aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + + return blobImpl.forget(); +} + void MultipartBlobImpl::GetInternalStream(nsIInputStream** aStream, ErrorResult& aRv) @@ -125,15 +156,19 @@ MultipartBlobImpl::CreateSlice(uint64_t aStart, uint64_t aLength, } // we can create our blob now - RefPtr impl = - new MultipartBlobImpl(blobImpls, aContentType); + RefPtr impl = Create(blobImpls, aContentType, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + return impl.forget(); } void -MultipartBlobImpl::InitializeBlob() +MultipartBlobImpl::InitializeBlob(ErrorResult& aRv) { - SetLengthAndModifiedDate(); + SetLengthAndModifiedDate(aRv); + NS_WARN_IF(aRv.Failed()); } void @@ -187,11 +222,12 @@ MultipartBlobImpl::InitializeBlob( mBlobImpls = blobSet.GetBlobImpls(); - SetLengthAndModifiedDate(); + SetLengthAndModifiedDate(aRv); + NS_WARN_IF(aRv.Failed()); } void -MultipartBlobImpl::SetLengthAndModifiedDate() +MultipartBlobImpl::SetLengthAndModifiedDate(ErrorResult& aRv) { MOZ_ASSERT(mLength == UINT64_MAX); MOZ_ASSERT(mLastModificationDate == INT64_MAX); @@ -208,16 +244,19 @@ MultipartBlobImpl::SetLengthAndModifiedDate() MOZ_ASSERT(!blob->IsDateUnknown()); #endif - ErrorResult error; - uint64_t subBlobLength = blob->GetSize(error); - MOZ_ALWAYS_TRUE(!error.Failed()); + uint64_t subBlobLength = blob->GetSize(aRv); + if (NS_WARN_IF(aRv.Failed())) { + return; + } MOZ_ASSERT(UINT64_MAX - subBlobLength >= totalLength); totalLength += subBlobLength; if (blob->IsFile()) { - int64_t partLastModified = blob->GetLastModified(error); - MOZ_ALWAYS_TRUE(!error.Failed()); + int64_t partLastModified = blob->GetLastModified(aRv); + if (NS_WARN_IF(aRv.Failed())) { + return; + } if (lastModified < partLastModified) { lastModified = partLastModified; @@ -314,7 +353,8 @@ MultipartBlobImpl::InitializeChromeFile(Blob& aBlob, blobSet.AppendBlobImpl(aBlob.Impl()); mBlobImpls = blobSet.GetBlobImpls(); - SetLengthAndModifiedDate(); + SetLengthAndModifiedDate(aRv); + NS_WARN_IF(aRv.Failed()); } void @@ -385,7 +425,8 @@ MultipartBlobImpl::InitializeChromeFile(nsPIDOMWindow* aWindow, blobSet.AppendBlobImpl(static_cast(blob.get())->Impl()); mBlobImpls = blobSet.GetBlobImpls(); - SetLengthAndModifiedDate(); + SetLengthAndModifiedDate(aRv); + NS_WARN_IF(aRv.Failed()); } void diff --git a/dom/base/MultipartBlobImpl.h b/dom/base/MultipartBlobImpl.h index 2a0778be28..2239a4936e 100644 --- a/dom/base/MultipartBlobImpl.h +++ b/dom/base/MultipartBlobImpl.h @@ -25,25 +25,17 @@ public: NS_DECL_ISUPPORTS_INHERITED // Create as a file - MultipartBlobImpl(const nsTArray>& aBlobImpls, - const nsAString& aName, - const nsAString& aContentType) - : BlobImplBase(aName, aContentType, UINT64_MAX), - mBlobImpls(aBlobImpls), - mIsFromNsIFile(false) - { - SetLengthAndModifiedDate(); - } + static already_AddRefed + Create(const nsTArray>& aBlobImpls, + const nsAString& aName, + const nsAString& aContentType, + ErrorResult& aRv); // Create as a blob - MultipartBlobImpl(const nsTArray>& aBlobImpls, - const nsAString& aContentType) - : BlobImplBase(aContentType, UINT64_MAX), - mBlobImpls(aBlobImpls), - mIsFromNsIFile(false) - { - SetLengthAndModifiedDate(); - } + static already_AddRefed + Create(const nsTArray>& aBlobImpls, + const nsAString& aContentType, + ErrorResult& aRv); // Create as a file to be later initialized explicit MultipartBlobImpl(const nsAString& aName) @@ -59,7 +51,7 @@ public: { } - void InitializeBlob(); + void InitializeBlob(ErrorResult& aRv); void InitializeBlob( JSContext* aCx, @@ -120,9 +112,26 @@ public: virtual bool MayBeClonedToOtherThreads() const override; protected: + MultipartBlobImpl(const nsTArray>& aBlobImpls, + const nsAString& aName, + const nsAString& aContentType) + : BlobImplBase(aName, aContentType, UINT64_MAX), + mBlobImpls(aBlobImpls), + mIsFromNsIFile(false) + { + } + + MultipartBlobImpl(const nsTArray>& aBlobImpls, + const nsAString& aContentType) + : BlobImplBase(aContentType, UINT64_MAX), + mBlobImpls(aBlobImpls), + mIsFromNsIFile(false) + { + } + virtual ~MultipartBlobImpl() {} - void SetLengthAndModifiedDate(); + void SetLengthAndModifiedDate(ErrorResult& aRv); nsTArray> mBlobImpls; bool mIsFromNsIFile; diff --git a/dom/base/ShadowRoot.cpp b/dom/base/ShadowRoot.cpp index 8c5a78d8cb..968602f6de 100644 --- a/dom/base/ShadowRoot.cpp +++ b/dom/base/ShadowRoot.cpp @@ -316,7 +316,7 @@ ShadowRoot::DistributeSingleNode(nsIContent* aContent) if (!isIndexFound) { // We have still not found an index in the insertion point, // thus it must be at the end. - MOZ_ASSERT(childIterator.Seek(aContent), + MOZ_ASSERT(childIterator.Seek(aContent, nullptr), "Trying to match a node that is not a candidate to be matched"); insertionPoint->AppendMatchedNode(aContent); } diff --git a/dom/base/StructuredCloneHolder.cpp b/dom/base/StructuredCloneHolder.cpp index 1c00f0191e..3b3659500f 100644 --- a/dom/base/StructuredCloneHolder.cpp +++ b/dom/base/StructuredCloneHolder.cpp @@ -556,7 +556,8 @@ namespace { // Recursive! already_AddRefed EnsureBlobForBackgroundManager(BlobImpl* aBlobImpl, - PBackgroundChild* aManager = nullptr) + PBackgroundChild* aManager, + ErrorResult& aRv) { MOZ_ASSERT(aBlobImpl); RefPtr blobImpl = aBlobImpl; @@ -605,7 +606,11 @@ EnsureBlobForBackgroundManager(BlobImpl* aBlobImpl, RefPtr& newSubBlobImpl = newSubBlobImpls[index]; - newSubBlobImpl = EnsureBlobForBackgroundManager(subBlobImpl, aManager); + newSubBlobImpl = EnsureBlobForBackgroundManager(subBlobImpl, aManager, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + MOZ_ASSERT(newSubBlobImpl); if (subBlobImpl != newSubBlobImpl) { @@ -621,9 +626,14 @@ EnsureBlobForBackgroundManager(BlobImpl* aBlobImpl, nsString name; blobImpl->GetName(name); - blobImpl = new MultipartBlobImpl(newSubBlobImpls, name, contentType); + blobImpl = MultipartBlobImpl::Create(newSubBlobImpls, name, + contentType, aRv); } else { - blobImpl = new MultipartBlobImpl(newSubBlobImpls, contentType); + blobImpl = MultipartBlobImpl::Create(newSubBlobImpls, contentType, aRv); + } + + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; } MOZ_ALWAYS_TRUE(NS_SUCCEEDED(blobImpl->SetMutable(false))); @@ -641,7 +651,13 @@ ReadBlob(JSContext* aCx, MOZ_ASSERT(aIndex < aHolder->BlobImpls().Length()); RefPtr blobImpl = aHolder->BlobImpls()[aIndex]; - blobImpl = EnsureBlobForBackgroundManager(blobImpl); + ErrorResult rv; + blobImpl = EnsureBlobForBackgroundManager(blobImpl, nullptr, rv); + if (NS_WARN_IF(rv.Failed())) { + rv.SuppressException(); + return nullptr; + } + MOZ_ASSERT(blobImpl); // RefPtr needs to go out of scope before toObjectOrNull() is @@ -669,7 +685,14 @@ WriteBlob(JSStructuredCloneWriter* aWriter, MOZ_ASSERT(aBlob); MOZ_ASSERT(aHolder); - RefPtr blobImpl = EnsureBlobForBackgroundManager(aBlob->Impl()); + ErrorResult rv; + RefPtr blobImpl = + EnsureBlobForBackgroundManager(aBlob->Impl(), nullptr, rv); + if (NS_WARN_IF(rv.Failed())) { + rv.SuppressException(); + return false; + } + MOZ_ASSERT(blobImpl); MOZ_ALWAYS_TRUE(NS_SUCCEEDED(blobImpl->SetMutable(false))); @@ -715,7 +738,13 @@ ReadFileList(JSContext* aCx, RefPtr blobImpl = aHolder->BlobImpls()[index]; MOZ_ASSERT(blobImpl->IsFile()); - blobImpl = EnsureBlobForBackgroundManager(blobImpl); + ErrorResult rv; + blobImpl = EnsureBlobForBackgroundManager(blobImpl, nullptr, rv); + if (NS_WARN_IF(rv.Failed())) { + rv.SuppressException(); + return nullptr; + } + MOZ_ASSERT(blobImpl); RefPtr file = File::Create(aHolder->ParentDuringRead(), blobImpl); @@ -754,14 +783,22 @@ WriteFileList(JSStructuredCloneWriter* aWriter, return false; } + ErrorResult rv; + nsTArray> blobImpls; + for (uint32_t i = 0; i < aFileList->Length(); ++i) { RefPtr blobImpl = - EnsureBlobForBackgroundManager(aFileList->Item(i)->Impl()); - MOZ_ASSERT(blobImpl); + EnsureBlobForBackgroundManager(aFileList->Item(i)->Impl(), nullptr, rv); + if (NS_WARN_IF(rv.Failed())) { + rv.SuppressException(); + return false; + } - aHolder->BlobImpls().AppendElement(blobImpl); + MOZ_ASSERT(blobImpl); + blobImpls.AppendElement(blobImpl); } + aHolder->BlobImpls().AppendElements(blobImpls); return true; } @@ -805,7 +842,12 @@ ReadFormData(JSContext* aCx, File::Create(aHolder->ParentDuringRead(), blobImpl); MOZ_ASSERT(file); - formData->Append(name, *file, thirdArg); + ErrorResult rv; + formData->Append(name, *file, thirdArg, rv); + if (NS_WARN_IF(rv.Failed())) { + return nullptr; + } + } else { MOZ_ASSERT(tag == 0); @@ -817,7 +859,11 @@ ReadFormData(JSContext* aCx, return nullptr; } - formData->Append(name, value); + ErrorResult rv; + formData->Append(name, value, rv); + if (NS_WARN_IF(rv.Failed())) { + return nullptr; + } } } @@ -866,16 +912,16 @@ WriteFormData(JSStructuredCloneWriter* aWriter, { } static bool - Write(const nsString& aName, bool isFile, const nsString& aValue, - File* aFile, void* aClosure) + Write(const nsString& aName, const OwningFileOrUSVString& aValue, + void* aClosure) { Closure* closure = static_cast(aClosure); if (!WriteString(closure->mWriter, aName)) { return false; } - if (isFile) { - BlobImpl* blobImpl = aFile->Impl(); + if (aValue.IsFile()) { + BlobImpl* blobImpl = aValue.GetAsFile()->Impl(); if (!JS_WriteUint32Pair(closure->mWriter, SCTAG_DOM_BLOB, closure->mHolder->BlobImpls().Length())) { return false; @@ -886,9 +932,10 @@ WriteFormData(JSStructuredCloneWriter* aWriter, } size_t charSize = sizeof(nsString::char_type); - if (!JS_WriteUint32Pair(closure->mWriter, 0, aValue.Length()) || - !JS_WriteBytes(closure->mWriter, aValue.get(), - aValue.Length() * charSize)) { + if (!JS_WriteUint32Pair(closure->mWriter, 0, + aValue.GetAsUSVString().Length()) || + !JS_WriteBytes(closure->mWriter, aValue.GetAsUSVString().get(), + aValue.GetAsUSVString().Length() * charSize)) { return false; } diff --git a/dom/base/WindowOrientationObserver.cpp b/dom/base/WindowOrientationObserver.cpp index d7ce628448..733a961adc 100644 --- a/dom/base/WindowOrientationObserver.cpp +++ b/dom/base/WindowOrientationObserver.cpp @@ -9,6 +9,8 @@ #include "nsGlobalWindow.h" #include "mozilla/Hal.h" +using namespace mozilla::dom; + /** * This class is used by nsGlobalWindow to implement window.orientation * and window.onorientationchange. This class is defined in its own file diff --git a/dom/base/moz.build b/dom/base/moz.build index b4e31153f0..a4b1a5bddc 100644 --- a/dom/base/moz.build +++ b/dom/base/moz.build @@ -16,7 +16,6 @@ XPIDL_SOURCES += [ 'nsIDOMDOMCursor.idl', 'nsIDOMDOMRequest.idl', 'nsIDOMFileList.idl', - 'nsIDOMFileReader.idl', 'nsIDOMFormData.idl', 'nsIDOMParser.idl', 'nsIDOMSerializer.idl', @@ -179,6 +178,7 @@ EXPORTS.mozilla.dom += [ 'EventSource.h', 'File.h', 'FileList.h', + 'FileReader.h', 'FragmentOrElement.h', 'FromParser.h', 'ImageEncoder.h', @@ -243,8 +243,8 @@ UNIFIED_SOURCES += [ 'Element.cpp', 'EventSource.cpp', 'File.cpp', - 'FileIOObject.cpp', 'FileList.cpp', + 'FileReader.cpp', 'FragmentOrElement.cpp', 'ImageEncoder.cpp', 'ImportManager.cpp', @@ -271,7 +271,6 @@ UNIFIED_SOURCES += [ 'nsDOMAttributeMap.cpp', 'nsDOMCaretPosition.cpp', 'nsDOMClassInfo.cpp', - 'nsDOMFileReader.cpp', 'nsDOMMutationObserver.cpp', 'nsDOMNavigationTiming.cpp', 'nsDOMScriptObjectFactory.cpp', diff --git a/dom/base/nsContentUtils.cpp b/dom/base/nsContentUtils.cpp index e2276ef643..3e608fd0ae 100644 --- a/dom/base/nsContentUtils.cpp +++ b/dom/base/nsContentUtils.cpp @@ -6616,22 +6616,6 @@ nsContentUtils::FindInternalContentViewer(const nsACString& aType, return nullptr; } -bool -nsContentUtils::GetContentSecurityPolicy(nsIContentSecurityPolicy** aCSP) -{ - NS_ASSERTION(NS_IsMainThread(), "Wrong thread!"); - - nsCOMPtr csp; - nsresult rv = SubjectPrincipal()->GetCsp(getter_AddRefs(csp)); - if (NS_FAILED(rv)) { - NS_ERROR("CSP: Failed to get CSP from principal."); - return false; - } - - csp.forget(aCSP); - return true; -} - // static bool nsContentUtils::IsPatternMatching(nsAString& aValue, nsAString& aPattern, diff --git a/dom/base/nsContentUtils.h b/dom/base/nsContentUtils.h index 0ce82d255c..70ea5510c0 100644 --- a/dom/base/nsContentUtils.h +++ b/dom/base/nsContentUtils.h @@ -505,11 +505,6 @@ public: return sSecurityManager; } - /** - * Get the ContentSecurityPolicy for a JS context. - **/ - static bool GetContentSecurityPolicy(nsIContentSecurityPolicy** aCSP); - // Returns the subject principal. Guaranteed to return non-null. May only // be called when nsContentUtils is initialized. static nsIPrincipal* SubjectPrincipal(); diff --git a/dom/base/nsDOMClassInfo.cpp b/dom/base/nsDOMClassInfo.cpp index 46191a06e8..f38329bb82 100644 --- a/dom/base/nsDOMClassInfo.cpp +++ b/dom/base/nsDOMClassInfo.cpp @@ -2282,8 +2282,7 @@ struct InterfaceShimEntry { // interface that has interface constants that sites might be getting off // of Ci. const InterfaceShimEntry kInterfaceShimMap[] = -{ { "nsIDOMFileReader", "FileReader" }, - { "nsIXMLHttpRequest", "XMLHttpRequest" }, +{ { "nsIXMLHttpRequest", "XMLHttpRequest" }, { "nsIDOMDOMException", "DOMException" }, { "nsIDOMNode", "Node" }, { "nsIDOMCSSPrimitiveValue", "CSSPrimitiveValue" }, diff --git a/dom/base/nsDOMMutationObserver.cpp b/dom/base/nsDOMMutationObserver.cpp index 27a8c04259..19f2125998 100644 --- a/dom/base/nsDOMMutationObserver.cpp +++ b/dom/base/nsDOMMutationObserver.cpp @@ -846,8 +846,7 @@ nsDOMMutationObserver::HandleMutation() } ClearPendingRecords(); - mozilla::ErrorResult rv; - mCallback->Call(this, mutations, *this, rv); + mCallback->Call(this, mutations, *this); } class AsyncMutationHandler : public nsRunnable diff --git a/dom/base/nsFormData.cpp b/dom/base/nsFormData.cpp index 6b16217b9e..ef93974699 100644 --- a/dom/base/nsFormData.cpp +++ b/dom/base/nsFormData.cpp @@ -25,7 +25,8 @@ namespace { // Implements steps 3 and 4 of the "create an entry" algorithm of FormData. already_AddRefed -CreateNewFileInstance(Blob& aBlob, const Optional& aFilename) +CreateNewFileInstance(Blob& aBlob, const Optional& aFilename, + ErrorResult& aRv) { // Step 3 "If value is a Blob object and not a File object, set value to // a new File object, representing the same bytes, whose name attribute value @@ -47,10 +48,15 @@ CreateNewFileInstance(Blob& aBlob, const Optional& aFilename) filename = NS_LITERAL_STRING("blob"); } - return aBlob.ToFile(filename); + RefPtr file = aBlob.ToFile(filename, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return nullptr; + } + + return file.forget(); } -} // anonymous namespace +} // namespace // ------------------------------------------------------------------------- // nsISupports @@ -61,7 +67,7 @@ NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(nsFormData) NS_IMPL_CYCLE_COLLECTION_UNLINK(mOwner) for (uint32_t i = 0, len = tmp->mFormData.Length(); i < len; ++i) { - ImplCycleCollectionUnlink(tmp->mFormData[i].fileValue); + ImplCycleCollectionUnlink(tmp->mFormData[i].value); } NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER @@ -71,8 +77,8 @@ NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(nsFormData) NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mOwner) for (uint32_t i = 0, len = tmp->mFormData.Length(); i < len; ++i) { - ImplCycleCollectionTraverse(cb,tmp->mFormData[i].fileValue, - "mFormData[i].fileValue", 0); + ImplCycleCollectionTraverse(cb, tmp->mFormData[i].value, + "mFormData[i].GetAsFile()", 0); } NS_IMPL_CYCLE_COLLECTION_TRAVERSE_SCRIPT_OBJECTS @@ -101,16 +107,22 @@ nsFormData::GetEncodedSubmission(nsIURI* aURI, } void -nsFormData::Append(const nsAString& aName, const nsAString& aValue) +nsFormData::Append(const nsAString& aName, const nsAString& aValue, + ErrorResult& aRv) { AddNameValuePair(aName, aValue); } void nsFormData::Append(const nsAString& aName, Blob& aBlob, - const Optional& aFilename) + const Optional& aFilename, + ErrorResult& aRv) { - RefPtr file = CreateNewFileInstance(aBlob, aFilename); + RefPtr file = CreateNewFileInstance(aBlob, aFilename, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return; + } + AddNameFilePair(aName, file); } @@ -126,24 +138,13 @@ nsFormData::Delete(const nsAString& aName) } } -void -nsFormData::ExtractValue(const FormDataTuple& aTuple, - OwningFileOrUSVString* aOutValue) -{ - if (aTuple.valueIsFile) { - aOutValue->SetAsFile() = aTuple.fileValue; - } else { - aOutValue->SetAsUSVString() = aTuple.stringValue; - } -} - void nsFormData::Get(const nsAString& aName, Nullable& aOutValue) { for (uint32_t i = 0; i < mFormData.Length(); ++i) { if (aName.Equals(mFormData[i].name)) { - ExtractValue(mFormData[i], &aOutValue.SetValue()); + aOutValue.SetValue() = mFormData[i].value; return; } } @@ -158,7 +159,7 @@ nsFormData::GetAll(const nsAString& aName, for (uint32_t i = 0; i < mFormData.Length(); ++i) { if (aName.Equals(mFormData[i].name)) { OwningFileOrUSVString* element = aValues.AppendElement(); - ExtractValue(mFormData[i], element); + *element = mFormData[i].value; } } } @@ -207,28 +208,54 @@ nsFormData::RemoveAllOthersAndGetFirstFormDataTuple(const nsAString& aName) void nsFormData::Set(const nsAString& aName, Blob& aBlob, - const Optional& aFilename) + const Optional& aFilename, + ErrorResult& aRv) { FormDataTuple* tuple = RemoveAllOthersAndGetFirstFormDataTuple(aName); if (tuple) { - RefPtr file = CreateNewFileInstance(aBlob, aFilename); + RefPtr file = CreateNewFileInstance(aBlob, aFilename, aRv); + if (NS_WARN_IF(aRv.Failed())) { + return; + } + SetNameFilePair(tuple, aName, file); } else { - Append(aName, aBlob, aFilename); + Append(aName, aBlob, aFilename, aRv); } } void -nsFormData::Set(const nsAString& aName, const nsAString& aValue) +nsFormData::Set(const nsAString& aName, const nsAString& aValue, + ErrorResult& aRv) { FormDataTuple* tuple = RemoveAllOthersAndGetFirstFormDataTuple(aName); if (tuple) { SetNameValuePair(tuple, aName, aValue); } else { - Append(aName, aValue); + Append(aName, aValue, aRv); } } +uint32_t +nsFormData::GetIterableLength() const +{ + return mFormData.Length(); +} + +const nsAString& +nsFormData::GetKeyAtIndex(uint32_t aIndex) const +{ + MOZ_ASSERT(aIndex < mFormData.Length()); + return mFormData[aIndex].name; +} + +const OwningFileOrUSVString& +nsFormData::GetValueAtIndex(uint32_t aIndex) const +{ + MOZ_ASSERT(aIndex < mFormData.Length()); + return mFormData[aIndex].value; +} + // ------------------------------------------------------------------------- // nsIDOMFormData @@ -252,7 +279,12 @@ nsFormData::Append(const nsAString& aName, nsIVariant* aValue) RefPtr blob = static_cast(domBlob.get()); if (domBlob) { Optional temp; - Append(aName, *blob, temp); + ErrorResult rv; + Append(aName, *blob, temp, rv); + if (NS_WARN_IF(rv.Failed())) { + return rv.StealNSResult(); + } + return NS_OK; } } @@ -265,7 +297,12 @@ nsFormData::Append(const nsAString& aName, nsIVariant* aValue) nsString valAsString; valAsString.Adopt(stringData, stringLen); - Append(aName, valAsString); + ErrorResult error; + Append(aName, valAsString, error); + if (NS_WARN_IF(error.Failed())) { + return error.StealNSResult(); + } + return NS_OK; } @@ -297,11 +334,13 @@ nsFormData::GetSendInfo(nsIInputStream** aBody, uint64_t* aContentLength, nsFSMultipartFormData fs(NS_LITERAL_CSTRING("UTF-8"), nullptr); for (uint32_t i = 0; i < mFormData.Length(); ++i) { - if (mFormData[i].valueIsFile) { - fs.AddNameFilePair(mFormData[i].name, mFormData[i].fileValue); - } - else { - fs.AddNameValuePair(mFormData[i].name, mFormData[i].stringValue); + if (mFormData[i].value.IsFile()) { + fs.AddNameFilePair(mFormData[i].name, mFormData[i].value.GetAsFile()); + } else if (mFormData[i].value.IsUSVString()) { + fs.AddNameValuePair(mFormData[i].name, + mFormData[i].value.GetAsUSVString()); + } else { + fs.AddNameFilePair(mFormData[i].name, nullptr); } } diff --git a/dom/base/nsFormData.h b/dom/base/nsFormData.h index 0ec96f5945..d6e06b05e5 100644 --- a/dom/base/nsFormData.h +++ b/dom/base/nsFormData.h @@ -37,13 +37,12 @@ private: typedef mozilla::dom::Blob Blob; typedef mozilla::dom::File File; + typedef mozilla::dom::OwningFileOrUSVString OwningFileOrUSVString; struct FormDataTuple { nsString name; - nsString stringValue; - RefPtr fileValue; - bool valueIsFile; + OwningFileOrUSVString value; }; // Returns the FormDataTuple to modify. This may be null, in which case @@ -57,8 +56,7 @@ private: { MOZ_ASSERT(aData); aData->name = aName; - aData->stringValue = aValue; - aData->valueIsFile = false; + aData->value.SetAsUSVString() = aValue; } void SetNameFilePair(FormDataTuple* aData, @@ -67,12 +65,11 @@ private: { MOZ_ASSERT(aData); aData->name = aName; - aData->fileValue = aFile; - aData->valueIsFile = true; + if (aFile) { + aData->value.SetAsFile() = aFile; + } } - void ExtractValue(const FormDataTuple& aTuple, - mozilla::dom::OwningFileOrUSVString* aOutValue); public: explicit nsFormData(nsISupports* aOwner = nullptr); @@ -96,16 +93,24 @@ public: Constructor(const mozilla::dom::GlobalObject& aGlobal, const mozilla::dom::Optional >& aFormElement, mozilla::ErrorResult& aRv); - void Append(const nsAString& aName, const nsAString& aValue); + void Append(const nsAString& aName, const nsAString& aValue, + mozilla::ErrorResult& aRv); void Append(const nsAString& aName, Blob& aBlob, - const mozilla::dom::Optional& aFilename); + const mozilla::dom::Optional& aFilename, + mozilla::ErrorResult& aRv); void Delete(const nsAString& aName); - void Get(const nsAString& aName, mozilla::dom::Nullable& aOutValue); - void GetAll(const nsAString& aName, nsTArray& aValues); + void Get(const nsAString& aName, mozilla::dom::Nullable& aOutValue); + void GetAll(const nsAString& aName, nsTArray& aValues); bool Has(const nsAString& aName); void Set(const nsAString& aName, Blob& aBlob, - const mozilla::dom::Optional& aFilename); - void Set(const nsAString& aName, const nsAString& aValue); + const mozilla::dom::Optional& aFilename, + mozilla::ErrorResult& aRv); + void Set(const nsAString& aName, const nsAString& aValue, + mozilla::ErrorResult& aRv); + + uint32_t GetIterableLength() const; + const nsAString& GetKeyAtIndex(uint32_t aIndex) const; + const OwningFileOrUSVString& GetValueAtIndex(uint32_t aIndex) const; // nsFormSubmission virtual nsresult GetEncodedSubmission(nsIURI* aURI, @@ -120,9 +125,9 @@ public: virtual nsresult AddNameFilePair(const nsAString& aName, File* aFile) override; - typedef bool (*FormDataEntryCallback)(const nsString& aName, bool aIsFile, - const nsString& aValue, - File* aFile, void* aClosure); + typedef bool (*FormDataEntryCallback)(const nsString& aName, + const OwningFileOrUSVString& aValue, + void* aClosure); uint32_t Length() const @@ -137,8 +142,7 @@ public: { for (uint32_t i = 0; i < mFormData.Length(); ++i) { FormDataTuple& tuple = mFormData[i]; - if (!aFunc(tuple.name, tuple.valueIsFile, tuple.stringValue, - tuple.fileValue, aClosure)) { + if (!aFunc(tuple.name, tuple.value, aClosure)) { return false; } } diff --git a/dom/base/nsIDOMFileReader.idl b/dom/base/nsIDOMFileReader.idl deleted file mode 100644 index 98b8559f87..0000000000 --- a/dom/base/nsIDOMFileReader.idl +++ /dev/null @@ -1,47 +0,0 @@ -/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ -/* This Source Code Form is subject to the terms of the Mozilla Public - * License, v. 2.0. If a copy of the MPL was not distributed with this - * file, You can obtain one at http://mozilla.org/MPL/2.0/. */ - -#include "nsIDOMEventTarget.idl" - -interface nsIDOMEventListener; -interface nsIDOMBlob; - -[builtinclass, uuid(2f34c719-bc14-4546-9fb9-2bab75e56e45)] -interface nsIDOMFileReader : nsIDOMEventTarget -{ - [implicit_jscontext] - void readAsArrayBuffer(in nsIDOMBlob filedata); - void readAsBinaryString(in nsIDOMBlob filedata); - void readAsText(in nsIDOMBlob filedata, [optional] in DOMString encoding); - void readAsDataURL(in nsIDOMBlob file); - - void abort(); - - const unsigned short EMPTY = 0; - const unsigned short LOADING = 1; - const unsigned short DONE = 2; - readonly attribute unsigned short readyState; - - [implicit_jscontext] - readonly attribute jsval result; - - // This is a DOMError - readonly attribute nsISupports error; - - [implicit_jscontext] attribute jsval onloadstart; - [implicit_jscontext] attribute jsval onprogress; - [implicit_jscontext] attribute jsval onload; - [implicit_jscontext] attribute jsval onabort; - [implicit_jscontext] attribute jsval onerror; - [implicit_jscontext] attribute jsval onloadend; -}; - -%{ C++ -#define NS_FILEREADER_CID \ -{0x06aa7c21, 0xfe05, 0x4cf2, \ -{0xb1, 0xc4, 0x0c, 0x71, 0x26, 0xa4, 0xf7, 0x13}} -#define NS_FILEREADER_CONTRACTID \ -"@mozilla.org/files/filereader;1" -%} diff --git a/dom/base/nsXMLHttpRequest.cpp b/dom/base/nsXMLHttpRequest.cpp index f96364a921..d18d161fe1 100644 --- a/dom/base/nsXMLHttpRequest.cpp +++ b/dom/base/nsXMLHttpRequest.cpp @@ -795,7 +795,7 @@ nsXMLHttpRequest::CreateResponseParsedJSON(JSContext* aCx) } void -nsXMLHttpRequest::CreatePartialBlob() +nsXMLHttpRequest::CreatePartialBlob(ErrorResult& aRv) { if (mDOMBlob) { // Use progress info to determine whether load is complete, but use @@ -804,9 +804,8 @@ nsXMLHttpRequest::CreatePartialBlob() if (mLoadTotal == mLoadTransferred) { mResponseBlob = mDOMBlob; } else { - ErrorResult rv; mResponseBlob = mDOMBlob->CreateSlice(0, mDataAvailable, - EmptyString(), rv); + EmptyString(), aRv); } return; } @@ -821,7 +820,7 @@ nsXMLHttpRequest::CreatePartialBlob() mChannel->GetContentType(contentType); } - mResponseBlob = mBlobSet->GetBlobInternal(GetOwner(), contentType); + mResponseBlob = mBlobSet->GetBlobInternal(GetOwner(), contentType, aRv); } NS_IMETHODIMP nsXMLHttpRequest::GetResponseType(nsAString& aResponseType) @@ -1015,7 +1014,7 @@ nsXMLHttpRequest::GetResponse(JSContext* aCx, } if (!mResponseBlob) { - CreatePartialBlob(); + CreatePartialBlob(aRv); } } @@ -1687,8 +1686,10 @@ nsXMLHttpRequest::Open(const nsACString& inMethod, const nsACString& url, nsILoadInfo::SEC_FORCE_INHERIT_PRINCIPAL; } - // If we have the document, use it - if (doc) { + // If we have the document, use it. Unfortunately, for dedicated workers + // 'doc' ends up being the parent document, which is not the document + // that we want to use. So make sure to avoid using 'doc' in that situation. + if (doc && doc->NodePrincipal() == mPrincipal) { rv = NS_NewChannel(getter_AddRefs(mChannel), uri, doc, @@ -2221,8 +2222,14 @@ nsXMLHttpRequest::OnStopRequest(nsIRequest *request, nsISupports *ctxt, nsresult // Also, no-store response cannot be written in persistent cache. nsAutoCString contentType; mChannel->GetContentType(contentType); - mResponseBlob = mBlobSet->GetBlobInternal(GetOwner(), contentType); + + ErrorResult rv; + mResponseBlob = mBlobSet->GetBlobInternal(GetOwner(), contentType, rv); mBlobSet = nullptr; + + if (NS_WARN_IF(rv.Failed())) { + return rv.StealNSResult(); + } } NS_ASSERTION(mResponseBody.IsEmpty(), "mResponseBody should be empty"); NS_ASSERTION(mResponseText.IsEmpty(), "mResponseText should be empty"); diff --git a/dom/base/nsXMLHttpRequest.h b/dom/base/nsXMLHttpRequest.h index 6f6a8179d8..373ba15ac0 100644 --- a/dom/base/nsXMLHttpRequest.h +++ b/dom/base/nsXMLHttpRequest.h @@ -605,7 +605,7 @@ protected: uint32_t count, uint32_t *writeCount); nsresult CreateResponseParsedJSON(JSContext* aCx); - void CreatePartialBlob(); + void CreatePartialBlob(ErrorResult& aRv); bool CreateDOMBlob(nsIRequest *request); // Change the state of the object with this. The broadcast argument // determines if the onreadystatechange listener should be called. diff --git a/dom/base/test/file_bug1198095.js b/dom/base/test/file_bug1198095.js new file mode 100644 index 0000000000..d85e474019 --- /dev/null +++ b/dom/base/test/file_bug1198095.js @@ -0,0 +1,26 @@ +var { classes: Cc, interfaces: Ci, utils: Cu } = Components; +Cu.importGlobalProperties(["File"]); + +function createFileWithData(message) { + var dirSvc = Cc["@mozilla.org/file/directory_service;1"].getService(Ci.nsIProperties); + var testFile = dirSvc.get("ProfD", Ci.nsIFile); + testFile.append("fileAPItestfileBug1198095"); + + var outStream = Cc["@mozilla.org/network/file-output-stream;1"].createInstance(Ci.nsIFileOutputStream); + outStream.init(testFile, 0x02 | 0x08 | 0x20, // write, create, truncate + 0666, 0); + + outStream.write(message, message.length); + outStream.close(); + + var domFile = new File(testFile); + return domFile; +} + +addMessageListener("file.open", function (message) { + sendAsyncMessage("file.opened", createFileWithData(message)); +}); + +addMessageListener("file.modify", function (message) { + sendAsyncMessage("file.modified", createFileWithData(message)); +}); diff --git a/dom/base/test/mochitest.ini b/dom/base/test/mochitest.ini index 7c294409e8..4b0b245cd8 100644 --- a/dom/base/test/mochitest.ini +++ b/dom/base/test/mochitest.ini @@ -257,6 +257,7 @@ support-files = referrer_change_server.sjs file_change_policy_redirect.html empty_worker.js + file_bug1198095.js [test_anonymousContent_api.html] [test_anonymousContent_append_after_reflow.html] @@ -846,3 +847,4 @@ skip-if = e10s || os != 'linux' || buildapp != 'browser' skip-if = buildapp == 'b2g' #no ssl support [test_document.all_iteration.html] [test_performance_translate.html] +[test_bug1198095.html] diff --git a/dom/base/test/test_bug1198095.html b/dom/base/test/test_bug1198095.html new file mode 100644 index 0000000000..dfdfa017f1 --- /dev/null +++ b/dom/base/test/test_bug1198095.html @@ -0,0 +1,77 @@ + + + + + Test for Bug 1198095 + + + + + +Mozilla Bug 1198095 + +
+
+
+ diff --git a/dom/base/test/test_ipc_messagemanager_blob.html b/dom/base/test/test_ipc_messagemanager_blob.html index c8872c93ee..fa8967888e 100644 --- a/dom/base/test/test_ipc_messagemanager_blob.html +++ b/dom/base/test/test_ipc_messagemanager_blob.html @@ -26,9 +26,7 @@ return; } - let reader = - Components.classes["@mozilla.org/files/filereader;1"] - .createInstance(Components.interfaces.nsIDOMFileReader); + let reader = new FileReader(); reader.addEventListener("load", function() { let response = reader.result == "this is a great success!" ? message.json : diff --git a/dom/bindings/Bindings.conf b/dom/bindings/Bindings.conf index 28ef7d1fd8..7c3a862f7b 100644 --- a/dom/bindings/Bindings.conf +++ b/dom/bindings/Bindings.conf @@ -513,7 +513,6 @@ DOMInterfaces = { }, 'FileReader': { - 'nativeType': 'nsDOMFileReader', 'implicitJSContext': [ 'readAsArrayBuffer' ], }, diff --git a/dom/bindings/Codegen.py b/dom/bindings/Codegen.py index ccf7ecf3ea..edd45d9326 100644 --- a/dom/bindings/Codegen.py +++ b/dom/bindings/Codegen.py @@ -14542,16 +14542,33 @@ class CGCallback(CGClass): assert args[0].name == "cx" and args[0].argType == "JSContext*" assert args[1].name == "aThisVal" and args[1].argType == "JS::Handle" args = args[2:] + + # Now remember which index the ErrorResult argument is at; + # we'll need this below. + assert args[-1].name == "aRv" and args[-1].argType == "ErrorResult&" + rvIndex = len(args) - 1 + assert rvIndex >= 0 + # Record the names of all the arguments, so we can use them when we call # the private method. argnames = [arg.name for arg in args] argnamesWithThis = ["s.GetContext()", "thisValJS"] + argnames argnamesWithoutThis = ["s.GetContext()", "JS::UndefinedHandleValue"] + argnames # Now that we've recorded the argnames for our call to our private - # method, insert our optional arguments for the execution reason and for - # deciding whether the CallSetup should re-throw exceptions on aRv. + # method, insert our optional argument for the execution reason. args.append(Argument("const char*", "aExecutionReason", "nullptr")) + + # Make copies of the arg list for the two "without rv" overloads. Note + # that those don't need aExceptionHandling or aCompartment arguments + # because those would make not sense anyway: the only sane thing to do + # with exceptions in the "without rv" cases is to report them. + argsWithoutRv = list(args) + argsWithoutRv.pop(rvIndex) + argsWithoutThisAndRv = list(argsWithoutRv) + + # Add the potional argument for deciding whether the CallSetup should + # re-throw exceptions on aRv. args.append(Argument("ExceptionHandling", "aExceptionHandling", "eReportExceptions")) # And the argument for communicating when exceptions should really be @@ -14563,6 +14580,22 @@ class CGCallback(CGClass): # And now insert our template argument. argsWithoutThis = list(args) args.insert(0, Argument("const T&", "thisVal")) + argsWithoutRv.insert(0, Argument("const T&", "thisVal")) + + argnamesWithoutThisAndRv = [arg.name for arg in argsWithoutThisAndRv] + argnamesWithoutThisAndRv.insert(rvIndex, "rv"); + # If we just leave things like that, and have no actual arguments in the + # IDL, we will end up trying to call the templated "without rv" overload + # with "rv" as the thisVal. That's no good. So explicitly append the + # aExceptionHandling and aCompartment values we need to end up matching + # the signature of our non-templated "with rv" overload. + argnamesWithoutThisAndRv.extend(["eReportExceptions", "nullptr"]) + + argnamesWithoutRv = [arg.name for arg in argsWithoutRv] + # Note that we need to insert at rvIndex + 1, since we inserted a + # thisVal arg at the start. + argnamesWithoutRv.insert(rvIndex + 1, "rv") + errorReturn = method.getDefaultRetval() setupCall = fill( @@ -14602,6 +14635,20 @@ class CGCallback(CGClass): errorReturn=errorReturn, methodName=method.name, callArgs=", ".join(argnamesWithoutThis)) + bodyWithThisWithoutRv = fill( + """ + IgnoredErrorResult rv; + return ${methodName}(${callArgs}); + """, + methodName=method.name, + callArgs=", ".join(argnamesWithoutRv)) + bodyWithoutThisAndRv = fill( + """ + IgnoredErrorResult rv; + return ${methodName}(${callArgs}); + """, + methodName=method.name, + callArgs=", ".join(argnamesWithoutThisAndRv)) return [ClassMethod(method.name, method.returnType, args, bodyInHeader=True, @@ -14610,6 +14657,13 @@ class CGCallback(CGClass): ClassMethod(method.name, method.returnType, argsWithoutThis, bodyInHeader=True, body=bodyWithoutThis), + ClassMethod(method.name, method.returnType, argsWithoutRv, + bodyInHeader=True, + templateArgs=["typename T"], + body=bodyWithThisWithoutRv), + ClassMethod(method.name, method.returnType, argsWithoutThisAndRv, + bodyInHeader=True, + body=bodyWithoutThisAndRv), method] def deps(self): diff --git a/dom/bindings/ErrorResult.h b/dom/bindings/ErrorResult.h index 6e24b93298..2eba6d4875 100644 --- a/dom/bindings/ErrorResult.h +++ b/dom/bindings/ErrorResult.h @@ -398,6 +398,16 @@ private: void operator=(const ErrorResult&) = delete; }; +// A class for use when an ErrorResult should just automatically be ignored. +class IgnoredErrorResult : public ErrorResult +{ +public: + ~IgnoredErrorResult() + { + SuppressException(); + } +}; + /****************************************************************************** ** Macros for checking results ******************************************************************************/ diff --git a/dom/fetch/FetchUtil.cpp b/dom/fetch/FetchUtil.cpp index 94f4dd99c5..943e63707b 100644 --- a/dom/fetch/FetchUtil.cpp +++ b/dom/fetch/FetchUtil.cpp @@ -203,7 +203,9 @@ public: bool URLParamsIterator(const nsString& aName, const nsString& aValue) override { - mFormData->Append(aName, aValue); + ErrorResult rv; + mFormData->Append(aName, aValue, rv); + MOZ_ASSERT(!rv.Failed()); return true; } @@ -392,7 +394,9 @@ private: NS_ConvertUTF8toUTF16 name(mName); if (mFilename.IsVoid()) { - mFormData->Append(name, NS_ConvertUTF8toUTF16(body)); + ErrorResult rv; + mFormData->Append(name, NS_ConvertUTF8toUTF16(body), rv); + MOZ_ASSERT(!rv.Failed()); } else { // Unfortunately we've to copy the data first since all our strings are // going to free it. We also need fallible alloc, so we can't just use @@ -417,7 +421,11 @@ private: NS_ConvertUTF8toUTF16(mFilename), NS_ConvertUTF8toUTF16(mContentType), /* aLastModifiedDate */ 0); Optional dummy; - mFormData->Append(name, *file, dummy); + ErrorResult rv; + mFormData->Append(name, *file, dummy, rv); + if (NS_WARN_IF(rv.Failed())) { + return false; + } } return true; diff --git a/dom/geolocation/nsGeolocation.cpp b/dom/geolocation/nsGeolocation.cpp index 60c6c1e28c..781c2f9c10 100644 --- a/dom/geolocation/nsGeolocation.cpp +++ b/dom/geolocation/nsGeolocation.cpp @@ -330,8 +330,7 @@ PositionError::NotifyCallback(const GeoPositionErrorCallback& aCallback) PositionErrorCallback* callback = aCallback.GetWebIDLCallback(); if (callback) { - ErrorResult err; - callback->Call(*this, err); + callback->Call(*this); } } else { nsIDOMGeoPositionErrorCallback* callback = aCallback.GetXPCOMCallback(); @@ -654,11 +653,10 @@ nsGeolocationRequest::SendLocation(nsIDOMGeoPosition* aPosition) nsAutoMicroTask mt; if (mCallback.HasWebIDLCallback()) { - ErrorResult err; PositionCallback* callback = mCallback.GetWebIDLCallback(); MOZ_ASSERT(callback); - callback->Call(*wrapped, err); + callback->Call(*wrapped); } else { nsIDOMGeoPositionCallback* callback = mCallback.GetXPCOMCallback(); diff --git a/dom/html/test/file_iframe_sandbox_g_if1.html b/dom/html/test/file_iframe_sandbox_g_if1.html index 90ca6074ac..9a985faf97 100644 --- a/dom/html/test/file_iframe_sandbox_g_if1.html +++ b/dom/html/test/file_iframe_sandbox_g_if1.html @@ -42,14 +42,9 @@ // test loading with relative url - this should fail since we are // sandboxed and have a null principal - try { - var worker_js = new Worker('file_iframe_sandbox_worker.js'); - } catch (e) { - ok(e.name === "SecurityError", "a worker in a sandboxed document should throw when loading from a relative URI"); - } - + var worker_js = new Worker('file_iframe_sandbox_worker.js'); worker_js.onerror = function(error) { - ok(false, "a worker in a sandboxed document should not tell the load error via error event"); + ok(true, "a worker in a sandboxed document should tell the load error via error event"); } worker_js.addEventListener('message', function(event) { diff --git a/dom/html/test/formData_test.js b/dom/html/test/formData_test.js index 613dc98238..4194f12332 100644 --- a/dom/html/test/formData_test.js +++ b/dom/html/test/formData_test.js @@ -105,6 +105,56 @@ function testFilename() { is(f.get("file3").name, "", "File's filename is returned even if empty."); } +function testIterable() { + var fd = new FormData(); + fd.set('1','2'); + fd.set('2','4'); + fd.set('3','6'); + fd.set('4','8'); + fd.set('5','10'); + + var key_iter = fd.keys(); + var value_iter = fd.values(); + var entries_iter = fd.entries(); + for (var i = 0; i < 5; ++i) { + var v = i + 1; + var key = key_iter.next(); + var value = value_iter.next(); + var entry = entries_iter.next(); + is(key.value, v.toString(), "Correct Key iterator: " + v.toString()); + ok(!key.done, "key.done is false"); + is(value.value, (v * 2).toString(), "Correct Value iterator: " + (v * 2).toString()); + ok(!value.done, "value.done is false"); + is(entry.value[0], v.toString(), "Correct Entry 0 iterator: " + v.toString()); + is(entry.value[1], (v * 2).toString(), "Correct Entry 1 iterator: " + (v * 2).toString()); + ok(!entry.done, "entry.done is false"); + } + + var last = key_iter.next(); + ok(last.done, "Nothing more to read."); + is(last.value, undefined, "Undefined is the last key"); + + last = value_iter.next(); + ok(last.done, "Nothing more to read."); + is(last.value, undefined, "Undefined is the last value"); + + last = entries_iter.next(); + ok(last.done, "Nothing more to read."); + + key_iter = fd.keys(); + key_iter.next(); + key_iter.next(); + fd.delete('1'); + fd.delete('2'); + fd.delete('3'); + fd.delete('4'); + fd.delete('5'); + + last = key_iter.next(); + ok(last.done, "Nothing more to read."); + is(last.value, undefined, "Undefined is the last key"); +} + function testSend(doneCb) { var xhr = new XMLHttpRequest(); xhr.open("POST", "form_submit_server.sjs"); @@ -161,6 +211,7 @@ function runTest(doneCb) { testSet(); testIterate(); testFilename(); + testIterable(); // Finally, send an XHR and verify the response matches. testSend(doneCb); } diff --git a/dom/indexedDB/test/test_message_manager_ipc.html b/dom/indexedDB/test/test_message_manager_ipc.html index 5a45b90a00..7fae1402ea 100644 --- a/dom/indexedDB/test/test_message_manager_ipc.html +++ b/dom/indexedDB/test/test_message_manager_ipc.html @@ -50,11 +50,6 @@ function childFrameScript() { sendAsyncMessage(mmName, { op: "done", result: result }); } - function createFileReader() { - return Cc["@mozilla.org/files/filereader;1"] - .createInstance(Ci.nsIDOMFileReader); - } - function grabAndContinue(arg) { testGenerator.send(arg); } @@ -78,7 +73,7 @@ function childFrameScript() { info("Reading blob"); - let reader = createFileReader(); + let reader = new FileReader(); reader.addEventListener("load", grabAndContinue); reader.readAsText(blob); @@ -94,7 +89,7 @@ function childFrameScript() { info("Reading slice"); - reader = createFileReader(); + reader = new FileReader(); reader.addEventListener("load", grabAndContinue); reader.readAsText(slice); @@ -155,7 +150,7 @@ function childFrameScript() { info("Reading blob"); - reader = createFileReader(); + reader = new FileReader(); reader.addEventListener("load", grabAndContinue); reader.readAsText(blob); @@ -190,7 +185,7 @@ function childFrameScript() { info("Reading Slice"); - reader = createFileReader(); + reader = new FileReader(); reader.addEventListener("load", grabAndContinue); reader.readAsText(slice); diff --git a/dom/indexedDB/test/unit/test_blob_file_backed.js b/dom/indexedDB/test/unit/test_blob_file_backed.js index 48857cecd1..40f49eeb13 100644 --- a/dom/indexedDB/test/unit/test_blob_file_backed.js +++ b/dom/indexedDB/test/unit/test_blob_file_backed.js @@ -3,14 +3,9 @@ * http://creativecommons.org/publicdomain/zero/1.0/ */ -let disableWorkerTest = "This test uses SpecialPowers"; +var disableWorkerTest = "This test uses SpecialPowers"; -let testGenerator = testSteps(); - -function createFileReader() { - return SpecialPowers.Cc["@mozilla.org/files/filereader;1"] - .createInstance(SpecialPowers.Ci.nsIDOMFileReader); -} +var testGenerator = testSteps(); function testSteps() { @@ -48,7 +43,7 @@ function testSteps() is(file.size, fileData.length, "Correct size"); is(file.type, fileType, "Correct type"); - let fileReader = createFileReader(); + let fileReader = new FileReader(); fileReader.onload = grabEventAndContinueHandler; fileReader.readAsText(file); @@ -85,7 +80,7 @@ function testSteps() is(file.size, fileData.length, "Correct size"); is(file.type, fileType, "Correct type"); - fileReader = createFileReader(); + fileReader = new FileReader(); fileReader.onload = grabEventAndContinueHandler; fileReader.readAsText(file); diff --git a/dom/indexedDB/test/unit/xpcshell-head-parent-process.js b/dom/indexedDB/test/unit/xpcshell-head-parent-process.js index bd1c0b0f7a..0a72f8511d 100644 --- a/dom/indexedDB/test/unit/xpcshell-head-parent-process.js +++ b/dom/indexedDB/test/unit/xpcshell-head-parent-process.js @@ -52,7 +52,7 @@ if (!this.runTest) { enableExperimental(); } - Cu.importGlobalProperties(["indexedDB", "Blob", "File"]); + Cu.importGlobalProperties(["indexedDB", "Blob", "File", "FileReader"]); do_test_pending(); testGenerator.next(); @@ -342,12 +342,6 @@ function getFile(name, type, str) return new File([str], name, {type: type}); } -function getFileReader() -{ - return SpecialPowers.Cc["@mozilla.org/files/filereader;1"] - .createInstance(SpecialPowers.Ci.nsIDOMFileReader); -} - function compareBuffers(buffer1, buffer2) { if (buffer1.byteLength != buffer2.byteLength) { @@ -391,7 +385,7 @@ function verifyBlob(blob1, blob2) } if (!buffer2) { - let reader = getFileReader(); + let reader = new FileReader(); reader.readAsArrayBuffer(blob2); reader.onload = function(event) { buffer2 = event.target.result; @@ -403,7 +397,7 @@ function verifyBlob(blob1, blob2) } } - let reader = getFileReader(); + let reader = new FileReader(); reader.readAsArrayBuffer(blob1); reader.onload = function(event) { buffer1 = event.target.result; diff --git a/dom/ipc/Blob.cpp b/dom/ipc/Blob.cpp index 8c5c89bd56..e2ae557277 100644 --- a/dom/ipc/Blob.cpp +++ b/dom/ipc/Blob.cpp @@ -904,13 +904,17 @@ CreateBlobImpl(const nsTArray& aBlobDatas, MOZ_ASSERT(!isMutable); } + ErrorResult rv; RefPtr blobImpl; if (!hasRecursed && aMetadata.IsFile()) { - blobImpl = - new MultipartBlobImpl(blobImpls, aMetadata.mName, aMetadata.mContentType); + blobImpl = MultipartBlobImpl::Create(blobImpls, aMetadata.mName, + aMetadata.mContentType, rv); } else { - blobImpl = - new MultipartBlobImpl(blobImpls, aMetadata.mContentType); + blobImpl = MultipartBlobImpl::Create(blobImpls, aMetadata.mContentType, rv); + } + + if (NS_WARN_IF(rv.Failed())) { + return nullptr; } MOZ_ALWAYS_TRUE(NS_SUCCEEDED(blobImpl->SetMutable(false))); diff --git a/dom/ipc/tests/test_blob_sliced_from_parent_process.html b/dom/ipc/tests/test_blob_sliced_from_parent_process.html index 00f8b13110..8a00ed5083 100644 --- a/dom/ipc/tests/test_blob_sliced_from_parent_process.html +++ b/dom/ipc/tests/test_blob_sliced_from_parent_process.html @@ -44,11 +44,6 @@ function childFrameScript() { sendAsyncMessage(messageName, { op: "done", result: result }); } - function createFileReader() { - return Cc["@mozilla.org/files/filereader;1"] - .createInstance(Ci.nsIDOMFileReader); - } - function grabAndContinue(arg) { testGenerator.send(arg); } @@ -65,7 +60,7 @@ function childFrameScript() { info("Reading blob"); - let reader = createFileReader(); + let reader = new FileReader(); reader.addEventListener("load", grabAndContinue); reader.readAsText(blob); @@ -84,7 +79,7 @@ function childFrameScript() { info("Reading slice"); - reader = createFileReader(); + reader = new FileReader(); reader.addEventListener("load", grabAndContinue); reader.readAsText(slice); diff --git a/dom/media/moz.build b/dom/media/moz.build index 1eb33dc6af..9e5acc1887 100644 --- a/dom/media/moz.build +++ b/dom/media/moz.build @@ -301,6 +301,11 @@ if CONFIG['MOZ_WEBRTC']: DEFINES['MOZILLA_INTERNAL_API'] = True +if CONFIG['OS_TARGET'] == 'WINNT': + DEFINES['WEBRTC_WIN'] = True +else: + DEFINES['WEBRTC_POSIX'] = True + if CONFIG['MOZ_OMX_DECODER']: DEFINES['MOZ_OMX_DECODER'] = True diff --git a/dom/media/omx/OMXCodecWrapper.cpp b/dom/media/omx/OMXCodecWrapper.cpp index d3df99ec41..df43dd8a42 100644 --- a/dom/media/omx/OMXCodecWrapper.cpp +++ b/dom/media/omx/OMXCodecWrapper.cpp @@ -45,12 +45,18 @@ enum BufferState bool OMXCodecReservation::ReserveOMXCodec() { - if (mClient) { - // Already tried reservation. - return false; + if (!mClient) { + mClient = new mozilla::MediaSystemResourceClient(mType); + } else { + if (mOwned) { + //CODEC_ERROR("OMX Reservation: (%d) already owned", (int) mType); + return true; + } + //CODEC_ERROR("OMX Reservation: (%d) already NOT owned", (int) mType); } - mClient = new mozilla::MediaSystemResourceClient(mType); - return mClient->AcquireSyncNoWait(); // don't wait if resrouce is not available + mOwned = mClient->AcquireSyncNoWait(); // don't wait if resource is not available + //CODEC_ERROR("OMX Reservation: (%d) Acquire was %s", (int) mType, mOwned ? "Successful" : "Failed"); + return mOwned; } void @@ -59,7 +65,12 @@ OMXCodecReservation::ReleaseOMXCodec() if (!mClient) { return; } - mClient->ReleaseResource(); + //CODEC_ERROR("OMX Reservation: Releasing resource: (%d) %s", (int) mType, mOwned ? "Owned" : "Not owned"); + if (mOwned) { + mClient->ReleaseResource(); + mClient = nullptr; + mOwned = false; + } } OMXAudioEncoder* diff --git a/dom/media/omx/OMXCodecWrapper.h b/dom/media/omx/OMXCodecWrapper.h index aee9157ad4..fe8a2d994d 100644 --- a/dom/media/omx/OMXCodecWrapper.h +++ b/dom/media/omx/OMXCodecWrapper.h @@ -26,7 +26,7 @@ namespace android { class OMXCodecReservation : public RefBase { public: - OMXCodecReservation(bool aEncoder) + OMXCodecReservation(bool aEncoder) : mOwned(false) { mType = aEncoder ? mozilla::MediaSystemResourceType::VIDEO_ENCODER : mozilla::MediaSystemResourceType::VIDEO_DECODER; @@ -45,6 +45,7 @@ public: private: mozilla::MediaSystemResourceType mType; + bool mOwned; // We already own this resource RefPtr mClient; }; diff --git a/dom/media/systemservices/CamerasChild.cpp b/dom/media/systemservices/CamerasChild.cpp index d4adf6f024..094887d21e 100644 --- a/dom/media/systemservices/CamerasChild.cpp +++ b/dom/media/systemservices/CamerasChild.cpp @@ -647,7 +647,7 @@ bool CamerasChild::RecvDeliverFrame(const int& capEngine, const int& capId, mozilla::ipc::Shmem&& shmem, - const int& size, + const size_t& size, const uint32_t& time_stamp, const int64_t& ntp_time, const int64_t& render_time) diff --git a/dom/media/systemservices/CamerasChild.h b/dom/media/systemservices/CamerasChild.h index 0ead231447..16202adc7b 100644 --- a/dom/media/systemservices/CamerasChild.h +++ b/dom/media/systemservices/CamerasChild.h @@ -85,7 +85,7 @@ public: // IPC messages recevied, received on the PBackground thread // these are the actual callbacks with data virtual bool RecvDeliverFrame(const int&, const int&, mozilla::ipc::Shmem&&, - const int&, const uint32_t&, const int64_t&, + const size_t&, const uint32_t&, const int64_t&, const int64_t&) override; virtual bool RecvFrameSizeChange(const int&, const int&, const int& w, const int& h) override; @@ -159,9 +159,9 @@ private: Mutex mRequestMutex; // Hold to wait for an async response to our calls Monitor mReplyMonitor; - // Async resposne valid? + // Async response valid? bool mReceivedReply; - // Aynsc reponses data contents; + // Async responses data contents; bool mReplySuccess; int mReplyInteger; webrtc::CaptureCapability mReplyCapability; diff --git a/dom/media/systemservices/CamerasParent.cpp b/dom/media/systemservices/CamerasParent.cpp index 6459ccab98..9bbc112ad3 100644 --- a/dom/media/systemservices/CamerasParent.cpp +++ b/dom/media/systemservices/CamerasParent.cpp @@ -16,6 +16,8 @@ #include "nsThreadUtils.h" #include "nsXPCOM.h" +#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" + #undef LOG #undef LOG_ENABLED mozilla::LazyLogModule gCamerasParentLog("CamerasParent"); @@ -90,7 +92,7 @@ public: int cap_id, ShmemBuffer buffer, unsigned char* altbuffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time) @@ -136,7 +138,7 @@ private: int mCapId; ShmemBuffer mBuffer; mozilla::UniquePtr mAlternateBuffer; - int mSize; + size_t mSize; uint32_t mTimeStamp; int64_t mNtpTime; int64_t mRenderTime; @@ -184,6 +186,7 @@ CamerasParent::DispatchToVideoCaptureThread(nsRunnable *event) void CamerasParent::StopVideoCapture() { + LOG((__PRETTY_FUNCTION__)); // We are called from the main thread (xpcom-shutdown) or // from PBackground (when the Actor shuts down). // Shut down the WebRTC stack (on the capture thread) @@ -232,7 +235,7 @@ CamerasParent::DeliverFrameOverIPC(CaptureEngine cap_engine, int cap_id, ShmemBuffer buffer, unsigned char* altbuffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time) @@ -280,7 +283,7 @@ CamerasParent::GetBuffer(size_t aSize) int CallbackHelper::DeliverFrame(unsigned char* buffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time, @@ -309,6 +312,17 @@ CallbackHelper::DeliverFrame(unsigned char* buffer, thread->Dispatch(runnable, NS_DISPATCH_NORMAL); return 0; } +// XXX!!! FIX THIS -- we should move to pure DeliverI420Frame +int +CallbackHelper::DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) +{ + return DeliverFrame(const_cast(webrtc_frame.buffer(webrtc::kYPlane)), + CalcBufferSize(webrtc::kI420, webrtc_frame.width(), webrtc_frame.height()), + webrtc_frame.timestamp(), + webrtc_frame.ntp_time_ms(), + webrtc_frame.render_time_ms(), + (void*) webrtc_frame.native_handle()); +} bool CamerasParent::RecvReleaseFrame(mozilla::ipc::Shmem&& s) { @@ -393,6 +407,7 @@ CamerasParent::SetupEngine(CaptureEngine aCapEngine) void CamerasParent::CloseEngines() { + LOG((__PRETTY_FUNCTION__)); if (!mWebRTCAlive) { return; } @@ -686,10 +701,12 @@ CamerasParent::RecvReleaseCaptureDevice(const int& aCapEngine, RefPtr ipc_runnable = media::NewRunnableFrom([self, error, numdev]() -> nsresult { if (self->IsShuttingDown()) { + LOG(("In Shutdown, not Releasing")); return NS_ERROR_FAILURE; } if (error) { Unused << self->SendReplyFailure(); + LOG(("Failed to free device nr %d", numdev)); return NS_ERROR_FAILURE; } else { Unused << self->SendReplySuccess(); diff --git a/dom/media/systemservices/CamerasParent.h b/dom/media/systemservices/CamerasParent.h index cbbfc79dc6..e3c95e9d2c 100644 --- a/dom/media/systemservices/CamerasParent.h +++ b/dom/media/systemservices/CamerasParent.h @@ -41,11 +41,12 @@ public: virtual int FrameSizeChange(unsigned int w, unsigned int h, unsigned int streams) override; virtual int DeliverFrame(unsigned char* buffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time, void *handle) override; + virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override; virtual bool IsTextureSupported() override { return false; }; friend CamerasParent; @@ -108,7 +109,7 @@ public: int cap_id, ShmemBuffer buffer, unsigned char* altbuffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time); diff --git a/dom/media/systemservices/LoadManager.cpp b/dom/media/systemservices/LoadManager.cpp index 55506bab8d..b090c80555 100644 --- a/dom/media/systemservices/LoadManager.cpp +++ b/dom/media/systemservices/LoadManager.cpp @@ -170,14 +170,6 @@ LoadManagerSingleton::AddObserver(webrtc::CPULoadStateObserver * aObserver) LOG(("LoadManager - Adding Observer")); MutexAutoLock lock(mLock); mObservers.AppendElement(aObserver); - if (mObservers.Length() == 1) { - if (!mLoadMonitor) { - mLoadMonitor = new LoadMonitor(mLoadMeasurementInterval); - mLoadMonitor->Init(mLoadMonitor); - mLoadMonitor->SetLoadChangeCallback(this); - mLastStateChange = TimeStamp::Now(); - } - } } void diff --git a/dom/media/systemservices/MediaUtils.h b/dom/media/systemservices/MediaUtils.h index d50472ffd6..bde0c913fa 100644 --- a/dom/media/systemservices/MediaUtils.h +++ b/dom/media/systemservices/MediaUtils.h @@ -10,6 +10,7 @@ #include "nsAutoPtr.h" #include "nsThreadUtils.h" #include "nsIAsyncShutdown.h" +#include "base/task.h" namespace mozilla { namespace media { diff --git a/dom/media/systemservices/PCameras.ipdl b/dom/media/systemservices/PCameras.ipdl index 637b04a95a..a71e054cb1 100644 --- a/dom/media/systemservices/PCameras.ipdl +++ b/dom/media/systemservices/PCameras.ipdl @@ -27,7 +27,7 @@ child: async FrameSizeChange(int capEngine, int cap_id, int w, int h); // transfers ownership of |buffer| from parent to child async DeliverFrame(int capEngine, int cap_id, - Shmem buffer, int size, uint32_t time_stamp, + Shmem buffer, size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time); async ReplyNumberOfCaptureDevices(int numdev); async ReplyNumberOfCapabilities(int numdev); diff --git a/dom/media/systemservices/moz.build b/dom/media/systemservices/moz.build index 27a5f68ac7..3645b57baf 100644 --- a/dom/media/systemservices/moz.build +++ b/dom/media/systemservices/moz.build @@ -26,6 +26,11 @@ if CONFIG['MOZ_WEBRTC']: '/media/webrtc/signaling', '/media/webrtc/trunk', ] +if CONFIG['OS_TARGET'] == 'WINNT': + DEFINES['WEBRTC_WIN'] = True +else: + DEFINES['WEBRTC_POSIX'] = True + if CONFIG['MOZ_WIDGET_TOOLKIT'] in ('android', 'gonk'): EXPORTS += [ diff --git a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp index ebb2bbb2ce..c527ac03a2 100644 --- a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp +++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp @@ -271,7 +271,7 @@ MediaEngineRemoteVideoSource::FrameSizeChange(unsigned int w, unsigned int h, int MediaEngineRemoteVideoSource::DeliverFrame(unsigned char* buffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time, @@ -283,7 +283,7 @@ MediaEngineRemoteVideoSource::DeliverFrame(unsigned char* buffer, return 0; } - if (mWidth*mHeight + 2*(((mWidth+1)/2)*((mHeight+1)/2)) != size) { + if ((size_t) (mWidth*mHeight + 2*(((mWidth+1)/2)*((mHeight+1)/2))) != size) { MOZ_ASSERT(false, "Wrong size frame in DeliverFrame!"); return 0; } diff --git a/dom/media/webrtc/MediaEngineRemoteVideoSource.h b/dom/media/webrtc/MediaEngineRemoteVideoSource.h index 3415c8b101..ab0b3cccc5 100644 --- a/dom/media/webrtc/MediaEngineRemoteVideoSource.h +++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.h @@ -38,6 +38,10 @@ #include "NullTransport.h" +namespace webrtc { +class I420VideoFrame; +} + namespace mozilla { /** @@ -53,11 +57,13 @@ public: virtual int FrameSizeChange(unsigned int w, unsigned int h, unsigned int streams) override; virtual int DeliverFrame(unsigned char* buffer, - int size, + size_t size, uint32_t time_stamp, int64_t ntp_time, int64_t render_time, void *handle) override; + // XXX!!!! FIX THIS + virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override { return 0; }; virtual bool IsTextureSupported() override { return false; }; // MediaEngineCameraVideoSource diff --git a/dom/media/webrtc/MediaEngineWebRTC.cpp b/dom/media/webrtc/MediaEngineWebRTC.cpp index 650656c5c0..23040d2182 100644 --- a/dom/media/webrtc/MediaEngineWebRTC.cpp +++ b/dom/media/webrtc/MediaEngineWebRTC.cpp @@ -251,7 +251,7 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource, JNIEnv* const env = jni::GetEnvForThread(); MOZ_ALWAYS_TRUE(!env->GetJavaVM(&jvm)); - if (webrtc::VoiceEngine::SetAndroidObjects(jvm, env, (void*)context) != 0) { + if (webrtc::VoiceEngine::SetAndroidObjects(jvm, (void*)context) != 0) { LOG(("VoiceEngine:SetAndroidObjects Failed")); return; } diff --git a/dom/media/webrtc/moz.build b/dom/media/webrtc/moz.build index 7789514afc..3e1a538ab4 100644 --- a/dom/media/webrtc/moz.build +++ b/dom/media/webrtc/moz.build @@ -20,6 +20,10 @@ EXPORTS += [ ] if CONFIG['MOZ_WEBRTC']: + if CONFIG['OS_TARGET'] == 'WINNT': + DEFINES['WEBRTC_WIN'] = True + else: + DEFINES['WEBRTC_POSIX'] = True EXPORTS += ['AudioOutputObserver.h', 'MediaEngineRemoteVideoSource.h', 'MediaEngineWebRTC.h'] diff --git a/dom/mobilemessage/gonk/MmsService.js b/dom/mobilemessage/gonk/MmsService.js index b9386c7de4..a58ea4eb6d 100644 --- a/dom/mobilemessage/gonk/MmsService.js +++ b/dom/mobilemessage/gonk/MmsService.js @@ -8,7 +8,7 @@ const {classes: Cc, interfaces: Ci, utils: Cu, results: Cr} = Components; -Cu.importGlobalProperties(['Blob']); +Cu.importGlobalProperties(['Blob', 'FileReader']); Cu.import("resource://gre/modules/XPCOMUtils.jsm"); Cu.import("resource://gre/modules/Services.jsm"); Cu.import("resource://gre/modules/PhoneNumberUtils.jsm"); @@ -1303,8 +1303,7 @@ SendTransaction.prototype = Object.create(CancellableTransaction.prototype, { return; } - let fileReader = Cc["@mozilla.org/files/filereader;1"] - .createInstance(Ci.nsIDOMFileReader); + let fileReader = new FileReader(); fileReader.addEventListener("loadend", (aEvent) => { let arrayBuffer = aEvent.target.result; aPart.content = new Uint8Array(arrayBuffer); diff --git a/dom/security/test/csp/file_child-src_iframe.html b/dom/security/test/csp/file_child-src_iframe.html new file mode 100644 index 0000000000..3534aa329b --- /dev/null +++ b/dom/security/test/csp/file_child-src_iframe.html @@ -0,0 +1,61 @@ + + + + Bug 1045891 + + + + + + diff --git a/dom/security/test/csp/file_child-src_inner_frame.html b/dom/security/test/csp/file_child-src_inner_frame.html new file mode 100644 index 0000000000..e42102430e --- /dev/null +++ b/dom/security/test/csp/file_child-src_inner_frame.html @@ -0,0 +1,21 @@ + + + + Bug 1045891 + + + + + + diff --git a/dom/security/test/csp/file_child-src_service_worker.html b/dom/security/test/csp/file_child-src_service_worker.html new file mode 100644 index 0000000000..c970dc4f48 --- /dev/null +++ b/dom/security/test/csp/file_child-src_service_worker.html @@ -0,0 +1,29 @@ + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_child-src_service_worker.js b/dom/security/test/csp/file_child-src_service_worker.js new file mode 100644 index 0000000000..53f7687077 --- /dev/null +++ b/dom/security/test/csp/file_child-src_service_worker.js @@ -0,0 +1,3 @@ +this.addEventListener('install', function(event) { + close(); +}); diff --git a/dom/security/test/csp/file_child-src_shared_worker-redirect.html b/dom/security/test/csp/file_child-src_shared_worker-redirect.html new file mode 100644 index 0000000000..313915302e --- /dev/null +++ b/dom/security/test/csp/file_child-src_shared_worker-redirect.html @@ -0,0 +1,47 @@ + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_child-src_shared_worker.html b/dom/security/test/csp/file_child-src_shared_worker.html new file mode 100644 index 0000000000..0e9a56a294 --- /dev/null +++ b/dom/security/test/csp/file_child-src_shared_worker.html @@ -0,0 +1,34 @@ + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_child-src_shared_worker.js b/dom/security/test/csp/file_child-src_shared_worker.js new file mode 100644 index 0000000000..0fca1394c3 --- /dev/null +++ b/dom/security/test/csp/file_child-src_shared_worker.js @@ -0,0 +1,8 @@ +onconnect = function(e) { + var port = e.ports[0]; + port.addEventListener('message', function(e) { + port.postMessage('success'); + }); + + port.start(); +} diff --git a/dom/security/test/csp/file_child-src_shared_worker_data.html b/dom/security/test/csp/file_child-src_shared_worker_data.html new file mode 100644 index 0000000000..a4befe4ca3 --- /dev/null +++ b/dom/security/test/csp/file_child-src_shared_worker_data.html @@ -0,0 +1,37 @@ + + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_child-src_worker-redirect.html b/dom/security/test/csp/file_child-src_worker-redirect.html new file mode 100644 index 0000000000..dbceec1cc9 --- /dev/null +++ b/dom/security/test/csp/file_child-src_worker-redirect.html @@ -0,0 +1,50 @@ + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_child-src_worker.html b/dom/security/test/csp/file_child-src_worker.html new file mode 100644 index 0000000000..9300d3f617 --- /dev/null +++ b/dom/security/test/csp/file_child-src_worker.html @@ -0,0 +1,32 @@ + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_child-src_worker.js b/dom/security/test/csp/file_child-src_worker.js new file mode 100644 index 0000000000..1d93cac6b9 --- /dev/null +++ b/dom/security/test/csp/file_child-src_worker.js @@ -0,0 +1,4 @@ +onmessage = function(e) { + postMessage('worker'); +}; + diff --git a/dom/security/test/csp/file_child-src_worker_data.html b/dom/security/test/csp/file_child-src_worker_data.html new file mode 100644 index 0000000000..e9e22f01da --- /dev/null +++ b/dom/security/test/csp/file_child-src_worker_data.html @@ -0,0 +1,33 @@ + + + + Bug 1045891 + + + + + diff --git a/dom/security/test/csp/file_main.html^headers^ b/dom/security/test/csp/file_main.html^headers^ index c7fdbae204..3338de389b 100644 --- a/dom/security/test/csp/file_main.html^headers^ +++ b/dom/security/test/csp/file_main.html^headers^ @@ -1 +1 @@ -Content-Security-Policy: default-src 'self' ; style-src 'unsafe-inline' 'self' +Content-Security-Policy: default-src 'self' blob: ; style-src 'unsafe-inline' 'self' diff --git a/dom/security/test/csp/file_main.js b/dom/security/test/csp/file_main.js index 864a5d230b..0bc15b6827 100644 --- a/dom/security/test/csp/file_main.js +++ b/dom/security/test/csp/file_main.js @@ -1,16 +1,28 @@ -// some javascript for the CSP XHR tests -// +function doXHR(uri) { + try { + var xhr = new XMLHttpRequest(); + xhr.open("GET", uri); + xhr.send(); + } catch(ex) {} +} +doXHR("http://mochi.test:8888/tests/dom/security/test/csp/file_CSP.sjs?testid=xhr_good"); +doXHR("http://example.com/tests/dom/security/test/csp/file_CSP.sjs?testid=xhr_bad"); +fetch("http://mochi.test:8888/tests/dom/security/test/csp/file_CSP.sjs?testid=fetch_good"); +fetch("http://example.com/tests/dom/security/test/csp/file_CSP.sjs?testid=fetch_bad"); +navigator.sendBeacon("http://mochi.test:8888/tests/dom/security/test/csp/file_CSP.sjs?testid=beacon_good"); try { - var xhr_good = new XMLHttpRequest(); - var xhr_good_uri ="http://mochi.test:8888/tests/dom/security/test/csp/file_CSP.sjs?testid=xhr_good"; - xhr_good.open("GET", xhr_good_uri, true); - xhr_good.send(null); -} catch(e) {} + navigator.sendBeacon("http://example.com/tests/dom/security/test/csp/file_CSP.sjs?testid=beacon_bad"); +} catch(ex) {} -try { - var xhr_bad = new XMLHttpRequest(); - var xhr_bad_uri ="http://example.com/tests/dom/security/test/csp/file_CSP.sjs?testid=xhr_bad"; - xhr_bad.open("GET", xhr_bad_uri, true); - xhr_bad.send(null); -} catch(e) {} + +new Worker("file_main_worker.js").postMessage({inherited : false}); + + +var blobxhr = new XMLHttpRequest(); +blobxhr.open("GET", "file_main_worker.js") +blobxhr.responseType = "blob"; +blobxhr.send(); +blobxhr.onload = () => { + new Worker(URL.createObjectURL(blobxhr.response)).postMessage({inherited : true}); +} diff --git a/dom/security/test/csp/file_main_worker.js b/dom/security/test/csp/file_main_worker.js new file mode 100644 index 0000000000..d8953eec74 --- /dev/null +++ b/dom/security/test/csp/file_main_worker.js @@ -0,0 +1,28 @@ +function doXHR(uri) { + try { + var xhr = new XMLHttpRequest(); + xhr.open("GET", uri); + xhr.send(); + } catch(ex) {} +} + +var sameBase = "http://mochi.test:8888/tests/dom/security/test/csp/file_CSP.sjs?testid="; +var crossBase = "http://example.com/tests/dom/security/test/csp/file_CSP.sjs?testid="; + +onmessage = (e) => { + for (base of [sameBase, crossBase]) { + var prefix; + var suffix; + if (e.data.inherited) { + prefix = base + "worker_inherited_" + suffix = base == sameBase ? "_good" : "_bad"; + } + else { + prefix = base + "worker_" + suffix = base == sameBase ? "_same_good" : "_cross_good"; + } + doXHR(prefix + "xhr" + suffix); + fetch(prefix + "fetch" + suffix); + try { importScripts(prefix + "script" + suffix); } catch(ex) {} + } +} diff --git a/dom/security/test/csp/file_redirect_worker.sjs b/dom/security/test/csp/file_redirect_worker.sjs new file mode 100644 index 0000000000..27df6a5fdf --- /dev/null +++ b/dom/security/test/csp/file_redirect_worker.sjs @@ -0,0 +1,34 @@ +// SJS file to serve resources for CSP redirect tests +// This file redirects to a specified resource. +const THIS_SITE = "http://mochi.test:8888"; +const OTHER_SITE = "http://example.com"; + +function handleRequest(request, response) +{ + var query = {}; + request.queryString.split('&').forEach(function (val) { + var [name, value] = val.split('='); + query[name] = unescape(value); + }); + + var resource = query['path']; + + response.setHeader("Cache-Control", "no-cache", false); + var loc = ''; + + // redirect to a resource on this site + if (query["redir"] == "same") { + loc = THIS_SITE+resource+"#"+query['page_id'] + } + + // redirect to a resource on a different site + else if (query["redir"] == "other") { + loc = OTHER_SITE+resource+"#"+query['page_id'] + } + + response.setStatusLine("1.1", 302, "Found"); + response.setHeader("Location", loc, false); + + response.write(''); + return; +} diff --git a/dom/security/test/csp/file_redirects_main.html b/dom/security/test/csp/file_redirects_main.html index 651f44d384..5c9affea04 100644 --- a/dom/security/test/csp/file_redirects_main.html +++ b/dom/security/test/csp/file_redirects_main.html @@ -11,17 +11,18 @@ var thisSite = "http://mochi.test:8888"; var otherSite = "http://example.com"; var page = "/tests/dom/security/test/csp/file_redirects_page.sjs"; -var tests = { "font-src": thisSite+page+"?testid=font-src&csp=1", - "frame-src": thisSite+page+"?testid=frame-src&csp=1", - "img-src": thisSite+page+"?testid=img-src&csp=1", - "media-src": thisSite+page+"?testid=media-src&csp=1", - "object-src": thisSite+page+"?testid=object-src&csp=1", - "script-src": thisSite+page+"?testid=script-src&csp=1", - "style-src": thisSite+page+"?testid=style-src&csp=1", - "worker": thisSite+page+"?testid=worker&csp=1", - "xhr-src": thisSite+page+"?testid=xhr-src&csp=1", - "script-src-from-worker": thisSite+page+"?testid=script-src-from-worker&csp=1", - "img-src-from-css": thisSite+page+"?testid=img-src-from-css&csp=1", +var tests = { "font-src": thisSite+page+"?testid=font-src", + "frame-src": thisSite+page+"?testid=frame-src", + "img-src": thisSite+page+"?testid=img-src", + "media-src": thisSite+page+"?testid=media-src", + "object-src": thisSite+page+"?testid=object-src", + "script-src": thisSite+page+"?testid=script-src", + "style-src": thisSite+page+"?testid=style-src", + "worker": thisSite+page+"?testid=worker", + "xhr-src": thisSite+page+"?testid=xhr-src", + "from-worker": thisSite+page+"?testid=from-worker", + "from-blob-worker": thisSite+page+"?testid=from-blob-worker", + "img-src-from-css": thisSite+page+"?testid=img-src-from-css", }; var container = document.getElementById("container"); diff --git a/dom/security/test/csp/file_redirects_page.sjs b/dom/security/test/csp/file_redirects_page.sjs index 58eff8af64..9e3c0d0350 100644 --- a/dom/security/test/csp/file_redirects_page.sjs +++ b/dom/security/test/csp/file_redirects_page.sjs @@ -14,9 +14,13 @@ function handleRequest(request, response) var resource = "/tests/dom/security/test/csp/file_redirects_resource.sjs"; // CSP header value - if (query["csp"] == 1) { - response.setHeader("Content-Security-Policy", "default-src 'self' ; style-src 'self' 'unsafe-inline'", false); + var additional = "" + if (query['testid'] == "worker") { + additional = "; script-src 'self' 'unsafe-inline'"; } + response.setHeader("Content-Security-Policy", + "default-src 'self' blob: ; style-src 'self' 'unsafe-inline'" + additional, + false); // downloadable font that redirects to another site if (query["testid"] == "font-src") { @@ -61,13 +65,13 @@ function handleRequest(request, response) // external stylesheet that redirects to another site if (query["testid"] == "style-src") { - response.write(''); + response.write(''); return; } // worker script resource that redirects to another site if (query["testid"] == "worker") { - response.write(''); + response.write(''); return; } @@ -84,13 +88,27 @@ function handleRequest(request, response) return; } - if (query["testid"] == "script-src-from-worker") { + if (query["testid"] == "from-worker") { // loads a script; launches a worker; that worker uses importscript; which then gets redirected // So it's: - // '); + // .. calls xhr("res=xhr-resp") + // .. calls fetch("res=xhr-resp") + response.write(''); + return; + } + + if (query["testid"] == "from-blob-worker") { + // loads a script; launches a worker; that worker uses importscript; which then gets redirected + // So it's: + // '); return; } } diff --git a/dom/security/test/csp/file_redirects_resource.sjs b/dom/security/test/csp/file_redirects_resource.sjs index b921bd608b..d281f19d8b 100644 --- a/dom/security/test/csp/file_redirects_resource.sjs +++ b/dom/security/test/csp/file_redirects_resource.sjs @@ -102,38 +102,45 @@ function handleRequest(request, response) // script that loads an internal worker that uses importScripts on a redirect // to an external script. - if (query["res"] == "loadWorkerThatImports") { + if (query["res"] == "loadWorkerThatMakesRequests") { // this creates a worker (same origin) that imports a redirecting script. - let workerURL = thisSite + resource + '?res=importScriptWorker&id=' + query["id"]; + let workerURL = thisSite + resource + '?res=makeRequestsWorker&id=' + query["id"]; response.setHeader("Content-Type", "application/javascript", false); - response.write("var w=new Worker('" + workerURL + "'); w.onmessage=function(event){ alert(event.data); }"); + response.write("new Worker('" + workerURL + "');"); + return; + } + + // script that loads an internal worker that uses importScripts on a redirect + // to an external script. + if (query["res"] == "loadBlobWorkerThatMakesRequests") { + // this creates a worker (same origin) that imports a redirecting script. + let workerURL = thisSite + resource + '?res=makeRequestsWorker&id=' + query["id"]; + response.setHeader("Content-Type", "application/javascript", false); + response.write("var x = new XMLHttpRequest(); x.open('GET', '" + workerURL + "'); "); + response.write("x.responseType = 'blob'; x.send(); "); + response.write("x.onload = () => { new Worker(URL.createObjectURL(x.response)); };"); return; } // source for a worker that simply calls importScripts on a script that // redirects. - if (query["res"] == "importScriptWorker") { + if (query["res"] == "makeRequestsWorker") { // this is code for a worker that imports a redirected script. - let scriptURL = thisSite + resource + "?redir=other&res=script&id=" + query["id"]; + let scriptURL = thisSite + resource + "?redir=other&res=script&id=script-src-redir-" + query["id"]; + let xhrURL = thisSite + resource + "?redir=other&res=xhr-resp&id=xhr-src-redir-" + query["id"]; + let fetchURL = thisSite + resource + "?redir=other&res=xhr-resp&id=fetch-src-redir-" + query["id"]; response.setHeader("Content-Type", "application/javascript", false); - response.write("importScripts('" + scriptURL + "');"); + response.write("try { importScripts('" + scriptURL + "'); } catch(ex) {} "); + response.write("var x = new XMLHttpRequest(); x.open('GET', '" + xhrURL + "'); x.send();"); + response.write("fetch('" + fetchURL + "');"); return; } // script that invokes XHR if (query["res"] == "xhr") { - response.setHeader("Content-Type", "text/html", false); - var resp = 'var x = new XMLHttpRequest(); x.open("GET", "' + otherSite + - resource+'?res=xhr-resp&testid=xhr-src-redir", false); ' + - 'x.send(null);'; - response.write(resp); - return; - } - - if (query["res"] == "xhr") { - response.setHeader("Content-Type", "text/html", false); - var resp = 'var x = new XMLHttpRequest(); x.open("GET", "' + otherSite + - resource+'?res=xhr-resp&testid=xhr-src-redir", false); ' + + response.setHeader("Content-Type", "application/javascript", false); + var resp = 'var x = new XMLHttpRequest();x.open("GET", "' + thisSite + + resource+'?redir=other&res=xhr-resp&id=xhr-src-redir", false);\n' + 'x.send(null);'; response.write(resp); return; diff --git a/dom/security/test/csp/file_service_worker.html b/dom/security/test/csp/file_service_worker.html new file mode 100644 index 0000000000..00a2b40206 --- /dev/null +++ b/dom/security/test/csp/file_service_worker.html @@ -0,0 +1,19 @@ + + + + Bug 1208559 - ServiceWorker registration not governed by CSP + + + + + diff --git a/dom/security/test/csp/file_service_worker.js b/dom/security/test/csp/file_service_worker.js new file mode 100644 index 0000000000..1bf583f4cc --- /dev/null +++ b/dom/security/test/csp/file_service_worker.js @@ -0,0 +1 @@ +dump("service workers: hello world"); diff --git a/dom/security/test/csp/file_worker_redirect.html b/dom/security/test/csp/file_worker_redirect.html deleted file mode 100644 index be80f5795f..0000000000 --- a/dom/security/test/csp/file_worker_redirect.html +++ /dev/null @@ -1,9 +0,0 @@ - - - - Bug 949706 - CSP: Correct handling of web workers importing scripts that get redirected - - - - - diff --git a/dom/security/test/csp/file_worker_redirect.sjs b/dom/security/test/csp/file_worker_redirect.sjs deleted file mode 100644 index 6c4acd6e7a..0000000000 --- a/dom/security/test/csp/file_worker_redirect.sjs +++ /dev/null @@ -1,37 +0,0 @@ -// testserver customized for the needs of: -// Bug 949706 - CSP: Correct handling of web workers importing scripts that get redirected - -function handleRequest(request, response) -{ - response.setHeader("Cache-Control", "no-cache", false); - response.setHeader("Content-Type", "text/html", false); - - var query = request.queryString; - - if (query === "stage_0_script_loads_worker") { - var newWorker = - "var myWorker = new Worker(\"file_worker_redirect.sjs?stage_1_worker_import_scripts\");" + - "myWorker.onmessage = function (event) { parent.checkResult(\"allowed\"); };" + - "myWorker.onerror = function (event) { parent.checkResult(\"blocked\"); };"; - response.write(newWorker); - return; - } - - if (query === "stage_1_worker_import_scripts") { - response.write("importScripts(\"file_worker_redirect.sjs?stage_2_redirect_imported_script\");"); - return; - } - - if (query === "stage_2_redirect_imported_script") { - var newLocation = - "http://test1.example.com/tests/dom/security/test/csp/file_worker_redirect.sjs?stage_3_target_script"; - response.setStatusLine("1.1", 302, "Found"); - response.setHeader("Location", newLocation, false); - return; - } - - if (query === "stage_3_target_script") { - response.write("postMessage(\"imported script loaded\");"); - return; - } -} diff --git a/dom/security/test/csp/mochitest.ini b/dom/security/test/csp/mochitest.ini index 53ea996acd..fae1bb1c7c 100644 --- a/dom/security/test/csp/mochitest.ini +++ b/dom/security/test/csp/mochitest.ini @@ -43,6 +43,7 @@ support-files = file_main.html file_main.html^headers^ file_main.js + file_main_worker.js file_web_manifest.html file_web_manifest_remote.html file_web_manifest_https.html @@ -115,8 +116,6 @@ support-files = file_multi_policy_injection_bypass_2.html^headers^ file_null_baseuri.html file_form-action.html - file_worker_redirect.html - file_worker_redirect.sjs file_referrerdirective.html referrerdirective.sjs file_upgrade_insecure.html @@ -128,6 +127,21 @@ support-files = file_upgrade_insecure_referrer_server.sjs file_upgrade_insecure_cors.html file_upgrade_insecure_cors_server.sjs + file_service_worker.html + file_service_worker.js + file_child-src_iframe.html + file_child-src_inner_frame.html + file_child-src_worker.html + file_child-src_worker_data.html + file_child-src_worker-redirect.html + file_child-src_worker.js + file_child-src_service_worker.html + file_child-src_service_worker.js + file_child-src_shared_worker.html + file_child-src_shared_worker_data.html + file_child-src_shared_worker-redirect.html + file_child-src_shared_worker.js + file_redirect_worker.sjs [test_base-uri.html] [test_blob_data_schemes.html] @@ -181,7 +195,6 @@ skip-if = buildapp == 'b2g' # intermittent orange (bug 1028490) [test_null_baseuri.html] [test_referrerdirective.html] skip-if = buildapp == 'b2g' #no ssl support -[test_worker_redirect.html] [test_upgrade_insecure.html] # no ssl support as well as websocket tests do not work (see test_websocket.html) skip-if = buildapp == 'b2g' || buildapp == 'mulet' || toolkit == 'gonk' || toolkit == 'android' @@ -191,3 +204,10 @@ skip-if = buildapp == 'b2g' || buildapp == 'mulet' || toolkit == 'gonk' || toolk skip-if = buildapp == 'b2g' || buildapp == 'mulet' || toolkit == 'gonk' || toolkit == 'android' [test_upgrade_insecure_cors.html] skip-if = buildapp == 'b2g' || buildapp == 'mulet' || toolkit == 'gonk' || toolkit == 'android' +[test_service_worker.html] +skip-if = buildapp == 'b2g' #no ssl support +[test_child-src_worker.html] +skip-if = buildapp == 'b2g' #investigate in bug 1222904 +[test_child-src_worker_data.html] +[test_child-src_worker-redirect.html] +[test_child-src_iframe.html] diff --git a/dom/security/test/csp/test_CSP.html b/dom/security/test/csp/test_CSP.html index 2b508bd0d1..253ea8d3a7 100644 --- a/dom/security/test/csp/test_CSP.html +++ b/dom/security/test/csp/test_CSP.html @@ -25,6 +25,22 @@ window.tests = { script_bad: -1, xhr_good: -1, xhr_bad: -1, + fetch_good: -1, + fetch_bad: -1, + beacon_good: -1, + beacon_bad: -1, + worker_xhr_same_good: -1, + worker_xhr_cross_good: -1, + worker_fetch_same_good: -1, + worker_fetch_cross_good: -1, + worker_script_same_good: -1, + worker_script_cross_good: -1, + worker_inherited_xhr_good: -1, + worker_inherited_xhr_bad: -1, + worker_inherited_fetch_good: -1, + worker_inherited_fetch_bad: -1, + worker_inherited_script_good: -1, + worker_inherited_script_bad: -1, media_good: -1, media_bad: -1, font_good: -1, @@ -81,10 +97,11 @@ examiner.prototype = { window.examiner = new examiner(); window.testResult = function(testname, result, msg) { - //test already complete.... forget it... remember the first result. + // test already complete.... forget it... remember the first result. if (window.tests[testname] != -1) return; + ok(testname in window.tests, "It's a real test"); window.tests[testname] = result; is(result, true, testname + ' test: ' + msg); diff --git a/dom/security/test/csp/test_child-src_iframe.html b/dom/security/test/csp/test_child-src_iframe.html new file mode 100644 index 0000000000..b4ba36f89f --- /dev/null +++ b/dom/security/test/csp/test_child-src_iframe.html @@ -0,0 +1,114 @@ + + + + Bug 1045891 + + + + + +

+ + + + + diff --git a/dom/security/test/csp/test_child-src_worker-redirect.html b/dom/security/test/csp/test_child-src_worker-redirect.html new file mode 100644 index 0000000000..dfb99149cc --- /dev/null +++ b/dom/security/test/csp/test_child-src_worker-redirect.html @@ -0,0 +1,125 @@ + + + + Bug 1045891 + + + + + +

+ + + + + diff --git a/dom/security/test/csp/test_child-src_worker.html b/dom/security/test/csp/test_child-src_worker.html new file mode 100644 index 0000000000..1216e4e2f0 --- /dev/null +++ b/dom/security/test/csp/test_child-src_worker.html @@ -0,0 +1,131 @@ + + + + Bug 1045891 + + + + + +

+ + + + + diff --git a/dom/security/test/csp/test_child-src_worker_data.html b/dom/security/test/csp/test_child-src_worker_data.html new file mode 100644 index 0000000000..089d32dbec --- /dev/null +++ b/dom/security/test/csp/test_child-src_worker_data.html @@ -0,0 +1,126 @@ + + + + Bug 1045891 + + + + + +

+ + + + + diff --git a/dom/security/test/csp/test_redirects.html b/dom/security/test/csp/test_redirects.html index 4656dd70de..4a588dcbe1 100644 --- a/dom/security/test/csp/test_redirects.html +++ b/dom/security/test/csp/test_redirects.html @@ -86,10 +86,16 @@ var testExpectedResults = { "font-src": true, "worker-redir": false, "xhr-src": true, "xhr-src-redir": false, - "script-src-from-worker": true, /* test runs */ - "script-src-redir-from-worker": false, /* redir is blocked */ - "img-src-from-css": true, /* test runs */ - "img-src-redir-from-css": false, /* redir is blocked */ + "from-worker": true, + "script-src-redir-from-worker": true, /* redir is allowed since policy isn't inherited */ + "xhr-src-redir-from-worker": true, /* redir is allowed since policy isn't inherited */ + "fetch-src-redir-from-worker": true, /* redir is allowed since policy isn't inherited */ + "from-blob-worker": true, + "script-src-redir-from-blob-worker": false, + "xhr-src-redir-from-blob-worker": false, + "fetch-src-redir-from-blob-worker": false, + "img-src-from-css": true, + "img-src-redir-from-css": false, }; // takes the name of the test, the URL that was tested, and whether the diff --git a/dom/security/test/csp/test_service_worker.html b/dom/security/test/csp/test_service_worker.html new file mode 100644 index 0000000000..d1dfac7301 --- /dev/null +++ b/dom/security/test/csp/test_service_worker.html @@ -0,0 +1,62 @@ + + + + Bug 1208559 - ServiceWorker registration not governed by CSP + + + + + + + + + + diff --git a/dom/security/test/csp/test_worker_redirect.html b/dom/security/test/csp/test_worker_redirect.html deleted file mode 100644 index 62a43c77b3..0000000000 --- a/dom/security/test/csp/test_worker_redirect.html +++ /dev/null @@ -1,76 +0,0 @@ - - - - Bug 949706 - CSP: Correct handling of web workers importing scripts that get redirected - - - - - - - - - - diff --git a/dom/tests/mochitest/general/frameStoragePrevented.html b/dom/tests/mochitest/general/frameStoragePrevented.html index b29cf2c9d0..3554d6082c 100644 --- a/dom/tests/mochitest/general/frameStoragePrevented.html +++ b/dom/tests/mochitest/general/frameStoragePrevented.html @@ -4,7 +4,7 @@ frame for storage prevented test - - - - -

- -

-
-
-
-
diff --git a/dom/workers/test/test_csp.html^headers^ b/dom/workers/test/test_csp.html^headers^
index b91ba384d9..1c93210799 100644
--- a/dom/workers/test/test_csp.html^headers^
+++ b/dom/workers/test/test_csp.html^headers^
@@ -1,2 +1,2 @@
 Cache-Control: no-cache
-Content-Security-Policy: default-src 'self'
+Content-Security-Policy: default-src 'self' blob:
diff --git a/dom/workers/test/test_csp.js b/dom/workers/test/test_csp.js
index cc6fcd3d04..dcbcd8c3a1 100644
--- a/dom/workers/test/test_csp.js
+++ b/dom/workers/test/test_csp.js
@@ -2,49 +2,47 @@
  * Any copyright is dedicated to the Public Domain.
  * http://creativecommons.org/publicdomain/zero/1.0/
  */
-msg = null;
-var errors = 5;
+var tests = 3;
 
-onerror = function(event) {
-  ok(true, msg);
-  if (!--errors) SimpleTest.finish();
-}
-
-msg = "No Eval allowed";
-worker = new Worker("csp_worker.js");
-worker.postMessage(0);
-worker.onmessage = function(event) {
-  ok(false, "Eval succeeded!");
-}
-
-msg = "No Eval allowed 2";
-worker = new Worker("csp_worker.js");
-worker.postMessage(4);
-worker.onmessage = function(event) {
-  ok(false, "Eval succeeded!");
-}
-
-msg = "ImportScripts data:";
-worker = new Worker("csp_worker.js");
-worker.postMessage(-1);
-worker.onmessage = function(event) {
-  ok(false, "Eval succeeded!");
-}
-
-msg = "ImportScripts javascript:";
-worker = new Worker("csp_worker.js");
-worker.postMessage(-2);
-worker.onmessage = function(event) {
-  ok(false, "Eval succeeded!");
-}
-
-msg = "Loading data:something";
-try {
-  worker = new Worker("data:application/javascript;base64,ZHVtcCgnaGVsbG8gd29ybGQnKQo=");
-  ok(false, "Should have thrown!");
-} catch (e) {
-  ok(true, "Threw as expected.");
-}
-
-worker = new Worker("javascript:dump(123);");
 SimpleTest.waitForExplicitFinish();
+
+testDone = function(event) {
+  if (!--tests) SimpleTest.finish();
+}
+
+// Workers don't inherit CSP
+worker = new Worker("csp_worker.js");
+worker.postMessage({ do: "eval" });
+worker.onmessage = function(event) {
+  is(event.data, 42, "Eval succeeded!");
+  testDone();
+}
+
+// blob: workers *do* inherit CSP
+xhr = new XMLHttpRequest;
+xhr.open("GET", "csp_worker.js");
+xhr.responseType = "blob";
+xhr.send();
+xhr.onload = (e) => {
+  uri = URL.createObjectURL(e.target.response);
+  worker = new Worker(uri);
+  worker.postMessage({ do: "eval" })
+  worker.onmessage = function(event) {
+    is(event.data, "Error: call to eval() blocked by CSP", "Eval threw");
+    testDone();
+  }
+}
+
+xhr = new XMLHttpRequest;
+xhr.open("GET", "csp_worker.js");
+xhr.responseType = "blob";
+xhr.send();
+xhr.onload = (e) => {
+  uri = URL.createObjectURL(e.target.response);
+  worker = new Worker(uri);
+  worker.postMessage({ do: "nest", uri: uri, level: 3 })
+  worker.onmessage = function(event) {
+    is(event.data, "Error: call to eval() blocked by CSP", "Eval threw in nested worker");
+    testDone();
+  }
+}
diff --git a/dom/workers/test/test_loadError.html b/dom/workers/test/test_loadError.html
index 355300c304..778b6ed771 100644
--- a/dom/workers/test/test_loadError.html
+++ b/dom/workers/test/test_loadError.html
@@ -13,55 +13,53 @@
 
-  
-
-
-  
-    
-      
-    
-    
-      
-      
-    
-  
Local Preview

-
-
diff --git a/media/webrtc/trunk/webrtc/test/manual/iframe-apprtc.html b/media/webrtc/trunk/webrtc/test/manual/iframe-apprtc.html
deleted file mode 100644
index 31e0db10c0..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/iframe-apprtc.html
+++ /dev/null
@@ -1,20 +0,0 @@
-
-
-
-
-  AppRTC web app in an IFRAME
-  
-
-
-  AppRTC in an <iframe> element:

-  
-
-
diff --git a/media/webrtc/trunk/webrtc/test/manual/iframe-video.html b/media/webrtc/trunk/webrtc/test/manual/iframe-video.html
deleted file mode 100644
index 3cb532b091..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/iframe-video.html
+++ /dev/null
@@ -1,19 +0,0 @@
-
-
-
-
-  IFRAME Single Local Preview (Video Only)
-  
-
-
-  
-
-
\ No newline at end of file
diff --git a/media/webrtc/trunk/webrtc/test/manual/multiple-audio.html b/media/webrtc/trunk/webrtc/test/manual/multiple-audio.html
deleted file mode 100644
index 80e4b646c6..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/multiple-audio.html
+++ /dev/null
@@ -1,56 +0,0 @@
-
-
-
-
-  Multiple Local Preview (Audio Only)
-  
-  
-  
-  
-
-
-  
-    
-      
-    
-    
-      
-      
-      
-      
-      
-      
-      
-      
-      
-      
-    
-  
Sound test

-
-
diff --git a/media/webrtc/trunk/webrtc/test/manual/multiple-video.html b/media/webrtc/trunk/webrtc/test/manual/multiple-video.html
deleted file mode 100644
index fd82bf82cf..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/multiple-video.html
+++ /dev/null
@@ -1,72 +0,0 @@
-
-
-
-
-  Multiple Local Preview (Video Only)
-  
-  
-  
-  
-
-
-  
-    
-      
-    
-    
-      
-      
-      
-    
-    
-      
-      
-      
-    
-    
-      
-      
-      
-    
-    
-      
-    
-  
Local Preview

-
-
diff --git a/media/webrtc/trunk/webrtc/test/manual/single-audio.html b/media/webrtc/trunk/webrtc/test/manual/single-audio.html
deleted file mode 100644
index b5d1ed464a..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/single-audio.html
+++ /dev/null
@@ -1,44 +0,0 @@
-
-
-
-
-  Single Local Preview (Audio Only)
-  
-  
-  
-  
-
-
-  
-    
-      
-    
-    
-      
-    
-  
Sound test

-
-
diff --git a/media/webrtc/trunk/webrtc/test/manual/single-video.html b/media/webrtc/trunk/webrtc/test/manual/single-video.html
deleted file mode 100644
index 7a88dcf97d..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/single-video.html
+++ /dev/null
@@ -1,45 +0,0 @@
-
-
-
-
-  Single Local Preview (Video Only)
-  
-  
-  
-  
-
-
-  
-    
-      
-    
-    
-      
-    
-  
Local Preview

-
-
diff --git a/media/webrtc/trunk/webrtc/test/manual/two-video-devices.html b/media/webrtc/trunk/webrtc/test/manual/two-video-devices.html
deleted file mode 100644
index 44c0947bdc..0000000000
--- a/media/webrtc/trunk/webrtc/test/manual/two-video-devices.html
+++ /dev/null
@@ -1,51 +0,0 @@
-
-
-
-
-  Single Local Preview (Video Only)
-  
-   
-  
-  
-
-
-  
-    
-      
-    
-    
-      
-      
-    
-  
Local Preview

-
-
\ No newline at end of file
diff --git a/media/webrtc/trunk/webrtc/test/metrics.gyp b/media/webrtc/trunk/webrtc/test/metrics.gyp
index 0486a7c9ea..eda0b054ee 100644
--- a/media/webrtc/trunk/webrtc/test/metrics.gyp
+++ b/media/webrtc/trunk/webrtc/test/metrics.gyp
@@ -21,7 +21,7 @@
       'type': 'static_library',
       'dependencies': [
         '<(webrtc_root)/common_video/common_video.gyp:common_video',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
         'testsupport/metrics/video_metrics.h',
diff --git a/media/webrtc/trunk/webrtc/test/mock_transport.h b/media/webrtc/trunk/webrtc/test/mock_transport.h
index 388ee92e9d..5b1cb8dfb6 100644
--- a/media/webrtc/trunk/webrtc/test/mock_transport.h
+++ b/media/webrtc/trunk/webrtc/test/mock_transport.h
@@ -19,9 +19,9 @@ namespace webrtc {
 class MockTransport : public webrtc::Transport {
  public:
   MOCK_METHOD3(SendPacket,
-      int(int channel, const void* data, int len));
+      int(int channel, const void* data, size_t len));
   MOCK_METHOD3(SendRTCPPacket,
-      int(int channel, const void* data, int len));
+      int(int channel, const void* data, size_t len));
 };
 }  // namespace webrtc
 #endif  // WEBRTC_TEST_MOCK_TRANSPORT_H_
diff --git a/media/webrtc/trunk/webrtc/test/null_transport.h b/media/webrtc/trunk/webrtc/test/null_transport.h
index e8d4d10275..b80f751b1e 100644
--- a/media/webrtc/trunk/webrtc/test/null_transport.h
+++ b/media/webrtc/trunk/webrtc/test/null_transport.h
@@ -19,8 +19,8 @@ class PacketReceiver;
 namespace test {
 class NullTransport : public newapi::Transport {
  public:
-  virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE;
-  virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE;
+  bool SendRtp(const uint8_t* packet, size_t length) override;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc b/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc
index 558bee379e..b9e430fe9f 100644
--- a/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc
+++ b/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/test/rtcp_packet_parser.h"
 
+#include "testing/gtest/include/gtest/gtest.h"
+
 namespace webrtc {
 namespace test {
 
@@ -17,9 +19,10 @@ RtcpPacketParser::RtcpPacketParser() {}
 
 RtcpPacketParser::~RtcpPacketParser() {}
 
-void RtcpPacketParser::Parse(const void *data, int len) {
+void RtcpPacketParser::Parse(const void *data, size_t len) {
   const uint8_t* packet = static_cast(data);
   RTCPUtility::RTCPParserV2 parser(packet, len, true);
+  EXPECT_TRUE(parser.IsValid());
   for (RTCPUtility::RTCPPacketTypes type = parser.Begin();
       type != RTCPUtility::kRtcpNotValidCode;
       type = parser.Iterate()) {
diff --git a/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.h b/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.h
index f7d36ba804..cc890b4f2f 100644
--- a/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.h
+++ b/media/webrtc/trunk/webrtc/test/rtcp_packet_parser.h
@@ -631,7 +631,7 @@ class RtcpPacketParser {
   RtcpPacketParser();
   ~RtcpPacketParser();
 
-  void Parse(const void *packet, int packet_len);
+  void Parse(const void *packet, size_t packet_len);
 
   SenderReport* sender_report() { return &sender_report_; }
   ReceiverReport* receiver_report() { return &receiver_report_; }
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc b/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc
index fd3116ebb3..26151bba1f 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_reader.cc
@@ -17,8 +17,9 @@
 #include 
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 namespace test {
@@ -42,11 +43,78 @@ static uint16_t kPacketHeaderSize = 8;
     }                                                  \
   } while (0)
 
+bool ReadUint32(uint32_t* out, FILE* file) {
+  *out = 0;
+  for (size_t i = 0; i < 4; ++i) {
+    *out <<= 8;
+    uint8_t tmp;
+    if (fread(&tmp, 1, sizeof(uint8_t), file) != sizeof(uint8_t))
+      return false;
+    *out |= tmp;
+  }
+  return true;
+}
+
+bool ReadUint16(uint16_t* out, FILE* file) {
+  *out = 0;
+  for (size_t i = 0; i < 2; ++i) {
+    *out <<= 8;
+    uint8_t tmp;
+    if (fread(&tmp, 1, sizeof(uint8_t), file) != sizeof(uint8_t))
+      return false;
+    *out |= tmp;
+  }
+  return true;
+}
+
 class RtpFileReaderImpl : public RtpFileReader {
  public:
   virtual bool Init(const std::string& filename) = 0;
 };
 
+class InterleavedRtpFileReader : public RtpFileReaderImpl {
+ public:
+  virtual ~InterleavedRtpFileReader() {
+    if (file_ != NULL) {
+      fclose(file_);
+      file_ = NULL;
+    }
+  }
+
+  virtual bool Init(const std::string& filename) {
+    file_ = fopen(filename.c_str(), "rb");
+    if (file_ == NULL) {
+      printf("ERROR: Can't open file: %s\n", filename.c_str());
+      return false;
+    }
+    return true;
+  }
+  virtual bool NextPacket(RtpPacket* packet) {
+    assert(file_ != NULL);
+    packet->length = RtpPacket::kMaxPacketBufferSize;
+    uint32_t len = 0;
+    TRY(ReadUint32(&len, file_));
+    if (packet->length < len) {
+      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
+              << packet->length
+              << " bytes allocated. Consider increasing the buffer "
+                 "size";
+    }
+    if (fread(packet->data, 1, len, file_) != len)
+      return false;
+
+    packet->length = len;
+    packet->original_length = len;
+    packet->time_ms = time_ms_;
+    time_ms_ += 5;
+    return true;
+  }
+
+ private:
+  FILE* file_ = NULL;
+  int64_t time_ms_ = 0;
+};
+
 // Read RTP packets from file in rtpdump format, as documented at:
 // http://www.cs.columbia.edu/irt/software/rtptools/
 class RtpDumpReader : public RtpFileReaderImpl {
@@ -91,25 +159,25 @@ class RtpDumpReader : public RtpFileReaderImpl {
     uint32_t source;
     uint16_t port;
     uint16_t padding;
-    TRY(Read(&start_sec));
-    TRY(Read(&start_usec));
-    TRY(Read(&source));
-    TRY(Read(&port));
-    TRY(Read(&padding));
+    TRY(ReadUint32(&start_sec, file_));
+    TRY(ReadUint32(&start_usec, file_));
+    TRY(ReadUint32(&source, file_));
+    TRY(ReadUint16(&port, file_));
+    TRY(ReadUint16(&padding, file_));
 
     return true;
   }
 
-  virtual bool NextPacket(Packet* packet) OVERRIDE {
+  bool NextPacket(RtpPacket* packet) override {
     uint8_t* rtp_data = packet->data;
-    packet->length = Packet::kMaxPacketBufferSize;
+    packet->length = RtpPacket::kMaxPacketBufferSize;
 
     uint16_t len;
     uint16_t plen;
     uint32_t offset;
-    TRY(Read(&len));
-    TRY(Read(&plen));
-    TRY(Read(&offset));
+    TRY(ReadUint16(&len, file_));
+    TRY(ReadUint16(&plen, file_));
+    TRY(ReadUint32(&offset, file_));
 
     // Use 'len' here because a 'plen' of 0 specifies rtcp.
     len -= kPacketHeaderSize;
@@ -130,30 +198,6 @@ class RtpDumpReader : public RtpFileReaderImpl {
   }
 
  private:
-  bool Read(uint32_t* out) {
-    *out = 0;
-    for (size_t i = 0; i < 4; ++i) {
-      *out <<= 8;
-      uint8_t tmp;
-      if (fread(&tmp, 1, sizeof(uint8_t), file_) != sizeof(uint8_t))
-        return false;
-      *out |= tmp;
-    }
-    return true;
-  }
-
-  bool Read(uint16_t* out) {
-    *out = 0;
-    for (size_t i = 0; i < 2; ++i) {
-      *out <<= 8;
-      uint8_t tmp;
-      if (fread(&tmp, 1, sizeof(uint8_t), file_) != sizeof(uint8_t))
-        return false;
-      *out |= tmp;
-    }
-    return true;
-  }
-
   FILE* file_;
 
   DISALLOW_COPY_AND_ASSIGN(RtpDumpReader);
@@ -221,7 +265,7 @@ class PcapReader : public RtpFileReaderImpl {
     }
   }
 
-  bool Init(const std::string& filename) OVERRIDE {
+  bool Init(const std::string& filename) override {
     return Initialize(filename) == kResultSuccess;
   }
 
@@ -259,15 +303,15 @@ class PcapReader : public RtpFileReaderImpl {
     }
 
     printf("Total packets in file: %d\n", total_packet_count);
-    printf("Total RTP/RTCP packets: %d\n", static_cast(packets_.size()));
+    printf("Total RTP/RTCP packets: %" PRIuS "\n", packets_.size());
 
     for (SsrcMapIterator mit = packets_by_ssrc_.begin();
         mit != packets_by_ssrc_.end(); ++mit) {
       uint32_t ssrc = mit->first;
       const std::vector& packet_numbers = mit->second;
       uint8_t pt = packets_[packet_numbers[0]].rtp_header.payloadType;
-      printf("SSRC: %08x, %d packets, pt=%d\n", ssrc,
-             static_cast(packet_numbers.size()), pt);
+      printf("SSRC: %08x, %" PRIuS " packets, pt=%d\n", ssrc,
+             packet_numbers.size(), pt);
     }
 
     // TODO(solenberg): Better validation of identified SSRC streams.
@@ -290,8 +334,8 @@ class PcapReader : public RtpFileReaderImpl {
     return kResultSuccess;
   }
 
-  virtual bool NextPacket(Packet* packet) OVERRIDE {
-    uint32_t length = Packet::kMaxPacketBufferSize;
+  bool NextPacket(RtpPacket* packet) override {
+    uint32_t length = RtpPacket::kMaxPacketBufferSize;
     if (NextPcap(packet->data, &length, &packet->time_ms) != kResultSuccess)
       return false;
     packet->length = static_cast(length);
@@ -597,6 +641,9 @@ RtpFileReader* RtpFileReader::Create(FileFormat format,
     case kRtpDump:
       reader = new RtpDumpReader();
       break;
+    case kLengthPacketInterleaved:
+      reader = new InterleavedRtpFileReader();
+      break;
   }
   if (!reader->Init(filename)) {
     delete reader;
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_reader.h b/media/webrtc/trunk/webrtc/test/rtp_file_reader.h
index 095ce76726..c302d4fbde 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_reader.h
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_reader.h
@@ -16,31 +16,29 @@
 
 namespace webrtc {
 namespace test {
+
+struct RtpPacket {
+  // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
+  // some overhead.
+  static const size_t kMaxPacketBufferSize = 3500;
+  uint8_t data[kMaxPacketBufferSize];
+  size_t length;
+  // The length the packet had on wire. Will be different from |length| when
+  // reading a header-only RTP dump.
+  size_t original_length;
+
+  uint32_t time_ms;
+};
+
 class RtpFileReader {
  public:
-  enum FileFormat {
-    kPcap,
-    kRtpDump,
-  };
-
-  struct Packet {
-    // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
-    // some overhead.
-    static const size_t kMaxPacketBufferSize = 3500;
-    uint8_t data[kMaxPacketBufferSize];
-    size_t length;
-    // The length the packet had on wire. Will be different from |length| when
-    // reading a header-only RTP dump.
-    size_t original_length;
-
-    uint32_t time_ms;
-  };
+  enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
 
   virtual ~RtpFileReader() {}
   static RtpFileReader* Create(FileFormat format,
                                const std::string& filename);
 
-  virtual bool NextPacket(Packet* packet) = 0;
+  virtual bool NextPacket(RtpPacket* packet) = 0;
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc b/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc
index 54fb874b63..929813f999 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_reader_unittest.cc
@@ -11,8 +11,8 @@
 #include 
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -30,7 +30,7 @@ class TestRtpFileReader : public ::testing::Test {
   }
 
   int CountRtpPackets() {
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     int c = 0;
     while (rtp_packet_source_->NextPacket(&packet)) {
       if (headers_only_file_)
@@ -43,7 +43,7 @@ class TestRtpFileReader : public ::testing::Test {
   }
 
  private:
-  scoped_ptr rtp_packet_source_;
+  rtc::scoped_ptr rtp_packet_source_;
   bool headers_only_file_;
 };
 
@@ -71,7 +71,7 @@ class TestPcapFileReader : public ::testing::Test {
 
   int CountRtpPackets() {
     int c = 0;
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     while (rtp_packet_source_->NextPacket(&packet)) {
       EXPECT_EQ(packet.length, packet.original_length);
       c++;
@@ -81,7 +81,7 @@ class TestPcapFileReader : public ::testing::Test {
 
   PacketsPerSsrc CountRtpPacketsPerSsrc() {
     PacketsPerSsrc pps;
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     while (rtp_packet_source_->NextPacket(&packet)) {
       RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
       webrtc::RTPHeader header;
@@ -93,7 +93,7 @@ class TestPcapFileReader : public ::testing::Test {
   }
 
  private:
-  scoped_ptr rtp_packet_source_;
+  rtc::scoped_ptr rtp_packet_source_;
 };
 
 TEST_F(TestPcapFileReader, TestEthernetIIFrame) {
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc b/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc
new file mode 100644
index 0000000000..4acaa27815
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_writer.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/test/rtp_file_writer.h"
+
+#include 
+
+#include 
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/constructormagic.h"
+
+namespace webrtc {
+namespace test {
+
+static const uint16_t kPacketHeaderSize = 8;
+static const char kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
+
+// Write RTP packets to file in rtpdump format, as documented at:
+// http://www.cs.columbia.edu/irt/software/rtptools/
+class RtpDumpWriter : public RtpFileWriter {
+ public:
+  explicit RtpDumpWriter(FILE* file) : file_(file) {
+    CHECK(file_ != NULL);
+    Init();
+  }
+  virtual ~RtpDumpWriter() {
+    if (file_ != NULL) {
+      fclose(file_);
+      file_ = NULL;
+    }
+  }
+
+  bool WritePacket(const RtpPacket* packet) override {
+    uint16_t len = static_cast(packet->length + kPacketHeaderSize);
+    CHECK_GE(packet->original_length, packet->length);
+    uint16_t plen = static_cast(packet->original_length);
+    uint32_t offset = packet->time_ms;
+    CHECK(WriteUint16(len));
+    CHECK(WriteUint16(plen));
+    CHECK(WriteUint32(offset));
+    return fwrite(packet->data, sizeof(uint8_t), packet->length, file_) ==
+           packet->length;
+  }
+
+ private:
+  bool Init() {
+    fprintf(file_, "%s", kFirstLine);
+
+    CHECK(WriteUint32(0));
+    CHECK(WriteUint32(0));
+    CHECK(WriteUint32(0));
+    CHECK(WriteUint16(0));
+    CHECK(WriteUint16(0));
+
+    return true;
+  }
+
+  bool WriteUint32(uint32_t in) {
+    // Loop through shifts = {24, 16, 8, 0}.
+    for (int shifts = 24; shifts >= 0; shifts -= 8) {
+      uint8_t tmp = static_cast((in >> shifts) & 0xFF);
+      if (fwrite(&tmp, sizeof(uint8_t), 1, file_) != 1)
+        return false;
+    }
+    return true;
+  }
+
+  bool WriteUint16(uint16_t in) {
+    // Write 8 MSBs.
+    uint8_t tmp = static_cast((in >> 8) & 0xFF);
+    if (fwrite(&tmp, sizeof(uint8_t), 1, file_) != 1)
+      return false;
+    // Write 8 LSBs.
+    tmp = static_cast(in & 0xFF);
+    if (fwrite(&tmp, sizeof(uint8_t), 1, file_) != 1)
+      return false;
+    return true;
+  }
+
+  FILE* file_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpDumpWriter);
+};
+
+RtpFileWriter* RtpFileWriter::Create(FileFormat format,
+                                     const std::string& filename) {
+  FILE* file = fopen(filename.c_str(), "wb");
+  if (file == NULL) {
+    printf("ERROR: Can't open file: %s\n", filename.c_str());
+    return NULL;
+  }
+  switch (format) {
+    case kRtpDump:
+      return new RtpDumpWriter(file);
+  }
+  fclose(file);
+  return NULL;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_writer.h b/media/webrtc/trunk/webrtc/test/rtp_file_writer.h
new file mode 100644
index 0000000000..453b27762e
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_writer.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_TEST_RTP_FILE_WRITER_H_
+#define WEBRTC_TEST_RTP_FILE_WRITER_H_
+
+#include 
+
+#include "webrtc/common_types.h"
+#include "webrtc/test/rtp_file_reader.h"
+
+namespace webrtc {
+namespace test {
+class RtpFileWriter {
+ public:
+  enum FileFormat {
+    kRtpDump,
+  };
+
+  virtual ~RtpFileWriter() {}
+  static RtpFileWriter* Create(FileFormat format, const std::string& filename);
+
+  virtual bool WritePacket(const RtpPacket* packet) = 0;
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_TEST_RTP_FILE_WRITER_H_
diff --git a/media/webrtc/trunk/webrtc/test/rtp_file_writer_unittest.cc b/media/webrtc/trunk/webrtc/test/rtp_file_writer_unittest.cc
new file mode 100644
index 0000000000..2c7c88cc3f
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/test/rtp_file_writer_unittest.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include 
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/test/rtp_file_reader.h"
+#include "webrtc/test/rtp_file_writer.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+class RtpFileWriterTest : public ::testing::Test {
+ public:
+  void Init(const std::string& filename) {
+    filename_ = test::OutputPath() + filename;
+    rtp_writer_.reset(
+        test::RtpFileWriter::Create(test::RtpFileWriter::kRtpDump, filename_));
+  }
+
+  void WriteRtpPackets(int num_packets) {
+    ASSERT_TRUE(rtp_writer_.get() != NULL);
+    test::RtpPacket packet;
+    for (int i = 1; i <= num_packets; ++i) {
+      packet.length = i;
+      packet.original_length = i;
+      packet.time_ms = i;
+      memset(packet.data, i, packet.length);
+      EXPECT_TRUE(rtp_writer_->WritePacket(&packet));
+    }
+  }
+
+  void CloseOutputFile() { rtp_writer_.reset(); }
+
+  void VerifyFileContents(int expected_packets) {
+    ASSERT_TRUE(rtp_writer_.get() == NULL)
+        << "Must call CloseOutputFile before VerifyFileContents";
+    rtc::scoped_ptr rtp_reader(
+        test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filename_));
+    ASSERT_TRUE(rtp_reader.get() != NULL);
+    test::RtpPacket packet;
+    int i = 0;
+    while (rtp_reader->NextPacket(&packet)) {
+      ++i;
+      EXPECT_EQ(static_cast(i), packet.length);
+      EXPECT_EQ(static_cast(i), packet.original_length);
+      EXPECT_EQ(static_cast(i), packet.time_ms);
+      for (int j = 0; j < i; ++j) {
+        EXPECT_EQ(i, packet.data[j]);
+      }
+    }
+    EXPECT_EQ(expected_packets, i);
+  }
+
+ private:
+  rtc::scoped_ptr rtp_writer_;
+  std::string filename_;
+};
+
+TEST_F(RtpFileWriterTest, WriteToRtpDump) {
+  Init("test_rtp_file_writer.rtp");
+  WriteRtpPackets(10);
+  CloseOutputFile();
+  VerifyFileContents(10);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h b/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
index 11531b3b2d..dd731f04db 100644
--- a/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
+++ b/media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
@@ -47,7 +47,6 @@ class RtpRtcpObserver {
 
   virtual EventTypeWrapper Wait() {
     EventTypeWrapper result = observation_complete_->Wait(timeout_ms_);
-    observation_complete_->Reset();
     return result;
   }
 
@@ -128,7 +127,7 @@ class RtpRtcpObserver {
           on_rtcp_(on_rtcp) {}
 
   private:
-    virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+   bool SendRtp(const uint8_t* packet, size_t length) override {
       EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
       Action action;
       {
@@ -145,7 +144,7 @@ class RtpRtcpObserver {
       return true;  // Will never happen, makes compiler happy.
     }
 
-    virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    bool SendRtcp(const uint8_t* packet, size_t length) override {
       EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
       Action action;
       {
@@ -170,9 +169,9 @@ class RtpRtcpObserver {
   };
 
  protected:
-  const scoped_ptr crit_;
-  const scoped_ptr observation_complete_;
-  const scoped_ptr parser_;
+  const rtc::scoped_ptr crit_;
+  const rtc::scoped_ptr observation_complete_;
+  const rtc::scoped_ptr parser_;
 
  private:
   PacketTransport send_transport_, receive_transport_;
diff --git a/media/webrtc/trunk/webrtc/test/test.gyp b/media/webrtc/trunk/webrtc/test/test.gyp
index 7aba2f5739..ea831a1471 100644
--- a/media/webrtc/trunk/webrtc/test/test.gyp
+++ b/media/webrtc/trunk/webrtc/test/test.gyp
@@ -17,7 +17,8 @@
       'type': 'static_library',
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
         'channel_transport/channel_transport.cc',
@@ -60,8 +61,12 @@
         'rtcp_packet_parser.h',
         'rtp_file_reader.cc',
         'rtp_file_reader.h',
+        'rtp_file_writer.cc',
+        'rtp_file_writer.h',
       ],
       'dependencies': [
+        '<(DEPTH)/webrtc/common.gyp:webrtc_common',
+        '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       ],
     },
@@ -73,7 +78,20 @@
         'field_trial.h',
       ],
       'dependencies': [
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+      ],
+    },
+    {
+      'target_name': 'histogram',
+      'type': 'static_library',
+      'sources': [
+        'histogram.cc',
+        'histogram.h',
+      ],
+      'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
     },
     {
@@ -84,9 +102,9 @@
       ],
       'dependencies': [
         'field_trial',
+        'histogram',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:metrics_default',
       ],
     },
     {
@@ -95,7 +113,8 @@
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/testing/gmock.gyp:gmock',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/common.gyp:gtest_prod',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
         'testsupport/fileutils.cc',
@@ -104,7 +123,6 @@
         'testsupport/frame_reader.h',
         'testsupport/frame_writer.cc',
         'testsupport/frame_writer.h',
-        'testsupport/gtest_prod_util.h',
         'testsupport/gtest_disable.h',
         'testsupport/mock/mock_frame_reader.h',
         'testsupport/mock/mock_frame_writer.h',
@@ -123,11 +141,11 @@
       'type': 'static_library',
       'dependencies': [
         'field_trial',
+        'histogram',
         'test_support',
         '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:metrics_default',
       ],
       'sources': [
         'run_all_unittests.cc',
diff --git a/media/webrtc/trunk/webrtc/test/test_suite.h b/media/webrtc/trunk/webrtc/test/test_suite.h
index e28fb23353..c166f283a5 100644
--- a/media/webrtc/trunk/webrtc/test/test_suite.h
+++ b/media/webrtc/trunk/webrtc/test/test_suite.h
@@ -18,7 +18,7 @@
 // any gtest based tests that are linked into your executable.
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 namespace test {
@@ -41,7 +41,7 @@ class TestSuite {
   DISALLOW_COPY_AND_ASSIGN(TestSuite);
 
  private:
-  scoped_ptr trace_to_stderr_;
+  rtc::scoped_ptr trace_to_stderr_;
 };
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/test/test_support_unittests.isolate b/media/webrtc/trunk/webrtc/test/test_support_unittests.isolate
index c1419e7e60..3fd89d602e 100644
--- a/media/webrtc/trunk/webrtc/test/test_support_unittests.isolate
+++ b/media/webrtc/trunk/webrtc/test/test_support_unittests.isolate
@@ -7,14 +7,6 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
-      'variables': {
-        'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
-        ],
-      },
-    }],
     ['OS=="linux" or OS=="mac" or OS=="win"', {
       'variables': {
         'command': [
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc b/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc
index 36ca391c41..8301e77165 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/fileutils.cc
@@ -23,7 +23,7 @@
 #else
 #include 
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #define GET_CURRENT_DIR getcwd
 #endif
 
@@ -168,7 +168,7 @@ std::string TempFilename(const std::string &dir, const std::string &prefix) {
   return "";
 #else
   int len = dir.size() + prefix.size() + 2 + 6;
-  scoped_ptr tempname(new char[len]);
+  rtc::scoped_ptr tempname(new char[len]);
 
   snprintf(tempname.get(), len, "%s/%sXXXXXX", dir.c_str(),
            prefix.c_str());
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/frame_reader.h b/media/webrtc/trunk/webrtc/test/testsupport/frame_reader.h
index 4ee90f90b4..9b50ec7bf4 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/frame_reader.h
+++ b/media/webrtc/trunk/webrtc/test/testsupport/frame_reader.h
@@ -54,12 +54,12 @@ class FrameReaderImpl : public FrameReader {
   //   frame_length_in_bytes   The size of each frame.
   //                           For YUV this is 3 * width * height / 2
   FrameReaderImpl(std::string input_filename, size_t frame_length_in_bytes);
-  virtual ~FrameReaderImpl();
-  virtual bool Init() OVERRIDE;
-  virtual bool ReadFrame(uint8_t* source_buffer) OVERRIDE;
-  virtual void Close() OVERRIDE;
-  virtual size_t FrameLength() OVERRIDE;
-  virtual int NumberOfFrames() OVERRIDE;
+  ~FrameReaderImpl() override;
+  bool Init() override;
+  bool ReadFrame(uint8_t* source_buffer) override;
+  void Close() override;
+  size_t FrameLength() override;
+  int NumberOfFrames() override;
 
  private:
   std::string input_filename_;
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/frame_writer.h b/media/webrtc/trunk/webrtc/test/testsupport/frame_writer.h
index daa3894a50..8a6b1c2152 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/frame_writer.h
+++ b/media/webrtc/trunk/webrtc/test/testsupport/frame_writer.h
@@ -52,11 +52,11 @@ class FrameWriterImpl : public FrameWriter {
   //   frame_length_in_bytes   The size of each frame.
   //                           For YUV: 3*width*height/2
   FrameWriterImpl(std::string output_filename, size_t frame_length_in_bytes);
-  virtual ~FrameWriterImpl();
-  virtual bool Init() OVERRIDE;
-  virtual bool WriteFrame(uint8_t* frame_buffer) OVERRIDE;
-  virtual void Close() OVERRIDE;
-  virtual size_t FrameLength() OVERRIDE;
+  ~FrameWriterImpl() override;
+  bool Init() override;
+  bool WriteFrame(uint8_t* frame_buffer) override;
+  void Close() override;
+  size_t FrameLength() override;
 
  private:
   std::string output_filename_;
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc b/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc
index 1e19806b4d..0202a71ebe 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/metrics/video_metrics.cc
@@ -111,8 +111,8 @@ int CalculateMetrics(VideoMetricsType video_metrics_type,
   const size_t frame_length = 3 * width * height >> 1;
   I420VideoFrame ref_frame;
   I420VideoFrame test_frame;
-  scoped_ptr ref_buffer(new uint8_t[frame_length]);
-  scoped_ptr test_buffer(new uint8_t[frame_length]);
+  rtc::scoped_ptr ref_buffer(new uint8_t[frame_length]);
+  rtc::scoped_ptr test_buffer(new uint8_t[frame_length]);
 
   // Set decoded image parameters.
   int half_width = (width + 1) / 2;
@@ -124,9 +124,9 @@ int CalculateMetrics(VideoMetricsType video_metrics_type,
   while (ref_bytes == frame_length && test_bytes == frame_length) {
     // Converting from buffer to plane representation.
     ConvertToI420(kI420, ref_buffer.get(), 0, 0, width, height, 0,
-                  kRotateNone, &ref_frame);
+                  kVideoRotation_0, &ref_frame);
     ConvertToI420(kI420, test_buffer.get(), 0, 0, width, height, 0,
-                  kRotateNone, &test_frame);
+                  kVideoRotation_0, &test_frame);
     switch (video_metrics_type) {
       case kPSNR:
         CalculateFrame(kPSNR, &ref_frame, &test_frame, frame_number,
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.cc b/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.cc
index 54552bd0f5..e27ec22f16 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.cc
@@ -12,6 +12,7 @@
 
 #include 
 #include 
+#include 
 
 namespace webrtc {
 namespace test {
@@ -22,10 +23,9 @@ PacketReader::PacketReader()
 PacketReader::~PacketReader() {}
 
 void PacketReader::InitializeReading(uint8_t* data,
-                                     int data_length_in_bytes,
-                                     int packet_size_in_bytes) {
+                                     size_t data_length_in_bytes,
+                                     size_t packet_size_in_bytes) {
   assert(data);
-  assert(data_length_in_bytes >= 0);
   assert(packet_size_in_bytes > 0);
   data_ = data;
   data_length_ = data_length_in_bytes;
@@ -40,16 +40,9 @@ int PacketReader::NextPacket(uint8_t** packet_pointer) {
     return -1;
   }
   *packet_pointer = data_ + currentIndex_;
-  // Check if we're about to read the last packet:
-  if (data_length_ - currentIndex_ <= packet_size_) {
-    int size = data_length_ - currentIndex_;
-    currentIndex_ = data_length_;
-    assert(size >= 0);
-    return size;
-  }
-  currentIndex_ += packet_size_;
-  assert(packet_size_ >= 0);
-  return packet_size_;
+  size_t old_index = currentIndex_;
+  currentIndex_ = std::min(currentIndex_ + packet_size_, data_length_);
+  return static_cast(currentIndex_ - old_index);
 }
 
 }  // namespace test
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.h b/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.h
index e84ecd96b5..b58db4d093 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.h
+++ b/media/webrtc/trunk/webrtc/test/testsupport/packet_reader.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
 #define WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
 
+#include 
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -24,12 +25,12 @@ class PacketReader {
 
   // Inizializes a new reading operation. Must be done before invoking the
   // NextPacket method.
-  // * data_length_in_bytes is the length of the data byte array. Must be >= 0.
+  // * data_length_in_bytes is the length of the data byte array.
   //   0 length will result in no packets are read.
   // * packet_size_in_bytes is the number of bytes to read in each NextPacket
   //   method call. Must be > 0
-  virtual void InitializeReading(uint8_t* data, int data_length_in_bytes,
-                                 int packet_size_in_bytes);
+  virtual void InitializeReading(uint8_t* data, size_t data_length_in_bytes,
+                                 size_t packet_size_in_bytes);
 
   // Moves the supplied pointer to the beginning of the next packet.
   // Returns:
@@ -41,9 +42,9 @@ class PacketReader {
 
  private:
   uint8_t* data_;
-  int data_length_;
-  int packet_size_;
-  int currentIndex_;
+  size_t data_length_;
+  size_t packet_size_;
+  size_t currentIndex_;
   bool initialized_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/packet_reader_unittest.cc b/media/webrtc/trunk/webrtc/test/testsupport/packet_reader_unittest.cc
index 83cde6957d..2679be473f 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/packet_reader_unittest.cc
+++ b/media/webrtc/trunk/webrtc/test/testsupport/packet_reader_unittest.cc
@@ -26,11 +26,11 @@ class PacketReaderTest: public PacketRelatedTest {
   void TearDown() {
     delete reader_;
   }
-  void VerifyPacketData(int expected_length,
+  void VerifyPacketData(size_t expected_length,
                         int actual_length,
                         uint8_t* original_data_pointer,
                         uint8_t* new_data_pointer) {
-    EXPECT_EQ(expected_length, actual_length);
+    EXPECT_EQ(static_cast(expected_length), actual_length);
     EXPECT_EQ(*original_data_pointer, *new_data_pointer);
     EXPECT_EQ(0, memcmp(original_data_pointer, new_data_pointer,
                         actual_length));
@@ -82,7 +82,7 @@ TEST_F(PacketReaderTest, NormalOnePacketData) {
   // Reading another one shall result in 0 bytes:
   length_to_read = reader_->NextPacket(&data_pointer);
   EXPECT_EQ(0, length_to_read);
-  EXPECT_EQ(kPacketSizeInBytes, data_pointer - data);
+  EXPECT_EQ(kPacketSizeInBytes, static_cast(data_pointer - data));
 }
 
 // Test with data length that will result in 3 packets
@@ -105,7 +105,8 @@ TEST_F(PacketReaderTest, NormalLargeData) {
   // Reading another one shall result in 0 bytes:
   length_to_read = reader_->NextPacket(&packet_data_pointer_);
   EXPECT_EQ(0, length_to_read);
-  EXPECT_EQ(kPacketDataLength, packet_data_pointer_ - packet_data_);
+  EXPECT_EQ(kPacketDataLength,
+            static_cast(packet_data_pointer_ - packet_data_));
 }
 
 // Test with empty data.
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h b/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h
index 12208349d8..88f1811740 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h
+++ b/media/webrtc/trunk/webrtc/test/testsupport/trace_to_stderr.h
@@ -28,7 +28,7 @@ class TraceToStderr : public TraceCallback {
   // This is useful for offline test tools, where the file time is much more
   // informative than the real time.
   explicit TraceToStderr(bool override_time);
-  virtual ~TraceToStderr();
+  ~TraceToStderr() override;
 
   // Every subsequent trace printout will use |time|. Has no effect if
   // |override_time| in the constructor was set to false.
@@ -39,9 +39,7 @@ class TraceToStderr : public TraceCallback {
   virtual void SetTimeSeconds(float time);
 
   // Implements TraceCallback.
-  virtual void Print(TraceLevel level,
-                     const char* msg_array,
-                     int length) OVERRIDE;
+  void Print(TraceLevel level, const char* msg_array, int length) override;
 
  private:
   bool override_time_;
diff --git a/media/webrtc/trunk/webrtc/test/testsupport/unittest_utils.h b/media/webrtc/trunk/webrtc/test/testsupport/unittest_utils.h
index bb244a441a..ba6db9883b 100644
--- a/media/webrtc/trunk/webrtc/test/testsupport/unittest_utils.h
+++ b/media/webrtc/trunk/webrtc/test/testsupport/unittest_utils.h
@@ -14,8 +14,8 @@
 namespace webrtc {
 namespace test {
 
-const int kPacketSizeInBytes = 1500;
-const int kPacketDataLength = kPacketSizeInBytes * 2 + 1;
+const size_t kPacketSizeInBytes = 1500;
+const size_t kPacketDataLength = kPacketSizeInBytes * 2 + 1;
 const int kPacketDataNumberOfPackets = 3;
 
 // A base test fixture for packet related tests. Contains
diff --git a/media/webrtc/trunk/webrtc/test/vcm_capturer.cc b/media/webrtc/trunk/webrtc/test/vcm_capturer.cc
index a5820bfe11..f9976d21c0 100644
--- a/media/webrtc/trunk/webrtc/test/vcm_capturer.cc
+++ b/media/webrtc/trunk/webrtc/test/vcm_capturer.cc
@@ -87,9 +87,9 @@ void VcmCapturer::Destroy() {
 VcmCapturer::~VcmCapturer() { Destroy(); }
 
 void VcmCapturer::OnIncomingCapturedFrame(const int32_t id,
-                                          I420VideoFrame& frame) {
+                                          const I420VideoFrame& frame) {
   if (started_)
-    input_->SwapFrame(&frame);
+    input_->IncomingCapturedFrame(frame);
 }
 
 void VcmCapturer::OnCaptureDelayChanged(const int32_t id, const int32_t delay) {
diff --git a/media/webrtc/trunk/webrtc/test/vcm_capturer.h b/media/webrtc/trunk/webrtc/test/vcm_capturer.h
index dde3edc2f7..c73eeb1ce9 100644
--- a/media/webrtc/trunk/webrtc/test/vcm_capturer.h
+++ b/media/webrtc/trunk/webrtc/test/vcm_capturer.h
@@ -24,13 +24,12 @@ class VcmCapturer : public VideoCapturer, public VideoCaptureDataCallback {
                              size_t height, size_t target_fps);
   virtual ~VcmCapturer();
 
-  virtual void Start() OVERRIDE;
-  virtual void Stop() OVERRIDE;
+  void Start() override;
+  void Stop() override;
 
-  virtual void OnIncomingCapturedFrame(
-      const int32_t id, I420VideoFrame& frame) OVERRIDE;  // NOLINT
-  virtual void OnCaptureDelayChanged(const int32_t id, const int32_t delay)
-      OVERRIDE;
+  void OnIncomingCapturedFrame(const int32_t id,
+                               const I420VideoFrame& frame) override;  // NOLINT
+  void OnCaptureDelayChanged(const int32_t id, const int32_t delay) override;
 
  private:
   explicit VcmCapturer(VideoSendStreamInput* input);
diff --git a/media/webrtc/trunk/webrtc/test/video_renderer.cc b/media/webrtc/trunk/webrtc/test/video_renderer.cc
index 1a0c6133ce..03c4948910 100644
--- a/media/webrtc/trunk/webrtc/test/video_renderer.cc
+++ b/media/webrtc/trunk/webrtc/test/video_renderer.cc
@@ -17,8 +17,9 @@ namespace webrtc {
 namespace test {
 
 class NullRenderer : public VideoRenderer {
-  virtual void RenderFrame(const I420VideoFrame& video_frame,
-                           int time_to_render_ms) OVERRIDE {}
+  void RenderFrame(const I420VideoFrame& video_frame,
+                   int time_to_render_ms) override {}
+  bool IsTextureSupported() const override { return false; }
 };
 
 VideoRenderer* VideoRenderer::Create(const char* window_title,
diff --git a/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp b/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp
index cd076639c4..ea6e4052db 100644
--- a/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp
+++ b/media/webrtc/trunk/webrtc/test/webrtc_test_common.gyp
@@ -57,8 +57,9 @@
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         '<(webrtc_root)/base/base.gyp:rtc_base',
+        '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:media_file',
-        '<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
+        '<(webrtc_root)/modules/modules.gyp:video_render',
         '<(webrtc_root)/test/test.gyp:frame_generator',
         '<(webrtc_root)/test/test.gyp:test_support',
         '<(webrtc_root)/test/test.gyp:rtp_test_utils',
@@ -151,12 +152,14 @@
             'webrtc_test_common',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/testing/gmock.gyp:gmock',
-            '<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
+            '<(webrtc_root)/modules/modules.gyp:video_capture',
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'sources': [
             'fake_network_pipe_unittest.cc',
+            'frame_generator_unittest.cc',
             'rtp_file_reader_unittest.cc',
+            'rtp_file_writer_unittest.cc',
           ],
         },
       ],  #targets
diff --git a/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h b/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
index 7f375351a2..4fa6c7ef46 100644
--- a/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
+++ b/media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
@@ -27,8 +27,9 @@ class D3dRenderer : public VideoRenderer {
                              size_t height);
   virtual ~D3dRenderer();
 
-  virtual void RenderFrame(const webrtc::I420VideoFrame& frame, int delta)
-      OVERRIDE;
+  void RenderFrame(const webrtc::I420VideoFrame& frame, int delta) override;
+  bool IsTextureSupported() const override { return false; }
+
  private:
   D3dRenderer(size_t width, size_t height);
 
diff --git a/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc b/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc
new file mode 100644
index 0000000000..a51216acd9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/activity_metric.cc
@@ -0,0 +1,384 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include 
+#include 
+#include 
+
+#include 
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_processing/agc/agc.h"
+#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
+#include "webrtc/modules/audio_processing/agc/common.h"
+#include "webrtc/modules/audio_processing/agc/histogram.h"
+#include "webrtc/modules/audio_processing/agc/pitch_based_vad.h"
+#include "webrtc/modules/audio_processing/agc/standalone_vad.h"
+#include "webrtc/modules/audio_processing/agc/utility.h"
+#include "webrtc/modules/interface/module_common_types.h"
+
+static const int kAgcAnalWindowSamples = 100;
+static const double kDefaultActivityThreshold = 0.3;
+
+DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
+DEFINE_string(true_vad, "", "name of a file containing true VAD in 'int'"
+              " format");
+DEFINE_string(video_vad, "", "name of a file containing video VAD (activity"
+              " probabilities) in double format. One activity per 10ms is"
+              " required. If no file is given the video information is not"
+              " incorporated. Negative activity is interpreted as video is"
+              " not adapted and the statistics are not computed during"
+              " the learning phase. Note that the negative video activities"
+              " are ONLY allowed at the beginning.");
+DEFINE_string(result, "", "name of a file to write the results. The results"
+              " will be appended to the end of the file. This is optional.");
+DEFINE_string(audio_content, "", "name of a file where audio content is written"
+              " to, in double format.");
+DEFINE_double(activity_threshold, kDefaultActivityThreshold,
+              "Activity threshold");
+
+namespace webrtc {
+
+// TODO(turajs) A new CL will be committed soon where ExtractFeatures will
+// notify the caller of "silence" input, instead of bailing out. We would not
+// need the following function when such a change is made.
+
+// Add some dither to quiet frames. This avoids the ExtractFeatures skip a
+// silence frame. Otherwise true VAD would drift with respect to the audio.
+// We only consider mono inputs.
+static void DitherSilence(AudioFrame* frame) {
+  ASSERT_EQ(1, frame->num_channels_);
+  const double kRmsSilence = 5;
+  const double sum_squared_silence = kRmsSilence * kRmsSilence *
+      frame->samples_per_channel_;
+  double sum_squared = 0;
+  for (int n = 0; n < frame->samples_per_channel_; n++)
+    sum_squared += frame->data_[n] * frame->data_[n];
+  if (sum_squared <= sum_squared_silence) {
+    for (int n = 0; n < frame->samples_per_channel_; n++)
+      frame->data_[n] = (rand() & 0xF) - 8;
+  }
+}
+
+class AgcStat {
+ public:
+  AgcStat()
+      : video_index_(0),
+        activity_threshold_(kDefaultActivityThreshold),
+        audio_content_(Histogram::Create(kAgcAnalWindowSamples)),
+        audio_processing_(new AgcAudioProc()),
+        vad_(new PitchBasedVad()),
+        standalone_vad_(StandaloneVad::Create()),
+        audio_content_fid_(NULL) {
+    for (int n = 0; n < kMaxNumFrames; n++)
+      video_vad_[n] = 0.5;
+  }
+
+  ~AgcStat() {
+    if (audio_content_fid_ != NULL) {
+      fclose(audio_content_fid_);
+    }
+  }
+
+  void set_audio_content_file(FILE* audio_content_fid) {
+    audio_content_fid_ = audio_content_fid;
+  }
+
+  int AddAudio(const AudioFrame& frame, double p_video,
+               int* combined_vad) {
+    if (frame.num_channels_ != 1 ||
+        frame.samples_per_channel_ !=
+            kSampleRateHz / 100 ||
+            frame.sample_rate_hz_ != kSampleRateHz)
+      return -1;
+    video_vad_[video_index_++] = p_video;
+    AudioFeatures features;
+    audio_processing_->ExtractFeatures(
+        frame.data_, frame.samples_per_channel_, &features);
+    if (FLAGS_standalone_vad) {
+      standalone_vad_->AddAudio(frame.data_,
+                                frame.samples_per_channel_);
+    }
+    if (features.num_frames > 0) {
+      double p[kMaxNumFrames] = {0.5, 0.5, 0.5, 0.5};
+      if (FLAGS_standalone_vad) {
+        standalone_vad_->GetActivity(p, kMaxNumFrames);
+      }
+      // TODO(turajs) combining and limiting are used in the source files as
+      // well they can be moved to utility.
+      // Combine Video and stand-alone VAD.
+      for (int n = 0; n < features.num_frames; n++) {
+        double p_active = p[n] * video_vad_[n];
+        double p_passive = (1 - p[n]) * (1 - video_vad_[n]);
+        p[n]  = p_active / (p_active + p_passive);
+        // Limit probabilities.
+        p[n] = std::min(std::max(p[n], 0.01), 0.99);
+      }
+      if (vad_->VoicingProbability(features, p) < 0)
+        return -1;
+      for (int n = 0; n < features.num_frames; n++) {
+        audio_content_->Update(features.rms[n], p[n]);
+        double ac = audio_content_->AudioContent();
+        if (audio_content_fid_ != NULL) {
+          fwrite(&ac, sizeof(ac), 1, audio_content_fid_);
+        }
+        if (ac > kAgcAnalWindowSamples * activity_threshold_) {
+          combined_vad[n] = 1;
+        } else {
+          combined_vad[n] = 0;
+        }
+      }
+      video_index_ = 0;
+    }
+    return features.num_frames;
+  }
+
+  void Reset() {
+    audio_content_->Reset();
+  }
+
+  void SetActivityThreshold(double activity_threshold) {
+    activity_threshold_ = activity_threshold;
+  }
+
+ private:
+  int video_index_;
+  double activity_threshold_;
+  double video_vad_[kMaxNumFrames];
+  rtc::scoped_ptr audio_content_;
+  rtc::scoped_ptr audio_processing_;
+  rtc::scoped_ptr vad_;
+  rtc::scoped_ptr standalone_vad_;
+
+  FILE* audio_content_fid_;
+};
+
+
+void void_main(int argc, char* argv[]) {
+  webrtc::AgcStat agc_stat;
+
+  FILE* pcm_fid = fopen(argv[1], "rb");
+  ASSERT_TRUE(pcm_fid != NULL) << "Cannot open PCM file " << argv[1];
+
+  if (argc < 2) {
+    fprintf(stderr, "\nNot Enough arguments\n");
+  }
+
+  FILE* true_vad_fid = NULL;
+  ASSERT_GT(FLAGS_true_vad.size(), 0u) << "Specify the file containing true "
+      "VADs using --true_vad flag.";
+  true_vad_fid = fopen(FLAGS_true_vad.c_str(), "rb");
+  ASSERT_TRUE(true_vad_fid != NULL) << "Cannot open the active list " <<
+      FLAGS_true_vad;
+
+  FILE* results_fid = NULL;
+  if (FLAGS_result.size() > 0) {
+    // True if this is the first time writing to this function and we add a
+    // header to the beginning of the file.
+    bool write_header;
+    // Open in the read mode. If it fails, the file doesn't exist and has to
+    // write a header for it. Otherwise no need to write a header.
+    results_fid = fopen(FLAGS_result.c_str(), "r");
+    if (results_fid == NULL) {
+      write_header = true;
+    } else {
+      fclose(results_fid);
+      write_header = false;
+    }
+    // Open in append mode.
+    results_fid = fopen(FLAGS_result.c_str(), "a");
+    ASSERT_TRUE(results_fid != NULL) << "Cannot open the file, " <<
+              FLAGS_result << ", to write the results.";
+    // Write the header if required.
+    if (write_header) {
+      fprintf(results_fid, "%% Total Active,  Misdetection,  "
+              "Total inactive,  False Positive,  On-sets,  Missed segments,  "
+              "Average response\n");
+    }
+  }
+
+  FILE* video_vad_fid = NULL;
+  if (FLAGS_video_vad.size() > 0) {
+    video_vad_fid = fopen(FLAGS_video_vad.c_str(), "rb");
+    ASSERT_TRUE(video_vad_fid != NULL) <<  "Cannot open the file, " <<
+              FLAGS_video_vad << " to read video-based VAD decisions.\n";
+  }
+
+  // AgsStat will be the owner of this file and will close it at its
+  // destructor.
+  FILE* audio_content_fid = NULL;
+  if (FLAGS_audio_content.size() > 0) {
+    audio_content_fid = fopen(FLAGS_audio_content.c_str(), "wb");
+    ASSERT_TRUE(audio_content_fid != NULL) << "Cannot open file, " <<
+              FLAGS_audio_content << " to write audio-content.\n";
+    agc_stat.set_audio_content_file(audio_content_fid);
+  }
+
+  webrtc::AudioFrame frame;
+  frame.num_channels_ = 1;
+  frame.sample_rate_hz_ = 16000;
+  frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
+  const size_t kSamplesToRead = frame.num_channels_ *
+      frame.samples_per_channel_;
+
+  agc_stat.SetActivityThreshold(FLAGS_activity_threshold);
+
+  int ret_val = 0;
+  int num_frames = 0;
+  int agc_vad[kMaxNumFrames];
+  uint8_t true_vad[kMaxNumFrames];
+  double p_video = 0.5;
+  int total_active = 0;
+  int total_passive = 0;
+  int total_false_positive = 0;
+  int total_missed_detection = 0;
+  int onset_adaptation = 0;
+  int num_onsets = 0;
+  bool onset = false;
+  uint8_t previous_true_vad = 0;
+  int num_not_adapted = 0;
+  int true_vad_index = 0;
+  bool in_false_positive_region = false;
+  int total_false_positive_duration = 0;
+  bool video_adapted = false;
+  while (kSamplesToRead == fread(frame.data_, sizeof(int16_t),
+                                 kSamplesToRead, pcm_fid)) {
+    assert(true_vad_index < kMaxNumFrames);
+    ASSERT_EQ(1u, fread(&true_vad[true_vad_index], sizeof(*true_vad), 1,
+                        true_vad_fid))
+        << "Size mismatch between True-VAD and the PCM file.\n";
+    if (video_vad_fid != NULL) {
+      ASSERT_EQ(1u, fread(&p_video, sizeof(p_video), 1, video_vad_fid)) <<
+          "Not enough video-based VAD probabilities.";
+    }
+
+    // Negative video activity indicates that the video-based VAD is not yet
+    // adapted. Disregards the learning phase in statistics.
+    if (p_video < 0) {
+      if (video_adapted) {
+        fprintf(stderr, "Negative video probabilities ONLY allowed at the "
+            "beginning of the sequence, not in the middle.\n");
+        exit(1);
+      }
+      continue;
+    } else {
+      video_adapted = true;
+    }
+
+    num_frames++;
+    uint8_t last_true_vad;
+    if (true_vad_index == 0) {
+      last_true_vad = previous_true_vad;
+    } else {
+      last_true_vad = true_vad[true_vad_index - 1];
+    }
+    if (last_true_vad == 1 && true_vad[true_vad_index] == 0) {
+      agc_stat.Reset();
+    }
+    true_vad_index++;
+
+    DitherSilence(&frame);
+
+    ret_val = agc_stat.AddAudio(frame, p_video, agc_vad);
+    ASSERT_GE(ret_val, 0);
+
+    if (ret_val > 0) {
+      ASSERT_TRUE(ret_val == true_vad_index);
+      for (int n = 0; n < ret_val; n++) {
+        if (true_vad[n] == 1) {
+          total_active++;
+          if (previous_true_vad == 0) {
+            num_onsets++;
+            onset = true;
+          }
+          if (agc_vad[n] == 0) {
+            total_missed_detection++;
+            if (onset)
+              onset_adaptation++;
+          } else {
+            in_false_positive_region = false;
+            onset = false;
+          }
+        } else if (true_vad[n] == 0) {
+          // Check if |on_set| flag is still up. If so it means that we totally
+          // missed an active region
+          if (onset)
+            num_not_adapted++;
+          onset = false;
+
+          total_passive++;
+          if (agc_vad[n] == 1) {
+            total_false_positive++;
+            in_false_positive_region = true;
+          }
+          if (in_false_positive_region) {
+            total_false_positive_duration++;
+          }
+        } else {
+          ASSERT_TRUE(false) << "Invalid value for true-VAD.\n";
+        }
+        previous_true_vad = true_vad[n];
+      }
+      true_vad_index = 0;
+    }
+  }
+
+  if (results_fid != NULL) {
+    fprintf(results_fid, "%4d  %4d  %4d  %4d  %4d  %4d  %4.0f %4.0f\n",
+            total_active,
+            total_missed_detection,
+            total_passive,
+            total_false_positive,
+            num_onsets,
+            num_not_adapted,
+            static_cast(onset_adaptation) / (num_onsets + 1e-12),
+            static_cast(total_false_positive_duration) /
+            (total_passive + 1e-12));
+  }
+  fprintf(stdout, "%4d %4d %4d %4d %4d %4d %4.0f %4.0f\n",
+          total_active,
+          total_missed_detection,
+          total_passive,
+          total_false_positive,
+          num_onsets,
+          num_not_adapted,
+          static_cast(onset_adaptation) / (num_onsets + 1e-12),
+          static_cast(total_false_positive_duration) /
+              (total_passive + 1e-12));
+
+  fclose(true_vad_fid);
+  fclose(pcm_fid);
+  if (video_vad_fid != NULL) {
+    fclose(video_vad_fid);
+  }
+  if (results_fid != NULL) {
+    fclose(results_fid);
+  }
+}
+
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  char kUsage[] =
+      "\nCompute the number of misdetected and false-positive frames. Not\n"
+      " that for each frame of audio (10 ms) there should be one true\n"
+      " activity. If any video-based activity is given, there should also be\n"
+      " one probability per frame.\n"
+      "\nUsage:\n\n"
+      "activity_metric input_pcm [options]\n"
+      "where 'input_pcm' is the input audio sampled at 16 kHz in 16 bits "
+      "format.\n\n";
+  google::SetUsageMessage(kUsage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::void_main(argc, argv);
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc
new file mode 100644
index 0000000000..ae8d942295
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_harness.cc
@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Refer to kUsage below for a description.
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+#include "webrtc/tools/agc/agc_manager.h"
+#include "webrtc/voice_engine/include/voe_audio_processing.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+#include "webrtc/voice_engine/include/voe_file.h"
+#include "webrtc/voice_engine/include/voe_hardware.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+
+DEFINE_bool(codecs, false, "print out available codecs");
+DEFINE_int32(pt, 103, "codec payload type (defaults to ISAC/16000/1)");
+DEFINE_bool(internal, true, "use the internal AGC in 'serial' mode, or as the "
+                            "first voice engine's AGC in parallel mode");
+DEFINE_bool(parallel, false, "run internal and public AGCs in parallel, with "
+    "left- and right-panning respectively. Not compatible with -aec.");
+DEFINE_bool(devices, false, "print out capture devices and indexes to be used "
+                            "with the capture flags");
+DEFINE_int32(capture1, 0, "capture device index for the first voice engine");
+DEFINE_int32(capture2, 0, "capture device index for second voice engine");
+DEFINE_int32(render1, 0, "render device index for first voice engine");
+DEFINE_int32(render2, 0, "render device index for second voice engine");
+DEFINE_bool(aec, false, "runs two voice engines in parallel, with the first "
+    "playing out a file and sending its captured signal to the second voice "
+    "engine. Also enables echo cancellation.");
+DEFINE_bool(ns, true, "enable noise suppression");
+DEFINE_bool(highpass, true, "enable high pass filter");
+DEFINE_string(filename, "", "filename for the -aec mode");
+
+namespace webrtc {
+namespace {
+
+const char kUsage[] =
+    "\nWithout additional flags, sets up a simple VoiceEngine loopback call\n"
+    "with the default audio devices and runs forever. The internal AGC is\n"
+    "enabled and the public disabled.\n\n"
+
+    "It can also run the public AGC in parallel with the internal, panned to\n"
+    "opposite stereo channels on the default render device. The capture\n"
+    "devices for each can be selected (recommended, because otherwise they\n"
+    "will fight for the level on the same device).\n\n"
+
+    "Lastly, it can be used for local AEC testing. In this mode, the first\n"
+    "voice engine plays out a file over the selected render device (normally\n"
+    "loudspeakers) and records from the selected capture device. The second\n"
+    "voice engine receives the capture signal and plays it out over the\n"
+    "selected render device (normally headphones). This allows the user to\n"
+    "test an echo scenario with the first voice engine, while monitoring the\n"
+    "result with the second.";
+
+class AgcVoiceEngine {
+ public:
+  enum Pan {
+    NoPan,
+    PanLeft,
+    PanRight
+  };
+
+  AgcVoiceEngine(bool internal, int tx_port, int rx_port, int capture_idx,
+                 int render_idx)
+      : voe_(VoiceEngine::Create()),
+        base_(VoEBase::GetInterface(voe_)),
+        hardware_(VoEHardware::GetInterface(voe_)),
+        codec_(VoECodec::GetInterface(voe_)),
+        manager_(new AgcManager(voe_)),
+        channel_(-1),
+        capture_idx_(capture_idx),
+        render_idx_(render_idx) {
+    SetUp(internal, tx_port, rx_port);
+  }
+
+  ~AgcVoiceEngine() {
+    TearDown();
+  }
+
+  void SetUp(bool internal, int tx_port, int rx_port) {
+    ASSERT_TRUE(voe_ != NULL);
+    ASSERT_TRUE(base_ != NULL);
+    ASSERT_TRUE(hardware_ != NULL);
+    ASSERT_TRUE(codec_ != NULL);
+    VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe_);
+    ASSERT_TRUE(audio != NULL);
+    VoENetwork* network = VoENetwork::GetInterface(voe_);
+    ASSERT_TRUE(network != NULL);
+
+    ASSERT_EQ(0, base_->Init());
+    channel_ = base_->CreateChannel();
+    ASSERT_NE(-1, channel_);
+
+    channel_transport_.reset(
+        new test::VoiceChannelTransport(network, channel_));
+    ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", tx_port));
+    ASSERT_EQ(0, channel_transport_->SetLocalReceiver(rx_port));
+
+    ASSERT_EQ(0, hardware_->SetRecordingDevice(capture_idx_));
+    ASSERT_EQ(0, hardware_->SetPlayoutDevice(render_idx_));
+
+    CodecInst codec_params = {0};
+    bool codec_found = false;
+    for (int i = 0; i < codec_->NumOfCodecs(); i++) {
+      ASSERT_EQ(0, codec_->GetCodec(i, codec_params));
+      if (FLAGS_pt == codec_params.pltype) {
+        codec_found = true;
+        break;
+      }
+    }
+    ASSERT_TRUE(codec_found);
+    ASSERT_EQ(0, codec_->SetSendCodec(channel_, codec_params));
+
+    ASSERT_EQ(0, audio->EnableHighPassFilter(FLAGS_highpass));
+    ASSERT_EQ(0, audio->SetNsStatus(FLAGS_ns));
+    ASSERT_EQ(0, audio->SetEcStatus(FLAGS_aec));
+
+    ASSERT_EQ(0, manager_->Enable(internal));
+    ASSERT_EQ(0, audio->SetAgcStatus(!internal));
+
+    audio->Release();
+    network->Release();
+  }
+
+  void TearDown() {
+    Stop();
+    channel_transport_.reset(NULL);
+    ASSERT_EQ(0, base_->DeleteChannel(channel_));
+    ASSERT_EQ(0, base_->Terminate());
+    // Don't test; the manager hasn't released its interfaces.
+    hardware_->Release();
+    base_->Release();
+    codec_->Release();
+    delete manager_;
+    ASSERT_TRUE(VoiceEngine::Delete(voe_));
+  }
+
+  void PrintDevices() {
+    int num_devices = 0;
+    char device_name[128] = {0};
+    char guid[128] = {0};
+    ASSERT_EQ(0, hardware_->GetNumOfRecordingDevices(num_devices));
+    printf("Capture devices:\n");
+    for (int i = 0; i < num_devices; i++) {
+      ASSERT_EQ(0, hardware_->GetRecordingDeviceName(i, device_name, guid));
+      printf("%d: %s\n", i, device_name);
+    }
+    ASSERT_EQ(0, hardware_->GetNumOfPlayoutDevices(num_devices));
+    printf("Render devices:\n");
+    for (int i = 0; i < num_devices; i++) {
+      ASSERT_EQ(0, hardware_->GetPlayoutDeviceName(i, device_name, guid));
+      printf("%d: %s\n", i, device_name);
+    }
+  }
+
+  void PrintCodecs() {
+    CodecInst params = {0};
+    printf("Codecs:\n");
+    for (int i = 0; i < codec_->NumOfCodecs(); i++) {
+      ASSERT_EQ(0, codec_->GetCodec(i, params));
+      printf("%d %s/%d/%d\n", params.pltype, params.plname, params.plfreq,
+             params.channels);
+    }
+  }
+
+  void StartSending() {
+    ASSERT_EQ(0, base_->StartSend(channel_));
+  }
+
+  void StartPlaying(Pan pan, const std::string& filename) {
+    VoEVolumeControl* volume = VoEVolumeControl::GetInterface(voe_);
+    VoEFile* file = VoEFile::GetInterface(voe_);
+    ASSERT_TRUE(volume != NULL);
+    ASSERT_TRUE(file != NULL);
+    if (pan == PanLeft) {
+      volume->SetOutputVolumePan(channel_, 1, 0);
+    } else if (pan == PanRight) {
+      volume->SetOutputVolumePan(channel_, 0, 1);
+    }
+    if (filename != "") {
+      printf("playing file\n");
+      ASSERT_EQ(0, file->StartPlayingFileLocally(channel_, filename.c_str(),
+          true, kFileFormatPcm16kHzFile, 1.0, 0, 0));
+    }
+    ASSERT_EQ(0, base_->StartReceive(channel_));
+    ASSERT_EQ(0, base_->StartPlayout(channel_));
+    volume->Release();
+    file->Release();
+  }
+
+  void Stop() {
+    ASSERT_EQ(0, base_->StopSend(channel_));
+    ASSERT_EQ(0, base_->StopPlayout(channel_));
+  }
+
+ private:
+  VoiceEngine* voe_;
+  VoEBase* base_;
+  VoEHardware* hardware_;
+  VoECodec* codec_;
+  AgcManager* manager_;
+  int channel_;
+  int capture_idx_;
+  int render_idx_;
+  rtc::scoped_ptr channel_transport_;
+};
+
+void RunHarness() {
+  rtc::scoped_ptr voe1(new AgcVoiceEngine(
+      FLAGS_internal, 2000, 2000, FLAGS_capture1, FLAGS_render1));
+  rtc::scoped_ptr voe2;
+  if (FLAGS_parallel) {
+    voe2.reset(new AgcVoiceEngine(!FLAGS_internal, 3000, 3000, FLAGS_capture2,
+                                  FLAGS_render2));
+    voe1->StartPlaying(AgcVoiceEngine::PanLeft, "");
+    voe1->StartSending();
+    voe2->StartPlaying(AgcVoiceEngine::PanRight, "");
+    voe2->StartSending();
+  } else if (FLAGS_aec) {
+    voe1.reset(new AgcVoiceEngine(FLAGS_internal, 2000, 4242, FLAGS_capture1,
+                                  FLAGS_render1));
+    voe2.reset(new AgcVoiceEngine(!FLAGS_internal, 4242, 2000, FLAGS_capture2,
+                                  FLAGS_render2));
+    voe1->StartPlaying(AgcVoiceEngine::NoPan, FLAGS_filename);
+    voe1->StartSending();
+    voe2->StartPlaying(AgcVoiceEngine::NoPan, "");
+  } else {
+    voe1->StartPlaying(AgcVoiceEngine::NoPan, "");
+    voe1->StartSending();
+  }
+
+  // Run forever...
+  SleepMs(0x7fffffff);
+}
+
+void PrintDevices() {
+  AgcVoiceEngine device_voe(false, 4242, 4242, 0, 0);
+  device_voe.PrintDevices();
+}
+
+void PrintCodecs() {
+  AgcVoiceEngine codec_voe(false, 4242, 4242, 0, 0);
+  codec_voe.PrintCodecs();
+}
+
+}  // namespace
+}  // namespace webrtc
+
+int main(int argc, char** argv) {
+  google::SetUsageMessage(webrtc::kUsage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::test::TraceToStderr trace_to_stderr;
+
+  if (FLAGS_parallel && FLAGS_aec) {
+    printf("-parallel and -aec are not compatible\n");
+    return 1;
+  }
+  if (FLAGS_devices) {
+    webrtc::PrintDevices();
+  }
+  if (FLAGS_codecs) {
+    webrtc::PrintCodecs();
+  }
+  if (!FLAGS_devices && !FLAGS_codecs) {
+    webrtc::RunHarness();
+  }
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_manager.cc
new file mode 100644
index 0000000000..83c0d0075b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_manager.cc
@@ -0,0 +1,252 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/tools/agc/agc_manager.h"
+
+#include 
+
+#include "webrtc/modules/audio_processing/agc/agc.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+
+namespace webrtc {
+
+class AgcManagerVolume : public VolumeCallbacks {
+ public:
+  // AgcManagerVolume acquires ownership of |volume|.
+  explicit AgcManagerVolume(VoEVolumeControl* volume)
+      : volume_(volume) {
+  }
+
+  ~AgcManagerVolume() {
+    if (volume_) {
+      volume_->Release();
+    }
+  }
+
+  virtual void SetMicVolume(int volume) {
+    if (volume_->SetMicVolume(volume) != 0) {
+      LOG_FERR1(LS_WARNING, SetMicVolume, volume);
+    }
+  }
+
+  int GetMicVolume() {
+    unsigned int volume = 0;
+    if (volume_->GetMicVolume(volume) != 0) {
+      LOG_FERR0(LS_WARNING, GetMicVolume);
+      return -1;
+    }
+    return volume;
+  }
+
+ private:
+  VoEVolumeControl* volume_;
+};
+
+class MediaCallback : public VoEMediaProcess {
+ public:
+  MediaCallback(AgcManagerDirect* direct, AudioProcessing* audioproc,
+                CriticalSectionWrapper* crit)
+      : direct_(direct),
+        audioproc_(audioproc),
+        crit_(crit),
+        frame_() {
+  }
+
+ protected:
+  virtual void Process(const int channel, const ProcessingTypes type,
+                       int16_t audio[], const int samples_per_channel,
+                       const int sample_rate_hz, const bool is_stereo) {
+    CriticalSectionScoped cs(crit_);
+    if (direct_->capture_muted()) {
+      return;
+    }
+
+    // Extract the first channel.
+    const int kMaxSampleRateHz = 48000;
+    const int kMaxSamplesPerChannel = kMaxSampleRateHz / 100;
+    assert(samples_per_channel < kMaxSamplesPerChannel &&
+           sample_rate_hz < kMaxSampleRateHz);
+    int16_t mono[kMaxSamplesPerChannel];
+    int16_t* mono_ptr = audio;
+    if (is_stereo) {
+      for (int n = 0; n < samples_per_channel; n++) {
+        mono[n] = audio[n * 2];
+      }
+      mono_ptr = mono;
+    }
+
+    direct_->Process(mono_ptr, samples_per_channel, sample_rate_hz);
+
+    // TODO(ajm): It's unfortunate we have to memcpy to this frame here, but
+    // it's needed for use with AudioProcessing.
+    frame_.num_channels_ = is_stereo ? 2 : 1;
+    frame_.samples_per_channel_ = samples_per_channel;
+    frame_.sample_rate_hz_ = sample_rate_hz;
+    const int length_samples = frame_.num_channels_ * samples_per_channel;
+    memcpy(frame_.data_, audio, length_samples * sizeof(int16_t));
+
+    // Apply compression to the audio.
+    if (audioproc_->ProcessStream(&frame_) != 0) {
+      LOG_FERR0(LS_ERROR, ProcessStream);
+    }
+
+    // Copy the compressed audio back to voice engine's array.
+    memcpy(audio, frame_.data_, length_samples * sizeof(int16_t));
+  }
+
+ private:
+  AgcManagerDirect* direct_;
+  AudioProcessing* audioproc_;
+  CriticalSectionWrapper* crit_;
+  AudioFrame frame_;
+};
+
+class PreprocCallback : public VoEMediaProcess {
+ public:
+  PreprocCallback(AgcManagerDirect* direct, CriticalSectionWrapper* crit)
+      : direct_(direct),
+        crit_(crit) {
+  }
+
+ protected:
+  virtual void Process(const int channel, const ProcessingTypes type,
+                       int16_t audio[], const int samples_per_channel,
+                       const int sample_rate_hz, const bool is_stereo) {
+    CriticalSectionScoped cs(crit_);
+    if (direct_->capture_muted()) {
+      return;
+    }
+    direct_->AnalyzePreProcess(audio, is_stereo ? 2 : 1, samples_per_channel);
+  }
+
+ private:
+  AgcManagerDirect* direct_;
+  CriticalSectionWrapper* crit_;
+};
+
+AgcManager::AgcManager(VoiceEngine* voe)
+    : media_(VoEExternalMedia::GetInterface(voe)),
+      volume_callbacks_(new AgcManagerVolume(VoEVolumeControl::GetInterface(
+          voe))),
+      crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      enabled_(false),
+      initialized_(false) {
+    Config config;
+    config.Set(new ExperimentalAgc(false));
+    audioproc_.reset(AudioProcessing::Create(config));
+    direct_.reset(new AgcManagerDirect(audioproc_->gain_control(),
+                                       volume_callbacks_.get()));
+    media_callback_.reset(new MediaCallback(direct_.get(),
+                                            audioproc_.get(),
+                                            crit_.get()));
+    preproc_callback_.reset(new PreprocCallback(direct_.get(), crit_.get()));
+}
+
+AgcManager::AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume,
+                       Agc* agc, AudioProcessing* audioproc)
+    : media_(media),
+      volume_callbacks_(new AgcManagerVolume(volume)),
+      crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      audioproc_(audioproc),
+      direct_(new AgcManagerDirect(agc,
+                                   audioproc_->gain_control(),
+                                   volume_callbacks_.get())),
+      media_callback_(new MediaCallback(direct_.get(),
+                                        audioproc_.get(),
+                                        crit_.get())),
+      preproc_callback_(new PreprocCallback(direct_.get(), crit_.get())),
+      enabled_(false),
+      initialized_(false) {
+}
+
+AgcManager::AgcManager()
+    : media_(NULL),
+      enabled_(false),
+      initialized_(false) {
+}
+
+AgcManager::~AgcManager() {
+  if (media_) {
+    if (enabled_) {
+      DeregisterCallbacks();
+    }
+    media_->Release();
+  }
+}
+
+int AgcManager::Enable(bool enable) {
+  if (enable == enabled_) {
+    return 0;
+  }
+  if (!initialized_) {
+    CriticalSectionScoped cs(crit_.get());
+    if (audioproc_->gain_control()->Enable(true) != 0) {
+      LOG_FERR1(LS_ERROR, gain_control()->Enable, true);
+      return -1;
+    }
+    if (direct_->Initialize() != 0) {
+      assert(false);
+      return -1;
+    }
+    initialized_ = true;
+  }
+
+  if (enable) {
+    if (media_->RegisterExternalMediaProcessing(0, kRecordingAllChannelsMixed,
+                                                *media_callback_) != 0) {
+      LOG(LS_ERROR) << "Failed to register postproc callback";
+      return -1;
+    }
+    if (media_->RegisterExternalMediaProcessing(0, kRecordingPreprocessing,
+                                                *preproc_callback_) != 0) {
+      LOG(LS_ERROR) << "Failed to register preproc callback";
+      return -1;
+    }
+  } else {
+    if (DeregisterCallbacks() != 0)
+      return -1;
+  }
+  enabled_ = enable;
+  return 0;
+}
+
+void AgcManager::CaptureDeviceChanged() {
+  CriticalSectionScoped cs(crit_.get());
+  direct_->Initialize();
+}
+
+void AgcManager::SetCaptureMuted(bool muted) {
+  CriticalSectionScoped cs(crit_.get());
+  direct_->SetCaptureMuted(muted);
+}
+
+int AgcManager::DeregisterCallbacks() {
+  // DeRegister shares a lock with the Process() callback. This call will block
+  // until the callback is finished and it's safe to continue teardown.
+  int err = 0;
+  if (media_->DeRegisterExternalMediaProcessing(0,
+          kRecordingAllChannelsMixed) != 0) {
+    LOG(LS_ERROR) << "Failed to deregister postproc callback";
+    err = -1;
+  }
+  if (media_->DeRegisterExternalMediaProcessing(0,
+          kRecordingPreprocessing) != 0) {
+    LOG(LS_ERROR) << "Failed to deregister preproc callback";
+    err = -1;
+  }
+  return err;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager.h b/media/webrtc/trunk/webrtc/tools/agc/agc_manager.h
new file mode 100644
index 0000000000..4f79f102d6
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_manager.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
+#define WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
+
+namespace webrtc {
+
+class Agc;
+class AudioProcessing;
+class CriticalSectionWrapper;
+class MediaCallback;
+class PreprocCallback;
+class VoEExternalMedia;
+class VoEVolumeControl;
+class VoiceEngine;
+class VolumeCallbacks;
+
+// Handles the interaction between VoiceEngine and the internal AGC. It hooks
+// into the capture stream through VoiceEngine's external media interface and
+// sends the audio to the AGC for analysis. It forwards requests for a capture
+// volume change from the AGC to the VoiceEngine volume interface.
+class AgcManager {
+ public:
+  explicit AgcManager(VoiceEngine* voe);
+  // Dependency injection for testing. Don't delete |agc| or |audioproc| as the
+  // memory is owned by the manager. If |media| or |volume| are non-fake
+  // reference counted classes, don't release them as this is handled by the
+  // manager.
+  AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume, Agc* agc,
+             AudioProcessing* audioproc);
+  virtual ~AgcManager();
+
+  // When enabled, registers external media processing callbacks with
+  // VoiceEngine to hook into the capture stream. Disabling deregisters the
+  // callbacks.
+  virtual int Enable(bool enable);
+  virtual bool enabled() const { return enabled_; }
+
+  // Call when the capture device has changed. This will trigger a retrieval of
+  // the initial capture volume on the next audio frame.
+  virtual void CaptureDeviceChanged();
+
+  // Call when the capture stream has been muted/unmuted. This causes the
+  // manager to disregard all incoming audio; chances are good it's background
+  // noise to which we'd like to avoid adapting.
+  virtual void SetCaptureMuted(bool muted);
+  virtual bool capture_muted() const { return direct_->capture_muted(); }
+
+ protected:
+  // Provide a default constructor for testing.
+  AgcManager();
+
+ private:
+  int DeregisterCallbacks();
+  int CheckVolumeAndReset();
+
+  VoEExternalMedia* media_;
+  rtc::scoped_ptr volume_callbacks_;
+  rtc::scoped_ptr crit_;
+  rtc::scoped_ptr audioproc_;
+  rtc::scoped_ptr direct_;
+  rtc::scoped_ptr media_callback_;
+  rtc::scoped_ptr preproc_callback_;
+  bool enabled_;
+  bool initialized_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_TOOLS_AGC_AGC_MANAGER_H_
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager_integrationtest.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_manager_integrationtest.cc
new file mode 100644
index 0000000000..4179e8c1c2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_manager_integrationtest.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/tools/agc/agc_manager.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_processing/agc/mock_agc.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+#include "webrtc/voice_engine/include/voe_network.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Mock;
+using ::testing::Return;
+
+namespace webrtc {
+
+class AgcManagerTest : public ::testing::Test {
+ protected:
+  AgcManagerTest()
+      : voe_(VoiceEngine::Create()),
+        base_(VoEBase::GetInterface(voe_)),
+        agc_(new MockAgc()),
+        manager_(new AgcManager(VoEExternalMedia::GetInterface(voe_),
+                                VoEVolumeControl::GetInterface(voe_),
+                                agc_,
+                                AudioProcessing::Create())),
+        channel_(-1) {
+  }
+
+  virtual void SetUp() {
+    ASSERT_TRUE(voe_ != NULL);
+    ASSERT_TRUE(base_ != NULL);
+    ASSERT_EQ(0, base_->Init());
+    channel_ = base_->CreateChannel();
+    ASSERT_NE(-1, channel_);
+
+    VoENetwork* network = VoENetwork::GetInterface(voe_);
+    ASSERT_TRUE(network != NULL);
+    channel_transport_.reset(
+        new test::VoiceChannelTransport(network, channel_));
+    ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", 1234));
+    network->Release();
+  }
+
+  virtual void TearDown() {
+    channel_transport_.reset(NULL);
+    ASSERT_EQ(0, base_->DeleteChannel(channel_));
+    ASSERT_EQ(0, base_->Terminate());
+    delete manager_;
+    // Test that the manager has released all VoE interfaces. The last
+    // reference is released in VoiceEngine::Delete.
+    EXPECT_EQ(1, base_->Release());
+    ASSERT_TRUE(VoiceEngine::Delete(voe_));
+  }
+
+  VoiceEngine* voe_;
+  VoEBase* base_;
+  MockAgc* agc_;
+  rtc::scoped_ptr channel_transport_;
+  // We use a pointer for the manager, so we can tear it down and test
+  // base_->Release() in the destructor.
+  AgcManager* manager_;
+  int channel_;
+};
+
+TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(EnableSucceeds)) {
+  EXPECT_EQ(0, manager_->Enable(true));
+  EXPECT_TRUE(manager_->enabled());
+  EXPECT_EQ(0, manager_->Enable(false));
+  EXPECT_FALSE(manager_->enabled());
+}
+
+TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(ProcessIsNotCalledByDefault)) {
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
+  EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
+  ASSERT_EQ(0, base_->StartSend(channel_));
+  SleepMs(100);
+  ASSERT_EQ(0, base_->StopSend(channel_));
+}
+
+TEST_F(AgcManagerTest, DISABLED_ProcessIsCalledOnlyWhenEnabled) {
+  EXPECT_CALL(*agc_, Reset());
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(0));
+  EXPECT_CALL(*agc_, Process(_, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(0));
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(false));
+  EXPECT_EQ(0, manager_->Enable(true));
+  ASSERT_EQ(0, base_->StartSend(channel_));
+  SleepMs(100);
+  EXPECT_EQ(0, manager_->Enable(false));
+  SleepMs(100);
+  Mock::VerifyAndClearExpectations(agc_);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
+  EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
+  SleepMs(100);
+  ASSERT_EQ(0, base_->StopSend(channel_));
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_manager_unittest.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_manager_unittest.cc
new file mode 100644
index 0000000000..fca8decc85
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_manager_unittest.cc
@@ -0,0 +1,736 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/tools/agc/agc_manager.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_processing/agc/mock_agc.h"
+#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
+#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+
+using ::testing::_;
+using ::testing::DoAll;
+using ::testing::Eq;
+using ::testing::Mock;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+using ::testing::SetArgReferee;
+
+namespace webrtc {
+namespace {
+
+const int kSampleRateHz = 32000;
+const int kNumChannels = 1;
+const int kSamplesPerChannel = kSampleRateHz / 100;
+const float kAboveClippedThreshold = 0.2f;
+
+}  // namespace
+
+class AgcManagerUnitTest : public ::testing::Test {
+ protected:
+  AgcManagerUnitTest()
+      : media_(),
+        volume_(),
+        agc_(new MockAgc),
+        audioproc_(new MockAudioProcessing),
+        gctrl_(audioproc_->gain_control()),
+        manager_(&media_, &volume_, agc_, audioproc_) {
+    EXPECT_CALL(*gctrl_, Enable(true));
+    ExpectInitialize();
+    manager_.Enable(true);
+    EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+        .WillOnce(Return(false));
+    // TODO(bjornv): Find a better solution that adds an initial volume here
+    // instead of applying SetVolumeAndProcess(128u) in each test, but at the
+    // same time can test a too low initial value.
+  }
+
+  void SetInitialVolume(unsigned int volume) {
+    ExpectInitialize();
+    manager_.CaptureDeviceChanged();
+    ExpectCheckVolumeAndReset(volume);
+    EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
+    PostProcCallback(1);
+  }
+
+  void SetVolumeAndProcess(unsigned int volume) {
+    // Volume is checked on first process call.
+    ExpectCheckVolumeAndReset(volume);
+    PostProcCallback(1);
+  }
+
+  void ExpectCheckVolumeAndReset(unsigned int volume) {
+    EXPECT_CALL(volume_, GetMicVolume(_))
+        .WillOnce(DoAll(SetArgReferee<0>(volume), Return(0)));
+    EXPECT_CALL(*agc_, Reset());
+  }
+
+  void ExpectVolumeChange(unsigned int current_volume,
+                          unsigned int new_volume) {
+    EXPECT_CALL(volume_, GetMicVolume(_))
+        .WillOnce(DoAll(SetArgReferee<0>(current_volume), Return(0)));
+    EXPECT_CALL(volume_, SetMicVolume(Eq(new_volume))).WillOnce(Return(0));
+  }
+
+  void ExpectInitialize() {
+    EXPECT_CALL(*gctrl_, set_mode(GainControl::kFixedDigital));
+    EXPECT_CALL(*gctrl_, set_target_level_dbfs(2));
+    EXPECT_CALL(*gctrl_, set_compression_gain_db(7));
+    EXPECT_CALL(*gctrl_, enable_limiter(true));
+  }
+
+  void PreProcCallback(int num_calls) {
+    for (int i = 0; i < num_calls; ++i) {
+      media_.CallProcess(kRecordingPreprocessing, NULL, kSamplesPerChannel,
+                         kSampleRateHz, kNumChannels);
+    }
+  }
+
+  void PostProcCallback(int num_calls) {
+    for (int i = 0; i < num_calls; ++i) {
+      EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return(0));
+      EXPECT_CALL(*audioproc_, ProcessStream(_)).WillOnce(Return(0));
+      media_.CallProcess(kRecordingAllChannelsMixed, NULL, kSamplesPerChannel,
+                         kSampleRateHz, kNumChannels);
+    }
+  }
+
+  ~AgcManagerUnitTest() {
+    EXPECT_CALL(volume_, Release()).WillOnce(Return(0));
+  }
+
+  FakeVoEExternalMedia media_;
+  MockVoEVolumeControl volume_;
+  MockAgc* agc_;
+  MockAudioProcessing* audioproc_;
+  MockGainControl* gctrl_;
+  AgcManager manager_;
+  test::TraceToStderr trace_to_stderr;
+};
+
+TEST_F(AgcManagerUnitTest, MicVolumeResponseToRmsError) {
+  SetVolumeAndProcess(128u);
+  // Compressor default; no residual error.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
+  PostProcCallback(1);
+
+  // Inside the compressor's window; no change of volume.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)));
+  PostProcCallback(1);
+
+  // Above the compressor's window; volume should be increased.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  ExpectVolumeChange(128u, 130u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  ExpectVolumeChange(130u, 168u);
+  PostProcCallback(1);
+
+  // Inside the compressor's window; no change of volume.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
+  PostProcCallback(1);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)));
+  PostProcCallback(1);
+
+  // Below the compressor's window; volume should be decreased.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  ExpectVolumeChange(168u, 167u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  ExpectVolumeChange(167u, 163u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-9), Return(true)));
+  ExpectVolumeChange(163u, 129u);
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, MicVolumeIsLimited) {
+  SetVolumeAndProcess(128u);
+  // Maximum upwards change is limited.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  ExpectVolumeChange(128u, 183u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  ExpectVolumeChange(183u, 243u);
+  PostProcCallback(1);
+
+  // Won't go higher than the maximum.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  ExpectVolumeChange(243u, 255u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  ExpectVolumeChange(255u, 254u);
+  PostProcCallback(1);
+
+  // Maximum downwards change is limited.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(254u, 194u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(194u, 137u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(137u, 88u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(88u, 54u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(54u, 33u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(33u, 18u);
+  PostProcCallback(1);
+
+  // Won't go lower than the minimum.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  ExpectVolumeChange(18u, 12u);
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, CompressorStepsTowardsTarget) {
+  SetVolumeAndProcess(128u);
+  // Compressor default; no call to set_compression_gain_db.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(20);
+
+  // Moves slowly upwards.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  PostProcCallback(1);
+
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  PostProcCallback(1);
+
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(20);
+
+  // Moves slowly downward, then reverses before reaching the original target.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  PostProcCallback(1);
+
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(20);
+}
+
+TEST_F(AgcManagerUnitTest, CompressorErrorIsDeemphasized) {
+  SetVolumeAndProcess(128u);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillRepeatedly(Return(false));
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  PostProcCallback(1);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(20);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillRepeatedly(Return(false));
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(7)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
+  PostProcCallback(1);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
+  PostProcCallback(20);
+}
+
+TEST_F(AgcManagerUnitTest, CompressorReachesMaximum) {
+  SetVolumeAndProcess(128u);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillRepeatedly(Return(false));
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, CompressorReachesMinimum) {
+  SetVolumeAndProcess(128u);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillRepeatedly(Return(false));
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(5)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(4)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(3)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(2)).WillOnce(Return(0));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, NoActionWhileMuted) {
+  SetVolumeAndProcess(128u);
+  manager_.SetCaptureMuted(true);
+  media_.CallProcess(kRecordingAllChannelsMixed, NULL, kSamplesPerChannel,
+                     kSampleRateHz, kNumChannels);
+}
+
+TEST_F(AgcManagerUnitTest, UnmutingChecksVolumeWithoutRaising) {
+  SetVolumeAndProcess(128u);
+  manager_.SetCaptureMuted(true);
+  manager_.SetCaptureMuted(false);
+  ExpectCheckVolumeAndReset(127u);
+  // SetMicVolume should not be called.
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(Return(false));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, UnmutingRaisesTooLowVolume) {
+  SetVolumeAndProcess(128u);
+  manager_.SetCaptureMuted(true);
+  manager_.SetCaptureMuted(false);
+  ExpectCheckVolumeAndReset(11u);
+  EXPECT_CALL(volume_, SetMicVolume(Eq(12u))).WillOnce(Return(0));
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(Return(false));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, ChangingDevicesChecksVolume) {
+  SetVolumeAndProcess(128u);
+  ExpectInitialize();
+  manager_.CaptureDeviceChanged();
+  ExpectCheckVolumeAndReset(128u);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(Return(false));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, LowInitialVolumeIsRaised) {
+  ExpectCheckVolumeAndReset(11u);
+  // Should set MicVolume to kMinInitMicLevel = 85.
+  EXPECT_CALL(volume_, SetMicVolume(Eq(85u))).WillOnce(Return(0));
+  PostProcCallback(1);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(Return(false));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, ManualLevelChangeResultsInNoSetMicCall) {
+  SetVolumeAndProcess(128u);
+  // Change outside of compressor's range, which would normally trigger a call
+  // to SetMicVolume.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  // GetMicVolume returns a value outside of the quantization slack, indicating
+  // a manual volume change.
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillOnce(DoAll(SetArgReferee<0>(154u), Return(0)));
+  // SetMicVolume should not be called.
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  PostProcCallback(1);
+
+  // Do the same thing, except downwards now.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillOnce(DoAll(SetArgReferee<0>(100u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  PostProcCallback(1);
+
+  // And finally verify the AGC continues working without a manual change.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  ExpectVolumeChange(100u, 99u);
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, RecoveryAfterManualLevelChangeFromMax) {
+  SetVolumeAndProcess(128u);
+  // Force the mic up to max volume. Takes a few steps due to the residual
+  // gain limitation.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  ExpectVolumeChange(128u, 183u);
+  PostProcCallback(1);
+  ExpectVolumeChange(183u, 243u);
+  PostProcCallback(1);
+  ExpectVolumeChange(243u, 255u);
+  PostProcCallback(1);
+
+  // Manual change does not result in SetMicVolume call.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillOnce(DoAll(SetArgReferee<0>(50u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  PostProcCallback(1);
+
+  // Continues working as usual afterwards.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  ExpectVolumeChange(50u, 69u);
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, RecoveryAfterManualLevelChangeBelowMin) {
+  SetVolumeAndProcess(128u);
+  // Manual change below min.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  // Don't set to zero, which will cause AGC to take no action.
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillOnce(DoAll(SetArgReferee<0>(1u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  PostProcCallback(1);
+
+  // Continues working as usual afterwards.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  ExpectVolumeChange(1u, 2u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  ExpectVolumeChange(2u, 11u);
+  PostProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  ExpectVolumeChange(11u, 18u);
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, NoClippingHasNoImpact) {
+  SetVolumeAndProcess(128u);
+  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillRepeatedly(Return(0));
+  PreProcCallback(100);
+}
+
+TEST_F(AgcManagerUnitTest, ClippingUnderThresholdHasNoImpact) {
+  SetVolumeAndProcess(128u);
+  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.099));
+  PreProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, ClippingLowersVolume) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(255u);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.101));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(255u, 240u);
+  PreProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, WaitingPeriodBetweenClippingChecks) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(255u);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(255u, 240u);
+  PreProcCallback(1);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillRepeatedly(Return(kAboveClippedThreshold));
+  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  PreProcCallback(300);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(240u, 225u);
+  PreProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, ClippingLoweringIsLimited) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(180u);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(180u, 170u);
+  PreProcCallback(1);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillRepeatedly(Return(kAboveClippedThreshold));
+  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  PreProcCallback(1000);
+}
+
+TEST_F(AgcManagerUnitTest, ClippingMaxIsRespectedWhenEqualToLevel) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(255u);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(255u, 240u);
+  PreProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillRepeatedly(DoAll(SetArgReferee<0>(240u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  PostProcCallback(10);
+}
+
+TEST_F(AgcManagerUnitTest, ClippingMaxIsRespectedWhenHigherThanLevel) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(200u);
+
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(200u, 185u);
+  PreProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+    .WillRepeatedly(DoAll(SetArgPointee<0>(40), Return(true)));
+  ExpectVolumeChange(185u, 240u);
+  PostProcCallback(1);
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillRepeatedly(DoAll(SetArgReferee<0>(240u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  PostProcCallback(10);
+}
+
+TEST_F(AgcManagerUnitTest, MaxCompressionIsIncreasedAfterClipping) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(210u);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(210u, 195u);
+  PreProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillRepeatedly(Return(false));
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(13)).WillOnce(Return(0));
+  PostProcCallback(1);
+
+  // Continue clipping until we hit the maximum surplus compression.
+  PreProcCallback(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(195u, 180u);
+  PreProcCallback(1);
+
+  PreProcCallback(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(180u, 170u);
+  PreProcCallback(1);
+
+  // Current level is now at the minimum, but the maximum allowed level still
+  // has more to decrease.
+  PreProcCallback(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  PreProcCallback(1);
+
+  PreProcCallback(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  PreProcCallback(1);
+
+  PreProcCallback(300);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  PreProcCallback(1);
+
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillRepeatedly(Return(false));
+  PostProcCallback(19);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(14)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(15)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(16)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(17)).WillOnce(Return(0));
+  PostProcCallback(20);
+  EXPECT_CALL(*gctrl_, set_compression_gain_db(18)).WillOnce(Return(0));
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, UserCanRaiseVolumeAfterClipping) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(225u);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  ExpectVolumeChange(225u, 210u);
+  PreProcCallback(1);
+
+  // High enough error to trigger a volume check.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(14), Return(true)));
+  // User changed the volume.
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillOnce(DoAll(SetArgReferee<0>(250u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(1);
+  PostProcCallback(1);
+
+  // Move down...
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-10), Return(true)));
+  ExpectVolumeChange(250u, 210u);
+  PostProcCallback(1);
+  // And back up to the new max established by the user.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(40), Return(true)));
+  ExpectVolumeChange(210u, 250u);
+  PostProcCallback(1);
+  // Will not move above new maximum.
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillRepeatedly(DoAll(SetArgReferee<0>(250u), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  PostProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, ClippingDoesNotPullLowVolumeBackUp) {
+  SetVolumeAndProcess(128u);
+  SetInitialVolume(80u);
+  EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
+      .WillOnce(Return(kAboveClippedThreshold));
+  EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  EXPECT_CALL(*agc_, Reset()).Times(0);
+  PreProcCallback(1);
+}
+
+TEST_F(AgcManagerUnitTest, TakesNoActionOnZeroMicVolume) {
+  SetVolumeAndProcess(128u);
+  EXPECT_CALL(*agc_, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  EXPECT_CALL(volume_, GetMicVolume(_))
+      .WillRepeatedly(DoAll(SetArgReferee<0>(0), Return(0)));
+  EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
+  PostProcCallback(10);
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/tools/agc/agc_test.cc b/media/webrtc/trunk/webrtc/tools/agc/agc_test.cc
new file mode 100644
index 0000000000..29769488c1
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/agc_test.cc
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include 
+#include 
+
+#include 
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_processing/agc/agc.h"
+#include "webrtc/modules/audio_processing/agc/utility.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/test/testsupport/trace_to_stderr.h"
+#include "webrtc/tools/agc/agc_manager.h"
+#include "webrtc/tools/agc/test_utils.h"
+#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
+#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
+
+DEFINE_string(in, "in.pcm", "input filename");
+DEFINE_string(out, "out.pcm", "output filename");
+DEFINE_int32(rate, 16000, "sample rate in Hz");
+DEFINE_int32(channels, 1, "number of channels");
+DEFINE_int32(level, -18, "target level in RMS dBFs [-100, 0]");
+DEFINE_bool(limiter, true, "enable a limiter for the compression stage");
+DEFINE_int32(cmp_level, 2, "target level in dBFs for the compression stage");
+DEFINE_int32(mic_gain, 80, "range of gain provided by the virtual mic in dB");
+DEFINE_int32(gain_offset, 0,
+             "an amount (in dB) to add to every entry in the gain map");
+DEFINE_string(gain_file, "",
+    "filename providing a mic gain mapping. The file should be text containing "
+    "a (floating-point) gain entry in dBFs per line corresponding to levels "
+    "from 0 to 255.");
+
+using ::testing::_;
+using ::testing::ByRef;
+using ::testing::DoAll;
+using ::testing::Mock;
+using ::testing::Return;
+using ::testing::SaveArg;
+using ::testing::SetArgReferee;
+
+namespace webrtc {
+namespace {
+
+const char kUsage[] = "\nProcess an audio file to simulate an analog agc.";
+
+void ReadGainMapFromFile(FILE* file, int offset, int gain_map[256]) {
+  for (int i = 0; i < 256; ++i) {
+    float gain = 0;
+    ASSERT_EQ(1, fscanf(file, "%f", &gain));
+    gain_map[i] = std::floor(gain + 0.5);
+  }
+
+  // Adjust from dBFs to gain in dB. We assume that level 127 provides 0 dB
+  // gain. This corresponds to the interpretation in MicLevel2Gain().
+  const int midpoint = gain_map[127];
+  printf("Gain map\n");
+  for (int i = 0; i < 256; ++i) {
+    gain_map[i] += offset - midpoint;
+    if (i % 5 == 0) {
+      printf("%d: %d dB\n", i, gain_map[i]);
+    }
+  }
+}
+
+void CalculateGainMap(int gain_range_db, int offset, int gain_map[256]) {
+  printf("Gain map\n");
+  for (int i = 0; i < 256; ++i) {
+    gain_map[i] = std::floor(MicLevel2Gain(gain_range_db, i) + 0.5) + offset;
+    if (i % 5 == 0) {
+      printf("%d: %d dB\n", i, gain_map[i]);
+    }
+  }
+}
+
+void RunAgc() {
+  test::TraceToStderr trace_to_stderr(true);
+  FILE* in_file = fopen(FLAGS_in.c_str(), "rb");
+  ASSERT_TRUE(in_file != NULL);
+  FILE* out_file = fopen(FLAGS_out.c_str(), "wb");
+  ASSERT_TRUE(out_file != NULL);
+
+  int gain_map[256];
+  if (FLAGS_gain_file != "") {
+    FILE* gain_file = fopen(FLAGS_gain_file.c_str(), "rt");
+    ASSERT_TRUE(gain_file != NULL);
+    ReadGainMapFromFile(gain_file, FLAGS_gain_offset, gain_map);
+    fclose(gain_file);
+  } else {
+    CalculateGainMap(FLAGS_mic_gain, FLAGS_gain_offset, gain_map);
+  }
+
+  FakeVoEExternalMedia media;
+  MockVoEVolumeControl volume;
+  Agc* agc = new Agc;
+  AudioProcessing* audioproc = AudioProcessing::Create();
+  ASSERT_TRUE(audioproc != NULL);
+  AgcManager manager(&media, &volume, agc, audioproc);
+
+  int mic_level = 128;
+  int last_mic_level = mic_level;
+  EXPECT_CALL(volume, GetMicVolume(_))
+      .WillRepeatedly(DoAll(SetArgReferee<0>(ByRef(mic_level)), Return(0)));
+  EXPECT_CALL(volume, SetMicVolume(_))
+      .WillRepeatedly(DoAll(SaveArg<0>(&mic_level), Return(0)));
+
+  manager.Enable(true);
+  ASSERT_EQ(0, agc->set_target_level_dbfs(FLAGS_level));
+  const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
+  GainControl* gctrl = audioproc->gain_control();
+  ASSERT_EQ(kNoErr, gctrl->set_target_level_dbfs(FLAGS_cmp_level));
+  ASSERT_EQ(kNoErr, gctrl->enable_limiter(FLAGS_limiter));
+
+  AudioFrame frame;
+  frame.num_channels_ = FLAGS_channels;
+  frame.sample_rate_hz_ = FLAGS_rate;
+  frame.samples_per_channel_ = FLAGS_rate / 100;
+  const size_t frame_length = frame.samples_per_channel_ * FLAGS_channels;
+  size_t sample_count = 0;
+  while (fread(frame.data_, sizeof(int16_t), frame_length, in_file) ==
+      frame_length) {
+    SimulateMic(gain_map, mic_level, last_mic_level, &frame);
+    last_mic_level = mic_level;
+    media.CallProcess(kRecordingAllChannelsMixed, frame.data_,
+                      frame.samples_per_channel_, FLAGS_rate, FLAGS_channels);
+    ASSERT_EQ(frame_length,
+              fwrite(frame.data_, sizeof(int16_t), frame_length, out_file));
+    sample_count += frame_length;
+    trace_to_stderr.SetTimeSeconds(static_cast(sample_count) /
+                                   FLAGS_channels / FLAGS_rate);
+  }
+  fclose(in_file);
+  fclose(out_file);
+  EXPECT_CALL(volume, Release());
+}
+
+}  // namespace
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  google::SetUsageMessage(webrtc::kUsage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::RunAgc();
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/tools/agc/fake_agc.h b/media/webrtc/trunk/webrtc/tools/agc/fake_agc.h
new file mode 100644
index 0000000000..6b39cd7c3b
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/fake_agc.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TOOLS_AGC_FAKE_AGC_H_
+#define WEBRTC_TOOLS_AGC_FAKE_AGC_H_
+
+#include "webrtc/modules/audio_processing/agc/agc.h"
+
+namespace webrtc {
+
+class FakeAgc : public Agc {
+ public:
+  FakeAgc()
+      : counter_(0),
+        volume_(kMaxVolume / 2) {
+  }
+
+  virtual int Process(const AudioFrame& audio_frame) {
+    const int kUpdateIntervalFrames = 10;
+    const int kMaxVolume = 255;
+    if (counter_ % kUpdateIntervalFrames == 0) {
+      volume_ = (++volume_) % kMaxVolume;
+    }
+    counter_++;
+    return 0;
+  }
+
+  virtual int FakeAgc::MicVolume() {
+    return volume_;
+  }
+
+ private:
+  int counter_;
+  int volume_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_TOOLS_AGC_FAKE_AGC_H_
diff --git a/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc b/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc
new file mode 100644
index 0000000000..3a26cb9ac7
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/test_utils.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/tools/agc/test_utils.h"
+
+#include 
+
+#include 
+
+#include "webrtc/modules/interface/module_common_types.h"
+
+namespace webrtc {
+
+float MicLevel2Gain(int gain_range_db, int level) {
+  return (level - 127.0f) / 128.0f * gain_range_db / 2;
+}
+
+float Db2Linear(float db) {
+  return powf(10.0f, db / 20.0f);
+}
+
+void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
+  const int frame_length = frame->samples_per_channel_ * frame->num_channels_;
+  // Smooth the transition between gain levels across the frame.
+  float smoothed_gain = last_gain;
+  float gain_step = (gain - last_gain) / (frame_length - 1);
+  for (int i = 0; i < frame_length; ++i) {
+    smoothed_gain += gain_step;
+    float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
+    sample = std::max(std::min(32767.0f, sample), -32768.0f);
+    frame->data_[i] = static_cast(sample);
+  }
+}
+
+void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame) {
+  ApplyGainLinear(Db2Linear(gain_db), Db2Linear(last_gain_db), frame);
+}
+
+void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
+                 AudioFrame* frame) {
+  assert(mic_level >= 0 && mic_level <= 255);
+  assert(last_mic_level >= 0 && last_mic_level <= 255);
+  ApplyGain(MicLevel2Gain(gain_range_db, mic_level),
+            MicLevel2Gain(gain_range_db, last_mic_level),
+            frame);
+}
+
+void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
+                 AudioFrame* frame) {
+  assert(mic_level >= 0 && mic_level <= 255);
+  assert(last_mic_level >= 0 && last_mic_level <= 255);
+  ApplyGain(gain_map[mic_level], gain_map[last_mic_level], frame);
+}
+
+}  // namespace webrtc
+
diff --git a/media/webrtc/trunk/webrtc/tools/agc/test_utils.h b/media/webrtc/trunk/webrtc/tools/agc/test_utils.h
new file mode 100644
index 0000000000..2aca9996f9
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/agc/test_utils.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_TOOLS_AGC_TEST_UTILS_H_
+#define WEBRTC_TOOLS_AGC_TEST_UTILS_H_
+namespace webrtc {
+
+class AudioFrame;
+
+float MicLevel2Gain(int gain_range_db, int level);
+float Db2Linear(float db);
+void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame);
+void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame);
+void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
+                 AudioFrame* frame);
+void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
+                 AudioFrame* frame);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_TOOLS_AGC_TEST_UTILS_H_
diff --git a/media/webrtc/trunk/webrtc/tools/converter/converter.h b/media/webrtc/trunk/webrtc/tools/converter/converter.h
index ce21c95db9..a23d5a14d4 100644
--- a/media/webrtc/trunk/webrtc/tools/converter/converter.h
+++ b/media/webrtc/trunk/webrtc/tools/converter/converter.h
@@ -13,8 +13,8 @@
 
 #include 
 
-#include "third_party/libyuv/include/libyuv/compare.h"
-#include "third_party/libyuv/include/libyuv/convert.h"
+#include "libyuv/compare.h"  // NOLINT
+#include "libyuv/convert.h"  // NOLINT
 
 namespace webrtc {
 namespace test {
diff --git a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
index 39e9974aa7..1eb8925537 100644
--- a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
+++ b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
@@ -15,7 +15,7 @@
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ -48,7 +48,7 @@ void RunHarness() {
   int channel = base->CreateChannel();
   ASSERT_NE(-1, channel);
 
-  scoped_ptr voice_channel_transport(
+  rtc::scoped_ptr voice_channel_transport(
       new VoiceChannelTransport(network, channel));
 
   ASSERT_EQ(0, voice_channel_transport->SetSendDestination("127.0.0.1", 1234));
diff --git a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/__init__.py b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/__init__.py
new file mode 100644
index 0000000000..e69de29bb2
diff --git a/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py
new file mode 100644
index 0000000000..77eda1e7a3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf/perf_utils.py
@@ -0,0 +1,31 @@
+# Copyright (c) 2012 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+# Copied from /src/chrome/test/pyautolib/pyauto_utils.py in Chromium.
+
+import sys
+
+def PrintPerfResult(graph_name, series_name, data_point, units,
+                    show_on_waterfall=False):
+  """Prints a line to stdout that is specially formatted for the perf bots.
+
+  Args:
+    graph_name: String name for the graph on which to plot the data.
+    series_name: String name for the series (line on the graph) associated with
+                 the data.  This is also the string displayed on the waterfall
+                 if |show_on_waterfall| is True.
+    data_point: Numeric data value to plot on the graph for the current build.
+                This can be a single value or an array of values.  If an array,
+                the graph will plot the average of the values, along with error
+                bars.
+    units: The string unit of measurement for the given |data_point|.
+    show_on_waterfall: Whether or not to display this result directly on the
+                       buildbot waterfall itself (in the buildbot step running
+                       this test on the waterfall page, not the stdio page).
+  """
+  waterfall_indicator = ['', '*'][show_on_waterfall]
+  print '%sRESULT %s: %s= %s %s' % (
+      waterfall_indicator, graph_name, series_name,
+      str(data_point).replace(' ', ''), units)
+  sys.stdout.flush()
\ No newline at end of file
diff --git a/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc b/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
index 11fb18aa3e..570fa0ad24 100644
--- a/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
+++ b/media/webrtc/trunk/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
@@ -12,7 +12,7 @@
 
 #include 
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
diff --git a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc
index c2a5e4991f..f5608c8343 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.cc
@@ -356,7 +356,7 @@ void PrintAnalysisResults(FILE* output, const std::string& label,
          ++iter) {
       fprintf(output, "%f,", iter->ssim_value);
     }
-    fprintf(output, "%f]\n", iter->ssim_value);
+    fprintf(output, "%f] score\n", iter->ssim_value);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h
index 31064a28f8..4704a8cde9 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h
+++ b/media/webrtc/trunk/webrtc/tools/frame_analyzer/video_quality_analysis.h
@@ -14,8 +14,8 @@
 #include 
 #include 
 
-#include "third_party/libyuv/include/libyuv/compare.h"
-#include "third_party/libyuv/include/libyuv/convert.h"
+#include "libyuv/compare.h"  // NOLINT
+#include "libyuv/convert.h"  // NOLINT
 
 namespace webrtc {
 namespace test {
diff --git a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc
index 93a548fe56..79c6033a30 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_lib.cc
@@ -13,8 +13,8 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 using std::string;
@@ -36,9 +36,9 @@ int EditFrames(const string& in_path, int width, int height,
   }
 
   // Frame size of I420.
-  int frame_length = CalcBufferSize(kI420, width, height);
+  size_t frame_length = CalcBufferSize(kI420, width, height);
 
-  webrtc::scoped_ptr temp_buffer(new uint8_t[frame_length]);
+  rtc::scoped_ptr temp_buffer(new uint8_t[frame_length]);
 
   FILE* out_fid = fopen(out_path.c_str(), "wb");
 
@@ -50,7 +50,7 @@ int EditFrames(const string& in_path, int width, int height,
 
   int num_frames_read = 0;
   int num_frames_read_between = 0;
-  int num_bytes_read;
+  size_t num_bytes_read;
 
   while ((num_bytes_read = fread(temp_buffer.get(), 1, frame_length, in_fid))
       == frame_length) {
diff --git a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_unittest.cc b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_unittest.cc
index 69468e18ce..31991b7175 100644
--- a/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_unittest.cc
+++ b/media/webrtc/trunk/webrtc/tools/frame_editing/frame_editing_unittest.cc
@@ -14,8 +14,8 @@
 #include 
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/tools/frame_editing/frame_editing_lib.h"
 
@@ -24,7 +24,7 @@ namespace test {
 
 const int kWidth = 352;
 const int kHeight = 288;
-const int kFrameSize = CalcBufferSize(kI420, kWidth, kHeight);
+const size_t kFrameSize = CalcBufferSize(kI420, kWidth, kHeight);
 
 class FrameEditingTest : public ::testing::Test {
  protected:
@@ -54,9 +54,10 @@ class FrameEditingTest : public ::testing::Test {
     remove(test_video_.c_str());
   }
   // Compares the frames in both streams to the end of one of the streams.
-  void CompareToTheEnd(FILE* test_video_fid, FILE* ref_video_fid,
-                       scoped_ptr* ref_buffer,
-                       scoped_ptr* test_buffer) {
+  void CompareToTheEnd(FILE* test_video_fid,
+                       FILE* ref_video_fid,
+                       rtc::scoped_ptr* ref_buffer,
+                       rtc::scoped_ptr* test_buffer) {
     while (!feof(test_video_fid) && !feof(ref_video_fid)) {
       num_bytes_read_ = fread(ref_buffer->get(), 1, kFrameSize, ref_video_fid);
       if (!feof(ref_video_fid)) {
@@ -79,9 +80,9 @@ class FrameEditingTest : public ::testing::Test {
   std::string test_video_;
   FILE* original_fid_;
   FILE* edited_fid_;
-  int num_bytes_read_;
-  scoped_ptr original_buffer_;
-  scoped_ptr edited_buffer_;
+  size_t num_bytes_read_;
+  rtc::scoped_ptr original_buffer_;
+  rtc::scoped_ptr edited_buffer_;
   int num_frames_read_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/tools/simple_command_line_parser_unittest.cc b/media/webrtc/trunk/webrtc/tools/simple_command_line_parser_unittest.cc
index f016b17d99..48640df172 100644
--- a/media/webrtc/trunk/webrtc/tools/simple_command_line_parser_unittest.cc
+++ b/media/webrtc/trunk/webrtc/tools/simple_command_line_parser_unittest.cc
@@ -16,7 +16,7 @@ namespace test {
 
 class CommandLineParserTest : public ::testing::Test {
  protected:
-  virtual void SetUp() {
+  void SetUp() override {
     parser_ = new CommandLineParser();
 
     test_flags_length_ = 3;
@@ -29,7 +29,7 @@ class CommandLineParserTest : public ::testing::Test {
     strncpy(test_flags_[1], "--foo", flag_size);
     strncpy(test_flags_[2], "--bar=1", flag_size);
   }
-  virtual void TearDown() {
+  void TearDown() override {
     for (int i = 0; i < test_flags_length_; ++i) {
       delete[] test_flags_[i];
     }
diff --git a/media/webrtc/trunk/webrtc/tools/tools.gyp b/media/webrtc/trunk/webrtc/tools/tools.gyp
index 102ba8ec30..e2a54212ec 100644
--- a/media/webrtc/trunk/webrtc/tools/tools.gyp
+++ b/media/webrtc/trunk/webrtc/tools/tools.gyp
@@ -15,10 +15,10 @@
       'target_name': 'video_quality_analysis',
       'type': 'static_library',
       'dependencies': [
-        '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
+        '<(webrtc_root)/common_video/common_video.gyp:common_video',
       ],
       'export_dependent_settings': [
-        '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
+        '<(webrtc_root)/common_video/common_video.gyp:common_video',
       ],
       'sources': [
         'frame_analyzer/video_quality_analysis.h',
@@ -51,8 +51,8 @@
       'target_name': 'rgba_to_i420_converter',
       'type': 'executable',
       'dependencies': [
+        '<(webrtc_root)/common_video/common_video.gyp:common_video',
         '<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
-        '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
       ],
       'sources': [
         'converter/converter.h',
@@ -91,7 +91,7 @@
       'type': 'executable',
       'dependencies': [
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
       ],
       'sources': [
         'force_mic_volume_max/force_mic_volume_max.cc',
@@ -101,13 +101,77 @@
   'conditions': [
     ['include_tests==1', {
       'targets' : [
+        {
+          'target_name': 'agc_manager',
+          'type': 'static_library',
+          'dependencies': [
+            '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+            '<(webrtc_root)/modules/modules.gyp:audio_processing',
+            '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+          ],
+          'sources': [
+            'agc/agc_manager.cc',
+            'agc/agc_manager.h',
+          ],
+        },
+        {
+          'target_name': 'agc_test_utils',
+          'type': 'static_library',
+          'sources': [
+            'agc/test_utils.cc',
+            'agc/test_utils.h',
+          ],
+        },
+        {
+          'target_name': 'agc_harness',
+          'type': 'executable',
+          'dependencies': [
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            '<(webrtc_root)/test/test.gyp:channel_transport',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            'agc_manager',
+          ],
+          'sources': [
+            'agc/agc_harness.cc',
+          ],
+        },  # agc_harness
+        {
+          'target_name': 'agc_proc',
+          'type': 'executable',
+          'dependencies': [
+            '<(DEPTH)/testing/gmock.gyp:gmock',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+            'agc_manager',
+            'agc_test_utils',
+          ],
+          'sources': [
+            'agc/agc_test.cc',
+          ],
+        },  # agc_proc
+        {
+          'target_name': 'activity_metric',
+          'type': 'executable',
+          'dependencies': [
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+            'agc_manager',
+          ],
+          'sources': [
+            'agc/activity_metric.cc',
+          ],
+        },  # activity_metric
         {
           'target_name': 'audio_e2e_harness',
           'type': 'executable',
           'dependencies': [
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
           ],
diff --git a/media/webrtc/trunk/webrtc/tools/tools_unittests.isolate b/media/webrtc/trunk/webrtc/tools/tools_unittests.isolate
index bf1fd0199a..ebe4714f7e 100644
--- a/media/webrtc/trunk/webrtc/tools/tools_unittests.isolate
+++ b/media/webrtc/trunk/webrtc/tools/tools_unittests.isolate
@@ -7,11 +7,10 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
+    ['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', {
       'variables': {
         'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
+          '<(DEPTH)/resources/foreman_cif.yuv',
         ],
       },
     }],
@@ -23,7 +22,6 @@
         ],
         'files': [
           '<(DEPTH)/DEPS',
-          '<(DEPTH)/resources/foreman_cif.yuv',
           '<(DEPTH)/testing/test_env.py',
           '<(PRODUCT_DIR)/tools_unittests<(EXECUTABLE_SUFFIX)',
         ],
diff --git a/media/webrtc/trunk/webrtc/typedefs.h b/media/webrtc/trunk/webrtc/typedefs.h
index 11f6b1ce5e..99ea82a994 100644
--- a/media/webrtc/trunk/webrtc/typedefs.h
+++ b/media/webrtc/trunk/webrtc/typedefs.h
@@ -155,28 +155,6 @@ typedef unsigned int        uint32_t;
 typedef unsigned __int64    uint64_t;
 #endif
 
-// Borrowed from Chromium's base/compiler_specific.h.
-// Annotate a virtual method indicating it must be overriding a virtual
-// method in the parent class.
-// Use like:
-//   virtual void foo() OVERRIDE;
-#if defined(_MSC_VER)
-#define OVERRIDE override
-#elif defined(__clang__)
-// Clang defaults to C++03 and warns about using override. Squelch that.
-// Intentionally no push/pop here so all users of OVERRIDE ignore the warning
-// too. This is like passing -Wno-c++11-extensions, except that GCC won't die
-// (because it won't see this pragma).
-#pragma clang diagnostic ignored "-Wc++11-extensions"
-#define OVERRIDE override
-#elif defined(__GNUC__) && __cplusplus >= 201103 && \
-    (__GNUC__ * 10000 + __GNUC_MINOR__ * 100) >= 40700
-// GCC 4.7 supports explicit virtual overrides when C++11 support is enabled.
-#define OVERRIDE override
-#else
-#define OVERRIDE
-#endif
-
 // Annotate a function indicating the caller must examine the return value.
 // Use like:
 //   int foo() WARN_UNUSED_RESULT;
@@ -191,13 +169,23 @@ typedef unsigned __int64    uint64_t;
 #endif  // WARN_UNUSED_RESULT
 
 // Put after a variable that might not be used, to prevent compiler warnings:
-//   int result UNUSED = DoSomething();
+//   int result ATTRIBUTE_UNUSED = DoSomething();
 //   assert(result == 17);
-#ifndef UNUSED
-#ifdef __GNUC__
-#define UNUSED __attribute__((unused))
+#ifndef ATTRIBUTE_UNUSED
+#if defined(__GNUC__) || defined(__clang__)
+#define ATTRIBUTE_UNUSED __attribute__((unused))
 #else
-#define UNUSED
+#define ATTRIBUTE_UNUSED
+#endif
+#endif
+
+// Macro to be used for switch-case fallthrough (required for enabling
+// -Wimplicit-fallthrough warning on Clang).
+#ifndef FALLTHROUGH
+#if defined(__clang__)
+#define FALLTHROUGH() [[clang::fallthrough]]
+#else
+#define FALLTHROUGH() do { } while (0)
 #endif
 #endif
 
diff --git a/media/webrtc/trunk/webrtc/video/BUILD.gn b/media/webrtc/trunk/webrtc/video/BUILD.gn
index df9b20d5fc..ee55105798 100644
--- a/media/webrtc/trunk/webrtc/video/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/video/BUILD.gn
@@ -34,6 +34,9 @@ source_set("video") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  deps = [ "../video_engine:video_engine_core" ]
+  deps = [
+    "..:webrtc_common",
+    "../video_engine:video_engine_core",
+  ]
 }
 
diff --git a/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc b/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc
index 29fb7300c0..c968b71b6b 100644
--- a/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/bitrate_estimator_tests.cc
@@ -13,11 +13,12 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
@@ -48,7 +49,7 @@ class TraceObserver {
   }
 
   ~TraceObserver() {
-    Trace::SetTraceCallback(NULL);
+    Trace::SetTraceCallback(nullptr);
     Trace::ReturnTrace();
   }
 
@@ -67,9 +68,7 @@ class TraceObserver {
         : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
           done_(EventWrapper::Create()) {}
 
-    virtual void Print(TraceLevel level,
-                       const char* message,
-                       int length) OVERRIDE {
+    void Print(TraceLevel level, const char* message, int length) override {
       CriticalSectionScoped lock(crit_sect_.get());
       std::string msg(message);
       if (msg.find("BitrateEstimator") != std::string::npos) {
@@ -103,10 +102,10 @@ class TraceObserver {
 
    private:
     typedef std::list Strings;
-    const scoped_ptr crit_sect_;
+    const rtc::scoped_ptr crit_sect_;
     Strings received_log_lines_ GUARDED_BY(crit_sect_);
     Strings expected_log_lines_ GUARDED_BY(crit_sect_);
-    scoped_ptr done_;
+    rtc::scoped_ptr done_;
   };
 
   Callback callback_;
@@ -145,7 +144,7 @@ class BitrateEstimatorTest : public test::CallTest {
     send_config_ = VideoSendStream::Config();
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
     // Encoders will be set separately per stream.
-    send_config_.encoder_settings.encoder = NULL;
+    send_config_.encoder_settings.encoder = nullptr;
     send_config_.encoder_settings.payload_name = "FAKE";
     send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
     encoder_config_.streams = test::CreateVideoStreams(1);
@@ -183,8 +182,8 @@ class BitrateEstimatorTest : public test::CallTest {
     explicit Stream(BitrateEstimatorTest* test)
         : test_(test),
           is_sending_receiving_(false),
-          send_stream_(NULL),
-          receive_stream_(NULL),
+          send_stream_(nullptr),
+          receive_stream_(nullptr),
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
@@ -192,7 +191,7 @@ class BitrateEstimatorTest : public test::CallTest {
       test_->send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
           test_->send_config_, test_->encoder_config_);
-      assert(test_->encoder_config_.streams.size() == 1);
+      DCHECK_EQ(1u, test_->encoder_config_.streams.size());
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
           send_stream_->Input(),
           test_->encoder_config_.streams[0].width,
@@ -217,11 +216,11 @@ class BitrateEstimatorTest : public test::CallTest {
     }
 
     ~Stream() {
-      frame_generator_capturer_.reset(NULL);
+      frame_generator_capturer_.reset(nullptr);
       test_->sender_call_->DestroyVideoSendStream(send_stream_);
-      send_stream_ = NULL;
+      send_stream_ = nullptr;
       test_->receiver_call_->DestroyVideoReceiveStream(receive_stream_);
-      receive_stream_ = NULL;
+      receive_stream_ = nullptr;
     }
 
     void StopSending() {
@@ -238,7 +237,7 @@ class BitrateEstimatorTest : public test::CallTest {
     bool is_sending_receiving_;
     VideoSendStream* send_stream_;
     VideoReceiveStream* receive_stream_;
-    scoped_ptr frame_generator_capturer_;
+    rtc::scoped_ptr frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
@@ -246,8 +245,8 @@ class BitrateEstimatorTest : public test::CallTest {
   TraceObserver receiver_trace_;
   test::DirectTransport send_transport_;
   test::DirectTransport receive_transport_;
-  scoped_ptr sender_call_;
-  scoped_ptr receiver_call_;
+  rtc::scoped_ptr sender_call_;
+  rtc::scoped_ptr receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector streams_;
 };
diff --git a/media/webrtc/trunk/webrtc/video/call.cc b/media/webrtc/trunk/webrtc/video/call.cc
index 2b4f76f4c5..d9dc334c00 100644
--- a/media/webrtc/trunk/webrtc/video/call.cc
+++ b/media/webrtc/trunk/webrtc/video/call.cc
@@ -8,23 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include 
 #include 
 
 #include 
 #include 
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
+#include "webrtc/modules/video_render/include/video_render.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video_engine/include/vie_base.h"
@@ -37,10 +41,12 @@ namespace webrtc {
 const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
 const char* RtpExtension::kAbsSendTime =
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
 
 bool RtpExtension::IsSupported(const std::string& name) {
   return name == webrtc::RtpExtension::kTOffset ||
-         name == webrtc::RtpExtension::kAbsSendTime;
+         name == webrtc::RtpExtension::kAbsSendTime ||
+         name == webrtc::RtpExtension::kVideoRotation;
 }
 
 VideoEncoder* VideoEncoder::Create(VideoEncoder::EncoderType codec_type) {
@@ -50,8 +56,8 @@ VideoEncoder* VideoEncoder::Create(VideoEncoder::EncoderType codec_type) {
     case kVp9:
       return VP9Encoder::Create();
   }
-  assert(false);
-  return NULL;
+  RTC_NOTREACHED();
+  return nullptr;
 }
 
 VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) {
@@ -61,8 +67,8 @@ VideoDecoder* VideoDecoder::Create(VideoDecoder::DecoderType codec_type) {
     case kVp9:
       return VP9Decoder::Create();
   }
-  assert(false);
-  return NULL;
+  RTC_NOTREACHED();
+  return nullptr;
 }
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
@@ -74,23 +80,23 @@ class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
   explicit CpuOveruseObserverProxy(LoadObserver* overuse_callback)
       : crit_(CriticalSectionWrapper::CreateCriticalSection()),
         overuse_callback_(overuse_callback) {
-    assert(overuse_callback != NULL);
+    DCHECK(overuse_callback != nullptr);
   }
 
   virtual ~CpuOveruseObserverProxy() {}
 
-  virtual void OveruseDetected() OVERRIDE {
+  void OveruseDetected() override {
     CriticalSectionScoped lock(crit_.get());
     overuse_callback_->OnLoadUpdate(LoadObserver::kOveruse);
   }
 
-  virtual void NormalUsage() OVERRIDE {
+  void NormalUsage() override {
     CriticalSectionScoped lock(crit_.get());
     overuse_callback_->OnLoadUpdate(LoadObserver::kUnderuse);
   }
 
  private:
-  const scoped_ptr crit_;
+  const rtc::scoped_ptr crit_;
   LoadObserver* overuse_callback_ GUARDED_BY(crit_);
 };
 
@@ -99,27 +105,27 @@ class Call : public webrtc::Call, public PacketReceiver {
   Call(webrtc::VideoEngine* video_engine, const Call::Config& config);
   virtual ~Call();
 
-  virtual PacketReceiver* Receiver() OVERRIDE;
+  PacketReceiver* Receiver() override;
 
-  virtual VideoSendStream* CreateVideoSendStream(
+  VideoSendStream* CreateVideoSendStream(
       const VideoSendStream::Config& config,
-      const VideoEncoderConfig& encoder_config) OVERRIDE;
+      const VideoEncoderConfig& encoder_config) override;
 
-  virtual void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream)
-      OVERRIDE;
+  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
 
-  virtual VideoReceiveStream* CreateVideoReceiveStream(
-      const VideoReceiveStream::Config& config) OVERRIDE;
+  VideoReceiveStream* CreateVideoReceiveStream(
+      const VideoReceiveStream::Config& config) override;
 
-  virtual void DestroyVideoReceiveStream(
-      webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
+  void DestroyVideoReceiveStream(
+      webrtc::VideoReceiveStream* receive_stream) override;
 
-  virtual Stats GetStats() const OVERRIDE;
+  Stats GetStats() const override;
 
-  virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                       size_t length) OVERRIDE;
+  DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) override;
 
-  virtual void SignalNetworkState(NetworkState state) OVERRIDE;
+  void SetBitrateConfig(
+      const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+  void SignalNetworkState(NetworkState state) override;
 
  private:
   DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length);
@@ -130,35 +136,39 @@ class Call : public webrtc::Call, public PacketReceiver {
   // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
   // ensures that we have a consistent network state signalled to all senders
   // and receivers.
-  scoped_ptr network_enabled_crit_;
+  rtc::scoped_ptr network_enabled_crit_;
   bool network_enabled_ GUARDED_BY(network_enabled_crit_);
 
-  scoped_ptr receive_crit_;
+  rtc::scoped_ptr receive_crit_;
   std::map receive_ssrcs_
       GUARDED_BY(receive_crit_);
 
-  scoped_ptr send_crit_;
+  rtc::scoped_ptr send_crit_;
   std::map send_ssrcs_ GUARDED_BY(send_crit_);
 
-  scoped_ptr overuse_observer_proxy_;
+  rtc::scoped_ptr overuse_observer_proxy_;
 
   VideoSendStream::RtpStateMap suspended_send_ssrcs_;
 
   VideoEngine* video_engine_;
   ViERTP_RTCP* rtp_rtcp_;
   ViECodec* codec_;
+  ViERender* render_;
   ViEBase* base_;
+  ViENetwork* network_;
   int base_channel_id_;
 
+  rtc::scoped_ptr external_render_;
+
   DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
-  VideoEngine* video_engine = config.webrtc_config != NULL
+  VideoEngine* video_engine = config.webrtc_config != nullptr
                                   ? VideoEngine::Create(*config.webrtc_config)
                                   : VideoEngine::Create();
-  assert(video_engine != NULL);
+  DCHECK(video_engine != nullptr);
 
   return new internal::Call(video_engine, config);
 }
@@ -172,36 +182,65 @@ Call::Call(webrtc::VideoEngine* video_engine, const Call::Config& config)
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       video_engine_(video_engine),
-      base_channel_id_(-1) {
-  assert(video_engine != NULL);
-  assert(config.send_transport != NULL);
+      base_channel_id_(-1),
+      external_render_(
+          VideoRender::CreateVideoRender(42, nullptr, false, kRenderExternal)) {
+  DCHECK(video_engine != nullptr);
+  DCHECK(config.send_transport != nullptr);
+
+  DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
+  DCHECK_GE(config.bitrate_config.start_bitrate_bps,
+            config.bitrate_config.min_bitrate_bps);
+  if (config.bitrate_config.max_bitrate_bps != -1) {
+    DCHECK_GE(config.bitrate_config.max_bitrate_bps,
+              config.bitrate_config.start_bitrate_bps);
+  }
 
   if (config.overuse_callback) {
     overuse_observer_proxy_.reset(
         new CpuOveruseObserverProxy(config.overuse_callback));
   }
 
+  render_ = ViERender::GetInterface(video_engine_);
+  DCHECK(render_ != nullptr);
+
+  render_->RegisterVideoRenderModule(*external_render_.get());
+
   rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_);
-  assert(rtp_rtcp_ != NULL);
+  DCHECK(rtp_rtcp_ != nullptr);
 
   codec_ = ViECodec::GetInterface(video_engine_);
-  assert(codec_ != NULL);
+  DCHECK(codec_ != nullptr);
+
+  network_ = ViENetwork::GetInterface(video_engine_);
 
   // As a workaround for non-existing calls in the old API, create a base
   // channel used as default channel when creating send and receive streams.
   base_ = ViEBase::GetInterface(video_engine_);
-  assert(base_ != NULL);
+  DCHECK(base_ != nullptr);
 
   base_->CreateChannel(base_channel_id_);
-  assert(base_channel_id_ != -1);
+  DCHECK(base_channel_id_ != -1);
+
+  network_->SetBitrateConfig(base_channel_id_,
+                             config_.bitrate_config.min_bitrate_bps,
+                             config_.bitrate_config.start_bitrate_bps,
+                             config_.bitrate_config.max_bitrate_bps);
 }
 
 Call::~Call() {
+  CHECK_EQ(0u, send_ssrcs_.size());
+  CHECK_EQ(0u, receive_ssrcs_.size());
   base_->DeleteChannel(base_channel_id_);
+
+  render_->DeRegisterVideoRenderModule(*external_render_.get());
+
   base_->Release();
+  network_->Release();
   codec_->Release();
+  render_->Release();
   rtp_rtcp_->Release();
-  webrtc::VideoEngine::Delete(video_engine_);
+  CHECK(webrtc::VideoEngine::Delete(video_engine_));
 }
 
 PacketReceiver* Call::Receiver() { return this; }
@@ -209,26 +248,22 @@ PacketReceiver* Call::Receiver() { return this; }
 VideoSendStream* Call::CreateVideoSendStream(
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
-  assert(config.rtp.ssrcs.size() > 0);
+  TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
+  LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
+  DCHECK(!config.rtp.ssrcs.empty());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
-  VideoSendStream* send_stream =
-      new VideoSendStream(config_.send_transport,
-                          overuse_observer_proxy_.get(),
-                          video_engine_,
-                          config,
-                          encoder_config,
-                          suspended_send_ssrcs_,
-                          base_channel_id_,
-                          config_.stream_start_bitrate_bps);
+  VideoSendStream* send_stream = new VideoSendStream(
+      config_.send_transport, overuse_observer_proxy_.get(), video_engine_,
+      config, encoder_config, suspended_send_ssrcs_, base_channel_id_);
 
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
   CriticalSectionScoped lock(network_enabled_crit_.get());
   WriteLockScoped write_lock(*send_crit_);
   for (size_t i = 0; i < config.rtp.ssrcs.size(); ++i) {
-    assert(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
+    DCHECK(send_ssrcs_.find(config.rtp.ssrcs[i]) == send_ssrcs_.end());
     send_ssrcs_[config.rtp.ssrcs[i]] = send_stream;
   }
   if (!network_enabled_)
@@ -237,11 +272,12 @@ VideoSendStream* Call::CreateVideoSendStream(
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
-  assert(send_stream != NULL);
+  TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
+  DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
-  VideoSendStream* send_stream_impl = NULL;
+  VideoSendStream* send_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*send_crit_);
     std::map::iterator it = send_ssrcs_.begin();
@@ -254,6 +290,7 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
       }
     }
   }
+  CHECK(send_stream_impl != nullptr);
 
   VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates();
 
@@ -263,12 +300,13 @@ void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
     suspended_send_ssrcs_[it->first] = it->second;
   }
 
-  assert(send_stream_impl != NULL);
   delete send_stream_impl;
 }
 
 VideoReceiveStream* Call::CreateVideoReceiveStream(
     const VideoReceiveStream::Config& config) {
+  TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
+  LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
   VideoReceiveStream* receive_stream =
       new VideoReceiveStream(video_engine_,
                              config,
@@ -280,7 +318,7 @@ VideoReceiveStream* Call::CreateVideoReceiveStream(
   // while changing network state.
   CriticalSectionScoped lock(network_enabled_crit_.get());
   WriteLockScoped write_lock(*receive_crit_);
-  assert(receive_ssrcs_.find(config.rtp.remote_ssrc) == receive_ssrcs_.end());
+  DCHECK(receive_ssrcs_.find(config.rtp.remote_ssrc) == receive_ssrcs_.end());
   receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
   // TODO(pbos): Configure different RTX payloads per receive payload.
   VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
@@ -295,9 +333,10 @@ VideoReceiveStream* Call::CreateVideoReceiveStream(
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
-  assert(receive_stream != NULL);
+  TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
+  DCHECK(receive_stream != nullptr);
 
-  VideoReceiveStream* receive_stream_impl = NULL;
+  VideoReceiveStream* receive_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*receive_crit_);
     // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
@@ -306,8 +345,8 @@ void Call::DestroyVideoReceiveStream(
         receive_ssrcs_.begin();
     while (it != receive_ssrcs_.end()) {
       if (it->second == static_cast(receive_stream)) {
-        assert(receive_stream_impl == NULL ||
-            receive_stream_impl == it->second);
+        if (receive_stream_impl != nullptr)
+          DCHECK(receive_stream_impl == it->second);
         receive_stream_impl = it->second;
         receive_ssrcs_.erase(it++);
       } else {
@@ -315,8 +354,7 @@ void Call::DestroyVideoReceiveStream(
       }
     }
   }
-
-  assert(receive_stream_impl != NULL);
+  CHECK(receive_stream_impl != nullptr);
   delete receive_stream_impl;
 }
 
@@ -337,11 +375,36 @@ Call::Stats Call::GetStats() const {
          ++it) {
       stats.pacer_delay_ms =
           std::max(it->second->GetPacerQueuingDelayMs(), stats.pacer_delay_ms);
+      int rtt_ms = it->second->GetRtt();
+      if (rtt_ms > 0)
+        stats.rtt_ms = rtt_ms;
     }
   }
   return stats;
 }
 
+void Call::SetBitrateConfig(
+    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
+  TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
+  DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
+  if (bitrate_config.max_bitrate_bps != -1)
+    DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
+  if (config_.bitrate_config.min_bitrate_bps ==
+          bitrate_config.min_bitrate_bps &&
+      (bitrate_config.start_bitrate_bps <= 0 ||
+       config_.bitrate_config.start_bitrate_bps ==
+           bitrate_config.start_bitrate_bps) &&
+      config_.bitrate_config.max_bitrate_bps ==
+          bitrate_config.max_bitrate_bps) {
+    // Nothing new to set, early abort to avoid encoder reconfigurations.
+    return;
+  }
+  config_.bitrate_config = bitrate_config;
+  network_->SetBitrateConfig(base_channel_id_, bitrate_config.min_bitrate_bps,
+                             bitrate_config.start_bitrate_bps,
+                             bitrate_config.max_bitrate_bps);
+}
+
 void Call::SignalNetworkState(NetworkState state) {
   // Take crit for entire function, it needs to be held while updating streams
   // to guarantee a consistent state across streams.
@@ -404,8 +467,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(const uint8_t* packet,
   if (length < 12)
     return DELIVERY_PACKET_ERROR;
 
-  const uint8_t* ptr = &packet[8];
-  uint32_t ssrc = ptr[0] << 24 | ptr[1] << 16 | ptr[2] << 8 | ptr[3];
+  uint32_t ssrc = ByteReader::ReadBigEndian(&packet[8]);
 
   ReadLockScoped read_lock(*receive_crit_);
   std::map::iterator it =
diff --git a/media/webrtc/trunk/webrtc/video/call_perf_tests.cc b/media/webrtc/trunk/webrtc/video/call_perf_tests.cc
index 9194b088dc..182a83edce 100644
--- a/media/webrtc/trunk/webrtc/video/call_perf_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/call_perf_tests.cc
@@ -7,14 +7,14 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include 
-
 #include 
 #include 
 #include 
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
@@ -22,7 +22,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/rtp_to_ntp.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
@@ -47,6 +46,8 @@ class CallPerfTest : public test::CallTest {
  protected:
   void TestAudioVideoSync(bool fec);
 
+  void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
+
   void TestMinTransmitBitrate(bool pad_to_min_bitrate);
 
   void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
@@ -61,7 +62,7 @@ class SyncRtcpObserver : public test::RtpRtcpObserver {
       : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs, config),
         crit_(CriticalSectionWrapper::CreateCriticalSection()) {}
 
-  virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+  Action OnSendRtcp(const uint8_t* packet, size_t length) override {
     RTCPUtility::RTCPParserV2 parser(packet, length, true);
     EXPECT_TRUE(parser.IsValid());
 
@@ -113,7 +114,7 @@ class SyncRtcpObserver : public test::RtpRtcpObserver {
     ntp_rtp_pairs_.push_front(ntp_rtp_pair);
   }
 
-  const scoped_ptr crit_;
+  const rtc::scoped_ptr crit_;
   RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
 };
 
@@ -135,8 +136,8 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
         creation_time_ms_(clock_->TimeInMilliseconds()),
         first_time_in_sync_(-1) {}
 
-  virtual void RenderFrame(const I420VideoFrame& video_frame,
-                           int time_to_render_ms) OVERRIDE {
+  void RenderFrame(const I420VideoFrame& video_frame,
+                   int time_to_render_ms) override {
     int64_t now_ms = clock_->TimeInMilliseconds();
     uint32_t playout_timestamp = 0;
     if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0)
@@ -178,6 +179,8 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
     }
   }
 
+  bool IsTextureSupported() const override { return false; }
+
  private:
   Clock* const clock_;
   int voe_channel_;
@@ -194,15 +197,14 @@ void CallPerfTest::TestAudioVideoSync(bool fec) {
         : channel_(channel),
           voe_network_(voe_network),
           parser_(RtpHeaderParser::Create()) {}
-    virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                         size_t length) OVERRIDE {
+    DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                 size_t length) override {
       int ret;
-      if (parser_->IsRtcp(packet, static_cast(length))) {
-        ret = voe_network_->ReceivedRTCPPacket(
-            channel_, packet, static_cast(length));
+      if (parser_->IsRtcp(packet, length)) {
+        ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
       } else {
-        ret = voe_network_->ReceivedRTPPacket(
-            channel_, packet, static_cast(length), PacketTime());
+        ret = voe_network_->ReceivedRTPPacket(channel_, packet, length,
+                                              PacketTime());
       }
       return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
     }
@@ -210,7 +212,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec) {
    private:
     int channel_;
     VoENetwork* voe_network_;
-    scoped_ptr parser_;
+    rtc::scoped_ptr parser_;
   };
 
   VoiceEngine* voice_engine = VoiceEngine::Create();
@@ -223,7 +225,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec) {
   ASSERT_STRNE("", audio_filename.c_str());
   test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
                                           audio_filename);
-  EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL));
+  EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
   int channel = voe_base->CreateChannel();
 
   FakeNetworkPipe::Config net_config;
@@ -327,13 +329,13 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
           start_time_ms_(start_time_ms),
           run_time_ms_(run_time_ms),
           creation_time_ms_(clock_->TimeInMilliseconds()),
-          capturer_(NULL),
+          capturer_(nullptr),
           rtp_start_timestamp_set_(false),
           rtp_start_timestamp_(0) {}
 
    private:
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int time_to_render_ms) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int time_to_render_ms) override {
       if (video_frame.ntp_time_ms() <= 0) {
         // Haven't got enough RTCP SR in order to calculate the capture ntp
         // time.
@@ -371,6 +373,8 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
       EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
     }
 
+    bool IsTextureSupported() const override { return false; }
+
     virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
@@ -391,21 +395,20 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
       return SEND_PACKET;
     }
 
-    virtual void OnFrameGeneratorCapturerCreated(
-        test::FrameGeneratorCapturer* frame_generator_capturer) OVERRIDE {
+    void OnFrameGeneratorCapturerCreated(
+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
       capturer_ = frame_generator_capturer;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       (*receive_configs)[0].renderer = this;
       // Enable the receiver side rtt calculation.
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
                                            "estimated capture NTP time to be "
                                            "within bounds.";
@@ -449,31 +452,54 @@ TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
   TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
 }
 
-TEST_F(CallPerfTest, RegisterCpuOveruseObserver) {
-  // Verifies that either a normal or overuse callback is triggered.
+void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
+                                  int encode_delay_ms) {
   class LoadObserver : public test::SendTest, public webrtc::LoadObserver {
    public:
-    LoadObserver() : SendTest(kLongTimeoutMs) {}
+    LoadObserver(LoadObserver::Load tested_load, int encode_delay_ms)
+        : SendTest(kLongTimeoutMs),
+          tested_load_(tested_load),
+          encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {}
 
-    virtual void OnLoadUpdate(Load load) OVERRIDE {
-      observation_complete_->Set();
+    void OnLoadUpdate(Load load) override {
+      if (load == tested_load_)
+        observation_complete_->Set();
     }
 
-    virtual Call::Config GetSenderCallConfig() OVERRIDE {
+    Call::Config GetSenderCallConfig() override {
       Call::Config config(SendTransport());
       config.overuse_callback = this;
       return config;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      send_config->encoder_settings.encoder = &encoder_;
+    }
+
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out before receiving an overuse callback.";
     }
-  } test;
+
+    LoadObserver::Load tested_load_;
+    test::DelayedEncoder encoder_;
+  } test(tested_load, encode_delay_ms);
 
   RunBaseTest(&test);
 }
 
+TEST_F(CallPerfTest, ReceivesCpuUnderuse) {
+  const int kEncodeDelayMs = 2;
+  TestCpuOveruse(LoadObserver::kUnderuse, kEncodeDelayMs);
+}
+
+TEST_F(CallPerfTest, ReceivesCpuOveruse) {
+  const int kEncodeDelayMs = 35;
+  TestCpuOveruse(LoadObserver::kOveruse, kEncodeDelayMs);
+}
+
 void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
   static const int kMaxEncodeBitrateKbps = 30;
   static const int kMinTransmitBitrateBps = 150000;
@@ -484,24 +510,23 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
    public:
     explicit BitrateObserver(bool using_min_transmit_bitrate)
         : EndToEndTest(kLongTimeoutMs),
-          send_stream_(NULL),
-          send_transport_receiver_(NULL),
+          send_stream_(nullptr),
+          send_transport_receiver_(nullptr),
           pad_to_min_bitrate_(using_min_transmit_bitrate),
           num_bitrate_observations_in_range_(0) {}
 
    private:
-    virtual void SetReceivers(PacketReceiver* send_transport_receiver,
-                              PacketReceiver* receive_transport_receiver)
-        OVERRIDE {
+    void SetReceivers(PacketReceiver* send_transport_receiver,
+                      PacketReceiver* receive_transport_receiver) override {
       send_transport_receiver_ = send_transport_receiver;
       test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
     }
 
-    virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                         size_t length) OVERRIDE {
+    DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                 size_t length) override {
       VideoSendStream::Stats stats = send_stream_->GetStats();
       if (stats.substreams.size() > 0) {
-        assert(stats.substreams.size() == 1);
+        DCHECK_EQ(1u, stats.substreams.size());
         int bitrate_kbps =
             stats.substreams.begin()->second.total_bitrate_bps / 1000;
         if (bitrate_kbps > 0) {
@@ -533,24 +558,23 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
       return send_transport_receiver_->DeliverPacket(packet, length);
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       send_stream_ = send_stream;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       if (pad_to_min_bitrate_) {
         encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
       } else {
-        assert(encoder_config->min_transmit_bitrate_bps == 0);
+        DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
       }
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timeout while waiting for send-bitrate stats.";
     }
@@ -584,9 +608,9 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
           time_to_reconfigure_(webrtc::EventWrapper::Create()),
           encoder_inits_(0) {}
 
-    virtual int32_t InitEncode(const VideoCodec* config,
-                               int32_t number_of_cores,
-                               uint32_t max_payload_size) OVERRIDE {
+    int32_t InitEncode(const VideoCodec* config,
+                       int32_t number_of_cores,
+                       size_t max_payload_size) override {
       if (encoder_inits_ == 0) {
         EXPECT_EQ(kInitialBitrateKbps, config->startBitrate)
             << "Encoder not initialized at expected bitrate.";
@@ -603,8 +627,8 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
       return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
     }
 
-    virtual int32_t SetRates(uint32_t new_target_bitrate_kbps,
-                             uint32_t framerate) OVERRIDE {
+    int32_t SetRates(uint32_t new_target_bitrate_kbps,
+                     uint32_t framerate) override {
       last_set_bitrate_ = new_target_bitrate_kbps;
       if (encoder_inits_ == 1 &&
           new_target_bitrate_kbps > kReconfigureThresholdKbps) {
@@ -613,16 +637,15 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
       return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
     }
 
-    Call::Config GetSenderCallConfig() OVERRIDE {
+    Call::Config GetSenderCallConfig() override {
       Call::Config config = EndToEndTest::GetSenderCallConfig();
-      config.stream_start_bitrate_bps = kInitialBitrateKbps * 1000;
+      config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
       return config;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config->streams[0].min_bitrate_bps = 50000;
       encoder_config->streams[0].target_bitrate_bps =
@@ -631,13 +654,13 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
       encoder_config_ = *encoder_config;
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       send_stream_ = send_stream;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       ASSERT_EQ(kEventSignaled, time_to_reconfigure_->Wait(kDefaultTimeoutMs))
           << "Timed out before receiving an initial high bitrate.";
       encoder_config_.streams[0].width *= 2;
@@ -649,7 +672,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
     }
 
    private:
-    scoped_ptr time_to_reconfigure_;
+    rtc::scoped_ptr time_to_reconfigure_;
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
diff --git a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc
index f5eca7ce44..1261ad5123 100644
--- a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc
+++ b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/video_coding/main/source/encoded_frame.h"
 
 namespace webrtc {
@@ -22,10 +23,10 @@ EncodedFrameCallbackAdapter::EncodedFrameCallbackAdapter(
 EncodedFrameCallbackAdapter::~EncodedFrameCallbackAdapter() {}
 
 int32_t EncodedFrameCallbackAdapter::Encoded(
-    EncodedImage& encodedImage,
+    const EncodedImage& encodedImage,
     const CodecSpecificInfo* codecSpecificInfo,
     const RTPFragmentationHeader* fragmentation) {
-  assert(observer_ != NULL);
+  DCHECK(observer_ != nullptr);
   FrameType frame_type =
         VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
   const EncodedFrame frame(encodedImage._buffer,
diff --git a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h
index d3814797ad..b39a8e2167 100644
--- a/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h
+++ b/media/webrtc/trunk/webrtc/video/encoded_frame_callback_adapter.h
@@ -22,7 +22,7 @@ class EncodedFrameCallbackAdapter : public EncodedImageCallback {
   explicit EncodedFrameCallbackAdapter(EncodedFrameObserver* observer);
   virtual ~EncodedFrameCallbackAdapter();
 
-  virtual int32_t Encoded(EncodedImage& encodedImage,
+  virtual int32_t Encoded(const EncodedImage& encodedImage,
                           const CodecSpecificInfo* codecSpecificInfo,
                           const RTPFragmentationHeader* fragmentation);
 
diff --git a/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc b/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc
index 2b3c00f5f5..956e8219fa 100644
--- a/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/end_to_end_tests.cc
@@ -7,8 +7,6 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include 
-
 #include 
 #include 
 #include 
@@ -16,6 +14,8 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@@ -24,7 +24,7 @@
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
@@ -34,7 +34,9 @@
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
+#include "webrtc/test/histogram.h"
 #include "webrtc/test/null_transport.h"
+#include "webrtc/test/rtcp_packet_parser.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
@@ -51,19 +53,19 @@ class EndToEndTest : public test::CallTest {
   EndToEndTest() {}
 
   virtual ~EndToEndTest() {
-    EXPECT_EQ(NULL, send_stream_);
+    EXPECT_EQ(nullptr, send_stream_);
     EXPECT_TRUE(receive_streams_.empty());
   }
 
  protected:
   class UnusedTransport : public newapi::Transport {
    private:
-    virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    bool SendRtp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected RTP sent.";
       return false;
     }
 
-    virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    bool SendRtcp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected RTCP sent.";
       return false;
     }
@@ -75,6 +77,7 @@ class EndToEndTest : public test::CallTest {
   void TestXrReceiverReferenceTimeReport(bool enable_rrtr);
   void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
   void TestRtpStatePreservation(bool use_rtx);
+  void TestReceivedFecPacketsNotNacked(const FakeNetworkPipe::Config& config);
 };
 
 TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
@@ -119,14 +122,16 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
    public:
     Renderer() : event_(EventWrapper::Create()) {}
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int /*time_to_render_ms*/) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int /*time_to_render_ms*/) override {
       event_->Set();
     }
 
+    bool IsTextureSupported() const override { return false; }
+
     EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
 
-    scoped_ptr event_;
+    rtc::scoped_ptr event_;
   } renderer;
 
   class TestFrameCallback : public I420FrameCallback {
@@ -136,12 +141,12 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
     EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
 
    private:
-    virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE {
+    void FrameCallback(I420VideoFrame* frame) override {
       SleepMs(kDelayRenderCallbackMs);
       event_->Set();
     }
 
-    scoped_ptr event_;
+    rtc::scoped_ptr event_;
   };
 
   test::DirectTransport sender_transport, receiver_transport;
@@ -164,9 +169,9 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
-  scoped_ptr frame_generator(
-      test::FrameGenerator::Create(kWidth, kHeight));
-  send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
+  rtc::scoped_ptr frame_generator(
+      test::FrameGenerator::CreateChromaGenerator(kWidth, kHeight));
+  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
   EXPECT_EQ(kEventSignaled, pre_render_callback.Wait())
       << "Timed out while waiting for pre-render callback.";
   EXPECT_EQ(kEventSignaled, renderer.Wait())
@@ -185,14 +190,15 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) {
    public:
     Renderer() : event_(EventWrapper::Create()) {}
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int /*time_to_render_ms*/) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int /*time_to_render_ms*/) override {
       event_->Set();
     }
+    bool IsTextureSupported() const override { return false; }
 
     EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
 
-    scoped_ptr event_;
+    rtc::scoped_ptr event_;
   } renderer;
 
   test::DirectTransport sender_transport, receiver_transport;
@@ -210,9 +216,10 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) {
   CreateStreams();
   Start();
 
-  scoped_ptr frame_generator(test::FrameGenerator::Create(
-      encoder_config_.streams[0].width, encoder_config_.streams[0].height));
-  send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
+  rtc::scoped_ptr frame_generator(
+      test::FrameGenerator::CreateChromaGenerator(
+          encoder_config_.streams[0].width, encoder_config_.streams[0].height));
+  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
 
   EXPECT_EQ(kEventSignaled, renderer.Wait())
       << "Timed out while waiting for the frame to render.";
@@ -234,15 +241,14 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) {
           decoder_(VP9Decoder::Create()),
           frame_counter_(0) {}
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for enough frames to be decoded.";
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "VP9";
       send_config->encoder_settings.payload_type = VCM_VP9_PAYLOAD_TYPE;
@@ -259,16 +265,18 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) {
       (*receive_configs)[0].decoders[0].decoder = decoder_.get();
     }
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int time_to_render_ms) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int time_to_render_ms) override {
       const int kRequiredFrames = 500;
       if (++frame_counter_ == kRequiredFrames)
         observation_complete_->Set();
     }
 
+    bool IsTextureSupported() const override { return false; }
+
    private:
-    scoped_ptr encoder_;
-    scoped_ptr decoder_;
+    rtc::scoped_ptr encoder_;
+    rtc::scoped_ptr decoder_;
     int frame_counter_;
   } test;
 
@@ -283,15 +291,16 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
           fake_encoder_(Clock::GetRealTimeClock()),
           frame_counter_(0) {}
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for enough frames to be decoded.";
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.nack.rtp_history_ms =
+          (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->encoder_settings.encoder = &fake_encoder_;
       send_config->encoder_settings.payload_name = "H264";
       send_config->encoder_settings.payload_type = kFakeSendPayloadType;
@@ -308,13 +317,15 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {
       (*receive_configs)[0].decoders[0].decoder = &fake_decoder_;
     }
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int time_to_render_ms) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int time_to_render_ms) override {
       const int kRequiredFrames = 500;
       if (++frame_counter_ == kRequiredFrames)
         observation_complete_->Set();
     }
 
+    bool IsTextureSupported() const override { return false; }
+
    private:
     test::FakeH264Decoder fake_decoder_;
     test::FakeH264Encoder fake_encoder_;
@@ -329,8 +340,7 @@ TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
    public:
     SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
       uint32_t ssrc = 0;
@@ -344,7 +354,7 @@ TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {
       return SEND_PACKET;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for a receiver RTCP packet to be sent.";
     }
@@ -367,7 +377,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
           nacks_left_(kNumberOfNacksToObserve) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
 
@@ -401,8 +411,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
@@ -417,21 +426,20 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out waiting for packets to be NACKed, retransmitted and "
              "rendered.";
     }
 
-    scoped_ptr rtp_parser_;
+    rtc::scoped_ptr rtp_parser_;
     std::set dropped_packets_;
     std::set retransmitted_packets_;
     uint64_t sent_rtp_packets_;
@@ -442,18 +450,14 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {
   RunBaseTest(&test);
 }
 
-// TODO(pbos): Flaky, webrtc:3269
-TEST_F(EndToEndTest, DISABLED_CanReceiveFec) {
+TEST_F(EndToEndTest, CanReceiveFec) {
   class FecRenderObserver : public test::EndToEndTest, public VideoRenderer {
    public:
     FecRenderObserver()
-        : EndToEndTest(kDefaultTimeoutMs),
-          state_(kFirstPacket),
-          protected_sequence_number_(0),
-          protected_frame_timestamp_(0) {}
+        : EndToEndTest(kDefaultTimeoutMs), state_(kFirstPacket) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE
+    Action OnSendRtp(const uint8_t* packet, size_t length) override
         EXCLUSIVE_LOCKS_REQUIRED(crit_) {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
@@ -464,6 +468,14 @@ TEST_F(EndToEndTest, DISABLED_CanReceiveFec) {
       if (encapsulated_payload_type != kFakeSendPayloadType)
         EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
 
+      if (protected_sequence_numbers_.count(header.sequenceNumber) != 0) {
+        // Retransmitted packet, should not count.
+        protected_sequence_numbers_.erase(header.sequenceNumber);
+        EXPECT_GT(protected_timestamps_.count(header.timestamp), 0u);
+        protected_timestamps_.erase(header.timestamp);
+        return SEND_PACKET;
+      }
+
       switch (state_) {
         case kFirstPacket:
           state_ = kDropEveryOtherPacketUntilFec;
@@ -478,43 +490,37 @@ TEST_F(EndToEndTest, DISABLED_CanReceiveFec) {
           break;
         case kDropNextMediaPacket:
           if (encapsulated_payload_type == kFakeSendPayloadType) {
-            protected_sequence_number_ = header.sequenceNumber;
-            protected_frame_timestamp_ = header.timestamp;
-            state_ = kProtectedPacketDropped;
+            protected_sequence_numbers_.insert(header.sequenceNumber);
+            protected_timestamps_.insert(header.timestamp);
+            state_ = kDropEveryOtherPacketUntilFec;
             return DROP_PACKET;
           }
           break;
-        case kProtectedPacketDropped:
-          EXPECT_NE(header.sequenceNumber, protected_sequence_number_)
-              << "Protected packet retransmitted. Should not happen with FEC.";
-          break;
       }
 
       return SEND_PACKET;
     }
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int time_to_render_ms) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int time_to_render_ms) override {
       CriticalSectionScoped lock(crit_.get());
-      // Rendering frame with timestamp associated with dropped packet -> FEC
+      // Rendering frame with timestamp of packet that was dropped -> FEC
       // protection worked.
-      if (state_ == kProtectedPacketDropped &&
-          video_frame.timestamp() == protected_frame_timestamp_) {
+      if (protected_timestamps_.count(video_frame.timestamp()) != 0)
         observation_complete_->Set();
-      }
     }
 
+    bool IsTextureSupported() const override { return false; }
+
     enum {
       kFirstPacket,
       kDropEveryOtherPacketUntilFec,
       kDropNextMediaPacket,
-      kProtectedPacketDropped,
     } state_;
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       // TODO(pbos): Run this test with combined NACK/FEC enabled as well.
       // int rtp_history_ms = 1000;
       // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms;
@@ -527,19 +533,142 @@ TEST_F(EndToEndTest, DISABLED_CanReceiveFec) {
       (*receive_configs)[0].renderer = this;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
-          << "Timed out while waiting for retransmitted NACKed frames to be "
-             "rendered again.";
+          << "Timed out waiting for dropped frames frames to be rendered.";
     }
 
-    uint32_t protected_sequence_number_ GUARDED_BY(crit_);
-    uint32_t protected_frame_timestamp_ GUARDED_BY(crit_);
+    std::set protected_sequence_numbers_ GUARDED_BY(crit_);
+    std::set protected_timestamps_ GUARDED_BY(crit_);
   } test;
 
   RunBaseTest(&test);
 }
 
+// Flacky on all platforms. See webrtc:4328.
+TEST_F(EndToEndTest, DISABLED_ReceivedFecPacketsNotNacked) {
+  // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
+  // Configure some network delay.
+  const int kNetworkDelayMs = 50;
+  FakeNetworkPipe::Config config;
+  config.queue_delay_ms = kNetworkDelayMs;
+  TestReceivedFecPacketsNotNacked(config);
+}
+
+void EndToEndTest::TestReceivedFecPacketsNotNacked(
+    const FakeNetworkPipe::Config& config) {
+  class FecNackObserver : public test::EndToEndTest {
+   public:
+    explicit FecNackObserver(const FakeNetworkPipe::Config& config)
+        : EndToEndTest(kDefaultTimeoutMs, config),
+          state_(kFirstPacket),
+          fec_sequence_number_(0),
+          has_last_sequence_number_(false),
+          last_sequence_number_(0) {}
+
+   private:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      EXPECT_EQ(kRedPayloadType, header.payloadType);
+
+      int encapsulated_payload_type =
+          static_cast(packet[header.headerLength]);
+      if (encapsulated_payload_type != kFakeSendPayloadType)
+        EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
+
+      if (has_last_sequence_number_ &&
+          !IsNewerSequenceNumber(header.sequenceNumber,
+                                 last_sequence_number_)) {
+        // Drop retransmitted packets.
+        return DROP_PACKET;
+      }
+      last_sequence_number_ = header.sequenceNumber;
+      has_last_sequence_number_ = true;
+
+      bool fec_packet = encapsulated_payload_type == kUlpfecPayloadType;
+      switch (state_) {
+        case kFirstPacket:
+          state_ = kDropEveryOtherPacketUntilFec;
+          break;
+        case kDropEveryOtherPacketUntilFec:
+          if (fec_packet) {
+            state_ = kDropAllMediaPacketsUntilFec;
+          } else if (header.sequenceNumber % 2 == 0) {
+            return DROP_PACKET;
+          }
+          break;
+        case kDropAllMediaPacketsUntilFec:
+          if (!fec_packet)
+            return DROP_PACKET;
+          fec_sequence_number_ = header.sequenceNumber;
+          state_ = kVerifyFecPacketNotInNackList;
+          break;
+        case kVerifyFecPacketNotInNackList:
+          // Continue to drop packets. Make sure no frame can be decoded.
+          if (fec_packet || header.sequenceNumber % 2 == 0)
+            return DROP_PACKET;
+          break;
+      }
+      return SEND_PACKET;
+    }
+
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      if (state_ == kVerifyFecPacketNotInNackList) {
+        test::RtcpPacketParser rtcp_parser;
+        rtcp_parser.Parse(packet, length);
+        std::vector nacks = rtcp_parser.nack_item()->last_nack_list();
+        if (!nacks.empty() &&
+            IsNewerSequenceNumber(nacks.back(), fec_sequence_number_)) {
+          EXPECT_TRUE(std::find(
+              nacks.begin(), nacks.end(), fec_sequence_number_) == nacks.end());
+          observation_complete_->Set();
+        }
+      }
+      return SEND_PACKET;
+    }
+
+    // TODO(holmer): Investigate why we don't send FEC packets when the bitrate
+    // is 10 kbps.
+    Call::Config GetSenderCallConfig() override {
+      Call::Config config(SendTransport());
+      const int kMinBitrateBps = 30000;
+      config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
+      return config;
+    }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      // Configure hybrid NACK/FEC.
+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      send_config->rtp.fec.red_payload_type = kRedPayloadType;
+      send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      (*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
+      (*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+    }
+
+    void PerformTest() override {
+      EXPECT_EQ(kEventSignaled, Wait())
+          << "Timed out while waiting for FEC packets to be received.";
+    }
+
+    enum {
+      kFirstPacket,
+      kDropEveryOtherPacketUntilFec,
+      kDropAllMediaPacketsUntilFec,
+      kVerifyFecPacketNotInNackList,
+    } state_;
+
+    uint16_t fec_sequence_number_;
+    bool has_last_sequence_number_;
+    uint16_t last_sequence_number_;
+  } test(config);
+
+  RunBaseTest(&test);
+}
+
 // This test drops second RTP packet with a marker bit set, makes sure it's
 // retransmitted and renders. Retransmission SSRCs are also checked.
 void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
@@ -557,7 +686,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
           frame_retransmitted_(false) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -581,7 +710,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
       return SEND_PACKET;
     }
 
-    virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE {
+    void FrameCallback(I420VideoFrame* frame) override {
       CriticalSectionScoped lock(crit_.get());
       if (frame->timestamp() == retransmitted_timestamp_) {
         EXPECT_TRUE(frame_retransmitted_);
@@ -589,10 +718,9 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
       }
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].pre_render_callback = this;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
@@ -606,7 +734,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) {
       }
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for retransmission to render.";
     }
@@ -637,16 +765,18 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
    public:
     Renderer() : event_(EventWrapper::Create()) {}
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int /*time_to_render_ms*/) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int /*time_to_render_ms*/) override {
       EXPECT_EQ(0, *video_frame.buffer(kYPlane))
           << "Rendered frame should have zero luma which is applied by the "
              "pre-render callback.";
       event_->Set();
     }
 
+    bool IsTextureSupported() const override { return false; }
+
     EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); }
-    scoped_ptr event_;
+    rtc::scoped_ptr event_;
   } renderer;
 
   class TestFrameCallback : public I420FrameCallback {
@@ -677,7 +807,7 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
       event_->Set();
     }
 
-    scoped_ptr event_;
+    rtc::scoped_ptr event_;
     int expected_luma_byte_;
     int next_luma_byte_;
   };
@@ -694,7 +824,7 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
   CreateSendConfig(1);
-  scoped_ptr encoder(
+  rtc::scoped_ptr encoder(
       VideoEncoder::Create(VideoEncoder::kVp8));
   send_config_.encoder_settings.encoder = encoder.get();
   send_config_.encoder_settings.payload_name = "VP8";
@@ -712,9 +842,9 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
-  scoped_ptr frame_generator(
-      test::FrameGenerator::Create(kWidth / 2, kHeight / 2));
-  send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
+  rtc::scoped_ptr frame_generator(
+      test::FrameGenerator::CreateChromaGenerator(kWidth / 2, kHeight / 2));
+  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
 
   EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait())
       << "Timed out while waiting for pre-encode callback.";
@@ -745,7 +875,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
           received_pli_(false) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -762,8 +892,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
@@ -781,8 +910,8 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
       return SEND_PACKET;
     }
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int time_to_render_ms) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int time_to_render_ms) override {
       CriticalSectionScoped lock(crit_.get());
       if (received_pli_ &&
           video_frame.timestamp() > highest_dropped_timestamp_) {
@@ -792,16 +921,17 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
         frames_to_drop_ = kPacketsToDrop;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    bool IsTextureSupported() const override { return false; }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
       (*receive_configs)[0].renderer = this;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait()) << "Timed out waiting for PLI to be "
                                            "received and a frame to be "
                                            "rendered afterwards.";
@@ -837,8 +967,8 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
     }
 
    private:
-    virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                         size_t length) OVERRIDE {
+    DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                 size_t length) override {
       if (RtpHeaderParser::IsRtcp(packet, length)) {
         return receiver_->DeliverPacket(packet, length);
       } else {
@@ -851,7 +981,7 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
     }
 
     PacketReceiver* receiver_;
-    scoped_ptr delivered_packet_;
+    rtc::scoped_ptr delivered_packet_;
   };
 
   test::DirectTransport send_transport, receive_transport;
@@ -894,15 +1024,14 @@ void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
           sent_rtcp_(0) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (++sent_rtp_ % 3 == 0)
         return DROP_PACKET;
 
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       ++sent_rtcp_;
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
@@ -939,16 +1068,15 @@ void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << (rtcp_mode_ == newapi::kRtcpCompound
                   ? "Timed out before observing enough compound packets."
@@ -988,21 +1116,23 @@ TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
           height_(height),
           done_(EventWrapper::Create()) {}
 
-    virtual void RenderFrame(const I420VideoFrame& video_frame,
-                             int time_to_render_ms) OVERRIDE {
+    void RenderFrame(const I420VideoFrame& video_frame,
+                     int time_to_render_ms) override {
       EXPECT_EQ(width_, video_frame.width());
       EXPECT_EQ(height_, video_frame.height());
       (*capturer_)->Stop();
       done_->Set();
     }
 
+    bool IsTextureSupported() const override { return false; }
+
     EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
 
    private:
     test::FrameGeneratorCapturer** capturer_;
     int width_;
     int height_;
-    scoped_ptr done_;
+    rtc::scoped_ptr done_;
   };
 
   struct {
@@ -1012,8 +1142,9 @@ TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
   } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}};
 
   test::DirectTransport sender_transport, receiver_transport;
-  scoped_ptr sender_call(Call::Create(Call::Config(&sender_transport)));
-  scoped_ptr receiver_call(
+  rtc::scoped_ptr sender_call(
+      Call::Create(Call::Config(&sender_transport)));
+  rtc::scoped_ptr receiver_call(
       Call::Create(Call::Config(&receiver_transport)));
   sender_transport.SetReceiver(receiver_call->Receiver());
   receiver_transport.SetReceiver(sender_call->Receiver());
@@ -1024,7 +1155,7 @@ TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) {
   VideoOutputObserver* observers[kNumStreams];
   test::FrameGeneratorCapturer* frame_generators[kNumStreams];
 
-  scoped_ptr encoders[kNumStreams];
+  rtc::scoped_ptr encoders[kNumStreams];
   for (size_t i = 0; i < kNumStreams; ++i)
     encoders[i].reset(VideoEncoder::Create(VideoEncoder::kVp8));
 
@@ -1115,10 +1246,10 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) {
     }
 
    private:
-    scoped_ptr buffer_;
+    rtc::scoped_ptr buffer_;
     size_t length_;
     FrameType frame_type_;
-    scoped_ptr called_;
+    rtc::scoped_ptr called_;
   };
 
   EncodedFrameTestObserver post_encode_observer;
@@ -1140,9 +1271,10 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) {
   CreateStreams();
   Start();
 
-  scoped_ptr frame_generator(test::FrameGenerator::Create(
-      encoder_config_.streams[0].width, encoder_config_.streams[0].height));
-  send_stream_->Input()->SwapFrame(frame_generator->NextFrame());
+  rtc::scoped_ptr frame_generator(
+      test::FrameGenerator::CreateChromaGenerator(
+          encoder_config_.streams[0].width, encoder_config_.streams[0].height));
+  send_stream_->Input()->IncomingCapturedFrame(*frame_generator->NextFrame());
 
   EXPECT_EQ(kEventSignaled, post_encode_observer.Wait())
       << "Timed out while waiting for send-side encoded-frame callback.";
@@ -1165,8 +1297,7 @@ TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
    public:
     RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
@@ -1191,7 +1322,7 @@ TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {
         observation_complete_->Set();
       return SEND_PACKET;
     }
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for a "
                                            "receiver RTCP REMB packet to be "
                                            "sent.";
@@ -1206,36 +1337,35 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {
    public:
     RtcpObserver()
         : EndToEndTest(kDefaultTimeoutMs),
-          sender_call_(NULL),
-          receiver_call_(NULL),
+          sender_call_(nullptr),
+          receiver_call_(nullptr),
           has_seen_pacer_delay_(false) {}
 
-    virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                         size_t length) OVERRIDE {
+    DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                 size_t length) override {
       Call::Stats sender_stats = sender_call_->GetStats();
       Call::Stats receiver_stats = receiver_call_->GetStats();
       if (!has_seen_pacer_delay_)
         has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
       if (sender_stats.send_bandwidth_bps > 0 &&
-          receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_)
+          receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_) {
         observation_complete_->Set();
+      }
       return receiver_call_->Receiver()->DeliverPacket(packet, length);
     }
 
-    virtual void OnCallsCreated(Call* sender_call,
-                                Call* receiver_call) OVERRIDE {
+    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
                                            "non-zero bandwidth stats.";
     }
 
-    virtual void SetReceivers(
-        PacketReceiver* send_transport_receiver,
-        PacketReceiver* receive_transport_receiver) OVERRIDE {
+    void SetReceivers(PacketReceiver* send_transport_receiver,
+                      PacketReceiver* receive_transport_receiver) override {
       test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
     }
 
@@ -1248,6 +1378,115 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {
   RunBaseTest(&test);
 }
 
+TEST_F(EndToEndTest, VerifyNackStats) {
+  static const int kPacketNumberToDrop = 200;
+  class NackObserver : public test::EndToEndTest {
+   public:
+    NackObserver()
+        : EndToEndTest(kLongTimeoutMs),
+          sent_rtp_packets_(0),
+          dropped_rtp_packet_(0),
+          dropped_rtp_packet_requested_(false),
+          send_stream_(nullptr),
+          start_runtime_ms_(-1) {}
+
+   private:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      if (++sent_rtp_packets_ == kPacketNumberToDrop) {
+        rtc::scoped_ptr parser(RtpHeaderParser::Create());
+        RTPHeader header;
+        EXPECT_TRUE(parser->Parse(packet, length, &header));
+        dropped_rtp_packet_ = header.sequenceNumber;
+        return DROP_PACKET;
+      }
+      VerifyStats();
+      return SEND_PACKET;
+    }
+
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
+      test::RtcpPacketParser rtcp_parser;
+      rtcp_parser.Parse(packet, length);
+      std::vector nacks = rtcp_parser.nack_item()->last_nack_list();
+      if (!nacks.empty() && std::find(
+          nacks.begin(), nacks.end(), dropped_rtp_packet_) != nacks.end()) {
+        dropped_rtp_packet_requested_ = true;
+      }
+      return SEND_PACKET;
+    }
+
+    void VerifyStats() {
+      if (!dropped_rtp_packet_requested_)
+        return;
+      int send_stream_nack_packets = 0;
+      int receive_stream_nack_packets = 0;
+      VideoSendStream::Stats stats = send_stream_->GetStats();
+      for (std::map::const_iterator it =
+           stats.substreams.begin(); it != stats.substreams.end(); ++it) {
+        const VideoSendStream::StreamStats& stream_stats = it->second;
+        send_stream_nack_packets +=
+            stream_stats.rtcp_packet_type_counts.nack_packets;
+      }
+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
+        VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
+        receive_stream_nack_packets +=
+            stats.rtcp_packet_type_counts.nack_packets;
+      }
+      if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
+        // NACK packet sent on receive stream and received on sent stream.
+        if (MinMetricRunTimePassed())
+          observation_complete_->Set();
+      }
+    }
+
+    bool MinMetricRunTimePassed() {
+      int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
+      if (start_runtime_ms_ == -1) {
+        start_runtime_ms_ = now;
+        return false;
+      }
+      int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
+      return elapsed_sec > metrics::kMinRunTimeInSeconds;
+    }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+    }
+
+    void OnStreamsCreated(
+        VideoSendStream* send_stream,
+        const std::vector& receive_streams) override {
+      send_stream_ = send_stream;
+      receive_streams_ = receive_streams;
+    }
+
+    void PerformTest() override {
+      EXPECT_EQ(kEventSignaled, Wait())
+          << "Timed out waiting for packet to be NACKed.";
+    }
+
+    uint64_t sent_rtp_packets_;
+    uint16_t dropped_rtp_packet_;
+    bool dropped_rtp_packet_requested_;
+    std::vector receive_streams_;
+    VideoSendStream* send_stream_;
+    int64_t start_runtime_ms_;
+  } test;
+
+  RunBaseTest(&test);
+
+  EXPECT_NE(-1, test::LastHistogramSample(
+      "WebRTC.Video.UniqueNackRequestsSentInPercent"));
+  EXPECT_NE(-1, test::LastHistogramSample(
+      "WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
+  EXPECT_GT(test::LastHistogramSample(
+      "WebRTC.Video.NackPacketsSentPerMinute"), 0);
+  EXPECT_GT(test::LastHistogramSample(
+      "WebRTC.Video.NackPacketsReceivedPerMinute"), 0);
+}
+
 void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
   static const int kNumRtcpReportPacketsToObserve = 5;
   class RtcpXrObserver : public test::EndToEndTest {
@@ -1262,8 +1501,7 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
 
    private:
     // Receive stream should send RR packets (and RRTR packets if enabled).
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
@@ -1310,16 +1548,15 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize;
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
           enable_rrtr_;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for RTCP SR/RR packets to be sent.";
     }
@@ -1351,7 +1588,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -1376,12 +1613,11 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
       return SEND_PACKET;
     }
 
-    virtual size_t GetNumStreams() const OVERRIDE { return num_ssrcs_; }
+    size_t GetNumStreams() const override { return num_ssrcs_; }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       if (num_ssrcs_ > 1) {
         // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
         for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
@@ -1396,13 +1632,13 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
         encoder_config->streams.resize(1);
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       send_stream_ = send_stream;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for "
           << (send_single_ssrc_first_ ? "first SSRC." : "SSRCs.");
@@ -1432,40 +1668,95 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
   RunBaseTest(&test);
 }
 
+TEST_F(EndToEndTest, ReportsSetEncoderRates) {
+  class EncoderRateStatsTest : public test::EndToEndTest,
+                               public test::FakeEncoder {
+   public:
+    EncoderRateStatsTest()
+        : EndToEndTest(kDefaultTimeoutMs),
+          FakeEncoder(Clock::GetRealTimeClock()) {}
+
+    void OnStreamsCreated(
+        VideoSendStream* send_stream,
+        const std::vector& receive_streams) override {
+      send_stream_ = send_stream;
+    }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      send_config->encoder_settings.encoder = this;
+    }
+
+    int32_t SetRates(uint32_t new_target_bitrate, uint32_t framerate) override {
+      // Make sure not to trigger on any default zero bitrates.
+      if (new_target_bitrate == 0)
+        return 0;
+      CriticalSectionScoped lock(crit_.get());
+      bitrate_kbps_ = new_target_bitrate;
+      observation_complete_->Set();
+      return 0;
+    }
+
+    void PerformTest() override {
+      ASSERT_EQ(kEventSignaled, Wait())
+          << "Timed out while waiting for encoder SetRates() call.";
+      // Wait for GetStats to report a corresponding bitrate.
+      for (unsigned int i = 0; i < kDefaultTimeoutMs; ++i) {
+        VideoSendStream::Stats stats = send_stream_->GetStats();
+        {
+          CriticalSectionScoped lock(crit_.get());
+          if ((stats.target_media_bitrate_bps + 500) / 1000 ==
+              static_cast(bitrate_kbps_)) {
+            return;
+          }
+        }
+        SleepMs(1);
+      }
+      FAIL()
+          << "Timed out waiting for stats reporting the currently set bitrate.";
+    }
+
+   private:
+    VideoSendStream* send_stream_;
+    uint32_t bitrate_kbps_ GUARDED_BY(crit_);
+  } test;
+
+  RunBaseTest(&test);
+}
+
 TEST_F(EndToEndTest, GetStats) {
+  static const int kStartBitrateBps = 3000000;
   class StatsObserver : public test::EndToEndTest, public I420FrameCallback {
    public:
-    StatsObserver()
-        : EndToEndTest(kLongTimeoutMs),
-          receive_stream_(NULL),
-          send_stream_(NULL),
-          expected_receive_ssrc_(),
+    explicit StatsObserver(const FakeNetworkPipe::Config& config)
+        : EndToEndTest(kLongTimeoutMs, config),
+          send_stream_(nullptr),
           expected_send_ssrcs_(),
           check_stats_event_(EventWrapper::Create()) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       check_stats_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       check_stats_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
       check_stats_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       check_stats_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
+    void FrameCallback(I420VideoFrame* video_frame) override {
       // Ensure that we have at least 5ms send side delay.
       int64_t render_time = video_frame->render_time_ms();
       if (render_time > 0)
@@ -1473,55 +1764,73 @@ TEST_F(EndToEndTest, GetStats) {
     }
 
     bool CheckReceiveStats() {
-      assert(receive_stream_ != NULL);
-      VideoReceiveStream::Stats stats = receive_stream_->GetStats();
-      EXPECT_EQ(expected_receive_ssrc_, stats.ssrc);
+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
+        VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
+        EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc);
 
-      // Make sure all fields have been populated.
+        // Make sure all fields have been populated.
+        // TODO(pbos): Use CompoundKey if/when we ever know that all stats are
+        // always filled for all receivers.
+        receive_stats_filled_["IncomingRate"] |=
+            stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
 
-      receive_stats_filled_["IncomingRate"] |=
-          stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
+        receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
 
-      receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
+        receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
 
-      receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
+        receive_stats_filled_["StatisticsUpdated"] |=
+            stats.rtcp_stats.cumulative_lost != 0 ||
+            stats.rtcp_stats.extended_max_sequence_number != 0 ||
+            stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
 
-      receive_stats_filled_["StatisticsUpdated"] |=
-          stats.rtcp_stats.cumulative_lost != 0 ||
-          stats.rtcp_stats.extended_max_sequence_number != 0 ||
-          stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
+        receive_stats_filled_["DataCountersUpdated"] |=
+            stats.rtp_stats.transmitted.payload_bytes != 0 ||
+            stats.rtp_stats.fec.packets != 0 ||
+            stats.rtp_stats.transmitted.header_bytes != 0 ||
+            stats.rtp_stats.transmitted.packets != 0 ||
+            stats.rtp_stats.transmitted.padding_bytes != 0 ||
+            stats.rtp_stats.retransmitted.packets != 0;
 
-      receive_stats_filled_["DataCountersUpdated"] |=
-          stats.rtp_stats.bytes != 0 || stats.rtp_stats.fec_packets != 0 ||
-          stats.rtp_stats.header_bytes != 0 || stats.rtp_stats.packets != 0 ||
-          stats.rtp_stats.padding_bytes != 0 ||
-          stats.rtp_stats.retransmitted_packets != 0;
+        receive_stats_filled_["CodecStats"] |=
+            stats.target_delay_ms != 0 || stats.discarded_packets != 0;
 
-      receive_stats_filled_["CodecStats"] |=
-          stats.avg_delay_ms != 0 || stats.discarded_packets != 0 ||
-          stats.key_frames != 0 || stats.delta_frames != 0;
+        receive_stats_filled_["FrameCounts"] |=
+            stats.frame_counts.key_frames != 0 ||
+            stats.frame_counts.delta_frames != 0;
+
+        receive_stats_filled_["CName"] |= stats.c_name != "";
+
+        receive_stats_filled_["RtcpPacketTypeCount"] |=
+            stats.rtcp_packet_type_counts.fir_packets != 0 ||
+            stats.rtcp_packet_type_counts.nack_packets != 0 ||
+            stats.rtcp_packet_type_counts.pli_packets != 0 ||
+            stats.rtcp_packet_type_counts.nack_requests != 0 ||
+            stats.rtcp_packet_type_counts.unique_nack_requests != 0;
+      }
 
       return AllStatsFilled(receive_stats_filled_);
     }
 
     bool CheckSendStats() {
-      assert(send_stream_ != NULL);
+      DCHECK(send_stream_ != nullptr);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       send_stats_filled_["NumStreams"] |=
           stats.substreams.size() == expected_send_ssrcs_.size();
 
-      for (std::map::const_iterator it =
+      send_stats_filled_["CpuOveruseMetrics"] |=
+          stats.avg_encode_time_ms != 0 || stats.encode_usage_percent != 0;
+
+      for (std::map::const_iterator it =
                stats.substreams.begin();
-           it != stats.substreams.end();
-           ++it) {
+           it != stats.substreams.end(); ++it) {
         EXPECT_TRUE(expected_send_ssrcs_.find(it->first) !=
                     expected_send_ssrcs_.end());
 
-        send_stats_filled_[CompoundKey("IncomingRate", it->first)] |=
+        send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
             stats.input_frame_rate != 0;
 
-        const SsrcStats& stream_stats = it->second;
+        const VideoSendStream::StreamStats& stream_stats = it->second;
 
         send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
             stream_stats.rtcp_stats.cumulative_lost != 0 ||
@@ -1529,23 +1838,33 @@ TEST_F(EndToEndTest, GetStats) {
             stream_stats.rtcp_stats.fraction_lost != 0;
 
         send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
-            stream_stats.rtp_stats.fec_packets != 0 ||
-            stream_stats.rtp_stats.padding_bytes != 0 ||
-            stream_stats.rtp_stats.retransmitted_packets != 0 ||
-            stream_stats.rtp_stats.packets != 0;
+            stream_stats.rtp_stats.fec.packets != 0 ||
+            stream_stats.rtp_stats.transmitted.padding_bytes != 0 ||
+            stream_stats.rtp_stats.retransmitted.packets != 0 ||
+            stream_stats.rtp_stats.transmitted.packets != 0;
 
         send_stats_filled_[CompoundKey("BitrateStatisticsObserver",
                                        it->first)] |=
             stream_stats.total_bitrate_bps != 0;
 
         send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
-            stream_stats.delta_frames != 0 || stream_stats.key_frames != 0;
+            stream_stats.frame_counts.delta_frames != 0 ||
+            stream_stats.frame_counts.key_frames != 0;
 
         send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
             stats.encode_frame_rate != 0;
 
         send_stats_filled_[CompoundKey("Delay", it->first)] |=
             stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
+
+        // TODO(pbos): Use CompoundKey when the test makes sure that all SSRCs
+        // report dropped packets.
+        send_stats_filled_["RtcpPacketTypeCount"] |=
+            stream_stats.rtcp_packet_type_counts.fir_packets != 0 ||
+            stream_stats.rtcp_packet_type_counts.nack_packets != 0 ||
+            stream_stats.rtcp_packet_type_counts.pli_packets != 0 ||
+            stream_stats.rtcp_packet_type_counts.nack_requests != 0 ||
+            stream_stats.rtcp_packet_type_counts.unique_nack_requests != 0;
       }
 
       return AllStatsFilled(send_stats_filled_);
@@ -1567,29 +1886,36 @@ TEST_F(EndToEndTest, GetStats) {
       return true;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    Call::Config GetSenderCallConfig() override {
+      Call::Config config = EndToEndTest::GetSenderCallConfig();
+      config.bitrate_config.start_bitrate_bps = kStartBitrateBps;
+      return config;
+    }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->pre_encode_callback = this;  // Used to inject delay.
-      send_config->rtp.c_name = "SomeCName";
+      expected_cname_ = send_config->rtp.c_name = "SomeCName";
 
-      expected_receive_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
       const std::vector& ssrcs = send_config->rtp.ssrcs;
-      for (size_t i = 0; i < ssrcs.size(); ++i)
+      for (size_t i = 0; i < ssrcs.size(); ++i) {
         expected_send_ssrcs_.insert(ssrcs[i]);
-
-      expected_cname_ = send_config->rtp.c_name;
+        expected_receive_ssrcs_.push_back(
+            (*receive_configs)[i].rtp.remote_ssrc);
+      }
     }
 
-    virtual void OnStreamsCreated(
+    size_t GetNumStreams() const override { return kNumSsrcs; }
+
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       send_stream_ = send_stream;
-      receive_stream_ = receive_streams[0];
+      receive_streams_ = receive_streams;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       Clock* clock = Clock::GetRealTimeClock();
       int64_t now = clock->TimeInMilliseconds();
       int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
@@ -1631,19 +1957,23 @@ TEST_F(EndToEndTest, GetStats) {
       }
     }
 
-    VideoReceiveStream* receive_stream_;
+    std::vector receive_streams_;
     std::map receive_stats_filled_;
 
     VideoSendStream* send_stream_;
     std::map send_stats_filled_;
 
-    uint32_t expected_receive_ssrc_;
+    std::vector expected_receive_ssrcs_;
     std::set expected_send_ssrcs_;
     std::string expected_cname_;
 
-    scoped_ptr check_stats_event_;
-  } test;
+    rtc::scoped_ptr check_stats_event_;
+  };
 
+  FakeNetworkPipe::Config network_config;
+  network_config.loss_percent = 5;
+
+  StatsObserver test(network_config);
   RunBaseTest(&test);
 }
 
@@ -1661,20 +1991,20 @@ TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
    public:
     ReceivedRtpStatsObserver()
         : EndToEndTest(kDefaultTimeoutMs),
-          receive_stream_(NULL),
+          receive_stream_(nullptr),
           sent_rtp_(0) {}
 
    private:
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       receive_stream_ = receive_streams[0];
     }
 
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       if (sent_rtp_ >= kNumRtpPacketsToSend) {
         VideoReceiveStream::Stats stats = receive_stream_->GetStats();
-        if (kNumRtpPacketsToSend == stats.rtp_stats.packets) {
+        if (kNumRtpPacketsToSend == stats.rtp_stats.transmitted.packets) {
           observation_complete_->Set();
         }
         return DROP_PACKET;
@@ -1683,7 +2013,7 @@ TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {
       return SEND_PACKET;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while verifying number of received RTP packets.";
     }
@@ -1716,18 +2046,16 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
         }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
       if (!registered_rtx_ssrc_[header.ssrc])
         return SEND_PACKET;
 
-      EXPECT_LE(static_cast(header.headerLength + header.paddingLength),
-                length);
+      EXPECT_LE(header.headerLength + header.paddingLength, length);
       const bool packet_is_redundant_payload =
-          static_cast(header.headerLength + header.paddingLength) <
-          length;
+          header.headerLength + header.paddingLength < length;
 
       if (!packet_is_redundant_payload)
         return SEND_PACKET;
@@ -1741,12 +2069,11 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
       return SEND_PACKET;
     }
 
-    virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; }
+    size_t GetNumStreams() const override { return kNumSsrcs; }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
       for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
         encoder_config->streams[i].min_bitrate_bps = 10000;
@@ -1755,7 +2082,6 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
       }
 
       send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
-      send_config->rtp.rtx.pad_with_redundant_payloads = true;
 
       for (size_t i = 0; i < kNumSsrcs; ++i)
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
@@ -1765,7 +2091,7 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
       encoder_config->min_transmit_bitrate_bps = 100000;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for redundant payloads on all SSRCs.";
     }
@@ -1802,15 +2128,14 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       const uint32_t ssrc = header.ssrc;
       const uint16_t sequence_number = header.sequenceNumber;
       const uint32_t timestamp = header.timestamp;
       const bool only_padding =
-          static_cast(header.headerLength + header.paddingLength) ==
-          length;
+          header.headerLength + header.paddingLength == length;
 
       EXPECT_TRUE(configured_ssrcs_[ssrc])
           << "Received SSRC that wasn't configured: " << ssrc;
@@ -1867,14 +2192,14 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {
     std::map last_observed_timestamp_;
     std::map configured_ssrcs_;
 
-    scoped_ptr crit_;
+    rtc::scoped_ptr crit_;
     size_t ssrcs_to_observe_ GUARDED_BY(crit_);
     std::map ssrc_observed_ GUARDED_BY(crit_);
   } observer(use_rtx);
 
   CreateCalls(Call::Config(observer.SendTransport()),
               Call::Config(observer.ReceiveTransport()));
-  observer.SetReceivers(sender_call_->Receiver(), NULL);
+  observer.SetReceivers(sender_call_->Receiver(), nullptr);
 
   CreateSendConfig(kNumSsrcs);
 
@@ -1981,124 +2306,85 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
           FakeEncoder(Clock::GetRealTimeClock()),
           test_crit_(CriticalSectionWrapper::CreateCriticalSection()),
           encoded_frames_(EventWrapper::Create()),
-          sender_packets_(EventWrapper::Create()),
-          receiver_packets_(EventWrapper::Create()),
+          packet_event_(EventWrapper::Create()),
           sender_state_(Call::kNetworkUp),
-          down_sender_rtp_(0),
-          down_sender_rtcp_(0),
-          receiver_state_(Call::kNetworkUp),
-          down_receiver_rtcp_(0),
+          sender_rtp_(0),
+          sender_rtcp_(0),
+          receiver_rtcp_(0),
           down_frames_(0) {}
 
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       CriticalSectionScoped lock(test_crit_.get());
-      if (sender_state_ == Call::kNetworkDown) {
-        ++down_sender_rtp_;
-        EXPECT_LE(down_sender_rtp_, kNumAcceptedDowntimeRtp)
-            << "RTP sent during sender-side downtime.";
-        if (down_sender_rtp_> kNumAcceptedDowntimeRtp)
-          sender_packets_->Set();
-      } else {
-        sender_packets_->Set();
-      }
+      ++sender_rtp_;
+      packet_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       CriticalSectionScoped lock(test_crit_.get());
-      if (sender_state_ == Call::kNetworkDown) {
-        ++down_sender_rtcp_;
-        EXPECT_LE(down_sender_rtcp_, kNumAcceptedDowntimeRtcp)
-            << "RTCP sent during sender-side downtime.";
-        if (down_sender_rtcp_ > kNumAcceptedDowntimeRtcp)
-          sender_packets_->Set();
-      } else {
-        sender_packets_->Set();
-      }
+      ++sender_rtcp_;
+      packet_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
       return SEND_PACKET;
     }
 
-    virtual Action OnReceiveRtcp(const uint8_t* packet,
-                                 size_t length) OVERRIDE {
+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       CriticalSectionScoped lock(test_crit_.get());
-      if (receiver_state_ == Call::kNetworkDown) {
-        ++down_receiver_rtcp_;
-        EXPECT_LE(down_receiver_rtcp_, kNumAcceptedDowntimeRtcp)
-            << "RTCP sent during receiver-side downtime.";
-        if (down_receiver_rtcp_ > kNumAcceptedDowntimeRtcp)
-          receiver_packets_->Set();
-      } else {
-        receiver_packets_->Set();
-      }
+      ++receiver_rtcp_;
+      packet_event_->Set();
       return SEND_PACKET;
     }
 
-    virtual void OnCallsCreated(Call* sender_call,
-                                Call* receiver_call) OVERRIDE {
+    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, encoded_frames_->Wait(kDefaultTimeoutMs))
           << "No frames received by the encoder.";
-      EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs))
-          << "Timed out waiting for send-side packets.";
-      EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
-          << "Timed out waiting for receiver-side packets.";
+      // Wait for packets from both sender/receiver.
+      WaitForPacketsOrSilence(false, false);
 
       // Sender-side network down.
       sender_call_->SignalNetworkState(Call::kNetworkDown);
       {
         CriticalSectionScoped lock(test_crit_.get());
-        sender_packets_->Reset();  // Earlier packets should not count.
+        // After network goes down we shouldn't be encoding more frames.
         sender_state_ = Call::kNetworkDown;
       }
-      EXPECT_EQ(kEventTimeout, sender_packets_->Wait(kSilenceTimeoutMs))
-          << "Packets sent during sender-network downtime.";
-      EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
-          << "Timed out waiting for receiver-side packets.";
+      // Wait for receiver-packets and no sender packets.
+      WaitForPacketsOrSilence(true, false);
+
       // Receiver-side network down.
       receiver_call_->SignalNetworkState(Call::kNetworkDown);
-      {
-        CriticalSectionScoped lock(test_crit_.get());
-        receiver_packets_->Reset();  // Earlier packets should not count.
-        receiver_state_ = Call::kNetworkDown;
-      }
-      EXPECT_EQ(kEventTimeout, receiver_packets_->Wait(kSilenceTimeoutMs))
-          << "Packets sent during receiver-network downtime.";
+      WaitForPacketsOrSilence(true, true);
 
       // Network back up again for both.
       {
         CriticalSectionScoped lock(test_crit_.get());
-        sender_packets_->Reset();  // Earlier packets should not count.
-        receiver_packets_->Reset();  // Earlier packets should not count.
-        sender_state_ = receiver_state_ = Call::kNetworkUp;
+        // It's OK to encode frames again, as we're about to bring up the
+        // network.
+        sender_state_ = Call::kNetworkUp;
       }
       sender_call_->SignalNetworkState(Call::kNetworkUp);
       receiver_call_->SignalNetworkState(Call::kNetworkUp);
-      EXPECT_EQ(kEventSignaled, sender_packets_->Wait(kDefaultTimeoutMs))
-          << "Timed out waiting for send-side packets.";
-      EXPECT_EQ(kEventSignaled, receiver_packets_->Wait(kDefaultTimeoutMs))
-          << "Timed out waiting for receiver-side packets.";
+      WaitForPacketsOrSilence(false, false);
     }
 
-    virtual int32_t Encode(const I420VideoFrame& input_image,
-                           const CodecSpecificInfo* codec_specific_info,
-                           const std::vector* frame_types)
-        OVERRIDE {
+    int32_t Encode(const I420VideoFrame& input_image,
+                   const CodecSpecificInfo* codec_specific_info,
+                   const std::vector* frame_types) override {
       {
         CriticalSectionScoped lock(test_crit_.get());
         if (sender_state_ == Call::kNetworkDown) {
@@ -2116,31 +2402,114 @@ TEST_F(EndToEndTest, RespectsNetworkState) {
     }
 
    private:
-    const scoped_ptr test_crit_;
-    scoped_ptr encoded_frames_;
-    scoped_ptr sender_packets_;
-    scoped_ptr receiver_packets_;
+    void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
+      int64_t initial_time_ms = clock_->TimeInMilliseconds();
+      int initial_sender_rtp;
+      int initial_sender_rtcp;
+      int initial_receiver_rtcp;
+      {
+        CriticalSectionScoped lock(test_crit_.get());
+        initial_sender_rtp = sender_rtp_;
+        initial_sender_rtcp = sender_rtcp_;
+        initial_receiver_rtcp = receiver_rtcp_;
+      }
+      bool sender_done = false;
+      bool receiver_done = false;
+      while(!sender_done || !receiver_done) {
+        packet_event_->Wait(kSilenceTimeoutMs);
+        int64_t time_now_ms = clock_->TimeInMilliseconds();
+        CriticalSectionScoped lock(test_crit_.get());
+        if (sender_down) {
+          ASSERT_LE(sender_rtp_ - initial_sender_rtp, kNumAcceptedDowntimeRtp)
+              << "RTP sent during sender-side downtime.";
+          ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
+                    kNumAcceptedDowntimeRtcp)
+              << "RTCP sent during sender-side downtime.";
+          if (time_now_ms - initial_time_ms >=
+              static_cast(kSilenceTimeoutMs)) {
+            sender_done = true;
+          }
+        } else {
+          if (sender_rtp_ > initial_sender_rtp)
+            sender_done = true;
+        }
+        if (receiver_down) {
+          ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
+                    kNumAcceptedDowntimeRtcp)
+              << "RTCP sent during receiver-side downtime.";
+          if (time_now_ms - initial_time_ms >=
+              static_cast(kSilenceTimeoutMs)) {
+            receiver_done = true;
+          }
+        } else {
+          if (receiver_rtcp_ > initial_receiver_rtcp)
+            receiver_done = true;
+        }
+      }
+    }
+
+    const rtc::scoped_ptr test_crit_;
+    const rtc::scoped_ptr encoded_frames_;
+    const rtc::scoped_ptr packet_event_;
     Call* sender_call_;
     Call* receiver_call_;
     Call::NetworkState sender_state_ GUARDED_BY(test_crit_);
-    int down_sender_rtp_ GUARDED_BY(test_crit_);
-    int down_sender_rtcp_ GUARDED_BY(test_crit_);
-    Call::NetworkState receiver_state_ GUARDED_BY(test_crit_);
-    int down_receiver_rtcp_ GUARDED_BY(test_crit_);
+    int sender_rtp_ GUARDED_BY(test_crit_);
+    int sender_rtcp_ GUARDED_BY(test_crit_);
+    int receiver_rtcp_ GUARDED_BY(test_crit_);
     int down_frames_ GUARDED_BY(test_crit_);
   } test;
 
   RunBaseTest(&test);
 }
 
+TEST_F(EndToEndTest, CallReportsRttForSender) {
+  static const int kSendDelayMs = 30;
+  static const int kReceiveDelayMs = 70;
+
+  FakeNetworkPipe::Config config;
+  config.queue_delay_ms = kSendDelayMs;
+  test::DirectTransport sender_transport(config);
+  config.queue_delay_ms = kReceiveDelayMs;
+  test::DirectTransport receiver_transport(config);
+
+  CreateCalls(Call::Config(&sender_transport),
+              Call::Config(&receiver_transport));
+
+  sender_transport.SetReceiver(receiver_call_->Receiver());
+  receiver_transport.SetReceiver(sender_call_->Receiver());
+
+  CreateSendConfig(1);
+  CreateMatchingReceiveConfigs();
+
+  CreateStreams();
+  CreateFrameGeneratorCapturer();
+  Start();
+
+  int64_t start_time_ms = clock_->TimeInMilliseconds();
+  while (true) {
+    Call::Stats stats = sender_call_->GetStats();
+    ASSERT_GE(start_time_ms + kDefaultTimeoutMs,
+              clock_->TimeInMilliseconds())
+        << "No RTT stats before timeout!";
+    if (stats.rtt_ms != -1) {
+      EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs);
+      break;
+    }
+    SleepMs(10);
+  }
+
+  Stop();
+  DestroyStreams();
+}
+
 TEST_F(EndToEndTest, NewSendStreamsRespectNetworkDown) {
   class UnusedEncoder : public test::FakeEncoder {
     public:
      UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
-    virtual int32_t Encode(const I420VideoFrame& input_image,
-                           const CodecSpecificInfo* codec_specific_info,
-                           const std::vector* frame_types)
-        OVERRIDE {
+     int32_t Encode(const I420VideoFrame& input_image,
+                    const CodecSpecificInfo* codec_specific_info,
+                    const std::vector* frame_types) override {
       ADD_FAILURE() << "Unexpected frame encode.";
       return test::FakeEncoder::Encode(
           input_image, codec_specific_info, frame_types);
@@ -2191,7 +2560,7 @@ TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {
 // a backend. This is to test that we hand channels back properly.
 TEST_F(EndToEndTest, CanCreateAndDestroyManyVideoStreams) {
   test::NullTransport transport;
-  scoped_ptr call(Call::Create(Call::Config(&transport)));
+  rtc::scoped_ptr call(Call::Create(Call::Config(&transport)));
   test::FakeDecoder fake_decoder;
   test::FakeEncoder fake_encoder(Clock::GetRealTimeClock());
   for (size_t i = 0; i < 100; ++i) {
diff --git a/media/webrtc/trunk/webrtc/video/full_stack.cc b/media/webrtc/trunk/webrtc/video/full_stack.cc
index 25e60eb8da..7f98f38a79 100644
--- a/media/webrtc/trunk/webrtc/video/full_stack.cc
+++ b/media/webrtc/trunk/webrtc/video/full_stack.cc
@@ -14,19 +14,21 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/cpu_info.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_encoder.h"
+#include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -43,11 +45,13 @@ struct FullStackTestParams {
     size_t width, height;
     int fps;
   } clip;
+  bool screenshare;
   int min_bitrate_bps;
   int target_bitrate_bps;
   int max_bitrate_bps;
   double avg_psnr_threshold;
   double avg_ssim_threshold;
+  int test_durations_secs;
   FakeNetworkPipe::Config link;
 };
 
@@ -69,42 +73,58 @@ class VideoAnalyzer : public PacketReceiver,
                 int duration_frames)
       : input_(input),
         transport_(transport),
-        receiver_(NULL),
+        receiver_(nullptr),
         test_label_(test_label),
-        frames_left_(duration_frames),
+        frames_to_process_(duration_frames),
+        frames_recorded_(0),
+        frames_processed_(0),
         dropped_frames_(0),
         last_render_time_(0),
         rtp_timestamp_delta_(0),
         crit_(CriticalSectionWrapper::CreateCriticalSection()),
-        first_send_frame_(NULL),
         avg_psnr_threshold_(avg_psnr_threshold),
         avg_ssim_threshold_(avg_ssim_threshold),
         comparison_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-        comparison_thread_(ThreadWrapper::CreateThread(&FrameComparisonThread,
-                                                       this)),
+        comparison_available_event_(EventWrapper::Create()),
         done_(EventWrapper::Create()) {
-    unsigned int id;
-    EXPECT_TRUE(comparison_thread_->Start(id));
+    // Create thread pool for CPU-expensive PSNR/SSIM calculations.
+
+    // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
+    // so that we don't accidentally starve "real" worker threads (codec etc).
+    // Also, don't allocate more than kMaxComparisonThreads, even if there are
+    // spare cores.
+
+    uint32_t num_cores = CpuInfo::DetectNumberOfCores();
+    DCHECK_GE(num_cores, 1u);
+    static const uint32_t kMinCoresLeft = 4;
+    static const uint32_t kMaxComparisonThreads = 8;
+
+    if (num_cores <= kMinCoresLeft) {
+      num_cores = 1;
+    } else {
+      num_cores -= kMinCoresLeft;
+      num_cores = std::min(num_cores, kMaxComparisonThreads);
+    }
+
+    for (uint32_t i = 0; i < num_cores; ++i) {
+      rtc::scoped_ptr thread =
+          ThreadWrapper::CreateThread(&FrameComparisonThread, this, "Analyzer");
+      EXPECT_TRUE(thread->Start());
+      comparison_thread_pool_.push_back(thread.release());
+    }
   }
 
   ~VideoAnalyzer() {
-    EXPECT_TRUE(comparison_thread_->Stop());
-
-    while (!frames_.empty()) {
-      delete frames_.back();
-      frames_.pop_back();
-    }
-    while (!frame_pool_.empty()) {
-      delete frame_pool_.back();
-      frame_pool_.pop_back();
+    for (ThreadWrapper* thread : comparison_thread_pool_) {
+      EXPECT_TRUE(thread->Stop());
+      delete thread;
     }
   }
 
   virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
 
-  virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                       size_t length) OVERRIDE {
-    scoped_ptr parser(RtpHeaderParser::Create());
+  DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) override {
+    rtc::scoped_ptr parser(RtpHeaderParser::Create());
     RTPHeader header;
     parser->Parse(packet, length, &header);
     {
@@ -116,34 +136,23 @@ class VideoAnalyzer : public PacketReceiver,
     return receiver_->DeliverPacket(packet, length);
   }
 
-  virtual void SwapFrame(I420VideoFrame* video_frame) OVERRIDE {
-    I420VideoFrame* copy = NULL;
-    {
-      CriticalSectionScoped lock(crit_.get());
-      if (frame_pool_.size() > 0) {
-        copy = frame_pool_.front();
-        frame_pool_.pop_front();
-      }
-    }
-    if (copy == NULL)
-      copy = new I420VideoFrame();
-
-    copy->CopyFrame(*video_frame);
-    copy->set_timestamp(copy->render_time_ms() * 90);
+  void IncomingCapturedFrame(const I420VideoFrame& video_frame) override {
+    I420VideoFrame copy = video_frame;
+    copy.set_timestamp(copy.ntp_time_ms() * 90);
 
     {
       CriticalSectionScoped lock(crit_.get());
-      if (first_send_frame_ == NULL && rtp_timestamp_delta_ == 0)
+      if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
         first_send_frame_ = copy;
 
       frames_.push_back(copy);
     }
 
-    input_->SwapFrame(video_frame);
+    input_->IncomingCapturedFrame(video_frame);
   }
 
-  virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
-    scoped_ptr parser(RtpHeaderParser::Create());
+  bool SendRtp(const uint8_t* packet, size_t length) override {
+    rtc::scoped_ptr parser(RtpHeaderParser::Create());
     RTPHeader header;
     parser->Parse(packet, length, &header);
 
@@ -151,8 +160,8 @@ class VideoAnalyzer : public PacketReceiver,
       CriticalSectionScoped lock(crit_.get());
       if (rtp_timestamp_delta_ == 0) {
         rtp_timestamp_delta_ =
-            header.timestamp - first_send_frame_->timestamp();
-        first_send_frame_ = NULL;
+            header.timestamp - first_send_frame_.timestamp();
+        first_send_frame_.Reset();
       }
       send_times_[header.timestamp - rtp_timestamp_delta_] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
@@ -161,38 +170,59 @@ class VideoAnalyzer : public PacketReceiver,
     return transport_->SendRtp(packet, length);
   }
 
-  virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+  bool SendRtcp(const uint8_t* packet, size_t length) override {
     return transport_->SendRtcp(packet, length);
   }
 
-  virtual void RenderFrame(const I420VideoFrame& video_frame,
-                           int time_to_render_ms) OVERRIDE {
+  void RenderFrame(const I420VideoFrame& video_frame,
+                   int time_to_render_ms) override {
     int64_t render_time_ms =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     uint32_t send_timestamp = video_frame.timestamp() - rtp_timestamp_delta_;
 
     CriticalSectionScoped lock(crit_.get());
-    while (frames_.front()->timestamp() < send_timestamp) {
-      AddFrameComparison(
-          frames_.front(), &last_rendered_frame_, true, render_time_ms);
-      frame_pool_.push_back(frames_.front());
+
+    while (frames_.front().timestamp() < send_timestamp) {
+      AddFrameComparison(frames_.front(), last_rendered_frame_, true,
+                         render_time_ms);
       frames_.pop_front();
     }
 
-    I420VideoFrame* reference_frame = frames_.front();
+    I420VideoFrame reference_frame = frames_.front();
     frames_.pop_front();
-    assert(reference_frame != NULL);
-    EXPECT_EQ(reference_frame->timestamp(), send_timestamp);
-    assert(reference_frame->timestamp() == send_timestamp);
+    assert(!reference_frame.IsZeroSize());
+    EXPECT_EQ(reference_frame.timestamp(), send_timestamp);
+    assert(reference_frame.timestamp() == send_timestamp);
 
-    AddFrameComparison(reference_frame, &video_frame, false, render_time_ms);
-    frame_pool_.push_back(reference_frame);
+    AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
 
-    last_rendered_frame_.CopyFrame(video_frame);
+    last_rendered_frame_ = video_frame;
   }
 
+  bool IsTextureSupported() const override { return false; }
+
   void Wait() {
-    EXPECT_EQ(kEventSignaled, done_->Wait(FullStackTest::kLongTimeoutMs));
+    // Frame comparisons can be very expensive. Wait for test to be done, but
+    // at time-out check if frames_processed is going up. If so, give it more
+    // time, otherwise fail. Hopefully this will reduce test flakiness.
+
+    int last_frames_processed = -1;
+    EventTypeWrapper eventType;
+    while ((eventType = done_->Wait(FullStackTest::kDefaultTimeoutMs)) !=
+           kEventSignaled) {
+      int frames_processed;
+      {
+        CriticalSectionScoped crit(comparison_lock_.get());
+        frames_processed = frames_processed_;
+      }
+      if (last_frames_processed == -1) {
+        last_frames_processed = frames_processed;
+        continue;
+      }
+      ASSERT_GT(frames_processed, last_frames_processed)
+          << "Analyzer stalled while waiting for test to finish.";
+      last_frames_processed = frames_processed;
+    }
   }
 
   VideoSendStreamInput* input_;
@@ -201,30 +231,21 @@ class VideoAnalyzer : public PacketReceiver,
 
  private:
   struct FrameComparison {
-    FrameComparison(const I420VideoFrame* reference,
-                    const I420VideoFrame* render,
+    FrameComparison()
+        : dropped(false), send_time_ms(0), recv_time_ms(0), render_time_ms(0) {}
+
+    FrameComparison(const I420VideoFrame& reference,
+                    const I420VideoFrame& render,
                     bool dropped,
                     int64_t send_time_ms,
                     int64_t recv_time_ms,
                     int64_t render_time_ms)
-        : dropped(dropped),
+        : reference(reference),
+          render(render),
+          dropped(dropped),
           send_time_ms(send_time_ms),
           recv_time_ms(recv_time_ms),
-          render_time_ms(render_time_ms) {
-      this->reference.CopyFrame(*reference);
-      this->render.CopyFrame(*render);
-    }
-
-    FrameComparison(const FrameComparison& compare)
-        : dropped(compare.dropped),
-          send_time_ms(compare.send_time_ms),
-          recv_time_ms(compare.recv_time_ms),
-          render_time_ms(compare.render_time_ms) {
-      this->reference.CopyFrame(compare.reference);
-      this->render.CopyFrame(compare.render);
-    }
-
-    ~FrameComparison() {}
+          render_time_ms(render_time_ms) {}
 
     I420VideoFrame reference;
     I420VideoFrame render;
@@ -234,15 +255,15 @@ class VideoAnalyzer : public PacketReceiver,
     int64_t render_time_ms;
   };
 
-  void AddFrameComparison(const I420VideoFrame* reference,
-                          const I420VideoFrame* render,
+  void AddFrameComparison(const I420VideoFrame& reference,
+                          const I420VideoFrame& render,
                           bool dropped,
                           int64_t render_time_ms)
       EXCLUSIVE_LOCKS_REQUIRED(crit_) {
-    int64_t send_time_ms = send_times_[reference->timestamp()];
-    send_times_.erase(reference->timestamp());
-    int64_t recv_time_ms = recv_times_[reference->timestamp()];
-    recv_times_.erase(reference->timestamp());
+    int64_t send_time_ms = send_times_[reference.timestamp()];
+    send_times_.erase(reference.timestamp());
+    int64_t recv_time_ms = recv_times_[reference.timestamp()];
+    recv_times_.erase(reference.timestamp());
 
     CriticalSectionScoped crit(comparison_lock_.get());
     comparisons_.push_back(FrameComparison(reference,
@@ -251,6 +272,7 @@ class VideoAnalyzer : public PacketReceiver,
                                            send_time_ms,
                                            recv_time_ms,
                                            render_time_ms));
+    comparison_available_event_->Set();
   }
 
   static bool FrameComparisonThread(void* obj) {
@@ -258,76 +280,110 @@ class VideoAnalyzer : public PacketReceiver,
   }
 
   bool CompareFrames() {
-    assert(frames_left_ > 0);
+    if (AllFramesRecorded())
+      return false;
 
     I420VideoFrame reference;
     I420VideoFrame render;
-    bool dropped;
-    int64_t send_time_ms;
-    int64_t recv_time_ms;
-    int64_t render_time_ms;
-
-    SleepMs(10);
-
-    while (true) {
-      {
-        CriticalSectionScoped crit(comparison_lock_.get());
-        if (comparisons_.empty())
-          return true;
-        reference.SwapFrame(&comparisons_.front().reference);
-        render.SwapFrame(&comparisons_.front().render);
-        dropped = comparisons_.front().dropped;
-        send_time_ms = comparisons_.front().send_time_ms;
-        recv_time_ms = comparisons_.front().recv_time_ms;
-        render_time_ms = comparisons_.front().render_time_ms;
-        comparisons_.pop_front();
-      }
-
-      PerformFrameComparison(&reference,
-                             &render,
-                             dropped,
-                             send_time_ms,
-                             recv_time_ms,
-                             render_time_ms);
-
-      if (--frames_left_ == 0) {
-        PrintResult("psnr", psnr_, " dB");
-        PrintResult("ssim", ssim_, "");
-        PrintResult("sender_time", sender_time_, " ms");
-        printf(
-            "RESULT dropped_frames: %s = %d\n", test_label_, dropped_frames_);
-        PrintResult("receiver_time", receiver_time_, " ms");
-        PrintResult("total_delay_incl_network", end_to_end_, " ms");
-        PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
-        EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
-        EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
-        done_->Set();
+    FrameComparison comparison;
 
+    if (!PopComparison(&comparison)) {
+      // Wait until new comparison task is available, or test is done.
+      // If done, wake up remaining threads waiting.
+      comparison_available_event_->Wait(1000);
+      if (AllFramesRecorded()) {
+        comparison_available_event_->Set();
         return false;
       }
+      return true;  // Try again.
     }
+
+    PerformFrameComparison(comparison);
+
+    if (FrameProcessed()) {
+      PrintResults();
+      done_->Set();
+      comparison_available_event_->Set();
+      return false;
+    }
+
+    return true;
   }
 
-  void PerformFrameComparison(const I420VideoFrame* reference,
-                              const I420VideoFrame* render,
-                              bool dropped,
-                              int64_t send_time_ms,
-                              int64_t recv_time_ms,
-                              int64_t render_time_ms) {
-    psnr_.AddSample(I420PSNR(reference, render));
-    ssim_.AddSample(I420SSIM(reference, render));
-    if (dropped) {
+  bool PopComparison(FrameComparison* comparison) {
+    CriticalSectionScoped crit(comparison_lock_.get());
+    // If AllFramesRecorded() is true, it means we have already popped
+    // frames_to_process_ frames from comparisons_, so there is no more work
+    // for this thread to be done. frames_processed_ might still be lower if
+    // all comparisons are not done, but those frames are currently being
+    // worked on by other threads.
+    if (comparisons_.empty() || AllFramesRecorded())
+      return false;
+
+    *comparison = comparisons_.front();
+    comparisons_.pop_front();
+
+    FrameRecorded();
+    return true;
+  }
+
+  // Increment counter for number of frames received for comparison.
+  void FrameRecorded() {
+    CriticalSectionScoped crit(comparison_lock_.get());
+    ++frames_recorded_;
+  }
+
+  // Returns true if all frames to be compared have been taken from the queue.
+  bool AllFramesRecorded() {
+    CriticalSectionScoped crit(comparison_lock_.get());
+    assert(frames_recorded_ <= frames_to_process_);
+    return frames_recorded_ == frames_to_process_;
+  }
+
+  // Increase count of number of frames processed. Returns true if this was the
+  // last frame to be processed.
+  bool FrameProcessed() {
+    CriticalSectionScoped crit(comparison_lock_.get());
+    ++frames_processed_;
+    assert(frames_processed_ <= frames_to_process_);
+    return frames_processed_ == frames_to_process_;
+  }
+
+  void PrintResults() {
+    CriticalSectionScoped crit(comparison_lock_.get());
+    PrintResult("psnr", psnr_, " dB");
+    PrintResult("ssim", ssim_, "");
+    PrintResult("sender_time", sender_time_, " ms");
+    printf("RESULT dropped_frames: %s = %d frames\n", test_label_,
+           dropped_frames_);
+    PrintResult("receiver_time", receiver_time_, " ms");
+    PrintResult("total_delay_incl_network", end_to_end_, " ms");
+    PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
+    EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
+    EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
+  }
+
+  void PerformFrameComparison(const FrameComparison& comparison) {
+    // Perform expensive psnr and ssim calculations while not holding lock.
+    double psnr = I420PSNR(&comparison.reference, &comparison.render);
+    double ssim = I420SSIM(&comparison.reference, &comparison.render);
+
+    CriticalSectionScoped crit(comparison_lock_.get());
+    psnr_.AddSample(psnr);
+    ssim_.AddSample(ssim);
+    if (comparison.dropped) {
       ++dropped_frames_;
       return;
     }
     if (last_render_time_ != 0)
-      rendered_delta_.AddSample(render_time_ms - last_render_time_);
-    last_render_time_ = render_time_ms;
+      rendered_delta_.AddSample(comparison.render_time_ms - last_render_time_);
+    last_render_time_ = comparison.render_time_ms;
 
-    int64_t input_time_ms = reference->render_time_ms();
-    sender_time_.AddSample(send_time_ms - input_time_ms);
-    receiver_time_.AddSample(render_time_ms - recv_time_ms);
-    end_to_end_.AddSample(render_time_ms - input_time_ms);
+    int64_t input_time_ms = comparison.reference.ntp_time_ms();
+    sender_time_.AddSample(comparison.send_time_ms - input_time_ms);
+    receiver_time_.AddSample(comparison.render_time_ms -
+                             comparison.recv_time_ms);
+    end_to_end_.AddSample(comparison.render_time_ms - input_time_ms);
   }
 
   void PrintResult(const char* result_type,
@@ -348,36 +404,35 @@ class VideoAnalyzer : public PacketReceiver,
   test::Statistics ssim_;
   test::Statistics end_to_end_;
   test::Statistics rendered_delta_;
-  int frames_left_;
+  const int frames_to_process_;
+  int frames_recorded_;
+  int frames_processed_;
   int dropped_frames_;
   int64_t last_render_time_;
   uint32_t rtp_timestamp_delta_;
 
-  const scoped_ptr crit_;
-  std::deque frames_ GUARDED_BY(crit_);
-  std::deque frame_pool_ GUARDED_BY(crit_);
+  const rtc::scoped_ptr crit_;
+  std::deque frames_ GUARDED_BY(crit_);
   I420VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
   std::map send_times_ GUARDED_BY(crit_);
   std::map recv_times_ GUARDED_BY(crit_);
-  I420VideoFrame* first_send_frame_ GUARDED_BY(crit_);
+  I420VideoFrame first_send_frame_ GUARDED_BY(crit_);
   const double avg_psnr_threshold_;
   const double avg_ssim_threshold_;
 
-  const scoped_ptr comparison_lock_;
-  const scoped_ptr comparison_thread_;
+  const rtc::scoped_ptr comparison_lock_;
+  std::vector comparison_thread_pool_;
+  const rtc::scoped_ptr comparison_available_event_;
   std::deque comparisons_ GUARDED_BY(comparison_lock_);
-  const scoped_ptr done_;
+  const rtc::scoped_ptr done_;
 };
 
 void FullStackTest::RunTest(const FullStackTestParams& params) {
   test::DirectTransport send_transport(params.link);
   test::DirectTransport recv_transport(params.link);
-  VideoAnalyzer analyzer(NULL,
-                         &send_transport,
-                         params.test_label,
-                         params.avg_psnr_threshold,
-                         params.avg_ssim_threshold,
-                         kFullStackTestDurationSecs * params.clip.fps);
+  VideoAnalyzer analyzer(nullptr, &send_transport, params.test_label,
+                         params.avg_psnr_threshold, params.avg_ssim_threshold,
+                         params.test_durations_secs * params.clip.fps);
 
   CreateCalls(Call::Config(&analyzer), Call::Config(&recv_transport));
 
@@ -387,12 +442,14 @@ void FullStackTest::RunTest(const FullStackTestParams& params) {
 
   CreateSendConfig(1);
 
-  scoped_ptr encoder(
+  rtc::scoped_ptr encoder(
       VideoEncoder::Create(VideoEncoder::kVp8));
   send_config_.encoder_settings.encoder = encoder.get();
   send_config_.encoder_settings.payload_name = "VP8";
   send_config_.encoder_settings.payload_type = 124;
   send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+  send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
+  send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
 
   VideoStream* stream = &encoder_config_.streams[0];
   stream->width = params.clip.width;
@@ -402,25 +459,56 @@ void FullStackTest::RunTest(const FullStackTestParams& params) {
   stream->max_bitrate_bps = params.max_bitrate_bps;
   stream->max_framerate = params.clip.fps;
 
+  if (params.screenshare) {
+    encoder_config_.content_type = VideoEncoderConfig::kScreenshare;
+    encoder_config_.min_transmit_bitrate_bps = 400 * 1000;
+    VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
+    vp8_settings.denoisingOn = false;
+    vp8_settings.frameDroppingOn = false;
+    vp8_settings.numberOfTemporalLayers = 2;
+    encoder_config_.encoder_specific_settings = &vp8_settings;
+
+    stream->temporal_layer_thresholds_bps.clear();
+    stream->temporal_layer_thresholds_bps.push_back(stream->target_bitrate_bps);
+  }
+
   CreateMatchingReceiveConfigs();
   receive_configs_[0].renderer = &analyzer;
   receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+  receive_configs_[0].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrcs[0];
+  receive_configs_[0].rtp.rtx[kSendRtxPayloadType].payload_type =
+      kSendRtxPayloadType;
 
   CreateStreams();
   analyzer.input_ = send_stream_->Input();
 
-  frame_generator_capturer_.reset(
-      test::FrameGeneratorCapturer::CreateFromYuvFile(
-          &analyzer,
-          test::ResourcePath(params.clip.name, "yuv").c_str(),
-          params.clip.width,
-          params.clip.height,
-          params.clip.fps,
-          Clock::GetRealTimeClock()));
+  if (params.screenshare) {
+    std::vector slides;
+    slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
+    slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
+    slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
+    slides.push_back(test::ResourcePath("difficult_photo_1850_1110", "yuv"));
 
-  ASSERT_TRUE(frame_generator_capturer_.get() != NULL)
-      << "Could not create capturer for " << params.clip.name
-      << ".yuv. Is this resource file present?";
+    rtc::scoped_ptr frame_generator(
+        test::FrameGenerator::CreateFromYuvFile(
+            slides, 1850, 1110,
+            10 * params.clip.fps)  // Cycle image every 10 seconds.
+        );
+    frame_generator_capturer_.reset(new test::FrameGeneratorCapturer(
+        Clock::GetRealTimeClock(), &analyzer, frame_generator.release(),
+        params.clip.fps));
+    ASSERT_TRUE(frame_generator_capturer_->Init());
+  } else {
+    frame_generator_capturer_.reset(
+        test::FrameGeneratorCapturer::CreateFromYuvFile(
+            &analyzer, test::ResourcePath(params.clip.name, "yuv"),
+            params.clip.width, params.clip.height, params.clip.fps,
+            Clock::GetRealTimeClock()));
+
+    ASSERT_TRUE(frame_generator_capturer_.get() != nullptr)
+        << "Could not create capturer for " << params.clip.name
+        << ".yuv. Is this resource file present?";
+  }
 
   Start();
 
@@ -437,12 +525,13 @@ void FullStackTest::RunTest(const FullStackTestParams& params) {
 TEST_F(FullStackTest, ParisQcifWithoutPacketLoss) {
   FullStackTestParams paris_qcif = {"net_delay_0_0_plr_0",
                                     {"paris_qcif", 176, 144, 30},
+                                    false,
                                     300000,
                                     300000,
                                     300000,
                                     36.0,
-                                    0.96
-                                   };
+                                    0.96,
+                                    kFullStackTestDurationSecs};
   RunTest(paris_qcif);
 }
 
@@ -450,24 +539,26 @@ TEST_F(FullStackTest, ForemanCifWithoutPacketLoss) {
   // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
   FullStackTestParams foreman_cif = {"foreman_cif_net_delay_0_0_plr_0",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      700000,
                                      700000,
                                      700000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   RunTest(foreman_cif);
 }
 
 TEST_F(FullStackTest, ForemanCifPlr5) {
   FullStackTestParams foreman_cif = {"foreman_cif_delay_50_0_plr_5",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      30000,
                                      500000,
                                      2000000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   foreman_cif.link.loss_percent = 5;
   foreman_cif.link.queue_delay_ms = 50;
   RunTest(foreman_cif);
@@ -476,12 +567,13 @@ TEST_F(FullStackTest, ForemanCifPlr5) {
 TEST_F(FullStackTest, ForemanCif500kbps) {
   FullStackTestParams foreman_cif = {"foreman_cif_500kbps",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      30000,
                                      500000,
                                      2000000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   foreman_cif.link.queue_length_packets = 0;
   foreman_cif.link.queue_delay_ms = 0;
   foreman_cif.link.link_capacity_kbps = 500;
@@ -491,12 +583,13 @@ TEST_F(FullStackTest, ForemanCif500kbps) {
 TEST_F(FullStackTest, ForemanCif500kbpsLimitedQueue) {
   FullStackTestParams foreman_cif = {"foreman_cif_500kbps_32pkts_queue",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      30000,
                                      500000,
                                      2000000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   foreman_cif.link.queue_length_packets = 32;
   foreman_cif.link.queue_delay_ms = 0;
   foreman_cif.link.link_capacity_kbps = 500;
@@ -506,12 +599,13 @@ TEST_F(FullStackTest, ForemanCif500kbpsLimitedQueue) {
 TEST_F(FullStackTest, ForemanCif500kbps100ms) {
   FullStackTestParams foreman_cif = {"foreman_cif_500kbps_100ms",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      30000,
                                      500000,
                                      2000000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   foreman_cif.link.queue_length_packets = 0;
   foreman_cif.link.queue_delay_ms = 100;
   foreman_cif.link.link_capacity_kbps = 500;
@@ -521,12 +615,13 @@ TEST_F(FullStackTest, ForemanCif500kbps100ms) {
 TEST_F(FullStackTest, ForemanCif500kbps100msLimitedQueue) {
   FullStackTestParams foreman_cif = {"foreman_cif_500kbps_100ms_32pkts_queue",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      30000,
                                      500000,
                                      2000000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   foreman_cif.link.queue_length_packets = 32;
   foreman_cif.link.queue_delay_ms = 100;
   foreman_cif.link.link_capacity_kbps = 500;
@@ -536,15 +631,30 @@ TEST_F(FullStackTest, ForemanCif500kbps100msLimitedQueue) {
 TEST_F(FullStackTest, ForemanCif1000kbps100msLimitedQueue) {
   FullStackTestParams foreman_cif = {"foreman_cif_1000kbps_100ms_32pkts_queue",
                                      {"foreman_cif", 352, 288, 30},
+                                     false,
                                      30000,
                                      2000000,
                                      2000000,
                                      0.0,
-                                     0.0
-                                    };
+                                     0.0,
+                                     kFullStackTestDurationSecs};
   foreman_cif.link.queue_length_packets = 32;
   foreman_cif.link.queue_delay_ms = 100;
   foreman_cif.link.link_capacity_kbps = 1000;
   RunTest(foreman_cif);
 }
+
+TEST_F(FullStackTest, ScreenshareSlides) {
+  FullStackTestParams screenshare_params = {
+      "screenshare_slides",
+      {"screenshare_slides", 1850, 1110, 5},
+      true,
+      50000,
+      100000,
+      1000000,
+      0.0,
+      0.0,
+      kFullStackTestDurationSecs};
+  RunTest(screenshare_params);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/loopback.cc b/media/webrtc/trunk/webrtc/video/loopback.cc
index 8013833dff..9ef0ec4224 100644
--- a/media/webrtc/trunk/webrtc/video/loopback.cc
+++ b/media/webrtc/trunk/webrtc/video/loopback.cc
@@ -12,120 +12,70 @@
 
 #include 
 
-#include "gflags/gflags.h"
+#include "webrtc/video/loopback.h"
+
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/field_trial.h"
 #include "webrtc/test/run_loop.h"
-#include "webrtc/test/run_test.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 #include "webrtc/test/video_capturer.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace test {
 
 static const int kAbsSendTimeExtensionId = 7;
 
-namespace flags {
-
-DEFINE_int32(width, 640, "Video width.");
-size_t Width() { return static_cast(FLAGS_width); }
-
-DEFINE_int32(height, 480, "Video height.");
-size_t Height() { return static_cast(FLAGS_height); }
-
-DEFINE_int32(fps, 30, "Frames per second.");
-int Fps() { return static_cast(FLAGS_fps); }
-
-DEFINE_int32(min_bitrate, 50, "Minimum video bitrate.");
-size_t MinBitrate() { return static_cast(FLAGS_min_bitrate); }
-
-DEFINE_int32(start_bitrate, 300, "Video starting bitrate.");
-size_t StartBitrate() { return static_cast(FLAGS_start_bitrate); }
-
-DEFINE_int32(max_bitrate, 800, "Maximum video bitrate.");
-size_t MaxBitrate() { return static_cast(FLAGS_max_bitrate); }
-
-DEFINE_string(codec, "VP8", "Video codec to use.");
-std::string Codec() { return static_cast(FLAGS_codec); }
-
-DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
-int LossPercent() {
-  return static_cast(FLAGS_loss_percent);
-}
-
-DEFINE_int32(link_capacity,
-             0,
-             "Capacity (kbps) of the fake link. 0 means infinite.");
-int LinkCapacity() {
-  return static_cast(FLAGS_link_capacity);
-}
-
-DEFINE_int32(queue_size, 0, "Size of the bottleneck link queue in packets.");
-int QueueSize() {
-  return static_cast(FLAGS_queue_size);
-}
-
-DEFINE_int32(avg_propagation_delay_ms,
-             0,
-             "Average link propagation delay in ms.");
-int AvgPropagationDelayMs() {
-  return static_cast(FLAGS_avg_propagation_delay_ms);
-}
-
-DEFINE_int32(std_propagation_delay_ms,
-             0,
-             "Link propagation delay standard deviation in ms.");
-int StdPropagationDelayMs() {
-  return static_cast(FLAGS_std_propagation_delay_ms);
-}
-
-DEFINE_bool(logs, false, "print logs to stderr");
-
-DEFINE_string(
-    force_fieldtrials,
-    "",
-    "Field trials control experimental feature code which can be forced. "
-    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
-    " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
-    "trials are separated by \"/\"");
-}  // namespace flags
-
 static const uint32_t kSendSsrc = 0x654321;
 static const uint32_t kSendRtxSsrc = 0x654322;
 static const uint32_t kReceiverLocalSsrc = 0x123456;
 
 static const uint8_t kRtxPayloadType = 96;
 
-void Loopback() {
-  scoped_ptr trace_to_stderr_;
-  if (webrtc::flags::FLAGS_logs)
+Loopback::Loopback(const Config& config)
+    : config_(config), clock_(Clock::GetRealTimeClock()) {
+}
+
+Loopback::~Loopback() {
+}
+
+void Loopback::Run() {
+  rtc::scoped_ptr trace_to_stderr_;
+  if (config_.logs)
     trace_to_stderr_.reset(new test::TraceToStderr);
 
-  scoped_ptr local_preview(test::VideoRenderer::Create(
-      "Local Preview", flags::Width(), flags::Height()));
-  scoped_ptr loopback_video(test::VideoRenderer::Create(
-      "Loopback Video", flags::Width(), flags::Height()));
+  rtc::scoped_ptr local_preview(
+      test::VideoRenderer::Create("Local Preview", config_.width,
+                                  config_.height));
+  rtc::scoped_ptr loopback_video(
+      test::VideoRenderer::Create("Loopback Video", config_.width,
+                                  config_.height));
 
   FakeNetworkPipe::Config pipe_config;
-  pipe_config.loss_percent = flags::LossPercent();
-  pipe_config.link_capacity_kbps = flags::LinkCapacity();
-  pipe_config.queue_length_packets = flags::QueueSize();
-  pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
-  pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
+  pipe_config.loss_percent = config_.loss_percent;
+  pipe_config.link_capacity_kbps = config_.link_capacity_kbps;
+  pipe_config.queue_length_packets = config_.queue_size;
+  pipe_config.queue_delay_ms = config_.avg_propagation_delay_ms;
+  pipe_config.delay_standard_deviation_ms = config_.std_propagation_delay_ms;
   test::DirectTransport transport(pipe_config);
   Call::Config call_config(&transport);
-  call_config.stream_start_bitrate_bps =
-      static_cast(flags::StartBitrate()) * 1000;
-  scoped_ptr call(Call::Create(call_config));
+
+  call_config.bitrate_config.min_bitrate_bps =
+      static_cast(config_.min_bitrate_kbps) * 1000;
+  call_config.bitrate_config.start_bitrate_bps =
+      static_cast(config_.start_bitrate_kbps) * 1000;
+  call_config.bitrate_config.max_bitrate_bps =
+      static_cast(config_.max_bitrate_kbps) * 1000;
+  rtc::scoped_ptr call(Call::Create(call_config));
 
   // Loopback, call sends to itself.
   transport.SetReceiver(call->Receiver());
@@ -139,41 +89,26 @@ void Loopback() {
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
 
   send_config.local_renderer = local_preview.get();
-  scoped_ptr encoder;
-  if (flags::Codec() == "VP8") {
+  rtc::scoped_ptr encoder;
+  if (config_.codec == "VP8") {
     encoder.reset(VideoEncoder::Create(VideoEncoder::kVp8));
-  } else if (flags::Codec() == "VP9") {
+  } else if (config_.codec == "VP9") {
     encoder.reset(VideoEncoder::Create(VideoEncoder::kVp9));
   } else {
     // Codec not supported.
-    assert(false && "Codec not supported!");
+    RTC_NOTREACHED() << "Codec not supported!";
     return;
   }
   send_config.encoder_settings.encoder = encoder.get();
-  send_config.encoder_settings.payload_name = flags::Codec();
+  send_config.encoder_settings.payload_name = config_.codec;
   send_config.encoder_settings.payload_type = 124;
-  VideoEncoderConfig encoder_config;
-  encoder_config.streams = test::CreateVideoStreams(1);
-  VideoStream* stream = &encoder_config.streams[0];
-  stream->width = flags::Width();
-  stream->height = flags::Height();
-  stream->min_bitrate_bps = static_cast(flags::MinBitrate()) * 1000;
-  stream->target_bitrate_bps = static_cast(flags::MaxBitrate()) * 1000;
-  stream->max_bitrate_bps = static_cast(flags::MaxBitrate()) * 1000;
-  stream->max_framerate = 30;
-  stream->max_qp = 56;
+
+  VideoEncoderConfig encoder_config(CreateEncoderConfig());
 
   VideoSendStream* send_stream =
       call->CreateVideoSendStream(send_config, encoder_config);
 
-  Clock* test_clock = Clock::GetRealTimeClock();
-
-  scoped_ptr camera(
-      test::VideoCapturer::Create(send_stream->Input(),
-                                  flags::Width(),
-                                  flags::Height(),
-                                  flags::Fps(),
-                                  test_clock));
+  rtc::scoped_ptr capturer(CreateCapturer(send_stream));
 
   VideoReceiveStream::Config receive_config;
   receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
@@ -193,11 +128,11 @@ void Loopback() {
 
   receive_stream->Start();
   send_stream->Start();
-  camera->Start();
+  capturer->Start();
 
   test::PressEnterToContinue();
 
-  camera->Stop();
+  capturer->Stop();
   send_stream->Stop();
   receive_stream->Stop();
 
@@ -208,13 +143,26 @@ void Loopback() {
 
   transport.StopSending();
 }
-}  // namespace webrtc
 
-int main(int argc, char* argv[]) {
-  ::testing::InitGoogleTest(&argc, argv);
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::test::InitFieldTrialsFromString(
-      webrtc::flags::FLAGS_force_fieldtrials);
-  webrtc::test::RunTest(webrtc::Loopback);
-  return 0;
+VideoEncoderConfig Loopback::CreateEncoderConfig() {
+  VideoEncoderConfig encoder_config;
+  encoder_config.streams = test::CreateVideoStreams(1);
+  VideoStream* stream = &encoder_config.streams[0];
+  stream->width = config_.width;
+  stream->height = config_.height;
+  stream->min_bitrate_bps = static_cast(config_.min_bitrate_kbps) * 1000;
+  stream->max_bitrate_bps = static_cast(config_.max_bitrate_kbps) * 1000;
+  stream->target_bitrate_bps =
+      static_cast(config_.max_bitrate_kbps) * 1000;
+  stream->max_framerate = config_.fps;
+  stream->max_qp = 56;
+  return encoder_config;
 }
+
+test::VideoCapturer* Loopback::CreateCapturer(VideoSendStream* send_stream) {
+  return test::VideoCapturer::Create(send_stream->Input(), config_.width,
+                                     config_.height, config_.fps, clock_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/loopback.h b/media/webrtc/trunk/webrtc/video/loopback.h
new file mode 100644
index 0000000000..d0aa591b4c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/loopback.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include 
+
+#include "webrtc/config.h"
+
+namespace webrtc {
+
+class VideoSendStream;
+class Clock;
+
+namespace test {
+
+class VideoCapturer;
+
+class Loopback {
+ public:
+  struct Config {
+    size_t width;
+    size_t height;
+    int32_t fps;
+    size_t min_bitrate_kbps;
+    size_t start_bitrate_kbps;
+    size_t max_bitrate_kbps;
+    int32_t min_transmit_bitrate_kbps;
+    std::string codec;
+    int32_t loss_percent;
+    int32_t link_capacity_kbps;
+    int32_t queue_size;
+    int32_t avg_propagation_delay_ms;
+    int32_t std_propagation_delay_ms;
+    bool logs;
+  };
+
+  explicit Loopback(const Config& config);
+  virtual ~Loopback();
+
+  void Run();
+
+ protected:
+  virtual VideoEncoderConfig CreateEncoderConfig();
+  virtual VideoCapturer* CreateCapturer(VideoSendStream* send_stream);
+
+  const Config config_;
+  Clock* const clock_;
+};
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/rampup_tests.cc b/media/webrtc/trunk/webrtc/video/rampup_tests.cc
index 011ef6af57..35d3297f39 100644
--- a/media/webrtc/trunk/webrtc/video/rampup_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/rampup_tests.cc
@@ -9,6 +9,8 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/common.h"
 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
@@ -91,7 +93,7 @@ void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) {
 void StreamObserver::OnReceiveBitrateChanged(
     const std::vector& ssrcs, unsigned int bitrate) {
   CriticalSectionScoped lock(crit_.get());
-  assert(expected_bitrate_bps_ > 0);
+  DCHECK_GT(expected_bitrate_bps_, 0u);
   if (start_bitrate_bps_ != 0) {
     // For tests with an explicitly set start bitrate, verify the first
     // bitrate estimate is close to the start bitrate and lower than the
@@ -108,19 +110,18 @@ void StreamObserver::OnReceiveBitrateChanged(
       TriggerTestDone();
     }
   }
-  rtp_rtcp_->SetREMBData(
-      bitrate, static_cast(ssrcs.size()), &ssrcs[0]);
+  rtp_rtcp_->SetREMBData(bitrate, ssrcs);
   rtp_rtcp_->Process();
 }
 
 bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
   CriticalSectionScoped lock(crit_.get());
   RTPHeader header;
-  EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast(length), &header));
+  EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
   payload_registry_->SetIncomingPayloadType(header);
   remote_bitrate_estimator_->IncomingPacket(
-      clock_->TimeInMilliseconds(), static_cast(length - 12), header);
+      clock_->TimeInMilliseconds(), length - 12, header);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
   }
@@ -135,7 +136,7 @@ bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
       ++rtx_media_packets_sent_;
     uint8_t restored_packet[kMaxPacketSize];
     uint8_t* restored_packet_ptr = restored_packet;
-    int restored_length = static_cast(length);
+    size_t restored_length = length;
     payload_registry_->RestoreOriginalPacket(&restored_packet_ptr,
                                              packet,
                                              &restored_length,
@@ -197,7 +198,7 @@ LowRateStreamObserver::LowRateStreamObserver(
       feedback_transport_(feedback_transport),
       receive_stats_(ReceiveStatistics::Create(clock)),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      send_stream_(NULL),
+      send_stream_(nullptr),
       test_state_(kFirstRampup),
       state_start_ms_(clock_->TimeInMilliseconds()),
       interval_start_ms_(state_start_ms_),
@@ -212,12 +213,12 @@ LowRateStreamObserver::LowRateStreamObserver(
   rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
   rtp_rtcp_->SetREMBStatus(true);
   rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
-  rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
-                                          kTransmissionTimeOffsetExtensionId);
+  rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                          kAbsSendTimeExtensionId);
   AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
   const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000;
   remote_bitrate_estimator_.reset(
-      rbe_factory.Create(this, clock, kMimdControl,
+      rbe_factory.Create(this, clock, kAimdControl,
                          kRemoteBitrateEstimatorMinBitrateBps));
   forward_transport_config_.link_capacity_kbps =
       kHighBandwidthLimitBps / 1000;
@@ -226,7 +227,7 @@ LowRateStreamObserver::LowRateStreamObserver(
   test::DirectTransport::SetReceiver(this);
 }
 
-void LowRateStreamObserver::SetSendStream(const VideoSendStream* send_stream) {
+void LowRateStreamObserver::SetSendStream(VideoSendStream* send_stream) {
   CriticalSectionScoped lock(crit_.get());
   send_stream_ = send_stream;
 }
@@ -235,8 +236,7 @@ void LowRateStreamObserver::OnReceiveBitrateChanged(
     const std::vector& ssrcs,
     unsigned int bitrate) {
   CriticalSectionScoped lock(crit_.get());
-  rtp_rtcp_->SetREMBData(
-      bitrate, static_cast(ssrcs.size()), &ssrcs[0]);
+  rtp_rtcp_->SetREMBData(bitrate, ssrcs);
   rtp_rtcp_->Process();
   last_remb_bps_ = bitrate;
 }
@@ -267,10 +267,10 @@ PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
     const uint8_t* packet, size_t length) {
   CriticalSectionScoped lock(crit_.get());
   RTPHeader header;
-  EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast(length), &header));
+  EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
   remote_bitrate_estimator_->IncomingPacket(
-      clock_->TimeInMilliseconds(), static_cast(length - 12), header);
+      clock_->TimeInMilliseconds(), length - 12, header);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
   }
@@ -299,7 +299,7 @@ std::string LowRateStreamObserver::GetModifierString() {
 void LowRateStreamObserver::EvolveTestState(unsigned int bitrate_bps) {
   int64_t now = clock_->TimeInMilliseconds();
   CriticalSectionScoped lock(crit_.get());
-  assert(send_stream_ != NULL);
+  DCHECK(send_stream_ != nullptr);
   switch (test_state_) {
     case kFirstRampup: {
       EXPECT_FALSE(suspended_in_stats_);
@@ -381,7 +381,7 @@ void RampUpTest::RunRampUpTest(bool rtx,
 
   CreateSendConfig(num_streams);
 
-  scoped_ptr rbe_factory;
+  rtc::scoped_ptr rbe_factory;
   RateControlType control_type;
   if (extension_type == RtpExtension::kAbsSendTime) {
     control_type = kAimdControl;
@@ -404,7 +404,7 @@ void RampUpTest::RunRampUpTest(bool rtx,
 
   Call::Config call_config(&stream_observer);
   if (start_bitrate_bps != 0) {
-    call_config.stream_start_bitrate_bps = start_bitrate_bps;
+    call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps;
     stream_observer.set_start_bitrate_bps(start_bitrate_bps);
   }
 
@@ -422,7 +422,6 @@ void RampUpTest::RunRampUpTest(bool rtx,
   if (rtx) {
     send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
     send_config_.rtp.rtx.ssrcs = rtx_ssrcs;
-    send_config_.rtp.rtx.pad_with_redundant_payloads = true;
   }
 
   if (num_streams == 1) {
@@ -463,12 +462,11 @@ void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
 
   send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   send_config_.rtp.extensions.push_back(RtpExtension(
-      RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId));
+      RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
   send_config_.suspend_below_min_bitrate = true;
   if (rtx) {
     send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
     send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200);
-    send_config_.rtp.rtx.pad_with_redundant_payloads = true;
   }
 
   CreateStreams();
@@ -508,4 +506,20 @@ TEST_F(RampUpTest, UpDownUpOneStreamRtx) { RunRampUpDownUpTest(1, true); }
 
 TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) { RunRampUpDownUpTest(3, true); }
 
+TEST_F(RampUpTest, AbsSendTimeSingleStream) {
+  RunRampUpTest(false, 1, 0, RtpExtension::kAbsSendTime);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcast) {
+  RunRampUpTest(false, 3, 0, RtpExtension::kAbsSendTime);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcastWithRtx) {
+  RunRampUpTest(true, 3, 0, RtpExtension::kAbsSendTime);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
+  RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps,
+                RtpExtension::kAbsSendTime);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/rampup_tests.h b/media/webrtc/trunk/webrtc/video/rampup_tests.h
index 69399b419c..4335fc13a8 100644
--- a/media/webrtc/trunk/webrtc/video/rampup_tests.h
+++ b/media/webrtc/trunk/webrtc/video/rampup_tests.h
@@ -15,10 +15,10 @@
 #include 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/video/transport_adapter.h"
 
@@ -49,12 +49,12 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
 
   void set_start_bitrate_bps(unsigned int start_bitrate_bps);
 
-  virtual void OnReceiveBitrateChanged(const std::vector& ssrcs,
-                                       unsigned int bitrate) OVERRIDE;
+  void OnReceiveBitrateChanged(const std::vector& ssrcs,
+                               unsigned int bitrate) override;
 
-  virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE;
+  bool SendRtp(const uint8_t* packet, size_t length) override;
 
-  virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
 
   EventTypeWrapper Wait();
 
@@ -65,15 +65,15 @@ class StreamObserver : public newapi::Transport, public RemoteBitrateObserver {
   void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   Clock* const clock_;
-  const scoped_ptr test_done_;
-  const scoped_ptr rtp_parser_;
-  scoped_ptr rtp_rtcp_;
+  const rtc::scoped_ptr test_done_;
+  const rtc::scoped_ptr rtp_parser_;
+  rtc::scoped_ptr rtp_rtcp_;
   internal::TransportAdapter feedback_transport_;
-  const scoped_ptr receive_stats_;
-  const scoped_ptr payload_registry_;
-  scoped_ptr remote_bitrate_estimator_;
+  const rtc::scoped_ptr receive_stats_;
+  const rtc::scoped_ptr payload_registry_;
+  rtc::scoped_ptr remote_bitrate_estimator_;
 
-  const scoped_ptr crit_;
+  const rtc::scoped_ptr crit_;
   unsigned int expected_bitrate_bps_ GUARDED_BY(crit_);
   unsigned int start_bitrate_bps_ GUARDED_BY(crit_);
   SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_);
@@ -96,17 +96,16 @@ class LowRateStreamObserver : public test::DirectTransport,
                         size_t number_of_streams,
                         bool rtx_used);
 
-  virtual void SetSendStream(const VideoSendStream* send_stream);
+  virtual void SetSendStream(VideoSendStream* send_stream);
 
   virtual void OnReceiveBitrateChanged(const std::vector& ssrcs,
                                        unsigned int bitrate);
 
-  virtual bool SendRtp(const uint8_t* data, size_t length) OVERRIDE;
+  bool SendRtp(const uint8_t* data, size_t length) override;
 
-  virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                       size_t length) OVERRIDE;
+  DeliveryStatus DeliverPacket(const uint8_t* packet, size_t length) override;
 
-  virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
 
   // Produces a string similar to "1stream_nortx", depending on the values of
   // number_of_streams_ and rtx_used_;
@@ -127,15 +126,15 @@ class LowRateStreamObserver : public test::DirectTransport,
   Clock* const clock_;
   const size_t number_of_streams_;
   const bool rtx_used_;
-  const scoped_ptr test_done_;
-  const scoped_ptr rtp_parser_;
-  scoped_ptr rtp_rtcp_;
+  const rtc::scoped_ptr test_done_;
+  const rtc::scoped_ptr rtp_parser_;
+  rtc::scoped_ptr rtp_rtcp_;
   internal::TransportAdapter feedback_transport_;
-  const scoped_ptr receive_stats_;
-  scoped_ptr remote_bitrate_estimator_;
+  const rtc::scoped_ptr receive_stats_;
+  rtc::scoped_ptr remote_bitrate_estimator_;
 
-  scoped_ptr crit_;
-  const VideoSendStream* send_stream_ GUARDED_BY(crit_);
+  rtc::scoped_ptr crit_;
+  VideoSendStream* send_stream_ GUARDED_BY(crit_);
   FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_);
   TestStates test_state_ GUARDED_BY(crit_);
   int64_t state_start_ms_ GUARDED_BY(crit_);
diff --git a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc
index fc4a6c29d0..b7a38535a4 100644
--- a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc
+++ b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.cc
@@ -14,17 +14,9 @@
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
-namespace internal {
 
-ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc,
-                                               Clock* clock,
-                                               ViERTP_RTCP* rtp_rtcp,
-                                               ViECodec* codec,
-                                               int channel)
-    : channel_(channel),
-      clock_(clock),
-      codec_(codec),
-      rtp_rtcp_(rtp_rtcp),
+ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
+    : clock_(clock),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
@@ -36,25 +28,8 @@ ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc,
 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {}
 
 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
-  VideoReceiveStream::Stats stats;
-  {
-    CriticalSectionScoped lock(crit_.get());
-    stats = stats_;
-  }
-  stats.c_name = GetCName();
-  codec_->GetReceiveSideDelay(channel_, &stats.avg_delay_ms);
-  stats.discarded_packets = codec_->GetNumDiscardedPackets(channel_);
-  codec_->GetReceiveCodecStatistics(
-      channel_, stats.key_frames, stats.delta_frames);
-
-  return stats;
-}
-
-std::string ReceiveStatisticsProxy::GetCName() const {
-  char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
-  if (rtp_rtcp_->GetRemoteRTCPCName(channel_, rtcp_cname) != 0)
-    rtcp_cname[0] = '\0';
-  return rtcp_cname;
+  CriticalSectionScoped lock(crit_.get());
+  return stats_;
 }
 
 void ReceiveStatisticsProxy::IncomingRate(const int video_channel,
@@ -71,19 +46,58 @@ void ReceiveStatisticsProxy::ReceiveStateChange(const int video_channel,
   receive_state_ = state;
 }
 
+void ReceiveStatisticsProxy::DecoderTiming(int decode_ms,
+                                           int max_decode_ms,
+                                           int current_delay_ms,
+                                           int target_delay_ms,
+                                           int jitter_buffer_ms,
+                                           int min_playout_delay_ms,
+                                           int render_delay_ms) {
+  CriticalSectionScoped lock(crit_.get());
+  stats_.decode_ms = decode_ms;
+  stats_.max_decode_ms = max_decode_ms;
+  stats_.current_delay_ms = current_delay_ms;
+  stats_.target_delay_ms = target_delay_ms;
+  stats_.jitter_buffer_ms = jitter_buffer_ms;
+  stats_.min_playout_delay_ms = min_playout_delay_ms;
+  stats_.render_delay_ms = render_delay_ms;
+}
+
+void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
+    uint32_t ssrc,
+    const RtcpPacketTypeCounter& packet_counter) {
+  CriticalSectionScoped lock(crit_.get());
+  if (stats_.ssrc != ssrc)
+    return;
+  stats_.rtcp_packet_type_counts = packet_counter;
+}
+
 void ReceiveStatisticsProxy::StatisticsUpdated(
     const webrtc::RtcpStatistics& statistics,
     uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-
+  // TODO(pbos): Handle both local and remote ssrcs here and DCHECK that we
+  // receive stats from one of them.
+  if (stats_.ssrc != ssrc)
+    return;
   stats_.rtcp_stats = statistics;
 }
 
+void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
+  CriticalSectionScoped lock(crit_.get());
+  // TODO(pbos): Handle both local and remote ssrcs here and DCHECK that we
+  // receive stats from one of them.
+  if (stats_.ssrc != ssrc)
+    return;
+  stats_.c_name = cname;
+}
+
 void ReceiveStatisticsProxy::DataCountersUpdated(
     const webrtc::StreamDataCounters& counters,
     uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-
+  if (stats_.ssrc != ssrc)
+    return;
   stats_.rtp_stats = counters;
 }
 
@@ -103,5 +117,19 @@ void ReceiveStatisticsProxy::OnRenderedFrame() {
   stats_.render_frame_rate = renders_fps_estimator_.Rate(now);
 }
 
-}  // namespace internal
+void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
+                                                   uint32_t frameRate) {
+}
+
+void ReceiveStatisticsProxy::OnFrameCountsUpdated(
+    const FrameCounts& frame_counts) {
+  CriticalSectionScoped lock(crit_.get());
+  stats_.frame_counts = frame_counts;
+}
+
+void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
+  CriticalSectionScoped lock(crit_.get());
+  stats_.discarded_packets = discarded_packets;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h
index 2bbd9db969..aaecb02a38 100644
--- a/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h
+++ b/media/webrtc/trunk/webrtc/video/receive_statistics_proxy.h
@@ -17,6 +17,7 @@
 #include "webrtc/common_types.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
+#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_engine/include/vie_rtp_rtcp.h"
 #include "webrtc/video_receive_stream.h"
@@ -29,17 +30,13 @@ class CriticalSectionWrapper;
 class ViECodec;
 class ViEDecoderObserver;
 
-namespace internal {
-
 class ReceiveStatisticsProxy : public ViEDecoderObserver,
+                               public VCMReceiveStatisticsCallback,
                                public RtcpStatisticsCallback,
+                               public RtcpPacketTypeCounterObserver,
                                public StreamDataCountersCallback {
  public:
-  ReceiveStatisticsProxy(uint32_t ssrc,
-                         Clock* clock,
-                         ViERTP_RTCP* rtp_rtcp,
-                         ViECodec* codec,
-                         int channel);
+  ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock);
   virtual ~ReceiveStatisticsProxy();
 
   VideoReceiveStream::Stats GetStats() const;
@@ -47,45 +44,49 @@ class ReceiveStatisticsProxy : public ViEDecoderObserver,
   void OnDecodedFrame();
   void OnRenderedFrame();
 
+  // Overrides VCMReceiveStatisticsCallback
+  void OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) override;
+  void OnFrameCountsUpdated(const FrameCounts& frame_counts) override;
+  void OnDiscardedPacketsUpdated(int discarded_packets) override;
+
   // Overrides ViEDecoderObserver.
-  virtual void IncomingCodecChanged(const int video_channel,
-                                    const VideoCodec& video_codec) OVERRIDE {}
-  virtual void IncomingRate(const int video_channel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate_bps) OVERRIDE;
-  virtual void DecoderTiming(int decode_ms,
-                             int max_decode_ms,
-                             int current_delay_ms,
-                             int target_delay_ms,
-                             int jitter_buffer_ms,
-                             int min_playout_delay_ms,
-                             int render_delay_ms) OVERRIDE {}
-  virtual void RequestNewKeyFrame(const int video_channel) OVERRIDE {}
-  virtual void ReceiveStateChange(const int video_channel, VideoReceiveState state) OVERRIDE;
+  void IncomingCodecChanged(const int video_channel,
+                            const VideoCodec& video_codec) override {}
+  void IncomingRate(const int video_channel,
+                    const unsigned int framerate,
+                    const unsigned int bitrate_bps) override;
+  void DecoderTiming(int decode_ms,
+                     int max_decode_ms,
+                     int current_delay_ms,
+                     int target_delay_ms,
+                     int jitter_buffer_ms,
+                     int min_playout_delay_ms,
+                     int render_delay_ms) override;
+  void RequestNewKeyFrame(const int video_channel) override {}
+  virtual void ReceiveStateChange(const int video_channel, VideoReceiveState state) override;
 
-  // Overrides RtcpStatisticsBallback.
-  virtual void StatisticsUpdated(const webrtc::RtcpStatistics& statistics,
-                                 uint32_t ssrc) OVERRIDE;
+  // Overrides RtcpStatisticsCallback.
+  void StatisticsUpdated(const webrtc::RtcpStatistics& statistics,
+                         uint32_t ssrc) override;
+  void CNameChanged(const char* cname, uint32_t ssrc) override;
 
+  // Overrides RtcpPacketTypeCounterObserver
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override;
   // Overrides StreamDataCountersCallback.
-  virtual void DataCountersUpdated(const webrtc::StreamDataCounters& counters,
-                                   uint32_t ssrc) OVERRIDE;
+  void DataCountersUpdated(const webrtc::StreamDataCounters& counters,
+                           uint32_t ssrc) override;
 
  private:
-  std::string GetCName() const;
-
-  const int channel_;
   Clock* const clock_;
-  ViECodec* const codec_;
-  ViERTP_RTCP* const rtp_rtcp_;
 
-  scoped_ptr crit_;
+  rtc::scoped_ptr crit_;
   VideoReceiveStream::Stats stats_ GUARDED_BY(crit_);
   RateStatistics decode_fps_estimator_ GUARDED_BY(crit_);
   RateStatistics renders_fps_estimator_ GUARDED_BY(crit_);
   VideoReceiveState receive_state_ GUARDED_BY(crit_);
 };
 
-}  // namespace internal
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_RECEIVE_STATISTICS_PROXY_H_
diff --git a/media/webrtc/trunk/webrtc/video/replay.cc b/media/webrtc/trunk/webrtc/video/replay.cc
index 5cfb06f8b8..740d149be8 100644
--- a/media/webrtc/trunk/webrtc/video/replay.cc
+++ b/media/webrtc/trunk/webrtc/video/replay.cc
@@ -16,14 +16,16 @@
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/null_transport.h"
+#include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/run_test.h"
@@ -106,6 +108,7 @@ bool ValidateInputFilenameNotEmpty(const char* flagname,
                                    const std::string& string) {
   return string != "";
 }
+
 DEFINE_string(input_file, "", "input file");
 static std::string InputFile() {
   return static_cast(FLAGS_input_file);
@@ -120,6 +123,11 @@ static std::string OutBase() {
   return static_cast(FLAGS_out_base);
 }
 
+DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file");
+static std::string DecoderBitstreamFilename() {
+  return static_cast(FLAGS_decoder_bitstream_filename);
+}
+
 // Flag for video codec.
 DEFINE_string(codec, "VP8", "Video codec");
 static std::string Codec() { return static_cast(FLAGS_codec); }
@@ -133,26 +141,26 @@ class FileRenderPassthrough : public VideoRenderer {
   FileRenderPassthrough(const std::string& basename, VideoRenderer* renderer)
       : basename_(basename),
         renderer_(renderer),
-        file_(NULL),
+        file_(nullptr),
         count_(0),
         last_width_(0),
         last_height_(0) {}
 
   ~FileRenderPassthrough() {
-    if (file_ != NULL)
+    if (file_ != nullptr)
       fclose(file_);
   }
 
  private:
-  virtual void RenderFrame(const I420VideoFrame& video_frame,
-                           int time_to_render_ms) OVERRIDE {
-    if (renderer_ != NULL)
+  void RenderFrame(const I420VideoFrame& video_frame,
+                   int time_to_render_ms) override {
+    if (renderer_ != nullptr)
       renderer_->RenderFrame(video_frame, time_to_render_ms);
     if (basename_ == "")
       return;
     if (last_width_ != video_frame.width() ||
         last_height_ != video_frame.height()) {
-      if (file_ != NULL)
+      if (file_ != nullptr)
         fclose(file_);
       std::stringstream filename;
       filename << basename_;
@@ -161,7 +169,7 @@ class FileRenderPassthrough : public VideoRenderer {
       filename << '_' << video_frame.width() << 'x' << video_frame.height()
                << ".yuv";
       file_ = fopen(filename.str().c_str(), "wb");
-      if (file_ == NULL) {
+      if (file_ == nullptr) {
         fprintf(stderr,
                 "Couldn't open file for writing: %s\n",
                 filename.str().c_str());
@@ -169,11 +177,13 @@ class FileRenderPassthrough : public VideoRenderer {
     }
     last_width_ = video_frame.width();
     last_height_ = video_frame.height();
-    if (file_ == NULL)
+    if (file_ == nullptr)
       return;
     PrintI420VideoFrame(video_frame, file_);
   }
 
+  bool IsTextureSupported() const override { return false; }
+
   const std::string basename_;
   VideoRenderer* const renderer_;
   FILE* file_;
@@ -182,9 +192,25 @@ class FileRenderPassthrough : public VideoRenderer {
   int last_height_;
 };
 
+class DecoderBitstreamFileWriter : public EncodedFrameObserver {
+ public:
+  explicit DecoderBitstreamFileWriter(const char* filename)
+      : file_(fopen(filename, "wb")) {
+    DCHECK(file_ != nullptr);
+  }
+  ~DecoderBitstreamFileWriter() { fclose(file_); }
+
+  virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
+    fwrite(encoded_frame.data_, 1, encoded_frame.length_, file_);
+  }
+
+ private:
+  FILE* file_;
+};
+
 void RtpReplay() {
-  scoped_ptr playback_video(test::VideoRenderer::Create(
-      "Playback Video", 640, 480));
+  rtc::scoped_ptr playback_video(
+      test::VideoRenderer::Create("Playback Video", 640, 480));
   FileRenderPassthrough file_passthrough(flags::OutBase(),
                                          playback_video.get());
 
@@ -192,7 +218,7 @@ void RtpReplay() {
   //             etc.
   test::NullTransport transport;
   Call::Config call_config(&transport);
-  scoped_ptr call(Call::Create(call_config));
+  rtc::scoped_ptr call(Call::Create(call_config));
 
   VideoReceiveStream::Config receive_config;
   receive_config.rtp.remote_ssrc = flags::Ssrc();
@@ -213,23 +239,42 @@ void RtpReplay() {
   VideoSendStream::Config::EncoderSettings encoder_settings;
   encoder_settings.payload_name = flags::Codec();
   encoder_settings.payload_type = flags::PayloadType();
-  VideoReceiveStream::Decoder decoder =
-      test::CreateMatchingDecoder(encoder_settings);
+  VideoReceiveStream::Decoder decoder;
+  rtc::scoped_ptr bitstream_writer;
+  if (flags::DecoderBitstreamFilename() != "") {
+    bitstream_writer.reset(new DecoderBitstreamFileWriter(
+        flags::DecoderBitstreamFilename().c_str()));
+    receive_config.pre_decode_callback = bitstream_writer.get();
+  }
+  decoder = test::CreateMatchingDecoder(encoder_settings);
+  if (flags::DecoderBitstreamFilename() != "") {
+    // Replace with a null decoder if we're writing the bitstream to a file
+    // instead.
+    delete decoder.decoder;
+    decoder.decoder = new test::FakeNullDecoder();
+  }
   receive_config.decoders.push_back(decoder);
 
   VideoReceiveStream* receive_stream =
       call->CreateVideoReceiveStream(receive_config);
 
-  scoped_ptr rtp_reader(test::RtpFileReader::Create(
+  rtc::scoped_ptr rtp_reader(test::RtpFileReader::Create(
       test::RtpFileReader::kRtpDump, flags::InputFile()));
-  if (rtp_reader.get() == NULL) {
+  if (rtp_reader.get() == nullptr) {
     rtp_reader.reset(test::RtpFileReader::Create(test::RtpFileReader::kPcap,
                                                  flags::InputFile()));
-    if (rtp_reader.get() == NULL) {
+    if (rtp_reader.get() == nullptr) {
       fprintf(stderr,
               "Couldn't open input file as either a rtpdump or .pcap. Note "
-              "that .pcapng is not supported.\n");
-      return;
+              "that .pcapng is not supported.\nTrying to interpret the file as "
+              "length/packet interleaved.\n");
+      rtp_reader.reset(test::RtpFileReader::Create(
+          test::RtpFileReader::kLengthPacketInterleaved, flags::InputFile()));
+      if (rtp_reader.get() == nullptr) {
+        fprintf(stderr,
+                "Unable to open input file with any supported format\n");
+        return;
+      }
     }
   }
   receive_stream->Start();
@@ -238,7 +283,7 @@ void RtpReplay() {
   int num_packets = 0;
   std::map unknown_packets;
   while (true) {
-    test::RtpFileReader::Packet packet;
+    test::RtpPacket packet;
     if (!rtp_reader->NextPacket(&packet))
       break;
     ++num_packets;
@@ -247,7 +292,7 @@ void RtpReplay() {
         break;
       case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
         RTPHeader header;
-        scoped_ptr parser(RtpHeaderParser::Create());
+        rtc::scoped_ptr parser(RtpHeaderParser::Create());
         parser->Parse(packet.data, packet.length, &header);
         if (unknown_packets[header.ssrc] == 0)
           fprintf(stderr, "Unknown SSRC: %u!\n", header.ssrc);
diff --git a/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc b/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc
new file mode 100644
index 0000000000..66bdf6ebc5
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/screenshare_loopback.cc
@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include 
+
+#include 
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/test/field_trial.h"
+#include "webrtc/test/frame_generator.h"
+#include "webrtc/test/frame_generator_capturer.h"
+#include "webrtc/test/run_test.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video/loopback.h"
+#include "webrtc/video/video_send_stream.h"
+
+namespace webrtc {
+namespace flags {
+
+// Fixed for prerecorded screenshare content.
+size_t Width() {
+  return 1850;
+}
+size_t Height() {
+  return 1110;
+}
+
+DEFINE_int32(fps, 5, "Frames per second.");
+int Fps() {
+  return static_cast(FLAGS_fps);
+}
+
+DEFINE_int32(min_bitrate, 50, "Minimum video bitrate.");
+size_t MinBitrate() {
+  return static_cast(FLAGS_min_bitrate);
+}
+
+DEFINE_int32(tl0_bitrate, 100, "Temporal layer 0 target bitrate.");
+size_t StartBitrate() {
+  return static_cast(FLAGS_tl0_bitrate);
+}
+
+DEFINE_int32(tl1_bitrate, 1000, "Temporal layer 1 target bitrate.");
+size_t MaxBitrate() {
+  return static_cast(FLAGS_tl1_bitrate);
+}
+
+DEFINE_int32(min_transmit_bitrate, 400, "Min transmit bitrate incl. padding.");
+int MinTransmitBitrate() {
+  return FLAGS_min_transmit_bitrate;
+}
+
+DEFINE_string(codec, "VP8", "Video codec to use.");
+std::string Codec() {
+  return static_cast(FLAGS_codec);
+}
+
+DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
+int LossPercent() {
+  return static_cast(FLAGS_loss_percent);
+}
+
+DEFINE_int32(link_capacity,
+             0,
+             "Capacity (kbps) of the fake link. 0 means infinite.");
+int LinkCapacity() {
+  return static_cast(FLAGS_link_capacity);
+}
+
+DEFINE_int32(queue_size, 0, "Size of the bottleneck link queue in packets.");
+int QueueSize() {
+  return static_cast(FLAGS_queue_size);
+}
+
+DEFINE_int32(avg_propagation_delay_ms,
+             0,
+             "Average link propagation delay in ms.");
+int AvgPropagationDelayMs() {
+  return static_cast(FLAGS_avg_propagation_delay_ms);
+}
+
+DEFINE_int32(std_propagation_delay_ms,
+             0,
+             "Link propagation delay standard deviation in ms.");
+int StdPropagationDelayMs() {
+  return static_cast(FLAGS_std_propagation_delay_ms);
+}
+
+DEFINE_bool(logs, false, "print logs to stderr");
+
+DEFINE_string(
+    force_fieldtrials,
+    "",
+    "Field trials control experimental feature code which can be forced. "
+    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+    " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
+    "trials are separated by \"/\"");
+}  // namespace flags
+
+class ScreenshareLoopback : public test::Loopback {
+ public:
+  explicit ScreenshareLoopback(const Config& config) : Loopback(config) {}
+  virtual ~ScreenshareLoopback() {}
+
+ protected:
+  VideoEncoderConfig CreateEncoderConfig() override {
+    VideoEncoderConfig encoder_config(test::Loopback::CreateEncoderConfig());
+    VideoStream* stream = &encoder_config.streams[0];
+    encoder_config.content_type = VideoEncoderConfig::kScreenshare;
+    encoder_config.min_transmit_bitrate_bps = flags::MinTransmitBitrate();
+    VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
+    vp8_settings.denoisingOn = false;
+    vp8_settings.frameDroppingOn = false;
+    vp8_settings.numberOfTemporalLayers = 2;
+    encoder_config.encoder_specific_settings = &vp8_settings;
+    stream->temporal_layer_thresholds_bps.clear();
+    stream->target_bitrate_bps =
+        static_cast(config_.start_bitrate_kbps) * 1000;
+    stream->temporal_layer_thresholds_bps.push_back(stream->target_bitrate_bps);
+    return encoder_config;
+  }
+
+  test::VideoCapturer* CreateCapturer(VideoSendStream* send_stream) override {
+    std::vector slides;
+    slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
+    slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
+    slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
+    slides.push_back(test::ResourcePath("difficult_photo_1850_1110", "yuv"));
+
+    test::FrameGenerator* frame_generator =
+        test::FrameGenerator::CreateFromYuvFile(
+            slides, flags::Width(), flags::Height(), 10 * flags::Fps());
+    test::FrameGeneratorCapturer* capturer(new test::FrameGeneratorCapturer(
+        clock_, send_stream->Input(), frame_generator, flags::Fps()));
+    EXPECT_TRUE(capturer->Init());
+    return capturer;
+  }
+};
+
+void Loopback() {
+  test::Loopback::Config config{flags::Width(),
+                                flags::Height(),
+                                flags::Fps(),
+                                flags::MinBitrate(),
+                                flags::StartBitrate(),
+                                flags::MaxBitrate(),
+                                flags::MinTransmitBitrate(),
+                                flags::Codec(),
+                                flags::LossPercent(),
+                                flags::LinkCapacity(),
+                                flags::QueueSize(),
+                                flags::AvgPropagationDelayMs(),
+                                flags::StdPropagationDelayMs(),
+                                flags::FLAGS_logs};
+  ScreenshareLoopback loopback(config);
+  loopback.Run();
+}
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  ::testing::InitGoogleTest(&argc, argv);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::test::InitFieldTrialsFromString(
+      webrtc::flags::FLAGS_force_fieldtrials);
+  webrtc::test::RunTest(webrtc::Loopback);
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc
index f2df0ed2f5..41a8a99727 100644
--- a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc
+++ b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.cc
@@ -12,13 +12,19 @@
 
 #include 
 
+#include "webrtc/base/checks.h"
+
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
 
-SendStatisticsProxy::SendStatisticsProxy(
-    const VideoSendStream::Config& config)
-    : config_(config),
+const int SendStatisticsProxy::kStatsTimeoutMs = 5000;
+
+SendStatisticsProxy::SendStatisticsProxy(Clock* clock,
+                                         const VideoSendStream::Config& config)
+    : clock_(clock),
+      config_(config),
       crit_(CriticalSectionWrapper::CreateCriticalSection()) {
 }
 
@@ -32,24 +38,45 @@ void SendStatisticsProxy::OutgoingRate(const int video_channel,
   stats_.media_bitrate_bps = bitrate;
 }
 
+void SendStatisticsProxy::CpuOveruseMetricsUpdated(
+    const CpuOveruseMetrics& metrics) {
+  CriticalSectionScoped lock(crit_.get());
+  stats_.avg_encode_time_ms = metrics.avg_encode_time_ms;
+  stats_.encode_usage_percent = metrics.encode_usage_percent;
+}
+
 void SendStatisticsProxy::SuspendChange(int video_channel, bool is_suspended) {
   CriticalSectionScoped lock(crit_.get());
   stats_.suspended = is_suspended;
 }
 
-void SendStatisticsProxy::CapturedFrameRate(const int capture_id,
-                                            const unsigned char frame_rate) {
-  CriticalSectionScoped lock(crit_.get());
-  stats_.input_frame_rate = frame_rate;
-}
-
-VideoSendStream::Stats SendStatisticsProxy::GetStats() const {
+VideoSendStream::Stats SendStatisticsProxy::GetStats() {
   CriticalSectionScoped lock(crit_.get());
+  PurgeOldStats();
+  stats_.input_frame_rate =
+      static_cast(input_frame_rate_tracker_.units_second());
   return stats_;
 }
 
-SsrcStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
-  std::map::iterator it = stats_.substreams.find(ssrc);
+void SendStatisticsProxy::PurgeOldStats() {
+  int64_t current_time_ms = clock_->TimeInMilliseconds();
+  for (std::map::iterator it =
+           stats_.substreams.begin();
+       it != stats_.substreams.end(); ++it) {
+    uint32_t ssrc = it->first;
+    if (update_times_[ssrc].resolution_update_ms + kStatsTimeoutMs >
+        current_time_ms)
+      continue;
+
+    it->second.width = 0;
+    it->second.height = 0;
+  }
+}
+
+VideoSendStream::StreamStats* SendStatisticsProxy::GetStatsEntry(
+    uint32_t ssrc) {
+  std::map::iterator it =
+      stats_.substreams.find(ssrc);
   if (it != stats_.substreams.end())
     return &it->second;
 
@@ -58,29 +85,75 @@ SsrcStats* SendStatisticsProxy::GetStatsEntry(uint32_t ssrc) {
       std::find(config_.rtp.rtx.ssrcs.begin(),
                 config_.rtp.rtx.ssrcs.end(),
                 ssrc) == config_.rtp.rtx.ssrcs.end()) {
-    return NULL;
+    return nullptr;
   }
 
   return &stats_.substreams[ssrc];  // Insert new entry and return ptr.
 }
 
+void SendStatisticsProxy::OnSetRates(uint32_t bitrate_bps, int framerate) {
+  CriticalSectionScoped lock(crit_.get());
+  stats_.target_media_bitrate_bps = bitrate_bps;
+}
+
+void SendStatisticsProxy::OnSendEncodedImage(
+    const EncodedImage& encoded_image,
+    const RTPVideoHeader* rtp_video_header) {
+  size_t simulcast_idx =
+      rtp_video_header != nullptr ? rtp_video_header->simulcastIdx : 0;
+  if (simulcast_idx >= config_.rtp.ssrcs.size()) {
+    LOG(LS_ERROR) << "Encoded image outside simulcast range (" << simulcast_idx
+                  << " >= " << config_.rtp.ssrcs.size() << ").";
+    return;
+  }
+  uint32_t ssrc = config_.rtp.ssrcs[simulcast_idx];
+
+  CriticalSectionScoped lock(crit_.get());
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
+    return;
+
+  stats->width = encoded_image._encodedWidth;
+  stats->height = encoded_image._encodedHeight;
+  update_times_[ssrc].resolution_update_ms = clock_->TimeInMilliseconds();
+}
+
+void SendStatisticsProxy::OnIncomingFrame() {
+  CriticalSectionScoped lock(crit_.get());
+  input_frame_rate_tracker_.Update(1);
+}
+
+void SendStatisticsProxy::RtcpPacketTypesCounterUpdated(
+    uint32_t ssrc,
+    const RtcpPacketTypeCounter& packet_counter) {
+  CriticalSectionScoped lock(crit_.get());
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
+    return;
+
+  stats->rtcp_packet_type_counts = packet_counter;
+}
+
 void SendStatisticsProxy::StatisticsUpdated(const RtcpStatistics& statistics,
                                             uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  SsrcStats* stats = GetStatsEntry(ssrc);
-  if (stats == NULL)
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
     return;
 
   stats->rtcp_stats = statistics;
 }
 
+void SendStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
+}
+
 void SendStatisticsProxy::DataCountersUpdated(
     const StreamDataCounters& counters,
     uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  SsrcStats* stats = GetStatsEntry(ssrc);
-  if (stats == NULL)
-    return;
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  DCHECK(stats != nullptr) << "DataCountersUpdated reported for unknown ssrc: "
+                           << ssrc;
 
   stats->rtp_stats = counters;
 }
@@ -89,42 +162,30 @@ void SendStatisticsProxy::Notify(const BitrateStatistics& total_stats,
                                  const BitrateStatistics& retransmit_stats,
                                  uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  SsrcStats* stats = GetStatsEntry(ssrc);
-  if (stats == NULL)
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
     return;
 
   stats->total_bitrate_bps = total_stats.bitrate_bps;
   stats->retransmit_bitrate_bps = retransmit_stats.bitrate_bps;
 }
 
-void SendStatisticsProxy::FrameCountUpdated(FrameType frame_type,
-                                            uint32_t frame_count,
-                                            const unsigned int ssrc) {
+void SendStatisticsProxy::FrameCountUpdated(const FrameCounts& frame_counts,
+                                            uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  SsrcStats* stats = GetStatsEntry(ssrc);
-  if (stats == NULL)
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
     return;
 
-  switch (frame_type) {
-    case kVideoFrameDelta:
-      stats->delta_frames = frame_count;
-      break;
-    case kVideoFrameKey:
-      stats->key_frames = frame_count;
-      break;
-    case kFrameEmpty:
-    case kAudioFrameSpeech:
-    case kAudioFrameCN:
-      break;
-  }
+  stats->frame_counts = frame_counts;
 }
 
 void SendStatisticsProxy::SendSideDelayUpdated(int avg_delay_ms,
                                                int max_delay_ms,
                                                uint32_t ssrc) {
   CriticalSectionScoped lock(crit_.get());
-  SsrcStats* stats = GetStatsEntry(ssrc);
-  if (stats == NULL)
+  VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
+  if (stats == nullptr)
     return;
   stats->avg_delay_ms = avg_delay_ms;
   stats->max_delay_ms = max_delay_ms;
diff --git a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h
index 2f645b1ebe..0a048a59c2 100644
--- a/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h
+++ b/media/webrtc/trunk/webrtc/video/send_statistics_proxy.h
@@ -13,75 +13,97 @@
 
 #include 
 
+#include "webrtc/base/ratetracker.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
-#include "webrtc/video_engine/include/vie_codec.h"
+#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/include/vie_capture.h"
+#include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_send_stream.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
 
-class SendStatisticsProxy : public RtcpStatisticsCallback,
+class SendStatisticsProxy : public CpuOveruseMetricsObserver,
+                            public RtcpStatisticsCallback,
+                            public RtcpPacketTypeCounterObserver,
                             public StreamDataCountersCallback,
                             public BitrateStatisticsObserver,
                             public FrameCountObserver,
                             public ViEEncoderObserver,
-                            public ViECaptureObserver,
+                            public VideoEncoderRateObserver,
                             public SendSideDelayObserver {
  public:
-  explicit SendStatisticsProxy(const VideoSendStream::Config& config);
+  static const int kStatsTimeoutMs;
+
+  SendStatisticsProxy(Clock* clock, const VideoSendStream::Config& config);
   virtual ~SendStatisticsProxy();
 
-  VideoSendStream::Stats GetStats() const;
+  VideoSendStream::Stats GetStats();
+
+  virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
+                                  const RTPVideoHeader* rtp_video_header);
+  // Used to update incoming frame rate.
+  void OnIncomingFrame();
+
+  // From VideoEncoderRateObserver.
+  void OnSetRates(uint32_t bitrate_bps, int framerate) override;
 
  protected:
+  // From CpuOveruseMetricsObserver.
+  void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override;
   // From RtcpStatisticsCallback.
-  virtual void StatisticsUpdated(const RtcpStatistics& statistics,
-                                 uint32_t ssrc) OVERRIDE;
+  void StatisticsUpdated(const RtcpStatistics& statistics,
+                         uint32_t ssrc) override;
+  void CNameChanged(const char* cname, uint32_t ssrc) override;
+  // From RtcpPacketTypeCounterObserver
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override;
   // From StreamDataCountersCallback.
-  virtual void DataCountersUpdated(const StreamDataCounters& counters,
-                                   uint32_t ssrc) OVERRIDE;
+  void DataCountersUpdated(const StreamDataCounters& counters,
+                           uint32_t ssrc) override;
 
   // From BitrateStatisticsObserver.
-  virtual void Notify(const BitrateStatistics& total_stats,
-                      const BitrateStatistics& retransmit_stats,
-                      uint32_t ssrc) OVERRIDE;
+  void Notify(const BitrateStatistics& total_stats,
+              const BitrateStatistics& retransmit_stats,
+              uint32_t ssrc) override;
 
   // From FrameCountObserver.
-  virtual void FrameCountUpdated(FrameType frame_type,
-                                 uint32_t frame_count,
-                                 const unsigned int ssrc) OVERRIDE;
+  void FrameCountUpdated(const FrameCounts& frame_counts,
+                         uint32_t ssrc) override;
 
   // From ViEEncoderObserver.
-  virtual void OutgoingRate(const int video_channel,
-                            const unsigned int framerate,
-                            const unsigned int bitrate) OVERRIDE;
+  void OutgoingRate(const int video_channel,
+                    const unsigned int framerate,
+                    const unsigned int bitrate) override;
 
-  virtual void SuspendChange(int video_channel, bool is_suspended) OVERRIDE;
+  void SuspendChange(int video_channel, bool is_suspended) override;
 
-  // From ViECaptureObserver.
-  virtual void BrightnessAlarm(const int capture_id,
-                               const Brightness brightness) OVERRIDE {}
-
-  virtual void CapturedFrameRate(const int capture_id,
-                                 const unsigned char frame_rate) OVERRIDE;
-
-  virtual void NoPictureAlarm(const int capture_id,
-                              const CaptureAlarm alarm) OVERRIDE {}
-
-  virtual void SendSideDelayUpdated(int avg_delay_ms,
-                                    int max_delay_ms,
-                                    uint32_t ssrc) OVERRIDE;
+  void SendSideDelayUpdated(int avg_delay_ms,
+                            int max_delay_ms,
+                            uint32_t ssrc) override;
 
  private:
-  SsrcStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  struct StatsUpdateTimes {
+    StatsUpdateTimes() : resolution_update_ms(0) {}
+    int64_t resolution_update_ms;
+  };
+  void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
+  Clock* const clock_;
   const VideoSendStream::Config config_;
-  scoped_ptr crit_;
+  rtc::scoped_ptr crit_;
   VideoSendStream::Stats stats_ GUARDED_BY(crit_);
+  rtc::RateTracker input_frame_rate_tracker_ GUARDED_BY(crit_);
+  std::map update_times_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc b/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc
index d7750f8aaf..0243add3b0 100644
--- a/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video/send_statistics_proxy_unittest.cc
@@ -21,13 +21,14 @@ namespace webrtc {
 
 class SendStatisticsProxyTest : public ::testing::Test {
  public:
-  SendStatisticsProxyTest() : avg_delay_ms_(0), max_delay_ms_(0) {}
+  SendStatisticsProxyTest()
+      : fake_clock_(1234), avg_delay_ms_(0), max_delay_ms_(0) {}
   virtual ~SendStatisticsProxyTest() {}
 
  protected:
   virtual void SetUp() {
     statistics_proxy_.reset(
-        new SendStatisticsProxy(GetTestConfig()));
+        new SendStatisticsProxy(&fake_clock_, GetTestConfig()));
     config_ = GetTestConfig();
     expected_ = VideoSendStream::Stats();
   }
@@ -48,29 +49,32 @@ class SendStatisticsProxyTest : public ::testing::Test {
     EXPECT_EQ(one.suspended, other.suspended);
 
     EXPECT_EQ(one.substreams.size(), other.substreams.size());
-    for (std::map::const_iterator it =
+    for (std::map::const_iterator it =
              one.substreams.begin();
-         it != one.substreams.end();
-         ++it) {
-      std::map::const_iterator corresponding_it =
-          other.substreams.find(it->first);
+         it != one.substreams.end(); ++it) {
+      std::map::const_iterator
+          corresponding_it = other.substreams.find(it->first);
       ASSERT_TRUE(corresponding_it != other.substreams.end());
-      const SsrcStats& a = it->second;
-      const SsrcStats& b = corresponding_it->second;
+      const VideoSendStream::StreamStats& a = it->second;
+      const VideoSendStream::StreamStats& b = corresponding_it->second;
 
-      EXPECT_EQ(a.key_frames, b.key_frames);
-      EXPECT_EQ(a.delta_frames, b.delta_frames);
+      EXPECT_EQ(a.frame_counts.key_frames, b.frame_counts.key_frames);
+      EXPECT_EQ(a.frame_counts.delta_frames, b.frame_counts.delta_frames);
       EXPECT_EQ(a.total_bitrate_bps, b.total_bitrate_bps);
       EXPECT_EQ(a.avg_delay_ms, b.avg_delay_ms);
       EXPECT_EQ(a.max_delay_ms, b.max_delay_ms);
 
-      EXPECT_EQ(a.rtp_stats.bytes, b.rtp_stats.bytes);
-      EXPECT_EQ(a.rtp_stats.header_bytes, b.rtp_stats.header_bytes);
-      EXPECT_EQ(a.rtp_stats.padding_bytes, b.rtp_stats.padding_bytes);
-      EXPECT_EQ(a.rtp_stats.packets, b.rtp_stats.packets);
-      EXPECT_EQ(a.rtp_stats.retransmitted_packets,
-                b.rtp_stats.retransmitted_packets);
-      EXPECT_EQ(a.rtp_stats.fec_packets, b.rtp_stats.fec_packets);
+      EXPECT_EQ(a.rtp_stats.transmitted.payload_bytes,
+                b.rtp_stats.transmitted.payload_bytes);
+      EXPECT_EQ(a.rtp_stats.transmitted.header_bytes,
+                b.rtp_stats.transmitted.header_bytes);
+      EXPECT_EQ(a.rtp_stats.transmitted.padding_bytes,
+                b.rtp_stats.transmitted.padding_bytes);
+      EXPECT_EQ(a.rtp_stats.transmitted.packets,
+                b.rtp_stats.transmitted.packets);
+      EXPECT_EQ(a.rtp_stats.retransmitted.packets,
+                b.rtp_stats.retransmitted.packets);
+      EXPECT_EQ(a.rtp_stats.fec.packets, b.rtp_stats.fec.packets);
 
       EXPECT_EQ(a.rtcp_stats.fraction_lost, b.rtcp_stats.fraction_lost);
       EXPECT_EQ(a.rtcp_stats.cumulative_lost, b.rtcp_stats.cumulative_lost);
@@ -80,12 +84,14 @@ class SendStatisticsProxyTest : public ::testing::Test {
     }
   }
 
-  scoped_ptr statistics_proxy_;
+  rtc::scoped_ptr statistics_proxy_;
+  SimulatedClock fake_clock_;
   VideoSendStream::Config config_;
   int avg_delay_ms_;
   int max_delay_ms_;
   VideoSendStream::Stats expected_;
-  typedef std::map::const_iterator StreamIterator;
+  typedef std::map::const_iterator
+      StreamIterator;
 };
 
 TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
@@ -94,7 +100,7 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
        it != config_.rtp.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
-    SsrcStats& ssrc_stats = expected_.substreams[ssrc];
+    VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc];
 
     // Add statistics with some arbitrary, but unique, numbers.
     uint32_t offset = ssrc * sizeof(RtcpStatistics);
@@ -108,7 +114,7 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
        it != config_.rtp.rtx.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
-    SsrcStats& ssrc_stats = expected_.substreams[ssrc];
+    VideoSendStream::StreamStats& ssrc_stats = expected_.substreams[ssrc];
 
     // Add statistics with some arbitrary, but unique, numbers.
     uint32_t offset = ssrc * sizeof(RtcpStatistics);
@@ -122,16 +128,6 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) {
   ExpectEqual(expected_, stats);
 }
 
-TEST_F(SendStatisticsProxyTest, CaptureFramerate) {
-  const int capture_fps = 31;
-
-  ViECaptureObserver* capture_observer = statistics_proxy_.get();
-  capture_observer->CapturedFrameRate(0, capture_fps);
-
-  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
-  EXPECT_EQ(capture_fps, stats.input_frame_rate);
-}
-
 TEST_F(SendStatisticsProxyTest, EncodedBitrateAndFramerate) {
   const int media_bitrate_bps = 500;
   const int encode_fps = 29;
@@ -165,24 +161,26 @@ TEST_F(SendStatisticsProxyTest, FrameCounts) {
        ++it) {
     const uint32_t ssrc = *it;
     // Add statistics with some arbitrary, but unique, numbers.
-    SsrcStats& stats = expected_.substreams[ssrc];
-    uint32_t offset = ssrc * sizeof(SsrcStats);
-    stats.key_frames = offset;
-    stats.delta_frames = offset + 1;
-    observer->FrameCountUpdated(kVideoFrameKey, stats.key_frames, ssrc);
-    observer->FrameCountUpdated(kVideoFrameDelta, stats.delta_frames, ssrc);
+    VideoSendStream::StreamStats& stats = expected_.substreams[ssrc];
+    uint32_t offset = ssrc * sizeof(VideoSendStream::StreamStats);
+    FrameCounts frame_counts;
+    frame_counts.key_frames = offset;
+    frame_counts.delta_frames = offset + 1;
+    stats.frame_counts = frame_counts;
+    observer->FrameCountUpdated(frame_counts, ssrc);
   }
   for (std::vector::const_iterator it = config_.rtp.rtx.ssrcs.begin();
        it != config_.rtp.rtx.ssrcs.end();
        ++it) {
     const uint32_t ssrc = *it;
     // Add statistics with some arbitrary, but unique, numbers.
-    SsrcStats& stats = expected_.substreams[ssrc];
-    uint32_t offset = ssrc * sizeof(SsrcStats);
-    stats.key_frames = offset;
-    stats.delta_frames = offset + 1;
-    observer->FrameCountUpdated(kVideoFrameKey, stats.key_frames, ssrc);
-    observer->FrameCountUpdated(kVideoFrameDelta, stats.delta_frames, ssrc);
+    VideoSendStream::StreamStats& stats = expected_.substreams[ssrc];
+    uint32_t offset = ssrc * sizeof(VideoSendStream::StreamStats);
+    FrameCounts frame_counts;
+    frame_counts.key_frames = offset;
+    frame_counts.delta_frames = offset + 1;
+    stats.frame_counts = frame_counts;
+    observer->FrameCountUpdated(frame_counts, ssrc);
   }
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
@@ -197,13 +195,14 @@ TEST_F(SendStatisticsProxyTest, DataCounters) {
     const uint32_t ssrc = *it;
     StreamDataCounters& counters = expected_.substreams[ssrc].rtp_stats;
     // Add statistics with some arbitrary, but unique, numbers.
-    uint32_t offset = ssrc * sizeof(StreamDataCounters);
-    counters.bytes = offset;
-    counters.header_bytes = offset + 1;
-    counters.fec_packets = offset + 2;
-    counters.padding_bytes = offset + 3;
-    counters.retransmitted_packets = offset + 4;
-    counters.packets = offset + 5;
+    size_t offset = ssrc * sizeof(StreamDataCounters);
+    uint32_t offset_uint32 = static_cast(offset);
+    counters.transmitted.payload_bytes = offset;
+    counters.transmitted.header_bytes = offset + 1;
+    counters.fec.packets = offset_uint32 + 2;
+    counters.transmitted.padding_bytes = offset + 3;
+    counters.retransmitted.packets = offset_uint32 + 4;
+    counters.transmitted.packets = offset_uint32 + 5;
     callback->DataCountersUpdated(counters, ssrc);
   }
   for (std::vector::const_iterator it = config_.rtp.rtx.ssrcs.begin();
@@ -212,13 +211,14 @@ TEST_F(SendStatisticsProxyTest, DataCounters) {
     const uint32_t ssrc = *it;
     StreamDataCounters& counters = expected_.substreams[ssrc].rtp_stats;
     // Add statistics with some arbitrary, but unique, numbers.
-    uint32_t offset = ssrc * sizeof(StreamDataCounters);
-    counters.bytes = offset;
-    counters.header_bytes = offset + 1;
-    counters.fec_packets = offset + 2;
-    counters.padding_bytes = offset + 3;
-    counters.retransmitted_packets = offset + 4;
-    counters.packets = offset + 5;
+    size_t offset = ssrc * sizeof(StreamDataCounters);
+    uint32_t offset_uint32 = static_cast(offset);
+    counters.transmitted.payload_bytes = offset;
+    counters.transmitted.header_bytes = offset + 1;
+    counters.fec.packets = offset_uint32 + 2;
+    counters.transmitted.padding_bytes = offset + 3;
+    counters.retransmitted.packets = offset_uint32 + 4;
+    counters.transmitted.packets = offset_uint32 + 5;
     callback->DataCountersUpdated(counters, ssrc);
   }
 
@@ -290,7 +290,7 @@ TEST_F(SendStatisticsProxyTest, SendSideDelay) {
 }
 
 TEST_F(SendStatisticsProxyTest, NoSubstreams) {
-  uint32_t exluded_ssrc =
+  uint32_t excluded_ssrc =
       std::max(
           *std::max_element(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end()),
           *std::max_element(config_.rtp.rtx.ssrcs.begin(),
@@ -299,25 +299,69 @@ TEST_F(SendStatisticsProxyTest, NoSubstreams) {
   // From RtcpStatisticsCallback.
   RtcpStatistics rtcp_stats;
   RtcpStatisticsCallback* rtcp_callback = statistics_proxy_.get();
-  rtcp_callback->StatisticsUpdated(rtcp_stats, exluded_ssrc);
-
-  // From StreamDataCountersCallback.
-  StreamDataCounters rtp_stats;
-  StreamDataCountersCallback* rtp_callback = statistics_proxy_.get();
-  rtp_callback->DataCountersUpdated(rtp_stats, exluded_ssrc);
+  rtcp_callback->StatisticsUpdated(rtcp_stats, excluded_ssrc);
 
   // From BitrateStatisticsObserver.
   BitrateStatistics total;
   BitrateStatistics retransmit;
   BitrateStatisticsObserver* bitrate_observer = statistics_proxy_.get();
-  bitrate_observer->Notify(total, retransmit, exluded_ssrc);
+  bitrate_observer->Notify(total, retransmit, excluded_ssrc);
 
   // From FrameCountObserver.
   FrameCountObserver* fps_observer = statistics_proxy_.get();
-  fps_observer->FrameCountUpdated(kVideoFrameKey, 1, exluded_ssrc);
+  FrameCounts frame_counts;
+  frame_counts.key_frames = 1;
+  fps_observer->FrameCountUpdated(frame_counts, excluded_ssrc);
 
   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
   EXPECT_TRUE(stats.substreams.empty());
 }
 
+TEST_F(SendStatisticsProxyTest, EncodedResolutionTimesOut) {
+  static const int kEncodedWidth = 123;
+  static const int kEncodedHeight = 81;
+  EncodedImage encoded_image;
+  encoded_image._encodedWidth = kEncodedWidth;
+  encoded_image._encodedHeight = kEncodedHeight;
+
+  RTPVideoHeader rtp_video_header;
+
+  rtp_video_header.simulcastIdx = 0;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+  rtp_video_header.simulcastIdx = 1;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+
+  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].width);
+  EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].height);
+  EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[1]].width);
+  EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].height);
+
+  // Forward almost to timeout, this should not have removed stats.
+  fake_clock_.AdvanceTimeMilliseconds(SendStatisticsProxy::kStatsTimeoutMs - 1);
+  stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[0]].width);
+  EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[0]].height);
+
+  // Update the first SSRC with bogus RTCP stats to make sure that encoded
+  // resolution still times out (no global timeout for all stats).
+  RtcpStatistics rtcp_statistics;
+  RtcpStatisticsCallback* rtcp_stats = statistics_proxy_.get();
+  rtcp_stats->StatisticsUpdated(rtcp_statistics, config_.rtp.ssrcs[0]);
+
+  // Report stats for second SSRC to make sure it's not outdated along with the
+  // first SSRC.
+  rtp_video_header.simulcastIdx = 1;
+  statistics_proxy_->OnSendEncodedImage(encoded_image, &rtp_video_header);
+
+  // Forward 1 ms, reach timeout, substream 0 should have no resolution
+  // reported, but substream 1 should.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  stats = statistics_proxy_->GetStats();
+  EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[0]].width);
+  EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[0]].height);
+  EXPECT_EQ(kEncodedWidth, stats.substreams[config_.rtp.ssrcs[1]].width);
+  EXPECT_EQ(kEncodedHeight, stats.substreams[config_.rtp.ssrcs[1]].height);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/transport_adapter.cc b/media/webrtc/trunk/webrtc/video/transport_adapter.cc
index 6f27d9972a..7b5a6962c4 100644
--- a/media/webrtc/trunk/webrtc/video/transport_adapter.cc
+++ b/media/webrtc/trunk/webrtc/video/transport_adapter.cc
@@ -18,24 +18,24 @@ TransportAdapter::TransportAdapter(newapi::Transport* transport)
 
 int TransportAdapter::SendPacket(int /*channel*/,
                                  const void* packet,
-                                 int length) {
+                                 size_t length) {
   if (enabled_.Value() == 0)
     return false;
 
   bool success = transport_->SendRtp(static_cast(packet),
-                                     static_cast(length));
-  return success ? length : -1;
+                                     length);
+  return success ? static_cast(length) : -1;
 }
 
 int TransportAdapter::SendRTCPPacket(int /*channel*/,
                                      const void* packet,
-                                     int length) {
+                                     size_t length) {
   if (enabled_.Value() == 0)
     return false;
 
   bool success = transport_->SendRtcp(static_cast(packet),
-                                      static_cast(length));
-  return success ? length : -1;
+                                      length);
+  return success ? static_cast(length) : -1;
 }
 
 void TransportAdapter::Enable() {
diff --git a/media/webrtc/trunk/webrtc/video/transport_adapter.h b/media/webrtc/trunk/webrtc/video/transport_adapter.h
index a9a72e1672..cd27d7cfe0 100644
--- a/media/webrtc/trunk/webrtc/video/transport_adapter.h
+++ b/media/webrtc/trunk/webrtc/video/transport_adapter.h
@@ -21,12 +21,10 @@ class TransportAdapter : public webrtc::Transport {
  public:
   explicit TransportAdapter(newapi::Transport* transport);
 
-  virtual int SendPacket(int /*channel*/,
-                         const void* packet,
-                         int length) OVERRIDE;
-  virtual int SendRTCPPacket(int /*channel*/,
-                             const void* packet,
-                             int length) OVERRIDE;
+  int SendPacket(int /*channel*/, const void* packet, size_t length) override;
+  int SendRTCPPacket(int /*channel*/,
+                     const void* packet,
+                     size_t length) override;
 
   void Enable();
   void Disable();
diff --git a/media/webrtc/trunk/webrtc/video/video_loopback.cc b/media/webrtc/trunk/webrtc/video/video_loopback.cc
new file mode 100644
index 0000000000..d183920b16
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video/video_loopback.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include 
+
+#include 
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/test/field_trial.h"
+#include "webrtc/test/run_test.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video/loopback.h"
+
+namespace webrtc {
+
+namespace flags {
+
+DEFINE_int32(width, 640, "Video width.");
+size_t Width() {
+  return static_cast(FLAGS_width);
+}
+
+DEFINE_int32(height, 480, "Video height.");
+size_t Height() {
+  return static_cast(FLAGS_height);
+}
+
+DEFINE_int32(fps, 30, "Frames per second.");
+int Fps() {
+  return static_cast(FLAGS_fps);
+}
+
+DEFINE_int32(min_bitrate, 50, "Minimum video bitrate.");
+size_t MinBitrate() {
+  return static_cast(FLAGS_min_bitrate);
+}
+
+DEFINE_int32(start_bitrate, 300, "Video starting bitrate.");
+size_t StartBitrate() {
+  return static_cast(FLAGS_start_bitrate);
+}
+
+DEFINE_int32(max_bitrate, 800, "Maximum video bitrate.");
+size_t MaxBitrate() {
+  return static_cast(FLAGS_max_bitrate);
+}
+
+int MinTransmitBitrate() {
+  return 0;
+}  // No min padding for regular video.
+
+DEFINE_string(codec, "VP8", "Video codec to use.");
+std::string Codec() {
+  return static_cast(FLAGS_codec);
+}
+
+DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
+int LossPercent() {
+  return static_cast(FLAGS_loss_percent);
+}
+
+DEFINE_int32(link_capacity,
+             0,
+             "Capacity (kbps) of the fake link. 0 means infinite.");
+int LinkCapacity() {
+  return static_cast(FLAGS_link_capacity);
+}
+
+DEFINE_int32(queue_size, 0, "Size of the bottleneck link queue in packets.");
+int QueueSize() {
+  return static_cast(FLAGS_queue_size);
+}
+
+DEFINE_int32(avg_propagation_delay_ms,
+             0,
+             "Average link propagation delay in ms.");
+int AvgPropagationDelayMs() {
+  return static_cast(FLAGS_avg_propagation_delay_ms);
+}
+
+DEFINE_int32(std_propagation_delay_ms,
+             0,
+             "Link propagation delay standard deviation in ms.");
+int StdPropagationDelayMs() {
+  return static_cast(FLAGS_std_propagation_delay_ms);
+}
+
+DEFINE_bool(logs, false, "print logs to stderr");
+
+DEFINE_string(
+    force_fieldtrials,
+    "",
+    "Field trials control experimental feature code which can be forced. "
+    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+    " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
+    "trials are separated by \"/\"");
+}  // namespace flags
+
+void Loopback() {
+  test::Loopback::Config config{flags::Width(),
+                                flags::Height(),
+                                flags::Fps(),
+                                flags::MinBitrate(),
+                                flags::StartBitrate(),
+                                flags::MaxBitrate(),
+                                0,  // No min transmit bitrate.
+                                flags::Codec(),
+                                flags::LossPercent(),
+                                flags::LinkCapacity(),
+                                flags::QueueSize(),
+                                flags::AvgPropagationDelayMs(),
+                                flags::StdPropagationDelayMs(),
+                                flags::FLAGS_logs};
+  test::Loopback loopback(config);
+  loopback.Run();
+}
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  ::testing::InitGoogleTest(&argc, argv);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::test::InitFieldTrialsFromString(
+      webrtc::flags::FLAGS_force_fieldtrials);
+  webrtc::test::RunTest(webrtc::Loopback);
+  return 0;
+}
diff --git a/media/webrtc/trunk/webrtc/video/video_receive_stream.cc b/media/webrtc/trunk/webrtc/video/video_receive_stream.cc
index 5b085bb856..126d485e58 100644
--- a/media/webrtc/trunk/webrtc/video/video_receive_stream.cc
+++ b/media/webrtc/trunk/webrtc/video/video_receive_stream.cc
@@ -10,11 +10,11 @@
 
 #include "webrtc/video/video_receive_stream.h"
 
-#include 
 #include 
 
 #include 
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/logging.h"
@@ -31,8 +31,77 @@
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
+std::string VideoReceiveStream::Decoder::ToString() const {
+  std::stringstream ss;
+  ss << "{decoder: " << (decoder != nullptr ? "(VideoDecoder)" : "nullptr");
+  ss << ", payload_type: " << payload_type;
+  ss << ", payload_name: " << payload_name;
+  ss << ", is_renderer: " << (is_renderer ? "yes" : "no");
+  ss << ", expected_delay_ms: " << expected_delay_ms;
+  ss << '}';
+
+  return ss.str();
+}
+
+std::string VideoReceiveStream::Config::ToString() const {
+  std::stringstream ss;
+  ss << "{decoders: [";
+  for (size_t i = 0; i < decoders.size(); ++i) {
+    ss << decoders[i].ToString();
+    if (i != decoders.size() - 1)
+      ss << ", ";
+  }
+  ss << ']';
+  ss << ", rtp: " << rtp.ToString();
+  ss << ", renderer: " << (renderer != nullptr ? "(renderer)" : "nullptr");
+  ss << ", render_delay_ms: " << render_delay_ms;
+  ss << ", audio_channel_id: " << audio_channel_id;
+  ss << ", pre_decode_callback: "
+     << (pre_decode_callback != nullptr ? "(EncodedFrameObserver)" : "nullptr");
+  ss << ", pre_render_callback: "
+     << (pre_render_callback != nullptr ? "(I420FrameCallback)" : "nullptr");
+  ss << ", target_delay_ms: " << target_delay_ms;
+  ss << '}';
+
+  return ss.str();
+}
+
+std::string VideoReceiveStream::Config::Rtp::ToString() const {
+  std::stringstream ss;
+  ss << "{remote_ssrc: " << remote_ssrc;
+  ss << ", local_ssrc: " << local_ssrc;
+  ss << ", rtcp_mode: " << (rtcp_mode == newapi::kRtcpCompound
+                                ? "kRtcpCompound"
+                                : "kRtcpReducedSize");
+  ss << ", rtcp_xr: ";
+  ss << "{receiver_reference_time_report: "
+     << (rtcp_xr.receiver_reference_time_report ? "on" : "off");
+  ss << '}';
+  ss << ", remb: " << (remb ? "on" : "off");
+  ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
+  ss << ", fec: " << fec.ToString();
+  ss << ", rtx: {";
+  for (auto& kv : rtx) {
+    ss << kv.first << " -> ";
+    ss << "{ssrc: " << kv.second.ssrc;
+    ss << ", payload_type: " << kv.second.payload_type;
+    ss << '}';
+  }
+  ss << '}';
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1)
+      ss << ", ";
+  }
+  ss << ']';
+  ss << '}';
+  return ss.str();
+}
+
 namespace internal {
 namespace {
+
 VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
   VideoCodec codec;
   memset(&codec, 0, sizeof(codec));
@@ -74,27 +143,27 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
       channel_(-1) {
   video_engine_base_ = ViEBase::GetInterface(video_engine);
   video_engine_base_->CreateReceiveChannel(channel_, base_channel);
-  assert(channel_ != -1);
+  DCHECK(channel_ != -1);
 
   rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
-  assert(rtp_rtcp_ != NULL);
+  DCHECK(rtp_rtcp_ != nullptr);
 
   // TODO(pbos): This is not fine grained enough...
   rtp_rtcp_->SetNACKStatus(channel_, config_.rtp.nack.rtp_history_ms > 0);
   rtp_rtcp_->SetKeyFrameRequestMethod(channel_, kViEKeyFrameRequestPliRtcp);
   SetRtcpMode(config_.rtp.rtcp_mode);
 
-  assert(config_.rtp.remote_ssrc != 0);
+  DCHECK(config_.rtp.remote_ssrc != 0);
   // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
-  assert(config_.rtp.local_ssrc != 0);
-  assert(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
+  DCHECK(config_.rtp.local_ssrc != 0);
+  DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
 
   rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.local_ssrc);
   // TODO(pbos): Support multiple RTX, per video payload.
   Config::Rtp::RtxMap::const_iterator it = config_.rtp.rtx.begin();
   if (it != config_.rtp.rtx.end()) {
-    assert(it->second.ssrc != 0);
-    assert(it->second.payload_type != 0);
+    DCHECK(it->second.ssrc != 0);
+    DCHECK(it->second.payload_type != 0);
 
     rtp_rtcp_->SetRemoteSSRCType(channel_, kViEStreamTypeRtx, it->second.ssrc);
     rtp_rtcp_->SetRtxReceivePayloadType(channel_, it->second.payload_type);
@@ -105,19 +174,24 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
+    // One-byte-extension local identifiers are in the range 1-14 inclusive.
+    DCHECK_GE(id, 1);
+    DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
-      if (rtp_rtcp_->SetReceiveTimestampOffsetStatus(channel_, true, id) != 0)
-        abort();
+      CHECK_EQ(0,
+               rtp_rtcp_->SetReceiveTimestampOffsetStatus(channel_, true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
-      if (rtp_rtcp_->SetReceiveAbsoluteSendTimeStatus(channel_, true, id) != 0)
-        abort();
+      CHECK_EQ(0,
+               rtp_rtcp_->SetReceiveAbsoluteSendTimeStatus(channel_, true, id));
+    } else if (extension == RtpExtension::kVideoRotation) {
+      CHECK_EQ(0, rtp_rtcp_->SetReceiveVideoRotationStatus(channel_, true, id));
     } else {
-      abort();  // Unsupported extension.
+      RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 
   network_ = ViENetwork::GetInterface(video_engine);
-  assert(network_ != NULL);
+  DCHECK(network_ != nullptr);
 
   network_->RegisterSendTransport(channel_, transport_adapter_);
 
@@ -125,16 +199,13 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
 
   if (config_.rtp.fec.ulpfec_payload_type != -1) {
     // ULPFEC without RED doesn't make sense.
-    assert(config_.rtp.fec.red_payload_type != -1);
+    DCHECK(config_.rtp.fec.red_payload_type != -1);
     VideoCodec codec;
     memset(&codec, 0, sizeof(codec));
     codec.codecType = kVideoCodecULPFEC;
     strcpy(codec.plName, "ulpfec");
     codec.plType = config_.rtp.fec.ulpfec_payload_type;
-    if (codec_->SetReceiveCodec(channel_, codec) != 0) {
-      LOG(LS_ERROR) << "Could not set ULPFEC codec. This shouldn't happen.";
-      abort();
-    }
+    CHECK_EQ(0, codec_->SetReceiveCodec(channel_, codec));
   }
   if (config_.rtp.fec.red_payload_type != -1) {
     VideoCodec codec;
@@ -142,53 +213,40 @@ VideoReceiveStream::VideoReceiveStream(webrtc::VideoEngine* video_engine,
     codec.codecType = kVideoCodecRED;
     strcpy(codec.plName, "red");
     codec.plType = config_.rtp.fec.red_payload_type;
-    if (codec_->SetReceiveCodec(channel_, codec) != 0) {
-      LOG(LS_ERROR) << "Could not set RED codec. This shouldn't happen.";
-      abort();
-    }
+    CHECK_EQ(0, codec_->SetReceiveCodec(channel_, codec));
   }
 
-  stats_proxy_.reset(new ReceiveStatisticsProxy(
-      config_.rtp.local_ssrc, clock_, rtp_rtcp_, codec_, channel_));
+  stats_proxy_.reset(
+      new ReceiveStatisticsProxy(config_.rtp.remote_ssrc, clock_));
 
-  if (rtp_rtcp_->RegisterReceiveChannelRtcpStatisticsCallback(
-          channel_, stats_proxy_.get()) != 0)
-    abort();
+  CHECK_EQ(0, rtp_rtcp_->RegisterReceiveChannelRtcpStatisticsCallback(
+                  channel_, stats_proxy_.get()));
+  CHECK_EQ(0, rtp_rtcp_->RegisterReceiveChannelRtpStatisticsCallback(
+                  channel_, stats_proxy_.get()));
+  CHECK_EQ(0, rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(
+                  channel_, stats_proxy_.get()));
+  CHECK_EQ(0, codec_->RegisterDecoderObserver(channel_, *stats_proxy_));
 
-  if (rtp_rtcp_->RegisterReceiveChannelRtpStatisticsCallback(
-          channel_, stats_proxy_.get()) != 0)
-    abort();
-
-  if (codec_->RegisterDecoderObserver(channel_, *stats_proxy_) != 0)
-    abort();
+  video_engine_base_->RegisterReceiveStatisticsProxy(channel_,
+                                                     stats_proxy_.get());
 
   external_codec_ = ViEExternalCodec::GetInterface(video_engine);
-  assert(!config_.decoders.empty());
+  DCHECK(!config_.decoders.empty());
   for (size_t i = 0; i < config_.decoders.size(); ++i) {
     const Decoder& decoder = config_.decoders[i];
-    if (external_codec_->RegisterExternalReceiveCodec(
-            channel_,
-            decoder.payload_type,
-            decoder.decoder,
-            decoder.renderer,
-            decoder.expected_delay_ms) != 0) {
-      // TODO(pbos): Abort gracefully? Can this be a runtime error?
-      abort();
-    }
+    CHECK_EQ(0, external_codec_->RegisterExternalReceiveCodec(
+                    channel_, decoder.payload_type, decoder.decoder,
+                    decoder.is_renderer, decoder.expected_delay_ms));
 
     VideoCodec codec = CreateDecoderVideoCodec(decoder);
 
-    if (codec_->SetReceiveCodec(channel_, codec) != 0) {
-      // TODO(pbos): Abort gracefully, this can be a runtime error.
-      //             Factor out to an Init() method.
-      abort();
-    }
+    CHECK_EQ(0, codec_->SetReceiveCodec(channel_, codec));
   }
 
   render_ = ViERender::GetInterface(video_engine);
-  assert(render_ != NULL);
+  DCHECK(render_ != nullptr);
 
-  render_->AddRenderCallback(channel_, this);
+  render_->AddRenderer(channel_, kVideoI420, this);
 
   if (voice_engine && config_.audio_channel_id != -1) {
     video_engine_base_->SetVoiceEngine(voice_engine);
@@ -220,7 +278,7 @@ VideoReceiveStream::~VideoReceiveStream() {
 
   network_->DeregisterSendTransport(channel_);
 
-  video_engine_base_->SetVoiceEngine(NULL);
+  video_engine_base_->SetVoiceEngine(nullptr);
   image_process_->Release();
   external_codec_->Release();
   codec_->DeregisterDecoderObserver(channel_);
@@ -228,6 +286,7 @@ VideoReceiveStream::~VideoReceiveStream() {
                                                            stats_proxy_.get());
   rtp_rtcp_->DeregisterReceiveChannelRtcpStatisticsCallback(channel_,
                                                             stats_proxy_.get());
+  rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(channel_, nullptr);
   codec_->Release();
   network_->Release();
   render_->Release();
@@ -238,17 +297,13 @@ VideoReceiveStream::~VideoReceiveStream() {
 
 void VideoReceiveStream::Start() {
   transport_adapter_.Enable();
-  if (render_->StartRender(channel_) != 0)
-    abort();
-  if (video_engine_base_->StartReceive(channel_) != 0)
-    abort();
+  CHECK_EQ(0, render_->StartRender(channel_));
+  CHECK_EQ(0, video_engine_base_->StartReceive(channel_));
 }
 
 void VideoReceiveStream::Stop() {
-  if (render_->StopRender(channel_) != 0)
-    abort();
-  if (video_engine_base_->StopReceive(channel_) != 0)
-    abort();
+  CHECK_EQ(0, render_->StopRender(channel_));
+  CHECK_EQ(0, video_engine_base_->StopReceive(channel_));
   transport_adapter_.Disable();
 }
 
@@ -257,13 +312,12 @@ VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
 }
 
 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return network_->ReceivedRTCPPacket(
-             channel_, packet, static_cast(length)) == 0;
+  return network_->ReceivedRTCPPacket(channel_, packet, length) == 0;
 }
 
 bool VideoReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) {
-  return network_->ReceivedRTPPacket(
-             channel_, packet, static_cast(length), PacketTime()) == 0;
+  return network_->ReceivedRTPPacket(channel_, packet, length, PacketTime()) ==
+      0;
 }
 
 void VideoReceiveStream::FrameCallback(I420VideoFrame* video_frame) {
@@ -273,9 +327,25 @@ void VideoReceiveStream::FrameCallback(I420VideoFrame* video_frame) {
     config_.pre_render_callback->FrameCallback(video_frame);
 }
 
-int32_t VideoReceiveStream::RenderFrame(const uint32_t stream_id,
-                                        I420VideoFrame& video_frame) {
-  if (config_.renderer != NULL)
+int VideoReceiveStream::FrameSizeChange(unsigned int width,
+                                        unsigned int height,
+                                        unsigned int number_of_streams) {
+  return 0;
+}
+
+int VideoReceiveStream::DeliverFrame(unsigned char* buffer,
+                                     size_t buffer_size,
+                                     uint32_t timestamp,
+                                     int64_t ntp_time_ms,
+                                     int64_t render_time_ms,
+                                     void* handle) {
+  CHECK(false) << "Renderer should be configured as kVideoI420 and never "
+                  "receive callbacks on DeliverFrame.";
+  return 0;
+}
+
+int VideoReceiveStream::DeliverI420Frame(const I420VideoFrame& video_frame) {
+  if (config_.renderer != nullptr)
     config_.renderer->RenderFrame(
         video_frame,
         video_frame.render_time_ms() - clock_->TimeInMilliseconds());
@@ -285,6 +355,12 @@ int32_t VideoReceiveStream::RenderFrame(const uint32_t stream_id,
   return 0;
 }
 
+bool VideoReceiveStream::IsTextureSupported() {
+  if (config_.renderer == nullptr)
+    return false;
+  return config_.renderer->IsTextureSupported();
+}
+
 void VideoReceiveStream::SignalNetworkState(Call::NetworkState state) {
   if (state == Call::kNetworkUp)
     SetRtcpMode(config_.rtp.rtcp_mode);
diff --git a/media/webrtc/trunk/webrtc/video/video_receive_stream.h b/media/webrtc/trunk/webrtc/video/video_receive_stream.h
index 2aa39e2389..dcc8122ee4 100644
--- a/media/webrtc/trunk/webrtc/video/video_receive_stream.h
+++ b/media/webrtc/trunk/webrtc/video/video_receive_stream.h
@@ -13,11 +13,11 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/video_render/include/video_render_defines.h"
 #include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/video/encoded_frame_callback_adapter.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video/transport_adapter.h"
@@ -40,7 +40,7 @@ namespace internal {
 
 class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public I420FrameCallback,
-                           public VideoRenderCallback {
+                           public ExternalRenderer {
  public:
   VideoReceiveStream(webrtc::VideoEngine* video_engine,
                      const VideoReceiveStream::Config& config,
@@ -49,16 +49,25 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
                      int base_channel);
   virtual ~VideoReceiveStream();
 
-  virtual void Start() OVERRIDE;
-  virtual void Stop() OVERRIDE;
-  virtual Stats GetStats() const OVERRIDE;
+  void Start() override;
+  void Stop() override;
+  Stats GetStats() const override;
 
   // Overrides I420FrameCallback.
-  virtual void FrameCallback(I420VideoFrame* video_frame) OVERRIDE;
+  void FrameCallback(I420VideoFrame* video_frame) override;
 
-  // Overrides VideoRenderCallback.
-  virtual int32_t RenderFrame(const uint32_t stream_id,
-                              I420VideoFrame& video_frame) OVERRIDE;
+  // Overrides ExternalRenderer.
+  int FrameSizeChange(unsigned int width,
+                      unsigned int height,
+                      unsigned int number_of_streams) override;
+  int DeliverFrame(unsigned char* buffer,
+                   size_t buffer_size,
+                   uint32_t timestamp,
+                   int64_t ntp_time_ms,
+                   int64_t render_time_ms,
+                   void* handle) override;
+  int DeliverI420Frame(const I420VideoFrame& webrtc_frame) override;
+  bool IsTextureSupported() override;
 
   void SignalNetworkState(Call::NetworkState state);
 
@@ -81,7 +90,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
   ViERTP_RTCP* rtp_rtcp_;
   ViEImageProcess* image_process_;
 
-  scoped_ptr stats_proxy_;
+  rtc::scoped_ptr stats_proxy_;
 
   int channel_;
 };
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream.cc b/media/webrtc/trunk/webrtc/video/video_send_stream.cc
index 489cd14081..9949f5017e 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream.cc
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream.cc
@@ -15,8 +15,10 @@
 #include 
 #include 
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/include/vie_capture.h"
 #include "webrtc/video_engine/include/vie_codec.h"
@@ -33,8 +35,7 @@ VideoSendStream::Config::EncoderSettings::ToString() const {
   std::stringstream ss;
   ss << "{payload_name: " << payload_name;
   ss << ", payload_type: " << payload_type;
-  if (encoder != NULL)
-    ss << ", encoder: " << (encoder != NULL ? "(encoder)" : "NULL");
+  ss << ", encoder: " << (encoder != nullptr ? "(VideoEncoder)" : "nullptr");
   ss << '}';
   return ss.str();
 }
@@ -42,13 +43,13 @@ VideoSendStream::Config::EncoderSettings::ToString() const {
 std::string VideoSendStream::Config::Rtp::Rtx::ToString()
     const {
   std::stringstream ss;
-  ss << "{ssrcs: {";
+  ss << "{ssrcs: [";
   for (size_t i = 0; i < ssrcs.size(); ++i) {
     ss << ssrcs[i];
     if (i != ssrcs.size() - 1)
-      ss << "}, {";
+      ss << ", ";
   }
-  ss << '}';
+  ss << ']';
 
   ss << ", payload_type: " << payload_type;
   ss << '}';
@@ -57,32 +58,26 @@ std::string VideoSendStream::Config::Rtp::Rtx::ToString()
 
 std::string VideoSendStream::Config::Rtp::ToString() const {
   std::stringstream ss;
-  ss << "{ssrcs: {";
+  ss << "{ssrcs: [";
   for (size_t i = 0; i < ssrcs.size(); ++i) {
     ss << ssrcs[i];
     if (i != ssrcs.size() - 1)
-      ss << "}, {";
+      ss << ", ";
   }
-  ss << '}';
-
+  ss << ']';
   ss << ", max_packet_size: " << max_packet_size;
-
-  ss << ", extensions: {";
+  ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1)
-      ss << "}, {";
+      ss << ", ";
   }
-  ss << '}';
+  ss << ']';
 
-  if (nack.rtp_history_ms != 0)
-    ss << ", nack.rtp_history_ms: " << nack.rtp_history_ms;
-  if (fec.ulpfec_payload_type != -1 || fec.red_payload_type != -1)
-    ss << ", fec: " << fec.ToString();
-  if (rtx.payload_type != 0 || !rtx.ssrcs.empty())
-    ss << ", rtx: " << rtx.ToString();
-  if (c_name != "")
-    ss << ", c_name: " << c_name;
+  ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
+  ss << ", fec: " << fec.ToString();
+  ss << ", rtx: " << rtx.ToString();
+  ss << ", c_name: " << c_name;
   ss << '}';
   return ss.str();
 }
@@ -91,17 +86,17 @@ std::string VideoSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{encoder_settings: " << encoder_settings.ToString();
   ss << ", rtp: " << rtp.ToString();
-  if (pre_encode_callback != NULL)
-    ss << ", (pre_encode_callback)";
-  if (post_encode_callback != NULL)
-    ss << ", (post_encode_callback)";
-  if (local_renderer != NULL) {
-    ss << ", (local_renderer, render_delay_ms: " << render_delay_ms << ")";
-  }
-  if (target_delay_ms > 0)
-    ss << ", target_delay_ms: " << target_delay_ms;
-  if (suspend_below_min_bitrate)
-    ss << ", suspend_below_min_bitrate: on";
+  ss << ", pre_encode_callback: "
+     << (pre_encode_callback != nullptr ? "(I420FrameCallback)" : "nullptr");
+  ss << ", post_encode_callback: " << (post_encode_callback != nullptr
+                                           ? "(EncodedFrameObserver)"
+                                           : "nullptr");
+  ss << "local_renderer: " << (local_renderer != nullptr ? "(VideoRenderer)"
+                                                         : "nullptr");
+  ss << ", render_delay_ms: " << render_delay_ms;
+  ss << ", target_delay_ms: " << target_delay_ms;
+  ss << ", suspend_below_min_bitrate: " << (suspend_below_min_bitrate ? "on"
+                                                                      : "off");
   ss << '}';
   return ss.str();
 }
@@ -114,38 +109,39 @@ VideoSendStream::VideoSendStream(
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
     const std::map& suspended_ssrcs,
-    int base_channel,
-    int start_bitrate_bps)
+    int base_channel)
     : transport_adapter_(transport),
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
-      start_bitrate_bps_(start_bitrate_bps),
       suspended_ssrcs_(suspended_ssrcs),
-      external_codec_(NULL),
+      external_codec_(nullptr),
       channel_(-1),
-      use_default_bitrate_(true),
-      stats_proxy_(config) {
+      use_config_bitrate_(true),
+      stats_proxy_(Clock::GetRealTimeClock(), config) {
   video_engine_base_ = ViEBase::GetInterface(video_engine);
-  video_engine_base_->CreateChannel(channel_, base_channel);
-  assert(channel_ != -1);
-  assert(start_bitrate_bps_ > 0);
+  video_engine_base_->CreateChannelWithoutDefaultEncoder(channel_,
+                                                         base_channel);
+  DCHECK(channel_ != -1);
 
   rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine);
-  assert(rtp_rtcp_ != NULL);
+  DCHECK(rtp_rtcp_ != nullptr);
 
-  assert(config_.rtp.ssrcs.size() > 0);
+  DCHECK(!config_.rtp.ssrcs.empty());
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
+    // One-byte-extension local identifiers are in the range 1-14 inclusive.
+    DCHECK_GE(id, 1);
+    DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
-      if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
-        abort();
+      CHECK_EQ(0, rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
-      if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
-        abort();
+      CHECK_EQ(0, rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id));
+    } else if (extension == RtpExtension::kVideoRotation) {
+      CHECK_EQ(0, rtp_rtcp_->SetSendVideoRotationStatus(channel_, true, id));
     } else {
-      abort();  // Unsupported extension.
+      RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 
@@ -153,7 +149,7 @@ VideoSendStream::VideoSendStream(
 
   // Enable NACK, FEC or both.
   if (config_.rtp.fec.red_payload_type != -1) {
-    assert(config_.rtp.fec.ulpfec_payload_type != -1);
+    DCHECK(config_.rtp.fec.ulpfec_payload_type != -1);
     if (config_.rtp.nack.rtp_history_ms > 0) {
       rtp_rtcp_->SetHybridNACKFECStatus(
           channel_,
@@ -174,7 +170,8 @@ VideoSendStream::VideoSendStream(
   ConfigureSsrcs();
 
   char rtcp_cname[ViERTP_RTCP::KMaxRTCPCNameLength];
-  assert(config_.rtp.c_name.length() < ViERTP_RTCP::KMaxRTCPCNameLength);
+  DCHECK_LT(config_.rtp.c_name.length(),
+            static_cast(ViERTP_RTCP::KMaxRTCPCNameLength));
   strncpy(rtcp_cname, config_.rtp.c_name.c_str(), sizeof(rtcp_cname) - 1);
   rtcp_cname[sizeof(rtcp_cname) - 1] = '\0';
 
@@ -185,33 +182,32 @@ VideoSendStream::VideoSendStream(
   capture_->ConnectCaptureDevice(capture_id_, channel_);
 
   network_ = ViENetwork::GetInterface(video_engine);
-  assert(network_ != NULL);
+  DCHECK(network_ != nullptr);
 
   network_->RegisterSendTransport(channel_, transport_adapter_);
   // 28 to match packet overhead in ModuleRtpRtcpImpl.
   network_->SetMTU(channel_,
                    static_cast(config_.rtp.max_packet_size + 28));
 
-  assert(config.encoder_settings.encoder != NULL);
-  assert(config.encoder_settings.payload_type >= 0);
-  assert(config.encoder_settings.payload_type <= 127);
+  DCHECK(config.encoder_settings.encoder != nullptr);
+  DCHECK_GE(config.encoder_settings.payload_type, 0);
+  DCHECK_LE(config.encoder_settings.payload_type, 127);
   external_codec_ = ViEExternalCodec::GetInterface(video_engine);
-  if (external_codec_->RegisterExternalSendCodec(
-          channel_,
-          config.encoder_settings.payload_type,
-          config.encoder_settings.encoder,
-          false) != 0) {
-    abort();
-  }
+  CHECK_EQ(0, external_codec_->RegisterExternalSendCodec(
+                  channel_, config.encoder_settings.payload_type,
+                  config.encoder_settings.encoder, false));
 
   codec_ = ViECodec::GetInterface(video_engine);
-  if (!ReconfigureVideoEncoder(encoder_config))
-    abort();
+  CHECK(ReconfigureVideoEncoder(encoder_config));
 
   if (overuse_observer)
     video_engine_base_->RegisterCpuOveruseObserver(channel_, overuse_observer);
+  // Registered regardless of monitoring, used for stats.
+  video_engine_base_->RegisterCpuOveruseMetricsObserver(channel_,
+                                                        &stats_proxy_);
 
   video_engine_base_->RegisterSendSideDelayObserver(channel_, &stats_proxy_);
+  video_engine_base_->RegisterSendStatisticsProxy(channel_, &stats_proxy_);
 
   image_process_ = ViEImageProcess::GetInterface(video_engine);
   image_process_->RegisterPreEncodeCallback(channel_,
@@ -228,11 +224,11 @@ VideoSendStream::VideoSendStream(
                                                        &stats_proxy_);
   rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
                                                       &stats_proxy_);
+  rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(channel_, &stats_proxy_);
   rtp_rtcp_->RegisterSendBitrateObserver(channel_, &stats_proxy_);
   rtp_rtcp_->RegisterSendFrameCountObserver(channel_, &stats_proxy_);
 
   codec_->RegisterEncoderObserver(channel_, stats_proxy_);
-  capture_->RegisterObserver(capture_id_, stats_proxy_);
 }
 
 VideoSendStream::~VideoSendStream() {
@@ -241,6 +237,7 @@ VideoSendStream::~VideoSendStream() {
 
   rtp_rtcp_->DeregisterSendFrameCountObserver(channel_, &stats_proxy_);
   rtp_rtcp_->DeregisterSendBitrateObserver(channel_, &stats_proxy_);
+  rtp_rtcp_->RegisterRtcpPacketTypeCounterObserver(channel_, nullptr);
   rtp_rtcp_->DeregisterSendChannelRtpStatisticsCallback(channel_,
                                                         &stats_proxy_);
   rtp_rtcp_->DeregisterSendChannelRtcpStatisticsCallback(channel_,
@@ -268,12 +265,13 @@ VideoSendStream::~VideoSendStream() {
   rtp_rtcp_->Release();
 }
 
-void VideoSendStream::SwapFrame(I420VideoFrame* frame) {
+void VideoSendStream::IncomingCapturedFrame(const I420VideoFrame& frame) {
   // TODO(pbos): Local rendering should not be done on the capture thread.
-  if (config_.local_renderer != NULL)
-    config_.local_renderer->RenderFrame(*frame, 0);
+  if (config_.local_renderer != nullptr)
+    config_.local_renderer->RenderFrame(frame, 0);
 
-  external_capture_->SwapFrame(frame);
+  stats_proxy_.OnIncomingFrame();
+  external_capture_->IncomingFrame(frame);
 }
 
 VideoSendStreamInput* VideoSendStream::Input() { return this; }
@@ -292,10 +290,11 @@ void VideoSendStream::Stop() {
 
 bool VideoSendStream::ReconfigureVideoEncoder(
     const VideoEncoderConfig& config) {
+  TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
   const std::vector& streams = config.streams;
-  assert(!streams.empty());
-  assert(config_.rtp.ssrcs.size() >= streams.size());
+  DCHECK(!streams.empty());
+  DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
 
   VideoCodec video_codec;
   memset(&video_codec, 0, sizeof(video_codec));
@@ -332,16 +331,30 @@ bool VideoSendStream::ReconfigureVideoEncoder(
   }
 
   if (video_codec.codecType == kVideoCodecVP8) {
-    if (config.encoder_specific_settings != NULL) {
+    if (config.encoder_specific_settings != nullptr) {
       video_codec.codecSpecific.VP8 = *reinterpret_cast(
                                           config.encoder_specific_settings);
     }
     video_codec.codecSpecific.VP8.numberOfTemporalLayers =
         static_cast(
             streams.back().temporal_layer_thresholds_bps.size() + 1);
+  } else if (video_codec.codecType == kVideoCodecVP9) {
+    if (config.encoder_specific_settings != nullptr) {
+      video_codec.codecSpecific.VP9 = *reinterpret_cast(
+                                          config.encoder_specific_settings);
+    }
+    video_codec.codecSpecific.VP9.numberOfTemporalLayers =
+        static_cast(
+            streams.back().temporal_layer_thresholds_bps.size() + 1);
+  } else if (video_codec.codecType == kVideoCodecH264) {
+    if (config.encoder_specific_settings != nullptr) {
+      video_codec.codecSpecific.H264 = *reinterpret_cast(
+                                           config.encoder_specific_settings);
+    }
   } else {
     // TODO(pbos): Support encoder_settings codec-agnostically.
-    assert(config.encoder_specific_settings == NULL);
+    DCHECK(config.encoder_specific_settings == nullptr)
+        << "Encoder-specific settings for codec type not wired up.";
   }
 
   strncpy(video_codec.plName,
@@ -352,18 +365,18 @@ bool VideoSendStream::ReconfigureVideoEncoder(
   video_codec.numberOfSimulcastStreams =
       static_cast(streams.size());
   video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
-  assert(streams.size() <= kMaxSimulcastStreams);
+  DCHECK_LE(streams.size(), static_cast(kMaxSimulcastStreams));
   for (size_t i = 0; i < streams.size(); ++i) {
     SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
-    assert(streams[i].width > 0);
-    assert(streams[i].height > 0);
-    assert(streams[i].max_framerate > 0);
+    DCHECK_GT(streams[i].width, 0u);
+    DCHECK_GT(streams[i].height, 0u);
+    DCHECK_GT(streams[i].max_framerate, 0);
     // Different framerates not supported per stream at the moment.
-    assert(streams[i].max_framerate == streams[0].max_framerate);
-    assert(streams[i].min_bitrate_bps >= 0);
-    assert(streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps);
-    assert(streams[i].max_bitrate_bps >= streams[i].target_bitrate_bps);
-    assert(streams[i].max_qp >= 0);
+    DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
+    DCHECK_GE(streams[i].min_bitrate_bps, 0);
+    DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
+    DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
+    DCHECK_GE(streams[i].max_qp, 0);
 
     sim_stream->width = static_cast(streams[i].width);
     sim_stream->height = static_cast(streams[i].height);
@@ -385,52 +398,41 @@ bool VideoSendStream::ReconfigureVideoEncoder(
     video_codec.qpMax = std::max(video_codec.qpMax,
                                  static_cast(streams[i].max_qp));
   }
-  unsigned int start_bitrate_bps;
-  if (codec_->GetCodecTargetBitrate(channel_, &start_bitrate_bps) != 0 ||
-      use_default_bitrate_) {
-    start_bitrate_bps = start_bitrate_bps_;
-  }
-  video_codec.startBitrate =
-      static_cast(start_bitrate_bps) / 1000;
+
+  // Set to zero to not update the bitrate controller from ViEEncoder, as
+  // the bitrate controller is already set from Call.
+  video_codec.startBitrate = 0;
 
   if (video_codec.minBitrate < kViEMinCodecBitrate)
     video_codec.minBitrate = kViEMinCodecBitrate;
   if (video_codec.maxBitrate < kViEMinCodecBitrate)
     video_codec.maxBitrate = kViEMinCodecBitrate;
-  if (video_codec.startBitrate < video_codec.minBitrate)
-    video_codec.startBitrate = video_codec.minBitrate;
-  if (video_codec.startBitrate > video_codec.maxBitrate)
-    video_codec.startBitrate = video_codec.maxBitrate;
 
-  if (video_codec.startBitrate < video_codec.minBitrate)
-    video_codec.startBitrate = video_codec.minBitrate;
-  if (video_codec.startBitrate > video_codec.maxBitrate)
-    video_codec.startBitrate = video_codec.maxBitrate;
-
-  assert(streams[0].max_framerate > 0);
+  DCHECK_GT(streams[0].max_framerate, 0);
   video_codec.maxFramerate = streams[0].max_framerate;
 
   if (codec_->SetSendCodec(channel_, video_codec) != 0)
     return false;
 
-  assert(config.min_transmit_bitrate_bps >= 0);
+  DCHECK_GE(config.min_transmit_bitrate_bps, 0);
   rtp_rtcp_->SetMinTransmitBitrate(channel_,
                                    config.min_transmit_bitrate_bps / 1000);
 
-  use_default_bitrate_ = false;
+  encoder_config_ = config;
+  use_config_bitrate_ = false;
   return true;
 }
 
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  return network_->ReceivedRTCPPacket(
-             channel_, packet, static_cast(length)) == 0;
+  return network_->ReceivedRTCPPacket(channel_, packet, length) == 0;
 }
 
-VideoSendStream::Stats VideoSendStream::GetStats() const {
+VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
 void VideoSendStream::ConfigureSsrcs() {
+  rtp_rtcp_->SetLocalSSRC(channel_, config_.rtp.ssrcs.front());
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
     rtp_rtcp_->SetLocalSSRC(
@@ -441,12 +443,11 @@ void VideoSendStream::ConfigureSsrcs() {
   }
 
   if (config_.rtp.rtx.ssrcs.empty()) {
-    assert(!config_.rtp.rtx.pad_with_redundant_payloads);
     return;
   }
 
   // Set up RTX.
-  assert(config_.rtp.rtx.ssrcs.size() == config_.rtp.ssrcs.size());
+  DCHECK_EQ(config_.rtp.rtx.ssrcs.size(), config_.rtp.ssrcs.size());
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
     rtp_rtcp_->SetLocalSSRC(channel_,
@@ -458,11 +459,7 @@ void VideoSendStream::ConfigureSsrcs() {
       rtp_rtcp_->SetRtpStateForSsrc(channel_, ssrc, it->second);
   }
 
-  if (config_.rtp.rtx.pad_with_redundant_payloads) {
-    rtp_rtcp_->SetPadWithRedundantPayloads(channel_, true);
-  }
-
-  assert(config_.rtp.rtx.payload_type >= 0);
+  DCHECK_GE(config_.rtp.rtx.payload_type, 0);
   rtp_rtcp_->SetRtxSendPayloadType(channel_, config_.rtp.rtx.payload_type);
 }
 
@@ -492,12 +489,22 @@ void VideoSendStream::SignalNetworkState(Call::NetworkState state) {
     rtp_rtcp_->SetRTCPStatus(channel_, kRtcpNone);
 }
 
-int VideoSendStream::GetPacerQueuingDelayMs() const {
-  int pacer_delay_ms = 0;
+int64_t VideoSendStream::GetPacerQueuingDelayMs() const {
+  int64_t pacer_delay_ms = 0;
   if (rtp_rtcp_->GetPacerQueuingDelayMs(channel_, &pacer_delay_ms) != 0) {
     return 0;
   }
   return pacer_delay_ms;
 }
+
+int64_t VideoSendStream::GetRtt() const {
+  webrtc::RtcpStatistics rtcp_stats;
+  int64_t rtt_ms;
+  if (rtp_rtcp_->GetSendChannelRtcpStatistics(channel_, rtcp_stats, rtt_ms) ==
+      0) {
+    return rtt_ms;
+  }
+  return -1;
+}
 }  // namespace internal
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream.h b/media/webrtc/trunk/webrtc/video/video_send_stream.h
index 873785dc9e..a5cd1ce77c 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream.h
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream.h
@@ -48,40 +48,40 @@ class VideoSendStream : public webrtc::VideoSendStream,
                   const VideoSendStream::Config& config,
                   const VideoEncoderConfig& encoder_config,
                   const std::map& suspended_ssrcs,
-                  int base_channel,
-                  int start_bitrate);
+                  int base_channel);
 
   virtual ~VideoSendStream();
 
-  virtual void Start() OVERRIDE;
-  virtual void Stop() OVERRIDE;
+  void Start() override;
+  void Stop() override;
 
-  virtual bool ReconfigureVideoEncoder(
-      const VideoEncoderConfig& config) OVERRIDE;
+  bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) override;
 
-  virtual Stats GetStats() const OVERRIDE;
+  Stats GetStats() override;
 
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // From VideoSendStreamInput.
-  virtual void SwapFrame(I420VideoFrame* frame) OVERRIDE;
+  void IncomingCapturedFrame(const I420VideoFrame& frame) override;
 
   // From webrtc::VideoSendStream.
-  virtual VideoSendStreamInput* Input() OVERRIDE;
+  VideoSendStreamInput* Input() override;
 
   typedef std::map RtpStateMap;
   RtpStateMap GetRtpStates() const;
 
   void SignalNetworkState(Call::NetworkState state);
 
-  int GetPacerQueuingDelayMs() const;
+  int64_t GetPacerQueuingDelayMs() const;
+
+  int64_t GetRtt() const;
 
  private:
   void ConfigureSsrcs();
   TransportAdapter transport_adapter_;
   EncodedFrameCallbackAdapter encoded_frame_proxy_;
   const VideoSendStream::Config config_;
-  const int start_bitrate_bps_;
+  VideoEncoderConfig encoder_config_;
   std::map suspended_ssrcs_;
 
   ViEBase* video_engine_base_;
@@ -99,7 +99,7 @@ class VideoSendStream : public webrtc::VideoSendStream,
   // Used as a workaround to indicate that we should be using the configured
   // start bitrate initially, instead of the one reported by VideoEngine (which
   // defaults to too high).
-  bool use_default_bitrate_;
+  bool use_config_bitrate_;
 
   SendStatisticsProxy stats_proxy_;
 };
diff --git a/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc b/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc
index 3ab1127444..428a2735c1 100644
--- a/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc
+++ b/media/webrtc/trunk/webrtc/video/video_send_stream_tests.cc
@@ -8,13 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include   // max
+#include 
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/common_video/interface/native_handle.h"
-#include "webrtc/common_video/interface/texture_video_frame.h"
 #include "webrtc/frame_callback.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
@@ -23,8 +25,6 @@
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/scoped_vector.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
@@ -32,6 +32,7 @@
 #include "webrtc/test/configurable_frame_size_encoder.h"
 #include "webrtc/test/null_transport.h"
 #include "webrtc/test/testsupport/perf_test.h"
+#include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video/transport_adapter.h"
 #include "webrtc/video_send_stream.h"
 
@@ -45,15 +46,15 @@ void ExpectEqualTextureFrames(const I420VideoFrame& frame1,
                               const I420VideoFrame& frame2);
 void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
                              const I420VideoFrame& frame2);
-void ExpectEqualFramesVector(const std::vector& frames1,
-                             const std::vector& frames2);
-I420VideoFrame* CreateI420VideoFrame(int width, int height, uint8_t data);
+void ExpectEqualFramesVector(const std::vector& frames1,
+                             const std::vector& frames2);
+I420VideoFrame CreateI420VideoFrame(int width, int height, uint8_t data);
 
 class FakeNativeHandle : public NativeHandle {
  public:
   FakeNativeHandle() {}
   virtual ~FakeNativeHandle() {}
-  virtual void* GetHandle() { return NULL; }
+  virtual void* GetHandle() { return nullptr; }
 };
 
 class VideoSendStreamTest : public test::CallTest {
@@ -94,7 +95,7 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
     CNameObserver() : SendTest(kDefaultTimeoutMs) {}
 
    private:
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
@@ -111,14 +112,13 @@ TEST_F(VideoSendStreamTest, SupportsCName) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.c_name = kCName;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for RTCP with CNAME.";
     }
@@ -136,7 +136,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
           kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
     }
 
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -149,15 +149,14 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for single RTP packet.";
     }
@@ -168,16 +167,18 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
 
 TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
   static const uint8_t kTOffsetExtensionId = 13;
+  static const int kEncodeDelayMs = 5;
   class TransmissionTimeOffsetObserver : public test::SendTest {
    public:
     TransmissionTimeOffsetObserver()
-        : SendTest(kDefaultTimeoutMs), encoder_(Clock::GetRealTimeClock()) {
+        : SendTest(kDefaultTimeoutMs),
+          encoder_(Clock::GetRealTimeClock(), kEncodeDelayMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId));
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -190,36 +191,20 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = &encoder_;
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for a single RTP packet.";
     }
 
-    class DelayedEncoder : public test::FakeEncoder {
-     public:
-      explicit DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {}
-      virtual int32_t Encode(
-          const I420VideoFrame& input_image,
-          const CodecSpecificInfo* codec_specific_info,
-          const std::vector* frame_types) OVERRIDE {
-        // A delay needs to be introduced to assure that we get a timestamp
-        // offset.
-        SleepMs(5);
-        return FakeEncoder::Encode(
-            input_image, codec_specific_info, frame_types);
-      }
-    };
-
-    DelayedEncoder encoder_;
+    test::DelayedEncoder encoder_;
   } test;
 
   RunBaseTest(&test);
@@ -237,11 +222,9 @@ class FakeReceiveStatistics : public NullReceiveStatistics {
     stats_map_[send_ssrc] = lossy_stats_.get();
   }
 
-  virtual StatisticianMap GetActiveStatisticians() const OVERRIDE {
-    return stats_map_;
-  }
+  StatisticianMap GetActiveStatisticians() const override { return stats_map_; }
 
-  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE {
+  StreamStatistician* GetStatistician(uint32_t ssrc) const override {
     return lossy_stats_.get();
   }
 
@@ -255,58 +238,35 @@ class FakeReceiveStatistics : public NullReceiveStatistics {
       stats_.cumulative_lost = cumulative_lost;
       stats_.extended_max_sequence_number = extended_max_sequence_number;
     }
-    virtual bool GetStatistics(RtcpStatistics* statistics,
-                               bool reset) OVERRIDE {
+    bool GetStatistics(RtcpStatistics* statistics, bool reset) override {
       *statistics = stats_;
       return true;
     }
-    virtual void GetDataCounters(uint32_t* bytes_received,
-                                 uint32_t* packets_received) const OVERRIDE {
+    void GetDataCounters(size_t* bytes_received,
+                         uint32_t* packets_received) const override {
       *bytes_received = 0;
       *packets_received = 0;
     }
-    virtual uint32_t BitrateReceived() const OVERRIDE { return 0; }
-    virtual void ResetStatistics() OVERRIDE {}
-    virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
-                                         int min_rtt) const OVERRIDE {
+    void GetReceiveStreamDataCounters(
+        StreamDataCounters* data_counters) const override {}
+    uint32_t BitrateReceived() const override { return 0; }
+    void ResetStatistics() override {}
+    bool IsRetransmitOfOldPacket(const RTPHeader& header,
+                                 int64_t min_rtt) const override {
       return false;
     }
 
-    virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE {
+    bool IsPacketInOrder(uint16_t sequence_number) const override {
       return true;
     }
 
     RtcpStatistics stats_;
   };
 
-  scoped_ptr lossy_stats_;
+  rtc::scoped_ptr lossy_stats_;
   StatisticianMap stats_map_;
 };
 
-TEST_F(VideoSendStreamTest, SwapsI420VideoFrames) {
-  static const size_t kWidth = 320;
-  static const size_t kHeight = 240;
-
-  test::NullTransport transport;
-  Call::Config call_config(&transport);
-  CreateSenderCall(call_config);
-
-  CreateSendConfig(1);
-  CreateStreams();
-  send_stream_->Start();
-
-  I420VideoFrame frame;
-  frame.CreateEmptyFrame(
-      kWidth, kHeight, kWidth, (kWidth + 1) / 2, (kWidth + 1) / 2);
-  uint8_t* old_y_buffer = frame.buffer(kYPlane);
-
-  send_stream_->Input()->SwapFrame(&frame);
-
-  EXPECT_NE(frame.buffer(kYPlane), old_y_buffer);
-
-  DestroyStreams();
-}
-
 TEST_F(VideoSendStreamTest, SupportsFec) {
   class FecObserver : public test::SendTest {
    public:
@@ -320,7 +280,7 @@ TEST_F(VideoSendStreamTest, SupportsFec) {
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -329,8 +289,8 @@ TEST_F(VideoSendStreamTest, SupportsFec) {
         // Receive statistics reporting having lost 50% of the packets.
         FakeReceiveStatistics lossy_receive_stats(
             kSendSsrcs[0], header.sequenceNumber, send_count_ / 2, 127);
-        RTCPSender rtcp_sender(
-            0, false, Clock::GetRealTimeClock(), &lossy_receive_stats);
+        RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
+                               &lossy_receive_stats, nullptr);
         EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
         rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
@@ -357,15 +317,14 @@ TEST_F(VideoSendStreamTest, SupportsFec) {
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.fec.red_payload_type = kRedPayloadType;
       send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out waiting for FEC and media packets.";
     }
 
@@ -395,7 +354,7 @@ void VideoSendStreamTest::TestNackRetransmission(
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -405,7 +364,7 @@ void VideoSendStreamTest::TestNackRetransmission(
         nacked_sequence_number_ = nack_sequence_number;
         NullReceiveStatistics null_stats;
         RTCPSender rtcp_sender(
-            0, false, Clock::GetRealTimeClock(), &null_stats);
+            0, false, Clock::GetRealTimeClock(), &null_stats, nullptr);
         EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
         rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
@@ -437,17 +396,16 @@ void VideoSendStreamTest::TestNackRetransmission(
       return SEND_PACKET;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->rtp.rtx.payload_type = retransmit_payload_type_;
       if (retransmit_ssrc_ != kSendSsrcs[0])
         send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for NACK retransmission.";
     }
@@ -477,18 +435,18 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
   // Use a fake encoder to output a frame of every size in the range [90, 290],
   // for each size making sure that the exact number of payload bytes received
   // is correct and that packets are fragmented to respect max packet size.
-  static const uint32_t kMaxPacketSize = 128;
-  static const uint32_t start = 90;
-  static const uint32_t stop = 290;
+  static const size_t kMaxPacketSize = 128;
+  static const size_t start = 90;
+  static const size_t stop = 290;
 
   // Observer that verifies that the expected number of packets and bytes
   // arrive for each frame size, from start_size to stop_size.
   class FrameFragmentationTest : public test::SendTest,
                                  public EncodedFrameObserver {
    public:
-    FrameFragmentationTest(uint32_t max_packet_size,
-                           uint32_t start_size,
-                           uint32_t stop_size,
+    FrameFragmentationTest(size_t max_packet_size,
+                           size_t start_size,
+                           size_t stop_size,
                            bool test_generic_packetization,
                            bool use_fec)
         : SendTest(kLongTimeoutMs),
@@ -503,16 +461,16 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
           accumulated_payload_(0),
           fec_packet_received_(false),
           current_size_rtp_(start_size),
-          current_size_frame_(start_size) {
+          current_size_frame_(static_cast(start_size)) {
       // Fragmentation required, this test doesn't make sense without it.
-      encoder_.SetFrameSize(start);
-      assert(stop_size > max_packet_size);
+      encoder_.SetFrameSize(start_size);
+      DCHECK_GT(stop_size, max_packet_size);
       transport_adapter_.Enable();
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t size) OVERRIDE {
-      uint32_t length = static_cast(size);
+    Action OnSendRtp(const uint8_t* packet, size_t size) override {
+      size_t length = size;
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
 
@@ -534,7 +492,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
         TriggerLossReport(header);
 
       if (test_generic_packetization_) {
-        uint32_t overhead = header.headerLength + header.paddingLength +
+        size_t overhead = header.headerLength + header.paddingLength +
                           (1 /* Generic header */);
         if (use_fec_)
           overhead += 1;  // RED for FEC header.
@@ -587,8 +545,8 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
         // Receive statistics reporting having lost 50% of the packets.
         FakeReceiveStatistics lossy_receive_stats(
             kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
-        RTCPSender rtcp_sender(
-            0, false, Clock::GetRealTimeClock(), &lossy_receive_stats);
+        RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
+                               &lossy_receive_stats, nullptr);
         EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
         rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
@@ -607,13 +565,19 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
           current_size_frame_.Value() < static_cast(stop_size_)) {
         ++current_size_frame_;
       }
-      encoder_.SetFrameSize(current_size_frame_.Value());
+      encoder_.SetFrameSize(static_cast(current_size_frame_.Value()));
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    Call::Config GetSenderCallConfig() override {
+      Call::Config config(SendTransport());
+      const int kMinBitrateBps = 30000;
+      config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
+      return config;
+    }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       if (use_fec_) {
         send_config->rtp.fec.red_payload_type = kRedPayloadType;
         send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
@@ -633,7 +597,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
           RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while observing incoming RTP packets.";
     }
@@ -641,17 +605,17 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
     internal::TransportAdapter transport_adapter_;
     test::ConfigurableFrameSizeEncoder encoder_;
 
-    const uint32_t max_packet_size_;
-    const uint32_t stop_size_;
+    const size_t max_packet_size_;
+    const size_t stop_size_;
     const bool test_generic_packetization_;
     const bool use_fec_;
 
     uint32_t packet_count_;
-    uint32_t accumulated_size_;
-    uint32_t accumulated_payload_;
+    size_t accumulated_size_;
+    size_t accumulated_payload_;
     bool fec_packet_received_;
 
-    uint32_t current_size_rtp_;
+    size_t current_size_rtp_;
     Atomic32 current_size_frame_;
   };
 
@@ -709,7 +673,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
     }
 
    private:
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       // Receive statistics reporting having lost 0% of the packets.
       // This is needed for the send-side bitrate controller to work properly.
       CriticalSectionScoped lock(crit_.get());
@@ -717,7 +681,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
       return SEND_PACKET;
     }
 
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       CriticalSectionScoped lock(crit_.get());
       ++rtp_count_;
       RTPHeader header;
@@ -752,7 +716,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
     }
 
     // This method implements the I420FrameCallback.
-    void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
+    void FrameCallback(I420VideoFrame* video_frame) override {
       CriticalSectionScoped lock(crit_.get());
       if (test_state_ == kDuringSuspend &&
           ++suspended_frame_count_ > kSuspendTimeFrames) {
@@ -773,22 +737,20 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
       high_remb_bps_ = value;
     }
 
-    virtual void SetReceivers(
-        PacketReceiver* send_transport_receiver,
-        PacketReceiver* receive_transport_receiver) OVERRIDE {
+    void SetReceivers(PacketReceiver* send_transport_receiver,
+                      PacketReceiver* receive_transport_receiver) override {
       transport_.SetReceiver(send_transport_receiver);
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       stream_ = send_stream;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       send_config->pre_encode_callback = this;
       send_config->suspend_below_min_bitrate = true;
@@ -800,7 +762,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
       set_high_remb_bps(min_bitrate_bps + threshold_window + 5000);
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out during suspend-below-min-bitrate test.";
       transport_.StopSending();
@@ -817,14 +779,14 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
         EXCLUSIVE_LOCKS_REQUIRED(crit_) {
       FakeReceiveStatistics receive_stats(
           kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
-      RTCPSender rtcp_sender(0, false, clock_, &receive_stats);
+      RTCPSender rtcp_sender(0, false, clock_, &receive_stats, nullptr);
       EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
       rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
       rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
       if (remb_value > 0) {
         rtcp_sender.SetREMBStatus(true);
-        rtcp_sender.SetREMBData(remb_value, 0, NULL);
+        rtcp_sender.SetREMBData(remb_value, std::vector());
       }
       RTCPSender::FeedbackState feedback_state;
       EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
@@ -835,7 +797,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
     Clock* const clock_;
     VideoSendStream* stream_;
 
-    const scoped_ptr crit_;
+    const rtc::scoped_ptr crit_;
     TestState test_state_ GUARDED_BY(crit_);
     int rtp_count_ GUARDED_BY(crit_);
     int last_sequence_number_ GUARDED_BY(crit_);
@@ -856,19 +818,19 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
           transport_adapter_(ReceiveTransport()),
           crit_(CriticalSectionWrapper::CreateCriticalSection()),
           last_packet_time_ms_(-1),
-          capturer_(NULL) {
+          capturer_(nullptr) {
       transport_adapter_.Enable();
     }
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       CriticalSectionScoped lock(crit_.get());
       last_packet_time_ms_ = clock_->TimeInMilliseconds();
       capturer_->Stop();
       return SEND_PACKET;
     }
 
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       CriticalSectionScoped lock(crit_.get());
       const int kVideoMutedThresholdMs = 10000;
       if (last_packet_time_ms_ > 0 &&
@@ -877,8 +839,8 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
         observation_complete_->Set();
       // Receive statistics reporting having lost 50% of the packets.
       FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
-      RTCPSender rtcp_sender(
-          0, false, Clock::GetRealTimeClock(), &receive_stats);
+      RTCPSender rtcp_sender(0, false, Clock::GetRealTimeClock(),
+                             &receive_stats, nullptr);
       EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
 
       rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
@@ -890,14 +852,13 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
       return SEND_PACKET;
     }
 
-    virtual void SetReceivers(
-        PacketReceiver* send_transport_receiver,
-        PacketReceiver* receive_transport_receiver) OVERRIDE {
+    void SetReceivers(PacketReceiver* send_transport_receiver,
+                      PacketReceiver* receive_transport_receiver) override {
       RtpRtcpObserver::SetReceivers(send_transport_receiver,
                                     send_transport_receiver);
     }
 
-    virtual size_t GetNumStreams() const OVERRIDE { return 3; }
+    size_t GetNumStreams() const override { return 3; }
 
     virtual void OnFrameGeneratorCapturerCreated(
         test::FrameGeneratorCapturer* frame_generator_capturer) {
@@ -905,14 +866,14 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
       capturer_ = frame_generator_capturer;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for RTP packets to stop being sent.";
     }
 
     Clock* const clock_;
     internal::TransportAdapter transport_adapter_;
-    const scoped_ptr crit_;
+    const rtc::scoped_ptr crit_;
     int64_t last_packet_time_ms_ GUARDED_BY(crit_);
     test::FrameGeneratorCapturer* capturer_ GUARDED_BY(crit_);
   } test;
@@ -920,86 +881,6 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
   RunBaseTest(&test);
 }
 
-TEST_F(VideoSendStreamTest, ProducesStats) {
-  class ProducesStats : public test::SendTest {
-   public:
-    ProducesStats()
-        : SendTest(kDefaultTimeoutMs),
-          stream_(NULL),
-          event_(EventWrapper::Create()) {}
-
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
-      event_->Set();
-
-      return SEND_PACKET;
-    }
-
-   private:
-    bool WaitForFilledStats() {
-      Clock* clock = Clock::GetRealTimeClock();
-      int64_t now = clock->TimeInMilliseconds();
-      int64_t stop_time = now + kDefaultTimeoutMs;
-      while (now < stop_time) {
-        int64_t time_left = stop_time - now;
-        if (time_left > 0 && event_->Wait(time_left) == kEventSignaled &&
-            CheckStats()) {
-          return true;
-        }
-        now = clock->TimeInMilliseconds();
-      }
-      return false;
-    }
-
-    bool CheckStats() {
-      VideoSendStream::Stats stats = stream_->GetStats();
-      // Check that all applicable data sources have been used.
-      if (stats.input_frame_rate > 0 && stats.encode_frame_rate > 0
-          && !stats.substreams.empty()) {
-        uint32_t ssrc = stats.substreams.begin()->first;
-        EXPECT_NE(
-            config_.rtp.ssrcs.end(),
-            std::find(
-                config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc));
-        // Check for data populated by various sources. RTCP excluded as this
-        // data is received from remote side. Tested in call tests instead.
-        const SsrcStats& entry = stats.substreams[ssrc];
-        if (entry.key_frames > 0u && entry.total_bitrate_bps > 0 &&
-            entry.rtp_stats.packets > 0u && entry.avg_delay_ms > 0 &&
-            entry.max_delay_ms > 0) {
-          return true;
-        }
-      }
-      return false;
-    }
-
-    void SetConfig(const VideoSendStream::Config& config) { config_ = config; }
-
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
-      SetConfig(*send_config);
-    }
-
-    virtual void OnStreamsCreated(
-        VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
-      stream_ = send_stream;
-    }
-
-    virtual void PerformTest() OVERRIDE {
-      EXPECT_TRUE(WaitForFilledStats())
-          << "Timed out waiting for filled statistics.";
-    }
-
-    VideoSendStream* stream_;
-    VideoSendStream::Config config_;
-    scoped_ptr event_;
-  } test;
-
-  RunBaseTest(&test);
-}
-
 // This test first observes "high" bitrate use at which point it sends a REMB to
 // indicate that it should be lowered significantly. The test then observes that
 // the bitrate observed is sinking well below the min-transmit-bitrate threshold
@@ -1026,22 +907,22 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
       rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       stream_ = send_stream;
     }
 
    private:
-    virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
-                                         size_t length) OVERRIDE {
+    DeliveryStatus DeliverPacket(const uint8_t* packet,
+                                 size_t length) override {
       if (RtpHeaderParser::IsRtcp(packet, length))
         return DELIVERY_OK;
 
       RTPHeader header;
       if (!parser_->Parse(packet, length, &header))
         return DELIVERY_PACKET_ERROR;
-      assert(stream_ != NULL);
+      DCHECK(stream_ != nullptr);
       VideoSendStream::Stats stats = stream_->GetStats();
       if (!stats.substreams.empty()) {
         EXPECT_EQ(1u, stats.substreams.size());
@@ -1054,7 +935,8 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
                           "bps",
                           false);
         if (total_bitrate_bps > kHighBitrateBps) {
-          rtp_rtcp_->SetREMBData(kRembBitrateBps, 1, &header.ssrc);
+          rtp_rtcp_->SetREMBData(kRembBitrateBps,
+                                 std::vector(1, header.ssrc));
           rtp_rtcp_->Process();
           bitrate_capped_ = true;
         } else if (bitrate_capped_ &&
@@ -1065,25 +947,23 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
       return DELIVERY_OK;
     }
 
-    virtual void SetReceivers(
-        PacketReceiver* send_transport_receiver,
-        PacketReceiver* receive_transport_receiver) OVERRIDE {
+    void SetReceivers(PacketReceiver* send_transport_receiver,
+                      PacketReceiver* receive_transport_receiver) override {
       RtpRtcpObserver::SetReceivers(this, send_transport_receiver);
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timeout while waiting for low bitrate stats after REMB.";
     }
 
-    scoped_ptr rtp_rtcp_;
+    rtc::scoped_ptr rtp_rtcp_;
     internal::TransportAdapter feedback_transport_;
     VideoSendStream* stream_;
     bool bitrate_capped_;
@@ -1092,14 +972,73 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
   RunBaseTest(&test);
 }
 
+TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
+  class StartBitrateObserver : public test::FakeEncoder {
+   public:
+    StartBitrateObserver()
+        : FakeEncoder(Clock::GetRealTimeClock()), start_bitrate_kbps_(0) {}
+    int32_t InitEncode(const VideoCodec* config,
+                       int32_t number_of_cores,
+                       size_t max_payload_size) override {
+      rtc::CritScope lock(&crit_);
+      start_bitrate_kbps_ = config->startBitrate;
+      return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
+    }
+
+    int32_t SetRates(uint32_t new_target_bitrate, uint32_t framerate) override {
+      rtc::CritScope lock(&crit_);
+      start_bitrate_kbps_ = new_target_bitrate;
+      return FakeEncoder::SetRates(new_target_bitrate, framerate);
+    }
+
+    int GetStartBitrateKbps() const {
+      rtc::CritScope lock(&crit_);
+      return start_bitrate_kbps_;
+    }
+
+   private:
+    mutable rtc::CriticalSection crit_;
+    int start_bitrate_kbps_ GUARDED_BY(crit_);
+  };
+
+  test::NullTransport transport;
+  CreateSenderCall(Call::Config(&transport));
+
+  CreateSendConfig(1);
+
+  Call::Config::BitrateConfig bitrate_config;
+  bitrate_config.start_bitrate_bps =
+      2 * encoder_config_.streams[0].max_bitrate_bps;
+  sender_call_->SetBitrateConfig(bitrate_config);
+
+  StartBitrateObserver encoder;
+  send_config_.encoder_settings.encoder = &encoder;
+
+  CreateStreams();
+
+  EXPECT_EQ(encoder_config_.streams[0].max_bitrate_bps / 1000,
+            encoder.GetStartBitrateKbps());
+
+  encoder_config_.streams[0].max_bitrate_bps =
+      2 * bitrate_config.start_bitrate_bps;
+  send_stream_->ReconfigureVideoEncoder(encoder_config_);
+
+  // New bitrate should be reconfigured above the previous max. As there's no
+  // network connection this shouldn't be flaky, as no bitrate should've been
+  // reported in between.
+  EXPECT_EQ(bitrate_config.start_bitrate_bps / 1000,
+            encoder.GetStartBitrateKbps());
+
+  DestroyStreams();
+}
+
 TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
   class FrameObserver : public I420FrameCallback {
    public:
     FrameObserver() : output_frame_event_(EventWrapper::Create()) {}
 
-    void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
-      // Clone the frame because the caller owns it.
-      output_frames_.push_back(video_frame->CloneFrame());
+    void FrameCallback(I420VideoFrame* video_frame) override {
+      output_frames_.push_back(*video_frame);
       output_frame_event_->Set();
     }
 
@@ -1109,16 +1048,16 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
           << "Timeout while waiting for output frames.";
     }
 
-    const std::vector& output_frames() const {
-      return output_frames_.get();
+    const std::vector& output_frames() const {
+      return output_frames_;
     }
 
    private:
     // Delivered output frames.
-    ScopedVector output_frames_;
+    std::vector output_frames_;
 
     // Indicate an output frame has arrived.
-    scoped_ptr output_frame_event_;
+    rtc::scoped_ptr output_frame_event_;
   };
 
   // Initialize send stream.
@@ -1130,9 +1069,9 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
   send_config_.pre_encode_callback = &observer;
   CreateStreams();
 
-  // Prepare five input frames. Send I420VideoFrame and TextureVideoFrame
+  // Prepare five input frames. Send ordinary I420VideoFrame and texture frames
   // alternatively.
-  ScopedVector input_frames;
+  std::vector input_frames;
   int width = static_cast(encoder_config_.streams[0].width);
   int height = static_cast(encoder_config_.streams[0].height);
   webrtc::RefCountImpl* handle1 =
@@ -1141,17 +1080,15 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
       new webrtc::RefCountImpl();
   webrtc::RefCountImpl* handle3 =
       new webrtc::RefCountImpl();
-  input_frames.push_back(new TextureVideoFrame(handle1, width, height, 1, 1));
-  input_frames.push_back(new TextureVideoFrame(handle2, width, height, 2, 2));
-  input_frames.push_back(CreateI420VideoFrame(width, height, 1));
-  input_frames.push_back(CreateI420VideoFrame(width, height, 2));
-  input_frames.push_back(new TextureVideoFrame(handle3, width, height, 3, 3));
+  input_frames.push_back(I420VideoFrame(handle1, width, height, 1, 1));
+  input_frames.push_back(I420VideoFrame(handle2, width, height, 2, 2));
+  input_frames.push_back(CreateI420VideoFrame(width, height, 3));
+  input_frames.push_back(CreateI420VideoFrame(width, height, 4));
+  input_frames.push_back(I420VideoFrame(handle3, width, height, 5, 5));
 
   send_stream_->Start();
   for (size_t i = 0; i < input_frames.size(); i++) {
-    // Make a copy of the input frame because the buffer will be swapped.
-    scoped_ptr frame(input_frames[i]->CloneFrame());
-    send_stream_->Input()->SwapFrame(frame.get());
+    send_stream_->Input()->IncomingCapturedFrame(input_frames[i]);
     // Do not send the next frame too fast, so the frame dropper won't drop it.
     if (i < input_frames.size() - 1)
       SleepMs(1000 / encoder_config_.streams[0].max_framerate);
@@ -1163,14 +1100,14 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
 
   // Test if the input and output frames are the same. render_time_ms and
   // timestamp are not compared because capturer sets those values.
-  ExpectEqualFramesVector(input_frames.get(), observer.output_frames());
+  ExpectEqualFramesVector(input_frames, observer.output_frames());
 
   DestroyStreams();
 }
 
 void ExpectEqualFrames(const I420VideoFrame& frame1,
                        const I420VideoFrame& frame2) {
-  if (frame1.native_handle() != NULL || frame2.native_handle() != NULL)
+  if (frame1.native_handle() != nullptr || frame2.native_handle() != nullptr)
     ExpectEqualTextureFrames(frame1, frame2);
   else
     ExpectEqualBufferFrames(frame1, frame2);
@@ -1181,6 +1118,7 @@ void ExpectEqualTextureFrames(const I420VideoFrame& frame1,
   EXPECT_EQ(frame1.native_handle(), frame2.native_handle());
   EXPECT_EQ(frame1.width(), frame2.width());
   EXPECT_EQ(frame1.height(), frame2.height());
+  EXPECT_EQ(frame1.render_time_ms(), frame2.render_time_ms());
 }
 
 void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
@@ -1190,7 +1128,7 @@ void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
   EXPECT_EQ(frame1.stride(kYPlane), frame2.stride(kYPlane));
   EXPECT_EQ(frame1.stride(kUPlane), frame2.stride(kUPlane));
   EXPECT_EQ(frame1.stride(kVPlane), frame2.stride(kVPlane));
-  EXPECT_EQ(frame1.ntp_time_ms(), frame2.ntp_time_ms());
+  EXPECT_EQ(frame1.render_time_ms(), frame2.render_time_ms());
   ASSERT_EQ(frame1.allocated_size(kYPlane), frame2.allocated_size(kYPlane));
   EXPECT_EQ(0,
             memcmp(frame1.buffer(kYPlane),
@@ -1208,33 +1146,22 @@ void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
                    frame1.allocated_size(kVPlane)));
 }
 
-void ExpectEqualFramesVector(const std::vector& frames1,
-                             const std::vector& frames2) {
+void ExpectEqualFramesVector(const std::vector& frames1,
+                             const std::vector& frames2) {
   EXPECT_EQ(frames1.size(), frames2.size());
   for (size_t i = 0; i < std::min(frames1.size(), frames2.size()); ++i)
-    ExpectEqualFrames(*frames1[i], *frames2[i]);
+    ExpectEqualFrames(frames1[i], frames2[i]);
 }
 
-I420VideoFrame* CreateI420VideoFrame(int width, int height, uint8_t data) {
-  I420VideoFrame* frame = new I420VideoFrame();
+I420VideoFrame CreateI420VideoFrame(int width, int height, uint8_t data) {
   const int kSizeY = width * height * 2;
-  const int kSizeUV = width * height;
-  scoped_ptr buffer(new uint8_t[kSizeY]);
+  rtc::scoped_ptr buffer(new uint8_t[kSizeY]);
   memset(buffer.get(), data, kSizeY);
-  frame->CreateFrame(kSizeY,
-                     buffer.get(),
-                     kSizeUV,
-                     buffer.get(),
-                     kSizeUV,
-                     buffer.get(),
-                     width,
-                     height,
-                     width,
-                     width / 2,
-                     width / 2);
-  frame->set_timestamp(data);
-  frame->set_ntp_time_ms(data);
-  frame->set_render_time_ms(data);
+  I420VideoFrame frame;
+  frame.CreateFrame(buffer.get(), buffer.get(), buffer.get(), width, height,
+                    width, width / 2, width / 2);
+  frame.set_timestamp(data);
+  frame.set_render_time_ms(data);
   return frame;
 }
 
@@ -1265,9 +1192,9 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
     }
 
    private:
-    virtual int32_t InitEncode(const VideoCodec* codecSettings,
-                               int32_t numberOfCores,
-                               uint32_t maxPayloadSize) OVERRIDE {
+    int32_t InitEncode(const VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize) override {
       CriticalSectionScoped lock(crit_.get());
       EXPECT_FALSE(initialized_);
       initialized_ = true;
@@ -1275,25 +1202,24 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
       return 0;
     }
 
-    virtual int32_t Encode(
-        const I420VideoFrame& inputImage,
-        const CodecSpecificInfo* codecSpecificInfo,
-        const std::vector* frame_types) OVERRIDE {
+    int32_t Encode(const I420VideoFrame& inputImage,
+                   const CodecSpecificInfo* codecSpecificInfo,
+                   const std::vector* frame_types) override {
       EXPECT_TRUE(IsReadyForEncode());
 
       observation_complete_->Set();
       return 0;
     }
 
-    virtual int32_t RegisterEncodeCompleteCallback(
-        EncodedImageCallback* callback) OVERRIDE {
+    int32_t RegisterEncodeCompleteCallback(
+        EncodedImageCallback* callback) override {
       CriticalSectionScoped lock(crit_.get());
       EXPECT_TRUE(initialized_);
       callback_registered_ = true;
       return 0;
     }
 
-    virtual int32_t Release() OVERRIDE {
+    int32_t Release() override {
       CriticalSectionScoped lock(crit_.get());
       EXPECT_TRUE(IsReadyForEncode());
       EXPECT_FALSE(released_);
@@ -1304,35 +1230,33 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
       return 0;
     }
 
-    virtual int32_t SetChannelParameters(uint32_t packetLoss,
-                                         int rtt) OVERRIDE {
+    int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override {
       EXPECT_TRUE(IsReadyForEncode());
       return 0;
     }
 
-    virtual int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) OVERRIDE {
+    int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override {
       EXPECT_TRUE(IsReadyForEncode());
       return 0;
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       // Encoder initialization should be done in stream construction before
       // starting.
       EXPECT_TRUE(IsReadyForEncode());
       stream_ = send_stream;
     }
 
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config_ = *encoder_config;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for Encode.";
       EXPECT_EQ(0u, num_releases());
@@ -1348,7 +1272,7 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
           << "Timed out while waiting for Encode.";
     }
 
-    scoped_ptr crit_;
+    rtc::scoped_ptr crit_;
     VideoSendStream* stream_;
     bool initialized_ GUARDED_BY(crit_);
     bool callback_registered_ GUARDED_BY(crit_);
@@ -1373,23 +1297,22 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
           num_initializations_(0) {}
 
    private:
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       encoder_config_ = *encoder_config;
     }
 
-    virtual void OnStreamsCreated(
+    void OnStreamsCreated(
         VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
+        const std::vector& receive_streams) override {
       stream_ = send_stream;
     }
 
-    virtual int32_t InitEncode(const VideoCodec* config,
-                               int32_t number_of_cores,
-                               uint32_t max_payload_size) OVERRIDE {
+    int32_t InitEncode(const VideoCodec* config,
+                       int32_t number_of_cores,
+                       size_t max_payload_size) override {
       if (num_initializations_ == 0) {
         // Verify default values.
         EXPECT_EQ(kRealtimeVideo, config->mode);
@@ -1401,7 +1324,7 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
       return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
 
       encoder_config_.content_type = VideoEncoderConfig::kScreenshare;
@@ -1419,131 +1342,188 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
   RunBaseTest(&test);
 }
 
+static const size_t kVideoCodecConfigObserverNumberOfTemporalLayers = 4;
+template 
+class VideoCodecConfigObserver : public test::SendTest,
+                                 public test::FakeEncoder {
+
+ public:
+  VideoCodecConfigObserver(VideoCodecType video_codec_type,
+                           const char* codec_name)
+      : SendTest(VideoSendStreamTest::kDefaultTimeoutMs),
+        FakeEncoder(Clock::GetRealTimeClock()),
+        video_codec_type_(video_codec_type),
+        codec_name_(codec_name),
+        num_initializations_(0) {
+    memset(&encoder_settings_, 0, sizeof(encoder_settings_));
+  }
+
+ private:
+  void ModifyConfigs(VideoSendStream::Config* send_config,
+                     std::vector* receive_configs,
+                     VideoEncoderConfig* encoder_config) override {
+    send_config->encoder_settings.encoder = this;
+    send_config->encoder_settings.payload_name = codec_name_;
+
+    for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
+      encoder_config->streams[i].temporal_layer_thresholds_bps.resize(
+          kVideoCodecConfigObserverNumberOfTemporalLayers - 1);
+    }
+
+    encoder_config->encoder_specific_settings = &encoder_settings_;
+    encoder_config_ = *encoder_config;
+  }
+
+  void OnStreamsCreated(
+      VideoSendStream* send_stream,
+      const std::vector& receive_streams) override {
+    stream_ = send_stream;
+  }
+
+  int32_t InitEncode(const VideoCodec* config,
+                     int32_t number_of_cores,
+                     size_t max_payload_size) override {
+    EXPECT_EQ(video_codec_type_, config->codecType);
+    VerifyCodecSpecifics(*config);
+    ++num_initializations_;
+    return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
+  }
+
+  void VerifyCodecSpecifics(const VideoCodec& config) const;
+
+  void PerformTest() override {
+    EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
+
+    encoder_settings_.frameDroppingOn = true;
+    stream_->ReconfigureVideoEncoder(encoder_config_);
+    EXPECT_EQ(2u, num_initializations_)
+        << "ReconfigureVideoEncoder did not reinitialize the encoder with "
+           "new encoder settings.";
+  }
+
+  int32_t Encode(const I420VideoFrame& input_image,
+                 const CodecSpecificInfo* codec_specific_info,
+                 const std::vector* frame_types) override {
+    // Silently skip the encode, FakeEncoder::Encode doesn't produce VP8.
+    return 0;
+  }
+
+  T encoder_settings_;
+  const VideoCodecType video_codec_type_;
+  const char* const codec_name_;
+  size_t num_initializations_;
+  VideoSendStream* stream_;
+  VideoEncoderConfig encoder_config_;
+};
+
+template <>
+void VideoCodecConfigObserver::VerifyCodecSpecifics(
+    const VideoCodec& config) const {
+  EXPECT_EQ(0, memcmp(&config.codecSpecific.H264, &encoder_settings_,
+                      sizeof(encoder_settings_)));
+}
+template <>
+void VideoCodecConfigObserver::VerifyCodecSpecifics(
+    const VideoCodec& config) const {
+  // Check that the number of temporal layers has propagated properly to
+  // VideoCodec.
+  EXPECT_EQ(kVideoCodecConfigObserverNumberOfTemporalLayers,
+            config.codecSpecific.VP8.numberOfTemporalLayers);
+
+  for (unsigned char i = 0; i < config.numberOfSimulcastStreams; ++i) {
+    EXPECT_EQ(kVideoCodecConfigObserverNumberOfTemporalLayers,
+              config.simulcastStream[i].numberOfTemporalLayers);
+  }
+
+  // Set expected temporal layers as they should have been set when
+  // reconfiguring the encoder and not match the set config.
+  VideoCodecVP8 encoder_settings = encoder_settings_;
+  encoder_settings.numberOfTemporalLayers =
+      kVideoCodecConfigObserverNumberOfTemporalLayers;
+  EXPECT_EQ(0, memcmp(&config.codecSpecific.VP8, &encoder_settings,
+                      sizeof(encoder_settings_)));
+}
+template <>
+void VideoCodecConfigObserver::VerifyCodecSpecifics(
+    const VideoCodec& config) const {
+  // Check that the number of temporal layers has propagated properly to
+  // VideoCodec.
+  EXPECT_EQ(kVideoCodecConfigObserverNumberOfTemporalLayers,
+            config.codecSpecific.VP9.numberOfTemporalLayers);
+
+  for (unsigned char i = 0; i < config.numberOfSimulcastStreams; ++i) {
+    EXPECT_EQ(kVideoCodecConfigObserverNumberOfTemporalLayers,
+              config.simulcastStream[i].numberOfTemporalLayers);
+  }
+
+  // Set expected temporal layers as they should have been set when
+  // reconfiguring the encoder and not match the set config.
+  VideoCodecVP9 encoder_settings = encoder_settings_;
+  encoder_settings.numberOfTemporalLayers =
+      kVideoCodecConfigObserverNumberOfTemporalLayers;
+  EXPECT_EQ(0, memcmp(&config.codecSpecific.VP9, &encoder_settings,
+                      sizeof(encoder_settings_)));
+}
+
 TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp8Config) {
-  static const size_t kNumberOfTemporalLayers = 4;
-  class VideoCodecConfigObserver : public test::SendTest,
-                                   public test::FakeEncoder {
-   public:
-    VideoCodecConfigObserver()
-        : SendTest(kDefaultTimeoutMs),
-          FakeEncoder(Clock::GetRealTimeClock()),
-          num_initializations_(0) {
-      memset(&vp8_settings_, 0, sizeof(vp8_settings_));
-    }
+  VideoCodecConfigObserver test(kVideoCodecVP8, "VP8");
+  RunBaseTest(&test);
+}
 
-   private:
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
-      send_config->encoder_settings.encoder = this;
-      send_config->encoder_settings.payload_name = "VP8";
-
-      for (size_t i = 0; i < encoder_config->streams.size(); ++i) {
-        encoder_config->streams[i].temporal_layer_thresholds_bps.resize(
-            kNumberOfTemporalLayers - 1);
-      }
-
-      encoder_config->encoder_specific_settings = &vp8_settings_;
-      encoder_config_ = *encoder_config;
-    }
-
-    virtual void OnStreamsCreated(
-        VideoSendStream* send_stream,
-        const std::vector& receive_streams) OVERRIDE {
-      stream_ = send_stream;
-    }
-
-    virtual int32_t InitEncode(const VideoCodec* config,
-                               int32_t number_of_cores,
-                               uint32_t max_payload_size) OVERRIDE {
-      EXPECT_EQ(kVideoCodecVP8, config->codecType);
-
-      // Check that the number of temporal layers has propagated properly to
-      // VideoCodec.
-      EXPECT_EQ(kNumberOfTemporalLayers,
-                config->codecSpecific.VP8.numberOfTemporalLayers);
-
-      for (unsigned char i = 0; i < config->numberOfSimulcastStreams; ++i) {
-        EXPECT_EQ(kNumberOfTemporalLayers,
-                  config->simulcastStream[i].numberOfTemporalLayers);
-      }
-
-      // Set expected temporal layers as they should have been set when
-      // reconfiguring the encoder and not match the set config.
-      vp8_settings_.numberOfTemporalLayers = kNumberOfTemporalLayers;
-      EXPECT_EQ(0,
-                memcmp(&config->codecSpecific.VP8,
-                       &vp8_settings_,
-                       sizeof(vp8_settings_)));
-      ++num_initializations_;
-      return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
-    }
-
-    virtual void PerformTest() OVERRIDE {
-      EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
-
-      vp8_settings_.denoisingOn = true;
-      stream_->ReconfigureVideoEncoder(encoder_config_);
-      EXPECT_EQ(2u, num_initializations_)
-          << "ReconfigureVideoEncoder did not reinitialize the encoder with "
-             "new encoder settings.";
-    }
-
-    int32_t Encode(const I420VideoFrame& input_image,
-                   const CodecSpecificInfo* codec_specific_info,
-                   const std::vector* frame_types) OVERRIDE {
-      // Silently skip the encode, FakeEncoder::Encode doesn't produce VP8.
-      return 0;
-    }
-
-    VideoCodecVP8 vp8_settings_;
-    size_t num_initializations_;
-    VideoSendStream* stream_;
-    VideoEncoderConfig encoder_config_;
-  } test;
+TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp9Config) {
+  VideoCodecConfigObserver test(kVideoCodecVP9, "VP9");
+  RunBaseTest(&test);
+}
 
+TEST_F(VideoSendStreamTest, EncoderSetupPropagatesH264Config) {
+  VideoCodecConfigObserver test(kVideoCodecH264, "H264");
   RunBaseTest(&test);
 }
 
 TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
-  class RtcpByeTest : public test::SendTest {
+  class RtcpSenderReportTest : public test::SendTest {
    public:
-    RtcpByeTest() : SendTest(kDefaultTimeoutMs), media_bytes_sent_(0) {}
+    RtcpSenderReportTest() : SendTest(kDefaultTimeoutMs),
+                             rtp_packets_sent_(0),
+                             media_bytes_sent_(0) {}
 
    private:
-    virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      ++rtp_packets_sent_;
       media_bytes_sent_ += length - header.headerLength - header.paddingLength;
       return SEND_PACKET;
     }
 
-    virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       RTCPUtility::RTCPParserV2 parser(packet, length, true);
       EXPECT_TRUE(parser.IsValid());
 
       RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
-      uint32_t sender_octet_count = 0;
       while (packet_type != RTCPUtility::kRtcpNotValidCode) {
         if (packet_type == RTCPUtility::kRtcpSrCode) {
-          sender_octet_count = parser.Packet().SR.SenderOctetCount;
-          EXPECT_EQ(sender_octet_count, media_bytes_sent_);
-          if (sender_octet_count > 0)
+          // Only compare sent media bytes if SenderPacketCount matches the
+          // number of sent rtp packets (a new rtp packet could be sent before
+          // the rtcp packet).
+          if (parser.Packet().SR.SenderOctetCount > 0 &&
+              parser.Packet().SR.SenderPacketCount == rtp_packets_sent_) {
+            EXPECT_EQ(media_bytes_sent_, parser.Packet().SR.SenderOctetCount);
             observation_complete_->Set();
+          }
         }
-
         packet_type = parser.Iterate();
       }
 
       return SEND_PACKET;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for RTCP sender report.";
     }
 
+    size_t rtp_packets_sent_;
     size_t media_bytes_sent_;
   } test;
 
@@ -1560,19 +1540,18 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
           test::FakeEncoder(Clock::GetRealTimeClock()) {}
 
    private:
-    virtual int32_t InitEncode(const VideoCodec* config,
-                               int32_t number_of_cores,
-                               uint32_t max_payload_size) {
+    int32_t InitEncode(const VideoCodec* config,
+                       int32_t number_of_cores,
+                       size_t max_payload_size) override {
       EXPECT_EQ(static_cast(kScreencastTargetBitrateKbps),
                 config->targetBitrate);
       observation_complete_->Set();
       return test::FakeEncoder::InitEncode(
           config, number_of_cores, max_payload_size);
     }
-    virtual void ModifyConfigs(
-        VideoSendStream::Config* send_config,
-        std::vector* receive_configs,
-        VideoEncoderConfig* encoder_config) OVERRIDE {
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
       EXPECT_EQ(1u, encoder_config->streams.size());
       EXPECT_TRUE(
@@ -1582,7 +1561,7 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
       encoder_config->content_type = VideoEncoderConfig::kScreenshare;
     }
 
-    virtual void PerformTest() OVERRIDE {
+    void PerformTest() override {
       EXPECT_EQ(kEventSignaled, Wait())
           << "Timed out while waiting for the encoder to be initialized.";
     }
@@ -1590,4 +1569,189 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
 
   RunBaseTest(&test);
 }
+
+TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
+  // These are chosen to be "kind of odd" to not be accidentally checked against
+  // default values.
+  static const int kMinBitrateKbps = 137;
+  static const int kStartBitrateKbps = 345;
+  static const int kLowerMaxBitrateKbps = 312;
+  static const int kMaxBitrateKbps = 413;
+  static const int kIncreasedStartBitrateKbps = 451;
+  static const int kIncreasedMaxBitrateKbps = 597;
+  class EncoderBitrateThresholdObserver : public test::SendTest,
+                                          public test::FakeEncoder {
+   public:
+    EncoderBitrateThresholdObserver()
+        : SendTest(kDefaultTimeoutMs),
+          FakeEncoder(Clock::GetRealTimeClock()),
+          num_initializations_(0) {}
+
+   private:
+    int32_t InitEncode(const VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize) override {
+      if (num_initializations_ == 0) {
+        EXPECT_EQ(static_cast(kMinBitrateKbps),
+                  codecSettings->minBitrate);
+        EXPECT_EQ(static_cast(kStartBitrateKbps),
+                  codecSettings->startBitrate);
+        EXPECT_EQ(static_cast(kMaxBitrateKbps),
+                  codecSettings->maxBitrate);
+        observation_complete_->Set();
+      } else if (num_initializations_ == 1) {
+        EXPECT_EQ(static_cast(kLowerMaxBitrateKbps),
+                  codecSettings->maxBitrate);
+        // The start bitrate should be kept (-1) and capped to the max bitrate.
+        // Since this is not an end-to-end call no receiver should have been
+        // returning a REMB that could lower this estimate.
+        EXPECT_EQ(codecSettings->startBitrate, codecSettings->maxBitrate);
+      } else if (num_initializations_ == 2) {
+        EXPECT_EQ(static_cast(kIncreasedMaxBitrateKbps),
+                  codecSettings->maxBitrate);
+        EXPECT_EQ(static_cast(kIncreasedStartBitrateKbps),
+                  codecSettings->startBitrate);
+      }
+      ++num_initializations_;
+      return FakeEncoder::InitEncode(codecSettings, numberOfCores,
+                                     maxPayloadSize);
+    }
+
+    Call::Config GetSenderCallConfig() override {
+      Call::Config config(SendTransport());
+      config.bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
+      config.bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
+      config.bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;
+      return config;
+    }
+
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      send_config->encoder_settings.encoder = this;
+      // Set bitrates lower/higher than min/max to make sure they are properly
+      // capped.
+      encoder_config->streams.front().min_bitrate_bps = kMinBitrateKbps * 1000;
+      encoder_config->streams.front().max_bitrate_bps = kMaxBitrateKbps * 1000;
+      encoder_config_ = *encoder_config;
+    }
+
+    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
+      call_ = sender_call;
+    }
+
+    void OnStreamsCreated(
+        VideoSendStream* send_stream,
+        const std::vector& receive_streams) override {
+      send_stream_ = send_stream;
+    }
+
+    void PerformTest() override {
+      Call::Config::BitrateConfig bitrate_config;
+      bitrate_config.start_bitrate_bps = kIncreasedStartBitrateKbps * 1000;
+      bitrate_config.max_bitrate_bps = kIncreasedMaxBitrateKbps * 1000;
+      call_->SetBitrateConfig(bitrate_config);
+      EXPECT_EQ(kEventSignaled, Wait())
+          << "Timed out while waiting encoder to be configured.";
+      encoder_config_.streams[0].min_bitrate_bps = 0;
+      encoder_config_.streams[0].max_bitrate_bps = kLowerMaxBitrateKbps * 1000;
+      send_stream_->ReconfigureVideoEncoder(encoder_config_);
+      EXPECT_EQ(2, num_initializations_)
+          << "Encoder should have been reconfigured with the new value.";
+      encoder_config_.streams[0].target_bitrate_bps =
+          encoder_config_.streams[0].min_bitrate_bps;
+      encoder_config_.streams[0].max_bitrate_bps =
+          kIncreasedMaxBitrateKbps * 1000;
+      send_stream_->ReconfigureVideoEncoder(encoder_config_);
+      EXPECT_EQ(3, num_initializations_)
+          << "Encoder should have been reconfigured with the new value.";
+    }
+
+    int num_initializations_;
+    webrtc::Call* call_;
+    webrtc::VideoSendStream* send_stream_;
+    webrtc::VideoEncoderConfig encoder_config_;
+  } test;
+
+  RunBaseTest(&test);
+}
+
+TEST_F(VideoSendStreamTest, ReportsSentResolution) {
+  static const size_t kNumStreams = 3;
+  // Unusual resolutions to make sure that they are the ones being reported.
+  static const struct {
+    int width;
+    int height;
+  } kEncodedResolution[kNumStreams] = {
+      {241, 181}, {300, 121}, {121, 221}};
+  class ScreencastTargetBitrateTest : public test::SendTest,
+                                      public test::FakeEncoder {
+   public:
+    ScreencastTargetBitrateTest()
+        : SendTest(kDefaultTimeoutMs),
+          test::FakeEncoder(Clock::GetRealTimeClock()) {}
+
+   private:
+    int32_t Encode(const I420VideoFrame& input_image,
+                   const CodecSpecificInfo* codecSpecificInfo,
+                   const std::vector* frame_types) override {
+      CodecSpecificInfo specifics;
+      memset(&specifics, 0, sizeof(specifics));
+      specifics.codecType = kVideoCodecGeneric;
+
+      uint8_t buffer[16] = {0};
+      EncodedImage encoded(buffer, sizeof(buffer), sizeof(buffer));
+      encoded._timeStamp = input_image.timestamp();
+      encoded.capture_time_ms_ = input_image.render_time_ms();
+
+      for (size_t i = 0; i < kNumStreams; ++i) {
+        specifics.codecSpecific.generic.simulcast_idx = static_cast(i);
+        encoded._frameType = (*frame_types)[i];
+        encoded._encodedWidth = kEncodedResolution[i].width;
+        encoded._encodedHeight = kEncodedResolution[i].height;
+        DCHECK(callback_ != nullptr);
+        if (callback_->Encoded(encoded, &specifics, nullptr) != 0)
+          return -1;
+      }
+
+      observation_complete_->Set();
+      return 0;
+    }
+    void ModifyConfigs(VideoSendStream::Config* send_config,
+                       std::vector* receive_configs,
+                       VideoEncoderConfig* encoder_config) override {
+      send_config->encoder_settings.encoder = this;
+      EXPECT_EQ(kNumStreams, encoder_config->streams.size());
+    }
+
+    size_t GetNumStreams() const override { return kNumStreams; }
+
+    void PerformTest() override {
+      EXPECT_EQ(kEventSignaled, Wait())
+          << "Timed out while waiting for the encoder to send one frame.";
+      VideoSendStream::Stats stats = send_stream_->GetStats();
+
+      for (size_t i = 0; i < kNumStreams; ++i) {
+        ASSERT_TRUE(stats.substreams.find(kSendSsrcs[i]) !=
+                    stats.substreams.end())
+            << "No stats for SSRC: " << kSendSsrcs[i]
+            << ", stats should exist as soon as frames have been encoded.";
+        VideoSendStream::StreamStats ssrc_stats =
+            stats.substreams[kSendSsrcs[i]];
+        EXPECT_EQ(kEncodedResolution[i].width, ssrc_stats.width);
+        EXPECT_EQ(kEncodedResolution[i].height, ssrc_stats.height);
+      }
+    }
+
+    void OnStreamsCreated(
+        VideoSendStream* send_stream,
+        const std::vector& receive_streams) override {
+      send_stream_ = send_stream;
+    }
+
+    VideoSendStream* send_stream_;
+  } test;
+
+  RunBaseTest(&test);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_encoder.h b/media/webrtc/trunk/webrtc/video_encoder.h
index 2bf52f3c32..c933a33469 100644
--- a/media/webrtc/trunk/webrtc/video_encoder.h
+++ b/media/webrtc/trunk/webrtc/video_encoder.h
@@ -29,11 +29,9 @@ class EncodedImageCallback {
   virtual ~EncodedImageCallback() {}
 
   // Callback function which is called when an image has been encoded.
-  // TODO(pbos): Make encoded_image const or pointer. Remove default arguments.
-  virtual int32_t Encoded(
-      EncodedImage& encoded_image,
-      const CodecSpecificInfo* codec_specific_info = NULL,
-      const RTPFragmentationHeader* fragmentation = NULL) = 0;
+  virtual int32_t Encoded(const EncodedImage& encoded_image,
+                          const CodecSpecificInfo* codec_specific_info,
+                          const RTPFragmentationHeader* fragmentation) = 0;
 };
 
 class VideoEncoder {
@@ -68,7 +66,7 @@ class VideoEncoder {
   //                                  WEBRTC_VIDEO_CODEC_ERROR
   virtual int32_t InitEncode(const VideoCodec* codec_settings,
                              int32_t number_of_cores,
-                             uint32_t max_payload_size) = 0;
+                             size_t max_payload_size) = 0;
 
   // Register an encode complete callback object.
   //
@@ -109,7 +107,7 @@ class VideoEncoder {
   //          - rtt         : Round-trip time in milliseconds
   // Return value           : WEBRTC_VIDEO_CODEC_OK if OK
   //                          <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR
-  virtual int32_t SetChannelParameters(uint32_t packet_loss, int rtt) = 0;
+  virtual int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) = 0;
 
   // Inform the encoder about the new target bit rate.
   //
diff --git a/media/webrtc/trunk/webrtc/video_engine/BUILD.gn b/media/webrtc/trunk/webrtc/video_engine/BUILD.gn
index debcd2f8db..2e496417a5 100644
--- a/media/webrtc/trunk/webrtc/video_engine/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/video_engine/BUILD.gn
@@ -29,6 +29,10 @@ source_set("video_engine_core") {
     "encoder_state_feedback.h",
     "overuse_frame_detector.cc",
     "overuse_frame_detector.h",
+    "payload_router.cc",
+    "payload_router.h",
+    "report_block_stats.cc",
+    "report_block_stats.h",
     "stream_synchronization.cc",
     "stream_synchronization.h",
     "vie_base_impl.cc",
@@ -99,18 +103,21 @@ source_set("video_engine_core") {
     cflags = [
       # TODO(jschuh): Bug 1348: fix size_t to int truncations.
       "/wd4267",  # size_t to int truncation.
+      # Bug 261.
+      "/wd4373",  # legacy warning for ignoring const / volatile in signatures.
     ]
   }
 
   deps = [
+    "..:webrtc_common",
     "../common_video",
     "../modules/bitrate_controller",
     "../modules/rtp_rtcp",
     "../modules/utility",
-    "../modules/video_capture",
+    "../modules/video_capture:video_capture_module",
     "../modules/video_coding",
     "../modules/video_processing",
-    "../modules/video_render",
+    "../modules/video_render:video_render_module",
     "../voice_engine",
     "../system_wrappers",
   ]
diff --git a/media/webrtc/trunk/webrtc/video_engine/OWNERS b/media/webrtc/trunk/webrtc/video_engine/OWNERS
index 7c5d371a7e..5e47b1ae06 100644
--- a/media/webrtc/trunk/webrtc/video_engine/OWNERS
+++ b/media/webrtc/trunk/webrtc/video_engine/OWNERS
@@ -1,6 +1,5 @@
 mflodman@webrtc.org
 stefan@webrtc.org
-mallinath@webrtc.org
 
 per-file *.isolate=kjellander@webrtc.org
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h b/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h
index e2585a4638..b45842709b 100644
--- a/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/browser_capture_impl.h
@@ -36,7 +36,7 @@ namespace webrtc {
                                   VideoCaptureCapability& capability) { return 0;};
 
     virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
-                                   VideoCaptureRotation& orientation) { return 0; }
+                                   VideoRotation& orientation) { return 0; }
 
     virtual int32_t GetBestMatchedCapability(const char* deviceUniqueIdUTF8,
                                              const VideoCaptureCapability& requested,
diff --git a/media/webrtc/trunk/webrtc/video_engine/call_stats.cc b/media/webrtc/trunk/webrtc/video_engine/call_stats.cc
index 05385c35ea..6e51eebe5a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/call_stats.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/call_stats.cc
@@ -17,23 +17,70 @@
 #include "webrtc/system_wrappers/interface/tick_util.h"
 
 namespace webrtc {
-
-// A rtt report is considered valid for this long.
-const int kRttTimeoutMs = 1500;
+namespace {
 // Time interval for updating the observers.
-const int kUpdateIntervalMs = 1000;
+const int64_t kUpdateIntervalMs = 1000;
+// Weight factor to apply to the average rtt.
+const float kWeightFactor = 0.3f;
+
+void RemoveOldReports(int64_t now, std::list* reports) {
+  // A rtt report is considered valid for this long.
+  const int64_t kRttTimeoutMs = 1500;
+  while (!reports->empty() &&
+         (now - reports->front().time) > kRttTimeoutMs) {
+    reports->pop_front();
+  }
+}
+
+int64_t GetMaxRttMs(std::list* reports) {
+  int64_t max_rtt_ms = 0;
+  for (std::list::const_iterator it = reports->begin();
+       it != reports->end(); ++it) {
+    max_rtt_ms = std::max(it->rtt, max_rtt_ms);
+  }
+  return max_rtt_ms;
+}
+
+int64_t GetAvgRttMs(std::list* reports) {
+  if (reports->empty()) {
+    return 0;
+  }
+  int64_t sum = 0;
+  for (std::list::const_iterator it = reports->begin();
+       it != reports->end(); ++it) {
+    sum += it->rtt;
+  }
+  return sum / reports->size();
+}
+
+void UpdateAvgRttMs(std::list* reports, int64_t* avg_rtt) {
+  uint32_t cur_rtt_ms = GetAvgRttMs(reports);
+  if (cur_rtt_ms == 0) {
+    // Reset.
+    *avg_rtt = 0;
+    return;
+  }
+  if (*avg_rtt == 0) {
+    // Initialize.
+    *avg_rtt = cur_rtt_ms;
+    return;
+  }
+  *avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
+}
+}  // namespace
 
 class RtcpObserver : public RtcpRttStats {
  public:
   explicit RtcpObserver(CallStats* owner) : owner_(owner) {}
   virtual ~RtcpObserver() {}
 
-  virtual void OnRttUpdate(uint32_t rtt) {
+  virtual void OnRttUpdate(int64_t rtt) {
     owner_->OnRttUpdate(rtt);
   }
 
-  virtual uint32_t LastProcessedRtt() const {
-    return owner_->last_processed_rtt_ms();
+  // Returns the average RTT.
+  virtual int64_t LastProcessedRtt() const {
+    return owner_->avg_rtt_ms();
   }
 
  private:
@@ -46,53 +93,45 @@ CallStats::CallStats()
     : crit_(CriticalSectionWrapper::CreateCriticalSection()),
       rtcp_rtt_stats_(new RtcpObserver(this)),
       last_process_time_(TickTime::MillisecondTimestamp()),
-      last_processed_rtt_ms_(0) {
+      max_rtt_ms_(0),
+      avg_rtt_ms_(0) {
 }
 
 CallStats::~CallStats() {
   assert(observers_.empty());
 }
 
-int32_t CallStats::TimeUntilNextProcess() {
+int64_t CallStats::TimeUntilNextProcess() {
   return last_process_time_ + kUpdateIntervalMs -
       TickTime::MillisecondTimestamp();
 }
 
 int32_t CallStats::Process() {
   CriticalSectionScoped cs(crit_.get());
-  if (TickTime::MillisecondTimestamp() < last_process_time_ + kUpdateIntervalMs)
+  int64_t now = TickTime::MillisecondTimestamp();
+  if (now < last_process_time_ + kUpdateIntervalMs)
     return 0;
 
-  // Remove invalid, as in too old, rtt values.
-  int64_t time_now = TickTime::MillisecondTimestamp();
-  while (!reports_.empty() && reports_.front().time + kRttTimeoutMs <
-         time_now) {
-    reports_.pop_front();
-  }
+  last_process_time_ = now;
 
-  // Find the max stored RTT.
-  uint32_t max_rtt = 0;
-  for (std::list::const_iterator it = reports_.begin();
-       it != reports_.end(); ++it) {
-    if (it->rtt > max_rtt)
-      max_rtt = it->rtt;
-  }
+  RemoveOldReports(now, &reports_);
+  max_rtt_ms_ = GetMaxRttMs(&reports_);
+  UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
 
-  // If there is a valid rtt, update all observers.
-  if (max_rtt > 0) {
+  // If there is a valid rtt, update all observers with the max rtt.
+  // TODO(asapersson): Consider changing this to report the average rtt.
+  if (max_rtt_ms_ > 0) {
     for (std::list::iterator it = observers_.begin();
          it != observers_.end(); ++it) {
-      (*it)->OnRttUpdate(max_rtt);
+      (*it)->OnRttUpdate(max_rtt_ms_);
     }
   }
-  last_processed_rtt_ms_ = max_rtt;
-  last_process_time_ = time_now;
   return 0;
 }
 
-uint32_t CallStats::last_processed_rtt_ms() const {
+int64_t CallStats::avg_rtt_ms() const {
   CriticalSectionScoped cs(crit_.get());
-  return last_processed_rtt_ms_;
+  return avg_rtt_ms_;
 }
 
 RtcpRttStats* CallStats::rtcp_rtt_stats() const {
@@ -120,10 +159,9 @@ void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) {
   }
 }
 
-void CallStats::OnRttUpdate(uint32_t rtt) {
+void CallStats::OnRttUpdate(int64_t rtt) {
   CriticalSectionScoped cs(crit_.get());
-  int64_t time_now = TickTime::MillisecondTimestamp();
-  reports_.push_back(RttTime(rtt, time_now));
+  reports_.push_back(RttTime(rtt, TickTime::MillisecondTimestamp()));
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/call_stats.h b/media/webrtc/trunk/webrtc/video_engine/call_stats.h
index 95f4fee71e..79a8af6267 100644
--- a/media/webrtc/trunk/webrtc/video_engine/call_stats.h
+++ b/media/webrtc/trunk/webrtc/video_engine/call_stats.h
@@ -14,8 +14,8 @@
 #include 
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -32,8 +32,8 @@ class CallStats : public Module {
   ~CallStats();
 
   // Implements Module, to use the process thread.
-  virtual int32_t TimeUntilNextProcess() OVERRIDE;
-  virtual int32_t Process() OVERRIDE;
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
 
   // Returns a RtcpRttStats to register at a statistics provider. The object
   // has the same lifetime as the CallStats instance.
@@ -43,28 +43,29 @@ class CallStats : public Module {
   void RegisterStatsObserver(CallStatsObserver* observer);
   void DeregisterStatsObserver(CallStatsObserver* observer);
 
- protected:
-  void OnRttUpdate(uint32_t rtt);
-
-  uint32_t last_processed_rtt_ms() const;
-
- private:
   // Helper struct keeping track of the time a rtt value is reported.
   struct RttTime {
-    RttTime(uint32_t new_rtt, int64_t rtt_time)
+    RttTime(int64_t new_rtt, int64_t rtt_time)
         : rtt(new_rtt), time(rtt_time) {}
-    const uint32_t rtt;
+    const int64_t rtt;
     const int64_t time;
   };
 
+ protected:
+  void OnRttUpdate(int64_t rtt);
+
+  int64_t avg_rtt_ms() const;
+
+ private:
   // Protecting all members.
-  scoped_ptr crit_;
+  rtc::scoped_ptr crit_;
   // Observer receiving statistics updates.
-  scoped_ptr rtcp_rtt_stats_;
+  rtc::scoped_ptr rtcp_rtt_stats_;
   // The last time 'Process' resulted in statistic update.
   int64_t last_process_time_;
   // The last RTT in the statistics update (zero if there is no valid estimate).
-  uint32_t last_processed_rtt_ms_;
+  int64_t max_rtt_ms_;
+  int64_t avg_rtt_ms_;
 
   // All Rtt reports within valid time interval, oldest first.
   std::list reports_;
diff --git a/media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc
index f504094ba2..0febbd0198 100644
--- a/media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/call_stats_unittest.cc
@@ -11,8 +11,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/video_engine/call_stats.h"
 
@@ -27,7 +27,7 @@ class MockStatsObserver : public CallStatsObserver {
   MockStatsObserver() {}
   virtual ~MockStatsObserver() {}
 
-  MOCK_METHOD1(OnRttUpdate, void(uint32_t));
+  MOCK_METHOD1(OnRttUpdate, void(int64_t));
 };
 
 class CallStatsTest : public ::testing::Test {
@@ -36,7 +36,7 @@ class CallStatsTest : public ::testing::Test {
     TickTime::UseFakeClock(12345);
     call_stats_.reset(new CallStats());
   }
-  scoped_ptr call_stats_;
+  rtc::scoped_ptr call_stats_;
 };
 
 TEST_F(CallStatsTest, AddAndTriggerCallback) {
@@ -44,21 +44,21 @@ TEST_F(CallStatsTest, AddAndTriggerCallback) {
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
   call_stats_->RegisterStatsObserver(&stats_observer);
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_EQ(0U, rtcp_rtt_stats->LastProcessedRtt());
+  EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
 
-  const uint32_t kRtt = 25;
+  const int64_t kRtt = 25;
   rtcp_rtt_stats->OnRttUpdate(kRtt);
   EXPECT_CALL(stats_observer, OnRttUpdate(kRtt))
       .Times(1);
   call_stats_->Process();
   EXPECT_EQ(kRtt, rtcp_rtt_stats->LastProcessedRtt());
 
-  const int kRttTimeOutMs = 1500 + 10;
+  const int64_t kRttTimeOutMs = 1500 + 10;
   TickTime::AdvanceFakeClock(kRttTimeOutMs);
   EXPECT_CALL(stats_observer, OnRttUpdate(_))
       .Times(0);
   call_stats_->Process();
-  EXPECT_EQ(0U, rtcp_rtt_stats->LastProcessedRtt());
+  EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
 
   call_stats_->DeregisterStatsObserver(&stats_observer);
 }
@@ -101,42 +101,42 @@ TEST_F(CallStatsTest, ProcessTime) {
 TEST_F(CallStatsTest, MultipleObservers) {
   MockStatsObserver stats_observer_1;
   call_stats_->RegisterStatsObserver(&stats_observer_1);
-  // Add the secondobserver twice, there should still be only one report to the
+  // Add the second observer twice, there should still be only one report to the
   // observer.
   MockStatsObserver stats_observer_2;
   call_stats_->RegisterStatsObserver(&stats_observer_2);
   call_stats_->RegisterStatsObserver(&stats_observer_2);
 
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
-  uint32_t rtt = 100;
-  rtcp_rtt_stats->OnRttUpdate(rtt);
+  const int64_t kRtt = 100;
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
 
   // Verify both observers are updated.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt))
       .Times(1);
-  EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt))
       .Times(1);
   call_stats_->Process();
 
   // Deregister the second observer and verify update is only sent to the first
   // observer.
   call_stats_->DeregisterStatsObserver(&stats_observer_2);
-  rtcp_rtt_stats->OnRttUpdate(rtt);
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt))
       .Times(1);
-  EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt))
       .Times(0);
   call_stats_->Process();
 
   // Deregister the first observer.
   call_stats_->DeregisterStatsObserver(&stats_observer_1);
-  rtcp_rtt_stats->OnRttUpdate(rtt);
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt))
       .Times(0);
-  EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt))
       .Times(0);
   call_stats_->Process();
 }
@@ -151,17 +151,17 @@ TEST_F(CallStatsTest, ChangeRtt) {
   TickTime::AdvanceFakeClock(1000);
 
   // Set a first value and verify the callback is triggered.
-  const uint32_t first_rtt = 100;
-  rtcp_rtt_stats->OnRttUpdate(first_rtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(first_rtt))
+  const int64_t kFirstRtt = 100;
+  rtcp_rtt_stats->OnRttUpdate(kFirstRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kFirstRtt))
       .Times(1);
   call_stats_->Process();
 
   // Increase rtt and verify the new value is reported.
   TickTime::AdvanceFakeClock(1000);
-  const uint32_t high_rtt = first_rtt + 20;
-  rtcp_rtt_stats->OnRttUpdate(high_rtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(high_rtt))
+  const int64_t kHighRtt = kFirstRtt + 20;
+  rtcp_rtt_stats->OnRttUpdate(kHighRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kHighRtt))
       .Times(1);
   call_stats_->Process();
 
@@ -169,20 +169,50 @@ TEST_F(CallStatsTest, ChangeRtt) {
   // rtt invalid. Report a lower rtt and verify the old/high value still is sent
   // in the callback.
   TickTime::AdvanceFakeClock(1000);
-  const uint32_t low_rtt = first_rtt - 20;
-  rtcp_rtt_stats->OnRttUpdate(low_rtt);
-  EXPECT_CALL(stats_observer, OnRttUpdate(high_rtt))
+  const int64_t kLowRtt = kFirstRtt - 20;
+  rtcp_rtt_stats->OnRttUpdate(kLowRtt);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kHighRtt))
       .Times(1);
   call_stats_->Process();
 
   // Advance time to make the high report invalid, the lower rtt should now be
   // in the callback.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(stats_observer, OnRttUpdate(low_rtt))
+  EXPECT_CALL(stats_observer, OnRttUpdate(kLowRtt))
       .Times(1);
   call_stats_->Process();
 
   call_stats_->DeregisterStatsObserver(&stats_observer);
 }
 
+TEST_F(CallStatsTest, LastProcessedRtt) {
+  MockStatsObserver stats_observer;
+  call_stats_->RegisterStatsObserver(&stats_observer);
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+  TickTime::AdvanceFakeClock(1000);
+
+  // Set a first values and verify that LastProcessedRtt initially returns the
+  // average rtt.
+  const int64_t kRttLow = 10;
+  const int64_t kRttHigh = 30;
+  const int64_t kAvgRtt = 20;
+  rtcp_rtt_stats->OnRttUpdate(kRttLow);
+  rtcp_rtt_stats->OnRttUpdate(kRttHigh);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kRttHigh))
+      .Times(1);
+  call_stats_->Process();
+  EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
+
+  // Update values and verify LastProcessedRtt.
+  TickTime::AdvanceFakeClock(1000);
+  rtcp_rtt_stats->OnRttUpdate(kRttLow);
+  rtcp_rtt_stats->OnRttUpdate(kRttHigh);
+  EXPECT_CALL(stats_observer, OnRttUpdate(kRttHigh))
+      .Times(1);
+  call_stats_->Process();
+  EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
+
+  call_stats_->DeregisterStatsObserver(&stats_observer);
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc
index 31861b53cf..0c0b69f1c8 100644
--- a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.cc
@@ -124,7 +124,7 @@ int32_t ScreenDeviceInfoImpl::GetBestMatchedCapability(const char* deviceUniqueI
 }
 
 int32_t ScreenDeviceInfoImpl::GetOrientation(const char* deviceUniqueIdUTF8,
-                                             VideoCaptureRotation& orientation) {
+                                             VideoRotation& orientation) {
   return 0;
 }
 
@@ -216,7 +216,7 @@ int32_t AppDeviceInfoImpl::GetBestMatchedCapability(
 }
 
 int32_t AppDeviceInfoImpl::GetOrientation(const char* deviceUniqueIdUTF8,
-                                          VideoCaptureRotation& orientation) {
+                                          VideoRotation& orientation) {
   return 0;
 }
 
@@ -320,7 +320,7 @@ int32_t WindowDeviceInfoImpl::GetBestMatchedCapability(const char* deviceUniqueI
 }
 
 int32_t WindowDeviceInfoImpl::GetOrientation(const char* deviceUniqueIdUTF8,
-                                             VideoCaptureRotation& orientation) {
+                                             VideoRotation& orientation) {
   return 0;
 }
 
@@ -355,11 +355,6 @@ const char* DesktopCaptureImpl::CurrentDeviceName() const {
   return _deviceUniqueId.c_str();
 }
 
-int32_t DesktopCaptureImpl::ChangeUniqueId(const int32_t id) {
-  _id = id;
-  return 0;
-}
-
 int32_t DesktopCaptureImpl::Init(const char* uniqueId,
                                  const CaptureDeviceType type) {
   DesktopCaptureOptions options = DesktopCaptureOptions::CreateDefault();
@@ -407,11 +402,11 @@ int32_t DesktopCaptureImpl::Init(const char* uniqueId,
 }
 
 // returns the number of milliseconds until the module want a worker thread to call Process
-int32_t DesktopCaptureImpl::TimeUntilNextProcess() {
+int64_t DesktopCaptureImpl::TimeUntilNextProcess() {
   CriticalSectionScoped cs(&_callBackCs);
 
-  int32_t timeToNormalProcess = kProcessInterval
-    - (int32_t)((TickTime::Now() - _lastProcessTime).Milliseconds());
+  int64_t timeToNormalProcess = 300
+    - (int64_t)((TickTime::Now() - _lastProcessTime).Milliseconds());
 
   return timeToNormalProcess;
 }
@@ -469,18 +464,19 @@ DesktopCaptureImpl::DesktopCaptureImpl(const int32_t id)
     _setCaptureDelay(0),
     _dataCallBack(NULL),
     _captureCallBack(NULL),
-  _lastProcessFrameCount(TickTime::Now()),
-  _rotateFrame(kRotateNone),
+    _lastProcessFrameCount(TickTime::Now()),
+    _rotateFrame(kVideoRotation_0),
   last_capture_time_(TickTime::MillisecondTimestamp()),
   delta_ntp_internal_ms_(
                          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() -
                          TickTime::MillisecondTimestamp()),
-  time_event_(*EventWrapper::Create()),
+  time_event_(EventWrapper::Create()),
 #if defined(_WIN32)
-  capturer_thread_(*ThreadWrapper::CreateUIThread(Run, this, kHighPriority, "ScreenCaptureThread")) {
+  capturer_thread_(ThreadWrapper::CreateUIThread(Run, this, "ScreenCaptureThread")) {
 #else
-  capturer_thread_(*ThreadWrapper::CreateThread(Run, this, kHighPriority, "ScreenCaptureThread")) {
+  capturer_thread_(ThreadWrapper::CreateThread(Run, this, "ScreenCaptureThread")) {
 #endif
+  capturer_thread_->SetPriority(kHighPriority);
   _requestedCapability.width = kDefaultWidth;
   _requestedCapability.height = kDefaultHeight;
   _requestedCapability.maxFPS = 30;
@@ -490,10 +486,8 @@ DesktopCaptureImpl::DesktopCaptureImpl(const int32_t id)
 }
 
 DesktopCaptureImpl::~DesktopCaptureImpl() {
-  time_event_.Set();
-  capturer_thread_.Stop();
-  delete &time_event_;
-  delete &capturer_thread_;
+  time_event_->Set();
+  capturer_thread_->Stop();
 
   DeRegisterCaptureDataCallback();
   DeRegisterCaptureCallback();
@@ -579,7 +573,7 @@ int32_t DesktopCaptureImpl::DeliverCapturedFrame(I420VideoFrame& captureFrame,
 
 // Copied from VideoCaptureImpl::IncomingFrame. See Bug 1038324
 int32_t DesktopCaptureImpl::IncomingFrame(uint8_t* videoFrame,
-                                          int32_t videoFrameLength,
+                                          size_t videoFrameLength,
                                           const VideoCaptureCapability& frameInfo,
                                           int64_t captureTime/*=0*/)
 {
@@ -614,7 +608,7 @@ int32_t DesktopCaptureImpl::IncomingFrame(uint8_t* videoFrame,
     int target_width = width;
     int target_height = abs(height);
     // Rotating resolution when for 90/270 degree rotations.
-    if (_rotateFrame == kRotate90 || _rotateFrame == kRotate270)  {
+    if (_rotateFrame == kVideoRotation_90 || _rotateFrame == kVideoRotation_270)  {
       target_height = width;
       target_width = abs(height);
     }
@@ -719,50 +713,15 @@ int32_t DesktopCaptureImpl::IncomingFrame(uint8_t* videoFrame,
   return 0;
 }
 
-int32_t DesktopCaptureImpl::IncomingI420VideoFrame(I420VideoFrame* video_frame, int64_t captureTime) {
-
-  CriticalSectionScoped cs(&_callBackCs);
-  int stride_y = video_frame->stride(kYPlane);
-  int stride_u = video_frame->stride(kUPlane);
-  int stride_v = video_frame->stride(kVPlane);
-  int size_y = video_frame->height() * stride_y;
-  int size_u = stride_u * ((video_frame->height() + 1) / 2);
-  int size_v =  stride_v * ((video_frame->height() + 1) / 2);
-  int ret = _captureFrame.CreateFrame(size_y, video_frame->buffer(kYPlane),
-                                      size_u, video_frame->buffer(kUPlane),
-                                      size_v, video_frame->buffer(kVPlane),
-                                      video_frame->width(), video_frame->height(),
-                                      stride_y, stride_u, stride_v);
-
-  if (ret < 0) {
-    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
-                 "Failed to create I420VideoFrame");
-    return -1;
-  }
-
-  DeliverCapturedFrame(_captureFrame, captureTime);
-
+int32_t DesktopCaptureImpl::SetCaptureRotation(VideoRotation rotation) {
+  CriticalSectionScoped cs(&_apiCs);
+  CriticalSectionScoped cs2(&_callBackCs);
+  _rotateFrame = rotation;
   return 0;
 }
 
-int32_t DesktopCaptureImpl::SetCaptureRotation(VideoCaptureRotation rotation) {
-  CriticalSectionScoped cs(&_apiCs);
-  CriticalSectionScoped cs2(&_callBackCs);
-  switch (rotation) {
-  case kCameraRotate0:
-    _rotateFrame = kRotateNone;
-    break;
-  case kCameraRotate90:
-    _rotateFrame = kRotate90;
-    break;
-  case kCameraRotate180:
-    _rotateFrame = kRotate180;
-    break;
-  case kCameraRotate270:
-    _rotateFrame = kRotate270;
-    break;
-  }
-  return 0;
+bool DesktopCaptureImpl::SetApplyRotation(bool enable) {
+  return true;
 }
 
 void DesktopCaptureImpl::EnableFrameRateCallback(const bool enable) {
@@ -817,15 +776,14 @@ uint32_t DesktopCaptureImpl::CalculateFrameRate(const TickTime& now) {
 
 int32_t DesktopCaptureImpl::StartCapture(const VideoCaptureCapability& capability) {
   _requestedCapability = capability;
-  desktop_capturer_cursor_composer_->Start(this);
-  unsigned int t_id =0;
-  capturer_thread_.Start(t_id);
-
 #if defined(_WIN32)
   uint32_t maxFPSNeeded = 1000/_requestedCapability.maxFPS;
-  capturer_thread_.RequestCallbackTimer(maxFPSNeeded);
+  capturer_thread_->RequestCallbackTimer(maxFPSNeeded);
 #endif
 
+  desktop_capturer_cursor_composer_->Start(this);
+  capturer_thread_->Start();
+
   return 0;
 }
 
@@ -852,7 +810,7 @@ void DesktopCaptureImpl::OnCaptureCompleted(DesktopFrame* frame) {
   // combine cursor in frame
   // Latest WebRTC already support it by DesktopFrameWithCursor/DesktopAndCursorComposer.
 
-  int32_t videoFrameLength = frameInfo.width * frameInfo.height * DesktopFrame::kBytesPerPixel;
+  size_t videoFrameLength = frameInfo.width * frameInfo.height * DesktopFrame::kBytesPerPixel;
   IncomingFrame(videoFrame, videoFrameLength, frameInfo);
   delete frame; //handled it, so we need delete it
 }
@@ -878,7 +836,7 @@ void DesktopCaptureImpl::process() {
   const uint32_t maxFPSNeeded = 1000/_requestedCapability.maxFPS;
   const float sleepTimeFactor = (100.0f / kMaxDesktopCaptureCpuUsage) - 1.0f;
   const uint32_t sleepTime = sleepTimeFactor * processTime;
-  time_event_.Wait(std::max(maxFPSNeeded, sleepTime));
+  time_event_->Wait(std::max(maxFPSNeeded, sleepTime));
 #endif
 }
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h
index 681a67ed85..dfa7a6465c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/desktop_capture_impl.h
@@ -71,10 +71,10 @@ public:
                                            const VideoCaptureCapability& requested,
                                            VideoCaptureCapability& resulting);
   virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
-                                 VideoCaptureRotation& orientation);
+                                 VideoRotation& orientation);
 protected:
   int32_t _id;
-  scoped_ptr desktop_device_info_;
+  rtc::scoped_ptr desktop_device_info_;
 
 };
 
@@ -109,10 +109,10 @@ public:
                                            const VideoCaptureCapability& requested,
                                            VideoCaptureCapability& resulting);
   virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
-                                 VideoCaptureRotation& orientation);
+                                 VideoRotation& orientation);
 protected:
   int32_t _id;
-  scoped_ptr desktop_device_info_;
+  rtc::scoped_ptr desktop_device_info_;
 };
 
 class WindowDeviceInfoImpl : public VideoCaptureModule::DeviceInfo {
@@ -146,10 +146,10 @@ public:
                                            const VideoCaptureCapability& requested,
                                            VideoCaptureCapability& resulting);
   virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
-                                 VideoCaptureRotation& orientation);
+                                 VideoRotation& orientation);
 protected:
   int32_t _id;
-  scoped_ptr desktop_device_info_;
+  rtc::scoped_ptr desktop_device_info_;
 
 };
 
@@ -168,48 +168,45 @@ public:
   static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(const int32_t id, const CaptureDeviceType type);
 
   int32_t Init(const char* uniqueId, const CaptureDeviceType type);
-  // Implements Module declared functions.
-  virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
 
   //Call backs
-  virtual void RegisterCaptureDataCallback(VideoCaptureDataCallback& dataCallback) OVERRIDE;
-  virtual void DeRegisterCaptureDataCallback() OVERRIDE;
-  virtual void RegisterCaptureCallback(VideoCaptureFeedBack& callBack) OVERRIDE;
-  virtual void DeRegisterCaptureCallback() OVERRIDE;
+  virtual void RegisterCaptureDataCallback(VideoCaptureDataCallback& dataCallback) override;
+  virtual void DeRegisterCaptureDataCallback() override;
+  virtual void RegisterCaptureCallback(VideoCaptureFeedBack& callBack) override;
+  virtual void DeRegisterCaptureCallback() override;
 
-  virtual void SetCaptureDelay(int32_t delayMS) OVERRIDE;
-  virtual int32_t CaptureDelay() OVERRIDE;
-  virtual int32_t SetCaptureRotation(VideoCaptureRotation rotation) OVERRIDE;
+  virtual void SetCaptureDelay(int32_t delayMS) override;
+  virtual int32_t CaptureDelay() override;
+  virtual int32_t SetCaptureRotation(VideoRotation rotation) override;
+  virtual bool SetApplyRotation(bool enable) override;
+  virtual bool GetApplyRotation() { return true; }
 
-  virtual void EnableFrameRateCallback(const bool enable) OVERRIDE;
-  virtual void EnableNoPictureAlarm(const bool enable) OVERRIDE;
+  virtual void EnableFrameRateCallback(const bool enable) override;
+  virtual void EnableNoPictureAlarm(const bool enable) override;
 
-  virtual const char* CurrentDeviceName() const OVERRIDE;
+  virtual const char* CurrentDeviceName() const override;
 
   // Module handling
-  virtual int32_t TimeUntilNextProcess() OVERRIDE;
-  virtual int32_t Process() OVERRIDE;
+  virtual int64_t TimeUntilNextProcess() override;
+  virtual int32_t Process() override;
 
   // Implement VideoCaptureExternal
   // |capture_time| must be specified in the NTP time format in milliseconds.
   virtual int32_t IncomingFrame(uint8_t* videoFrame,
-                                int32_t videoFrameLength,
+                                size_t videoFrameLength,
                                 const VideoCaptureCapability& frameInfo,
-                                int64_t captureTime = 0) OVERRIDE;
-  virtual int32_t IncomingI420VideoFrame(
-                                         I420VideoFrame* video_frame,
-                                         int64_t captureTime = 0) OVERRIDE;
+                                int64_t captureTime = 0) override;
 
   // Platform dependent
-  virtual int32_t StartCapture(const VideoCaptureCapability& capability) OVERRIDE;
-  virtual int32_t StopCapture() OVERRIDE;
-  virtual bool CaptureStarted() OVERRIDE;
-  virtual int32_t CaptureSettings(VideoCaptureCapability& settings) OVERRIDE;
-  VideoCaptureEncodeInterface* GetEncodeInterface(const VideoCodec& codec) OVERRIDE { return NULL; }
+  virtual int32_t StartCapture(const VideoCaptureCapability& capability) override;
+  virtual int32_t StopCapture() override;
+  virtual bool CaptureStarted() override;
+  virtual int32_t CaptureSettings(VideoCaptureCapability& settings) override;
+  VideoCaptureEncodeInterface* GetEncodeInterface(const VideoCodec& codec) override { return NULL; }
 
   //ScreenCapturer::Callback
-  virtual SharedMemory* CreateSharedMemory(size_t size) OVERRIDE;
-  virtual void OnCaptureCompleted(DesktopFrame* frame) OVERRIDE;
+  virtual SharedMemory* CreateSharedMemory(size_t size) override;
+  virtual void OnCaptureCompleted(DesktopFrame* frame) override;
 
 protected:
   DesktopCaptureImpl(const int32_t id);
@@ -243,10 +240,9 @@ private:
 
   TickTime _lastProcessFrameCount;
   TickTime _incomingFrameTimes[kFrameRateCountHistorySize];// timestamp for local captured frames
-  VideoRotationMode _rotateFrame; //Set if the frame should be rotated by the capture module.
+  VideoRotation _rotateFrame; //Set if the frame should be rotated by the capture module.
 
   I420VideoFrame _captureFrame;
-  VideoFrame _capture_encoded_frame;
 
   // Used to make sure incoming timestamp is increasing for every frame.
   int64_t last_capture_time_;
@@ -262,9 +258,9 @@ public:
   void process();
 
 private:
-  scoped_ptr desktop_capturer_cursor_composer_;
-  EventWrapper& time_event_;
-  ThreadWrapper&  capturer_thread_;
+  rtc::scoped_ptr desktop_capturer_cursor_composer_;
+  rtc::scoped_ptr time_event_;
+  rtc::scoped_ptr capturer_thread_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h b/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h
index a58532ce7f..998793a254 100644
--- a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h
+++ b/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback.h
@@ -17,7 +17,7 @@
 #include 
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -54,10 +54,10 @@ class EncoderStateFeedback {
  private:
   typedef std::map SsrcEncoderMap;
 
-  scoped_ptr crit_;
+  rtc::scoped_ptr crit_;
 
   // Instance registered at the class requesting new key frames.
-  scoped_ptr observer_;
+  rtc::scoped_ptr observer_;
 
   // Maps a unique ssrc to the given encoder.
   SsrcEncoderMap encoders_;
diff --git a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc
index 4e15752fde..eab33d407a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/encoder_state_feedback_unittest.cc
@@ -15,10 +15,13 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
+#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/vie_encoder.h"
 
 using ::testing::NiceMock;
@@ -27,8 +30,8 @@ namespace webrtc {
 
 class MockVieEncoder : public ViEEncoder {
  public:
-  explicit MockVieEncoder(ProcessThread* process_thread)
-      : ViEEncoder(1, 1, 1, config_, *process_thread, NULL) {}
+  explicit MockVieEncoder(ProcessThread* process_thread, PacedSender* pacer)
+      : ViEEncoder(1, 1, config_, *process_thread, pacer, NULL, NULL, false) {}
   ~MockVieEncoder() {}
 
   MOCK_METHOD1(OnReceivedIntraFrameRequest,
@@ -45,17 +48,26 @@ class MockVieEncoder : public ViEEncoder {
 
 class VieKeyRequestTest : public ::testing::Test {
  protected:
+  VieKeyRequestTest()
+      : pacer_(Clock::GetRealTimeClock(),
+               &router_,
+               BitrateController::kDefaultStartBitrateKbps,
+               PacedSender::kDefaultPaceMultiplier *
+                   BitrateController::kDefaultStartBitrateKbps,
+               0) {}
   virtual void SetUp() {
     process_thread_.reset(new NiceMock);
     encoder_state_feedback_.reset(new EncoderStateFeedback());
   }
-  scoped_ptr process_thread_;
-  scoped_ptr encoder_state_feedback_;
+  rtc::scoped_ptr process_thread_;
+  rtc::scoped_ptr encoder_state_feedback_;
+  PacketRouter router_;
+  PacedSender pacer_;
 };
 
 TEST_F(VieKeyRequestTest, CreateAndTriggerRequests) {
   const int ssrc = 1234;
-  MockVieEncoder encoder(process_thread_.get());
+  MockVieEncoder encoder(process_thread_.get(), &pacer_);
   EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
 
   EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(ssrc))
@@ -83,8 +95,8 @@ TEST_F(VieKeyRequestTest, CreateAndTriggerRequests) {
 TEST_F(VieKeyRequestTest, MultipleEncoders) {
   const int ssrc_1 = 1234;
   const int ssrc_2 = 5678;
-  MockVieEncoder encoder_1(process_thread_.get());
-  MockVieEncoder encoder_2(process_thread_.get());
+  MockVieEncoder encoder_1(process_thread_.get(), &pacer_);
+  MockVieEncoder encoder_2(process_thread_.get(), &pacer_);
   EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc_1, &encoder_1));
   EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc_2, &encoder_2));
 
@@ -129,7 +141,7 @@ TEST_F(VieKeyRequestTest, MultipleEncoders) {
 
 TEST_F(VieKeyRequestTest, AddTwiceError) {
   const int ssrc = 1234;
-  MockVieEncoder encoder(process_thread_.get());
+  MockVieEncoder encoder(process_thread_.get(), &pacer_);
   EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
   EXPECT_FALSE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
   encoder_state_feedback_->RemoveEncoder(&encoder);
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h
index 203a5c0ab1..4e0b15355b 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_base.h
@@ -29,6 +29,8 @@ namespace webrtc {
 
 class Config;
 class VoiceEngine;
+class ReceiveStatisticsProxy;
+class SendStatisticsProxy;
 
 // CpuOveruseObserver is called when a system overuse is detected and
 // VideoEngine cannot keep up the encoding frequency.
@@ -45,12 +47,12 @@ class CpuOveruseObserver {
 
 struct CpuOveruseOptions {
   CpuOveruseOptions()
-      : enable_capture_jitter_method(true),
+      : enable_capture_jitter_method(false),
         low_capture_jitter_threshold_ms(20.0f),
         high_capture_jitter_threshold_ms(30.0f),
-        enable_encode_usage_method(false),
-        low_encode_usage_threshold_percent(60),
-        high_encode_usage_threshold_percent(90),
+        enable_encode_usage_method(true),
+        low_encode_usage_threshold_percent(55),
+        high_encode_usage_threshold_percent(85),
         low_encode_time_rsd_threshold(-1),
         high_encode_time_rsd_threshold(-1),
         enable_extended_processing_usage(true),
@@ -114,7 +116,6 @@ struct CpuOveruseMetrics {
       : capture_jitter_ms(-1),
         avg_encode_time_ms(-1),
         encode_usage_percent(-1),
-        encode_rsd(-1),
         capture_queue_delay_ms_per_s(-1) {}
 
   int capture_jitter_ms;  // The current estimated jitter in ms based on
@@ -122,14 +123,18 @@ struct CpuOveruseMetrics {
   int avg_encode_time_ms;   // The average encode time in ms.
   int encode_usage_percent; // The average encode time divided by the average
                             // time difference between incoming captured frames.
-  // TODO(asapersson): Remove metric, not used.
-  int encode_rsd;           // The relative std dev of encode time of frames.
   int capture_queue_delay_ms_per_s;  // The current time delay between an
                                      // incoming captured frame until the frame
                                      // is being processed. The delay is
                                      // expressed in ms delay per second.
 };
 
+class CpuOveruseMetricsObserver {
+ public:
+  virtual ~CpuOveruseMetricsObserver() {}
+  virtual void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) = 0;
+};
+
 class WEBRTC_DLLEXPORT VideoEngine {
  public:
   // Creates a VideoEngine object, which can then be used to acquire sub‐APIs.
@@ -191,6 +196,9 @@ class WEBRTC_DLLEXPORT ViEBase {
   virtual int CreateChannel(int& video_channel,
                             int original_channel) = 0;
 
+  virtual int CreateChannelWithoutDefaultEncoder(int& video_channel,
+                                                 int original_channel) = 0;
+
   // Creates a new channel grouped together with |original_channel|. The channel
   // can only receive video and it is assumed the remote end-point is the same
   // as for |original_channel|.
@@ -212,6 +220,9 @@ class WEBRTC_DLLEXPORT ViEBase {
 
   // Gets cpu overuse measures.
   virtual int GetCpuOveruseMetrics(int channel, CpuOveruseMetrics* metrics) = 0;
+  virtual void RegisterCpuOveruseMetricsObserver(
+      int channel,
+      CpuOveruseMetricsObserver* observer) = 0;
 
   // Registers a callback which is called when send-side delay statistics has
   // been updated.
@@ -251,6 +262,14 @@ class WEBRTC_DLLEXPORT ViEBase {
   // Returns the last VideoEngine error code.
   virtual int LastError() = 0;
 
+  virtual void RegisterSendStatisticsProxy(
+      int channel,
+      SendStatisticsProxy* send_statistics_proxy) = 0;
+
+  virtual void RegisterReceiveStatisticsProxy(
+      int channel,
+      ReceiveStatisticsProxy* receive_statistics_proxy) = 0;
+
  protected:
   ViEBase() {}
   virtual ~ViEBase() {}
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h
index 8150725a88..3c722b0af2 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_capture.h
@@ -20,6 +20,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/interface/i420_video_frame.h"
+#include "webrtc/common_video/rotation.h"
 
 namespace webrtc {
 
@@ -74,37 +75,6 @@ enum CaptureAlarm {
   AlarmCleared = 1
 };
 
-enum RotateCapturedFrame {
-  RotateCapturedFrame_0 = 0,
-  RotateCapturedFrame_90 = 90,
-  RotateCapturedFrame_180 = 180,
-  RotateCapturedFrame_270 = 270
-};
-
-struct ViEVideoFrameI420 {
-  ViEVideoFrameI420() {
-    y_plane = NULL;
-    u_plane = NULL;
-    v_plane = NULL;
-    y_pitch = 0;
-    u_pitch = 0;
-    v_pitch = 0;
-    width = 0;
-    height = 0;
-  }
-
-  unsigned char* y_plane;
-  unsigned char* u_plane;
-  unsigned char* v_plane;
-
-  int y_pitch;
-  int u_pitch;
-  int v_pitch;
-
-  unsigned short width;
-  unsigned short height;
-};
-
 // This class declares an abstract interface to be used when implementing
 // a user-defined capture device. This interface is not meant to be
 // implemented by the user. Instead, the user should call AllocateCaptureDevice
@@ -118,22 +88,7 @@ class WEBRTC_DLLEXPORT ViEExternalCapture {
 
   // This method is called by the user to deliver a new captured frame to
   // VideoEngine.
-  // |capture_time| must be specified in the NTP time format in milliseconds.
-  virtual int IncomingFrame(unsigned char* video_frame,
-                            unsigned int video_frame_length,
-                            unsigned short width,
-                            unsigned short height,
-                            RawVideoType video_type,
-                            unsigned long long capture_time = 0) = 0;
-
-  // This method is specifically for delivering a new captured I420 frame to
-  // VideoEngine.
-  // |capture_time| must be specified in the NTP time format in milliseconds.
-  virtual int IncomingFrameI420(
-      const ViEVideoFrameI420& video_frame,
-      unsigned long long capture_time = 0) = 0;
-
-  virtual void SwapFrame(I420VideoFrame* frame) {}
+  virtual void IncomingFrame(const I420VideoFrame& frame) = 0;
 };
 
 // This class declares an abstract interface for a user defined observer. It is
@@ -217,8 +172,8 @@ class WEBRTC_DLLEXPORT ViECapture {
 
   // Rotates captured frames before encoding and sending.
   // Used on mobile devices with rotates cameras.
-  virtual int SetRotateCapturedFrames(const int capture_id,
-                                      const RotateCapturedFrame rotation) = 0;
+  virtual int SetVideoRotation(const int capture_id,
+                               const VideoRotation rotation) = 0;
 
   // This function sets the expected delay from when a video frame is captured
   // to when that frame is delivered to VideoEngine.
@@ -251,7 +206,7 @@ class WEBRTC_DLLEXPORT ViECapture {
   // order to display the frames correctly if the display is rotated in its
   // natural orientation.
   virtual int GetOrientation(const char* unique_id_utf8,
-                             RotateCapturedFrame& orientation) = 0;
+                             VideoRotation& orientation) = 0;
 
   // Enables brightness alarm detection and the brightness alarm callback.
   virtual int EnableBrightnessAlarm(const int capture_id,
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h
index bf4869892e..dbdbde91cb 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_codec.h
@@ -147,6 +147,8 @@ class WEBRTC_DLLEXPORT ViECodec {
   virtual int GetReceiveSideDelay(const int video_channel,
                                   int* delay_ms) const = 0;
 
+  // Current target bitrate for this channel.
+  virtual uint32_t GetLastObservedBitrateBps(int video_channel) const = 0;
   // Gets the bitrate targeted by the video codec rate control in kbit/s.
   virtual int GetCodecTargetBitrate(const int video_channel,
                                     unsigned int* bitrate) const = 0;
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h
index 536995ee0a..99018dbcd2 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_external_codec.h
@@ -34,7 +34,7 @@ class WEBRTC_DLLEXPORT ViEExternalCodec {
                                           const unsigned char pl_type) = 0;
 
   virtual int RegisterExternalReceiveCodec(const int video_channel,
-                                           const unsigned int pl_type,
+                                           const unsigned char pl_type,
                                            VideoDecoder* decoder,
                                            bool decoder_render = false,
                                            int render_delay = 0) = 0;
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h
index 8bb895fb58..adf0c199c9 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_image_process.h
@@ -32,7 +32,7 @@ class WEBRTC_DLLEXPORT ViEEffectFilter {
  public:
   // This method is called with an I420 video frame allowing the user to
   // modify the video frame.
-  virtual int Transform(int size,
+  virtual int Transform(size_t size,
                         unsigned char* frame_buffer,
                         int64_t ntp_time_ms,
                         unsigned int timestamp,
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h
index bb368182cf..e962e729aa 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_network.h
@@ -47,6 +47,11 @@ class WEBRTC_DLLEXPORT ViENetwork {
   // for all sub-API:s before the VideoEngine object can be safely deleted.
   virtual int Release() = 0;
 
+  virtual void SetBitrateConfig(int video_channel,
+                                int min_bitrate_bps,
+                                int start_bitrate_bps,
+                                int max_bitrate_bps) = 0;
+
   // Inform the engine about if the network adapter is currently transmitting
   // packets or not.
   virtual void SetNetworkTransmissionState(const int video_channel,
@@ -65,14 +70,14 @@ class WEBRTC_DLLEXPORT ViENetwork {
   // the RTP header and payload.
   virtual int ReceivedRTPPacket(const int video_channel,
                                 const void* data,
-                                const int length,
+                                const size_t length,
                                 const PacketTime& packet_time) = 0;
 
   // When using external transport for a channel, received RTCP packets should
   // be passed to VideoEngine using this function.
   virtual int ReceivedRTCPPacket(const int video_channel,
                                  const void* data,
-                                 const int length) = 0;
+                                 const size_t length) = 0;
 
   // This function sets the Maximum Transition Unit (MTU) for a channel. The
   // RTP packet will be packetized based on this MTU to optimize performance
@@ -82,17 +87,10 @@ class WEBRTC_DLLEXPORT ViENetwork {
   // Forward (audio) packet to bandwidth estimator for the given video channel,
   // for aggregated audio+video BWE.
   virtual int ReceivedBWEPacket(const int video_channel,
-      int64_t arrival_time_ms, int payload_size, const RTPHeader& header) {
+      int64_t arrival_time_ms, size_t payload_size, const RTPHeader& header) {
     return 0;
   }
 
-  // TODO(holmer): Remove the default implementation when this has been fixed
-  // in fakewebrtcvideoengine.cc.
-  virtual bool SetBandwidthEstimationConfig(int video_channel,
-                                            const webrtc::Config& config) {
-    return false;
-  }
-
  protected:
   ViENetwork() {}
   virtual ~ViENetwork() {}
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h
index 2c9b0f55b3..8d3f2b5a1a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_render.h
@@ -20,6 +20,7 @@
 
 namespace webrtc {
 
+class I420VideoFrame;
 class VideoEngine;
 class VideoRender;
 class VideoRenderCallback;
@@ -36,7 +37,7 @@ class ExternalRenderer {
 
   // This method is called when a new frame should be rendered.
   virtual int DeliverFrame(unsigned char* buffer,
-                           int buffer_size,
+                           size_t buffer_size,
                            // RTP timestamp in 90kHz.
                            uint32_t timestamp,
                            // NTP time of the capture time in local timebase
@@ -47,6 +48,9 @@ class ExternalRenderer {
                            // Handle of the underlying video frame.
                            void* handle) = 0;
 
+  // Alternative interface for I420 frames.
+  virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) = 0;
+
   // Returns true if the renderer supports textures. DeliverFrame can be called
   // with NULL |buffer| and non-NULL |handle|.
   virtual bool IsTextureSupported() = 0;
@@ -104,24 +108,11 @@ class ViERender {
                               const float right,
                               const float bottom) = 0;
 
-  // This function mirrors the rendered stream left and right or up and down.
-  virtual int MirrorRenderStream(const int render_id,
-                                 const bool enable,
-                                 const bool mirror_xaxis,
-                                 const bool mirror_yaxis) = 0;
-
   // External render.
   virtual int AddRenderer(const int render_id,
                           RawVideoType video_input_format,
                           ExternalRenderer* renderer) = 0;
 
-  // Propagating VideoRenderCallback down to the VideoRender module for new API.
-  // Contains default-implementation not to break code mocking this interface.
-  // (Ugly, but temporary.)
-  virtual int AddRenderCallback(int render_id, VideoRenderCallback* callback) {
-    return 0;
-  }
-
  protected:
   ViERender() {}
   virtual ~ViERender() {}
diff --git a/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h b/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h
index 5cd799225e..3898d5b1fe 100644
--- a/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h
+++ b/media/webrtc/trunk/webrtc/video_engine/include/vie_rtp_rtcp.h
@@ -69,25 +69,6 @@ class WEBRTC_DLLEXPORT ViERTPObserver {
   virtual ~ViERTPObserver() {}
 };
 
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterRTCPObserver() and
-// deregistered using DeregisterRTCPObserver().
-
-class WEBRTC_DLLEXPORT ViERTCPObserver {
- public:
-  // This method is called if a application-defined RTCP packet has been
-  // received.
-  virtual void OnApplicationDataReceived(
-      const int video_channel,
-      const unsigned char sub_type,
-      const unsigned int name,
-      const char* data,
-      const unsigned short data_length_in_bytes) = 0;
- protected:
-  virtual ~ViERTCPObserver() {}
-};
-
 struct SenderInfo;
 
 class WEBRTC_DLLEXPORT ViERTP_RTCP {
@@ -137,15 +118,6 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int SetRtxSendPayloadType(const int video_channel,
                                     const uint8_t payload_type) = 0;
 
-  // This enables sending redundant payloads when padding up the bitrate instead
-  // of sending dummy padding packets. This feature is off by default and will
-  // only have an effect if RTX is also enabled.
-  // TODO(holmer): Remove default implementation once this has been implemented
-  // in libjingle.
-  virtual int SetPadWithRedundantPayloads(int video_channel, bool enable) {
-    return 0;
-  }
-
   virtual int SetRtxReceivePayloadType(const int video_channel,
                                        const uint8_t payload_type) = 0;
 
@@ -285,6 +257,14 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
                                                bool enable,
                                                int id) = 0;
 
+  virtual int SetSendVideoRotationStatus(int video_channel,
+                                         bool enable,
+                                         int id) = 0;
+
+  virtual int SetReceiveVideoRotationStatus(int video_channel,
+                                            bool enable,
+                                            int id) = 0;
+
   // Enables/disables RTCP Receiver Reference Time Report Block extension/
   // DLRR Report Block extension (RFC 3611).
   virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) = 0;
@@ -315,7 +295,7 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   // stream.
   virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
                                               RtcpStatistics& basic_stats,
-                                              int& rtt_ms) const = 0;
+                                              int64_t& rtt_ms) const = 0;
 
   // This function returns statistics reported by the remote client in RTCP
   // report blocks. If several streams are reported, the statistics will be
@@ -324,16 +304,16 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   // and will always be set to 0.
   virtual int GetSendChannelRtcpStatistics(const int video_channel,
                                            RtcpStatistics& basic_stats,
-                                           int& rtt_ms) const = 0;
+                                           int64_t& rtt_ms) const = 0;
 
   // TODO(sprang): Temporary hacks to prevent libjingle build from failing,
   // remove when libjingle has been lifted to support webrtc issue 2589
   virtual int GetReceivedRTCPStatistics(const int video_channel,
-                                unsigned short& fraction_lost,
-                                unsigned int& cumulative_lost,
-                                unsigned int& extended_max,
-                                unsigned int& jitter,
-                                int& rtt_ms) const {
+                                        unsigned short& fraction_lost,
+                                        unsigned int& cumulative_lost,
+                                        unsigned int& extended_max,
+                                        unsigned int& jitter,
+                                        int64_t& rtt_ms) const {
     RtcpStatistics stats;
     int ret_code = GetReceiveChannelRtcpStatistics(video_channel,
                                              stats,
@@ -345,11 +325,11 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
     return ret_code;
   }
   virtual int GetSentRTCPStatistics(const int video_channel,
-                            unsigned short& fraction_lost,
-                            unsigned int& cumulative_lost,
-                            unsigned int& extended_max,
-                            unsigned int& jitter,
-                            int& rtt_ms) const {
+                                    unsigned short& fraction_lost,
+                                    unsigned int& cumulative_lost,
+                                    unsigned int& extended_max,
+                                    unsigned int& jitter,
+                                    int64_t& rtt_ms) const {
     RtcpStatistics stats;
     int ret_code = GetSendChannelRtcpStatistics(video_channel,
                                                 stats,
@@ -382,17 +362,17 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   // TODO(sprang): Temporary hacks to prevent libjingle build from failing,
   // remove when libjingle has been lifted to support webrtc issue 2589
   virtual int GetRTPStatistics(const int video_channel,
-                       unsigned int& bytes_sent,
+                       size_t& bytes_sent,
                        unsigned int& packets_sent,
-                       unsigned int& bytes_received,
+                       size_t& bytes_received,
                        unsigned int& packets_received) const {
     StreamDataCounters sent;
     StreamDataCounters received;
     int ret_code = GetRtpStatistics(video_channel, sent, received);
-    bytes_sent = sent.bytes;
-    packets_sent = sent.packets;
-    bytes_received = received.bytes;
-    packets_received = received.packets;
+    bytes_sent = sent.transmitted.payload_bytes;
+    packets_sent = sent.transmitted.packets;
+    bytes_received = received.transmitted.payload_bytes;
+    packets_received = received.transmitted.packets;
     return ret_code;
   }
 
@@ -409,12 +389,14 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
       int video_channel, StreamDataCountersCallback* callback) = 0;
 
 
-  // Gets sent and received RTCP packet types.
-  // TODO(asapersson): Remove default implementation.
-  virtual int GetRtcpPacketTypeCounters(
+  // Gets RTCP packet type statistics from a sent/received stream.
+  virtual int GetSendRtcpPacketTypeCounter(
       int video_channel,
-      RtcpPacketTypeCounter* packets_sent,
-      RtcpPacketTypeCounter* packets_received) const { return -1; }
+      RtcpPacketTypeCounter* packet_counter) const = 0;
+
+  virtual int GetReceiveRtcpPacketTypeCounter(
+      int video_channel,
+      RtcpPacketTypeCounter* packet_counter) const = 0;
 
   // Gets the sender info part of the last received RTCP Sender Report (SR)
   virtual int GetRemoteRTCPSenderInfo(const int video_channel,
@@ -450,17 +432,10 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
       const int video_channel,
       unsigned int* estimated_bandwidth) const = 0;
 
-  // This function gets the receive-side bandwidth esitmator statistics.
-  // TODO(jiayl): remove the default impl when libjingle's FakeWebRtcVideoEngine
-  // is updated.
-  virtual int GetReceiveBandwidthEstimatorStats(
-      const int video_channel,
-      ReceiveBandwidthEstimatorStats* output) const { return -1; }
-
   // This function gets the PacedSender queuing delay for the last sent frame.
   // TODO(jiayl): remove the default impl when libjingle is updated.
   virtual int GetPacerQueuingDelayMs(
-      const int video_channel, int* delay_ms) const {
+      const int video_channel, int64_t* delay_ms) const {
     return -1;
   }
 
@@ -484,13 +459,6 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   // Removes a registered instance of ViERTPObserver.
   virtual int DeregisterRTPObserver(const int video_channel) = 0;
 
-  // Registers an instance of a user implementation of the ViERTCPObserver.
-  virtual int RegisterRTCPObserver(const int video_channel,
-                                   ViERTCPObserver& observer) = 0;
-
-  // Removes a registered instance of ViERTCPObserver.
-  virtual int DeregisterRTCPObserver(const int video_channel) = 0;
-
   // Registers and instance of a user implementation of ViEFrameCountObserver
   virtual int RegisterSendFrameCountObserver(
       int video_channel, FrameCountObserver* observer) = 0;
@@ -499,6 +467,12 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   virtual int DeregisterSendFrameCountObserver(
       int video_channel, FrameCountObserver* observer) = 0;
 
+  // Called when RTCP packet type counters might have been changed. User has to
+  // filter on SSRCs to determine whether it's status sent or received.
+  virtual int RegisterRtcpPacketTypeCounterObserver(
+      int video_channel,
+      RtcpPacketTypeCounterObserver* observer) = 0;
+
  protected:
   virtual ~ViERTP_RTCP() {}
 };
diff --git a/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h b/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h
index d4e17f4d27..a59fdbfddb 100644
--- a/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h
+++ b/media/webrtc/trunk/webrtc/video_engine/mock/mock_vie_frame_provider_base.h
@@ -17,11 +17,10 @@ namespace webrtc {
 
 class MockViEFrameCallback : public ViEFrameCallback {
  public:
-  MOCK_METHOD4(DeliverFrame,
+  MOCK_METHOD3(DeliverFrame,
                void(int id,
                     I420VideoFrame* video_frame,
-                    int num_csrcs,
-                    const uint32_t CSRC[kRtpCsrcSize]));
+                    const std::vector& csrcs));
   MOCK_METHOD2(DelayChanged, void(int id, int frame_delay));
   MOCK_METHOD3(GetPreferedFrameSettings,
                int(int* width, int* height, int* frame_rate));
diff --git a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc
index 32b0d2554e..88d5d052bc 100644
--- a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.cc
@@ -17,9 +17,9 @@
 #include 
 #include 
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/exp_filter.h"
 #include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
@@ -135,7 +135,7 @@ class OveruseFrameDetector::EncodeTimeAvg {
  private:
   const float kWeightFactor;
   const float kInitialAvgEncodeTimeMs;
-  scoped_ptr filtered_encode_time_ms_;
+  rtc::scoped_ptr filtered_encode_time_ms_;
 };
 
 // Class for calculating the processing usage on the send-side (the average
@@ -208,8 +208,8 @@ class OveruseFrameDetector::SendProcessingUsage {
   const float kMaxSampleDiffMs;
   uint64_t count_;
   CpuOveruseOptions options_;
-  scoped_ptr filtered_processing_ms_;
-  scoped_ptr filtered_frame_diff_ms_;
+  rtc::scoped_ptr filtered_processing_ms_;
+  rtc::scoped_ptr filtered_frame_diff_ms_;
 };
 
 // Class for calculating the processing time of frames.
@@ -314,12 +314,14 @@ class OveruseFrameDetector::CaptureQueueDelay {
   const float kWeightFactor;
   std::list frames_;
   int delay_ms_;
-  scoped_ptr filtered_delay_ms_per_s_;
+  rtc::scoped_ptr filtered_delay_ms_per_s_;
 };
 
-OveruseFrameDetector::OveruseFrameDetector(Clock* clock)
-    : crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      observer_(NULL),
+OveruseFrameDetector::OveruseFrameDetector(
+    Clock* clock,
+    CpuOveruseMetricsObserver* metrics_observer)
+    : observer_(NULL),
+      metrics_observer_(metrics_observer),
       clock_(clock),
       next_process_time_(clock_->TimeInMilliseconds()),
       num_process_times_(0),
@@ -337,19 +339,21 @@ OveruseFrameDetector::OveruseFrameDetector(Clock* clock)
       frame_queue_(new FrameQueue()),
       last_sample_time_ms_(0),
       capture_queue_delay_(new CaptureQueueDelay()) {
+  DCHECK(metrics_observer != nullptr);
+  processing_thread_.DetachFromThread();
 }
 
 OveruseFrameDetector::~OveruseFrameDetector() {
 }
 
 void OveruseFrameDetector::SetObserver(CpuOveruseObserver* observer) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   observer_ = observer;
 }
 
 void OveruseFrameDetector::SetOptions(const CpuOveruseOptions& options) {
   assert(options.min_frame_samples > 0);
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   if (options_.Equals(options)) {
     return;
   }
@@ -360,32 +364,31 @@ void OveruseFrameDetector::SetOptions(const CpuOveruseOptions& options) {
 }
 
 int OveruseFrameDetector::CaptureQueueDelayMsPerS() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return capture_queue_delay_->delay_ms();
 }
 
 int OveruseFrameDetector::LastProcessingTimeMs() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return frame_queue_->last_processing_time_ms();
 }
 
 int OveruseFrameDetector::FramesInQueue() const {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   return frame_queue_->NumFrames();
 }
 
-void OveruseFrameDetector::GetCpuOveruseMetrics(
-    CpuOveruseMetrics* metrics) const {
-  CriticalSectionScoped cs(crit_.get());
-  metrics->capture_jitter_ms = static_cast(capture_deltas_.StdDev() + 0.5);
-  metrics->avg_encode_time_ms = encode_time_->Value();
-  metrics->encode_rsd = 0;
-  metrics->encode_usage_percent = usage_->Value();
-  metrics->capture_queue_delay_ms_per_s = capture_queue_delay_->Value();
+void OveruseFrameDetector::UpdateCpuOveruseMetrics() {
+  metrics_.capture_jitter_ms = static_cast(capture_deltas_.StdDev() + 0.5);
+  metrics_.avg_encode_time_ms = encode_time_->Value();
+  metrics_.encode_usage_percent = usage_->Value();
+  metrics_.capture_queue_delay_ms_per_s = capture_queue_delay_->Value();
+
+  metrics_observer_->CpuOveruseMetricsUpdated(metrics_);
 }
 
-int32_t OveruseFrameDetector::TimeUntilNextProcess() {
-  CriticalSectionScoped cs(crit_.get());
+int64_t OveruseFrameDetector::TimeUntilNextProcess() {
+  DCHECK(processing_thread_.CalledOnValidThread());
   return next_process_time_ - clock_->TimeInMilliseconds();
 }
 
@@ -411,12 +414,13 @@ void OveruseFrameDetector::ResetAll(int num_pixels) {
   capture_queue_delay_->ClearFrames();
   last_capture_time_ = 0;
   num_process_times_ = 0;
+  UpdateCpuOveruseMetrics();
 }
 
 void OveruseFrameDetector::FrameCaptured(int width,
                                          int height,
                                          int64_t capture_time_ms) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
 
   int64_t now = clock_->TimeInMilliseconds();
   if (FrameSizeChanged(width * height) || FrameTimeoutDetected(now)) {
@@ -434,15 +438,16 @@ void OveruseFrameDetector::FrameCaptured(int width,
   if (options_.enable_extended_processing_usage) {
     frame_queue_->Start(capture_time_ms, now);
   }
+  UpdateCpuOveruseMetrics();
 }
 
 void OveruseFrameDetector::FrameProcessingStarted() {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   capture_queue_delay_->FrameProcessingStarted(clock_->TimeInMilliseconds());
 }
 
 void OveruseFrameDetector::FrameEncoded(int encode_time_ms) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   int64_t now = clock_->TimeInMilliseconds();
   if (last_encode_sample_ms_ != 0) {
     int64_t diff_ms = now - last_encode_sample_ms_;
@@ -453,10 +458,11 @@ void OveruseFrameDetector::FrameEncoded(int encode_time_ms) {
   if (!options_.enable_extended_processing_usage) {
     AddProcessingTime(encode_time_ms);
   }
+  UpdateCpuOveruseMetrics();
 }
 
 void OveruseFrameDetector::FrameSent(int64_t capture_time_ms) {
-  CriticalSectionScoped cs(crit_.get());
+  rtc::CritScope cs(&crit_);
   if (!options_.enable_extended_processing_usage) {
     return;
   }
@@ -465,6 +471,7 @@ void OveruseFrameDetector::FrameSent(int64_t capture_time_ms) {
   if (delay_ms > 0) {
     AddProcessingTime(delay_ms);
   }
+  UpdateCpuOveruseMetrics();
 }
 
 void OveruseFrameDetector::AddProcessingTime(int elapsed_ms) {
@@ -477,7 +484,7 @@ void OveruseFrameDetector::AddProcessingTime(int elapsed_ms) {
 }
 
 int32_t OveruseFrameDetector::Process() {
-  CriticalSectionScoped cs(crit_.get());
+  DCHECK(processing_thread_.CalledOnValidThread());
 
   int64_t now = clock_->TimeInMilliseconds();
 
@@ -487,9 +494,12 @@ int32_t OveruseFrameDetector::Process() {
 
   int64_t diff_ms = now - next_process_time_ + kProcessIntervalMs;
   next_process_time_ = now + kProcessIntervalMs;
+
+  rtc::CritScope cs(&crit_);
   ++num_process_times_;
 
   capture_queue_delay_->CalculateDelayChange(diff_ms);
+  UpdateCpuOveruseMetrics();
 
   if (num_process_times_ <= options_.min_process_count) {
     return 0;
@@ -535,6 +545,7 @@ int32_t OveruseFrameDetector::Process() {
                   << " encode usage " << usage_->Value()
                   << " overuse detections " << num_overuse_detections_
                   << " rampup delay " << rampup_delay;
+
   return 0;
 }
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h
index f90a4f8250..7c04cf7752 100644
--- a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h
+++ b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector.h
@@ -12,16 +12,18 @@
 #define WEBRTC_VIDEO_ENGINE_OVERUSE_FRAME_DETECTOR_H_
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/exp_filter.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/video_engine/include/vie_base.h"
 
 namespace webrtc {
 
 class Clock;
 class CpuOveruseObserver;
-class CriticalSectionWrapper;
 
 // TODO(pbos): Move this somewhere appropriate.
 class Statistics {
@@ -43,14 +45,15 @@ class Statistics {
   float sum_;
   uint64_t count_;
   CpuOveruseOptions options_;
-  scoped_ptr filtered_samples_;
-  scoped_ptr filtered_variance_;
+  rtc::scoped_ptr filtered_samples_;
+  rtc::scoped_ptr filtered_variance_;
 };
 
 // Use to detect system overuse based on jitter in incoming frames.
 class OveruseFrameDetector : public Module {
  public:
-  explicit OveruseFrameDetector(Clock* clock);
+  OveruseFrameDetector(Clock* clock,
+                       CpuOveruseMetricsObserver* metrics_observer);
   ~OveruseFrameDetector();
 
   // Registers an observer receiving overuse and underuse callbacks. Set
@@ -72,34 +75,14 @@ class OveruseFrameDetector : public Module {
   // Called for each sent frame.
   void FrameSent(int64_t capture_time_ms);
 
-  // Accessors.
-
-  // Returns CpuOveruseMetrics where
-  // capture_jitter_ms: The estimated jitter based on incoming captured frames.
-  // avg_encode_time_ms: Running average of reported encode time
-  //                     (FrameEncoded()). Only used for stats.
-  // TODO(asapersson): Rename metric.
-  // encode_usage_percent: The average processing time of a frame on the
-  //                       send-side divided by the average time difference
-  //                       between incoming captured frames.
-  // capture_queue_delay_ms_per_s: The current time delay between an incoming
-  //                               captured frame (FrameCaptured()) until the
-  //                               frame is being processed
-  //                               (FrameProcessingStarted()). (Note: if a new
-  //                               frame is received before an old frame has
-  //                               been processed, the old frame is skipped).
-  //                               The delay is expressed in ms delay per sec.
-  //                               Only used for stats.
-  void GetCpuOveruseMetrics(CpuOveruseMetrics* metrics) const;
-
   // Only public for testing.
   int CaptureQueueDelayMsPerS() const;
   int LastProcessingTimeMs() const;
   int FramesInQueue() const;
 
   // Implements Module.
-  virtual int32_t TimeUntilNextProcess() OVERRIDE;
-  virtual int32_t Process() OVERRIDE;
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
 
  private:
   class EncodeTimeAvg;
@@ -107,31 +90,47 @@ class OveruseFrameDetector : public Module {
   class CaptureQueueDelay;
   class FrameQueue;
 
-  void AddProcessingTime(int elapsed_ms);
+  void UpdateCpuOveruseMetrics() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
-  bool IsOverusing();
-  bool IsUnderusing(int64_t time_now);
+  // TODO(asapersson): This method is only used on one thread, so it shouldn't
+  // need a guard.
+  void AddProcessingTime(int elapsed_ms) EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
-  bool FrameTimeoutDetected(int64_t now) const;
-  bool FrameSizeChanged(int num_pixels) const;
+  // TODO(asapersson): This method is always called on the processing thread.
+  // If locking is required, consider doing that locking inside the
+  // implementation and reduce scope as much as possible.  We should also
+  // see if we can avoid calling out to other methods while holding the lock.
+  bool IsOverusing() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  bool IsUnderusing(int64_t time_now) EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
-  void ResetAll(int num_pixels);
+  bool FrameTimeoutDetected(int64_t now) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  bool FrameSizeChanged(int num_pixels) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
-  // Protecting all members.
-  scoped_ptr crit_;
+  void ResetAll(int num_pixels) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Protecting all members except const and those that are only accessed on the
+  // processing thread.
+  // TODO(asapersson): See if we can reduce locking.  As is, video frame
+  // processing contends with reading stats and the processing thread.
+  mutable rtc::CriticalSection crit_;
 
   // Observer getting overuse reports.
-  CpuOveruseObserver* observer_;
+  CpuOveruseObserver* observer_ GUARDED_BY(crit_);
 
-  CpuOveruseOptions options_;
+  CpuOveruseOptions options_ GUARDED_BY(crit_);
 
-  Clock* clock_;
-  int64_t next_process_time_;
-  int64_t num_process_times_;
+  // Stats metrics.
+  CpuOveruseMetricsObserver* const metrics_observer_;
+  CpuOveruseMetrics metrics_ GUARDED_BY(crit_);
 
-  Statistics capture_deltas_;
-  int64_t last_capture_time_;
+  Clock* const clock_;
+  int64_t next_process_time_;  // Only accessed on the processing thread.
+  int64_t num_process_times_ GUARDED_BY(crit_);
 
+  Statistics capture_deltas_ GUARDED_BY(crit_);
+  int64_t last_capture_time_ GUARDED_BY(crit_);
+
+  // These six members are only accessed on the processing thread.
   int64_t last_overuse_time_;
   int checks_above_threshold_;
   int num_overuse_detections_;
@@ -141,15 +140,22 @@ class OveruseFrameDetector : public Module {
   int current_rampup_delay_ms_;
 
   // Number of pixels of last captured frame.
-  int num_pixels_;
+  int num_pixels_ GUARDED_BY(crit_);
 
-  int64_t last_encode_sample_ms_;
-  scoped_ptr encode_time_;
-  scoped_ptr usage_;
-  scoped_ptr frame_queue_;
-  int64_t last_sample_time_ms_;
+  int64_t last_encode_sample_ms_;  // Only accessed by one thread.
 
-  scoped_ptr capture_queue_delay_;
+  // TODO(asapersson): Can these be regular members (avoid separate heap
+  // allocs)?
+  const rtc::scoped_ptr encode_time_ GUARDED_BY(crit_);
+  const rtc::scoped_ptr usage_ GUARDED_BY(crit_);
+  const rtc::scoped_ptr frame_queue_ GUARDED_BY(crit_);
+
+  int64_t last_sample_time_ms_;  // Only accessed by one thread.
+
+  const rtc::scoped_ptr capture_queue_delay_
+      GUARDED_BY(crit_);
+
+  rtc::ThreadChecker processing_thread_;
 
   DISALLOW_COPY_AND_ASSIGN(OveruseFrameDetector);
 };
diff --git a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc
index e2361695b3..cfb7f01d4f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/overuse_frame_detector_unittest.cc
@@ -11,8 +11,8 @@
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/clock.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/overuse_frame_detector.h"
 
@@ -47,12 +47,13 @@ class CpuOveruseObserverImpl : public CpuOveruseObserver {
   int normaluse_;
 };
 
-class OveruseFrameDetectorTest : public ::testing::Test {
+class OveruseFrameDetectorTest : public ::testing::Test,
+                                 public CpuOveruseMetricsObserver {
  protected:
   virtual void SetUp() {
     clock_.reset(new SimulatedClock(1234));
     observer_.reset(new MockCpuOveruseObserver());
-    overuse_detector_.reset(new OveruseFrameDetector(clock_.get()));
+    overuse_detector_.reset(new OveruseFrameDetector(clock_.get(), this));
 
     options_.low_capture_jitter_threshold_ms = 10.0f;
     options_.high_capture_jitter_threshold_ms = 15.0f;
@@ -61,6 +62,10 @@ class OveruseFrameDetectorTest : public ::testing::Test {
     overuse_detector_->SetObserver(observer_.get());
   }
 
+  void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override {
+    metrics_ = metrics;
+  }
+
   int InitialJitter() {
     return ((options_.low_capture_jitter_threshold_ms +
              options_.high_capture_jitter_threshold_ms) / 2.0f) + 0.5;
@@ -124,40 +129,37 @@ class OveruseFrameDetectorTest : public ::testing::Test {
     overuse_detector_->Process();
   }
 
-  int CaptureJitterMs() {
-    CpuOveruseMetrics metrics;
-    overuse_detector_->GetCpuOveruseMetrics(&metrics);
-    return metrics.capture_jitter_ms;
-  }
+  int CaptureJitterMs() { return metrics_.capture_jitter_ms; }
 
-  int AvgEncodeTimeMs() {
-    CpuOveruseMetrics metrics;
-    overuse_detector_->GetCpuOveruseMetrics(&metrics);
-    return metrics.avg_encode_time_ms;
-  }
+  int AvgEncodeTimeMs() { return metrics_.avg_encode_time_ms; }
 
-  int UsagePercent() {
-    CpuOveruseMetrics metrics;
-    overuse_detector_->GetCpuOveruseMetrics(&metrics);
-    return metrics.encode_usage_percent;
-  }
+  int UsagePercent() { return metrics_.encode_usage_percent; }
 
   CpuOveruseOptions options_;
-  scoped_ptr clock_;
-  scoped_ptr observer_;
-  scoped_ptr overuse_detector_;
+  rtc::scoped_ptr clock_;
+  rtc::scoped_ptr observer_;
+  rtc::scoped_ptr overuse_detector_;
+  CpuOveruseMetrics metrics_;
 };
 
 // enable_capture_jitter_method = true;
 // CaptureJitterMs() > high_capture_jitter_threshold_ms => overuse.
 // CaptureJitterMs() < low_capture_jitter_threshold_ms => underuse.
 TEST_F(OveruseFrameDetectorTest, TriggerOveruse) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
+  overuse_detector_->SetOptions(options_);
   // capture_jitter > high => overuse
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
   TriggerOveruse(options_.high_threshold_consecutive_count);
 }
 
 TEST_F(OveruseFrameDetectorTest, OveruseAndRecover) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
+  overuse_detector_->SetOptions(options_);
   // capture_jitter > high => overuse
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
   TriggerOveruse(options_.high_threshold_consecutive_count);
@@ -167,6 +169,10 @@ TEST_F(OveruseFrameDetectorTest, OveruseAndRecover) {
 }
 
 TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithNoObserver) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
+  overuse_detector_->SetOptions(options_);
   overuse_detector_->SetObserver(NULL);
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
   TriggerOveruse(options_.high_threshold_consecutive_count);
@@ -177,6 +183,7 @@ TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithNoObserver) {
 TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithMethodDisabled) {
   options_.enable_capture_jitter_method = false;
   options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
   overuse_detector_->SetOptions(options_);
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
   TriggerOveruse(options_.high_threshold_consecutive_count);
@@ -185,6 +192,10 @@ TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithMethodDisabled) {
 }
 
 TEST_F(OveruseFrameDetectorTest, DoubleOveruseAndRecover) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
+  overuse_detector_->SetOptions(options_);
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(2);
   TriggerOveruse(options_.high_threshold_consecutive_count);
   TriggerOveruse(options_.high_threshold_consecutive_count);
@@ -193,6 +204,9 @@ TEST_F(OveruseFrameDetectorTest, DoubleOveruseAndRecover) {
 }
 
 TEST_F(OveruseFrameDetectorTest, TriggerUnderuseWithMinProcessCount) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
   CpuOveruseObserverImpl overuse_observer_;
   overuse_detector_->SetObserver(&overuse_observer_);
   options_.min_process_count = 1;
@@ -206,6 +220,10 @@ TEST_F(OveruseFrameDetectorTest, TriggerUnderuseWithMinProcessCount) {
 }
 
 TEST_F(OveruseFrameDetectorTest, ConstantOveruseGivesNoNormalUsage) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
+  overuse_detector_->SetOptions(options_);
   EXPECT_CALL(*(observer_.get()), NormalUsage()).Times(0);
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(64);
   for(size_t i = 0; i < 64; ++i) {
@@ -214,6 +232,9 @@ TEST_F(OveruseFrameDetectorTest, ConstantOveruseGivesNoNormalUsage) {
 }
 
 TEST_F(OveruseFrameDetectorTest, ConsecutiveCountTriggersOveruse) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(1);
   options_.high_threshold_consecutive_count = 2;
   overuse_detector_->SetOptions(options_);
@@ -221,22 +242,15 @@ TEST_F(OveruseFrameDetectorTest, ConsecutiveCountTriggersOveruse) {
 }
 
 TEST_F(OveruseFrameDetectorTest, IncorrectConsecutiveCountTriggersNoOveruse) {
+  options_.enable_capture_jitter_method = true;
+  options_.enable_encode_usage_method = false;
+  options_.enable_extended_processing_usage = false;
   EXPECT_CALL(*(observer_.get()), OveruseDetected()).Times(0);
   options_.high_threshold_consecutive_count = 2;
   overuse_detector_->SetOptions(options_);
   TriggerOveruse(1);
 }
 
-TEST_F(OveruseFrameDetectorTest, GetCpuOveruseMetrics) {
-  CpuOveruseMetrics metrics;
-  overuse_detector_->GetCpuOveruseMetrics(&metrics);
-  EXPECT_GT(metrics.capture_jitter_ms, 0);
-  EXPECT_GT(metrics.avg_encode_time_ms, 0);
-  EXPECT_GT(metrics.encode_usage_percent, 0);
-  EXPECT_GE(metrics.capture_queue_delay_ms_per_s, 0);
-  EXPECT_GE(metrics.encode_rsd, 0);
-}
-
 TEST_F(OveruseFrameDetectorTest, CaptureJitter) {
   EXPECT_EQ(InitialJitter(), CaptureJitterMs());
   InsertFramesWithInterval(1000, kFrameInterval33ms, kWidth, kHeight);
diff --git a/media/webrtc/trunk/webrtc/video_engine/payload_router.cc b/media/webrtc/trunk/webrtc/video_engine/payload_router.cc
new file mode 100644
index 0000000000..958cdd869c
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video_engine/payload_router.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_engine/payload_router.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+
+namespace webrtc {
+
+PayloadRouter::PayloadRouter()
+    : crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      active_(false) {}
+
+PayloadRouter::~PayloadRouter() {}
+
+size_t PayloadRouter::DefaultMaxPayloadLength() {
+  const size_t kIpUdpSrtpLength = 44;
+  return IP_PACKET_SIZE - kIpUdpSrtpLength;
+}
+
+void PayloadRouter::SetSendingRtpModules(
+    const std::list& rtp_modules) {
+  CriticalSectionScoped cs(crit_.get());
+  rtp_modules_.clear();
+  rtp_modules_.reserve(rtp_modules.size());
+  for (auto* rtp_module : rtp_modules) {
+    rtp_modules_.push_back(rtp_module);
+  }
+}
+
+void PayloadRouter::set_active(bool active) {
+  CriticalSectionScoped cs(crit_.get());
+  active_ = active;
+}
+
+bool PayloadRouter::active() {
+  CriticalSectionScoped cs(crit_.get());
+  return active_ && !rtp_modules_.empty();
+}
+
+bool PayloadRouter::RoutePayload(FrameType frame_type,
+                                 int8_t payload_type,
+                                 uint32_t time_stamp,
+                                 int64_t capture_time_ms,
+                                 const uint8_t* payload_data,
+                                 size_t payload_length,
+                                 const RTPFragmentationHeader* fragmentation,
+                                 const RTPVideoHeader* rtp_video_hdr) {
+  CriticalSectionScoped cs(crit_.get());
+  if (!active_ || rtp_modules_.empty())
+    return false;
+
+  // The simulcast index might actually be larger than the number of modules in
+  // case the encoder was processing a frame during a codec reconfig.
+  if (rtp_video_hdr != NULL &&
+      rtp_video_hdr->simulcastIdx >= rtp_modules_.size())
+    return false;
+
+  int stream_idx = 0;
+  if (rtp_video_hdr != NULL)
+    stream_idx = rtp_video_hdr->simulcastIdx;
+  return rtp_modules_[stream_idx]->SendOutgoingData(
+      frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
+      payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
+}
+
+void PayloadRouter::SetTargetSendBitrates(
+    const std::vector& stream_bitrates) {
+  CriticalSectionScoped cs(crit_.get());
+  if (stream_bitrates.size() < rtp_modules_.size()) {
+    // There can be a size mis-match during codec reconfiguration.
+    return;
+  }
+  int idx = 0;
+  for (auto* rtp_module : rtp_modules_) {
+    rtp_module->SetTargetSendBitrate(stream_bitrates[idx++]);
+  }
+}
+
+size_t PayloadRouter::MaxPayloadLength() const {
+  size_t min_payload_length = DefaultMaxPayloadLength();
+  CriticalSectionScoped cs(crit_.get());
+  for (auto* rtp_module : rtp_modules_) {
+    size_t module_payload_length = rtp_module->MaxDataPayloadLength();
+    if (module_payload_length < min_payload_length)
+      min_payload_length = module_payload_length;
+  }
+  return min_payload_length;
+}
+
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/payload_router.h b/media/webrtc/trunk/webrtc/video_engine/payload_router.h
new file mode 100644
index 0000000000..b96defd5e3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video_engine/payload_router.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+#define WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
+
+#include 
+#include 
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/system_wrappers/interface/atomic32.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+class RTPFragmentationHeader;
+class RtpRtcp;
+struct RTPVideoHeader;
+
+// PayloadRouter routes outgoing data to the correct sending RTP module, based
+// on the simulcast layer in RTPVideoHeader.
+class PayloadRouter {
+ public:
+  PayloadRouter();
+  ~PayloadRouter();
+
+  static size_t DefaultMaxPayloadLength();
+
+  // Rtp modules are assumed to be sorted in simulcast index order.
+  void SetSendingRtpModules(const std::list& rtp_modules);
+
+  // PayloadRouter will only route packets if being active, all packets will be
+  // dropped otherwise.
+  void set_active(bool active);
+  bool active();
+
+  // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
+  // Returns true if the packet was routed / sent, false otherwise.
+  bool RoutePayload(FrameType frame_type,
+                    int8_t payload_type,
+                    uint32_t time_stamp,
+                    int64_t capture_time_ms,
+                    const uint8_t* payload_data,
+                    size_t payload_size,
+                    const RTPFragmentationHeader* fragmentation,
+                    const RTPVideoHeader* rtp_video_hdr);
+
+  // Configures current target bitrate per module. 'stream_bitrates' is assumed
+  // to be in the same order as 'SetSendingRtpModules'.
+  void SetTargetSendBitrates(const std::vector& stream_bitrates);
+
+  // Returns the maximum allowed data payload length, given the configured MTU
+  // and RTP headers.
+  size_t MaxPayloadLength() const;
+
+  void AddRef() { ++ref_count_; }
+  void Release() { if (--ref_count_ == 0) { delete this; } }
+
+ private:
+  // TODO(mflodman): When the new video API has launched, remove crit_ and
+  // assume rtp_modules_ will never change during a call.
+  rtc::scoped_ptr crit_;
+
+  // Active sending RTP modules, in layer order.
+  std::vector rtp_modules_ GUARDED_BY(crit_.get());
+  bool active_ GUARDED_BY(crit_.get());
+
+  Atomic32 ref_count_;
+
+  DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
diff --git a/media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc
new file mode 100644
index 0000000000..de391576d8
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video_engine/payload_router_unittest.cc
@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+#include 
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/video_engine/payload_router.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace webrtc {
+
+class PayloadRouterTest : public ::testing::Test {
+ protected:
+  virtual void SetUp() {
+    payload_router_.reset(new PayloadRouter());
+  }
+  rtc::scoped_ptr payload_router_;
+};
+
+TEST_F(PayloadRouterTest, SendOnOneModule) {
+  MockRtpRtcp rtp;
+  std::list modules(1, &rtp);
+
+  payload_router_->SetSendingRtpModules(modules);
+
+  uint8_t payload = 'a';
+  FrameType frame_type = kVideoFrameKey;
+  int8_t payload_type = 96;
+
+  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
+                                    NULL))
+      .Times(0);
+  EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
+                                             &payload, 1, NULL, NULL));
+
+  payload_router_->set_active(true);
+  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
+                                    NULL))
+      .Times(1);
+  EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
+                                            &payload, 1, NULL, NULL));
+
+  payload_router_->set_active(false);
+  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
+                                    NULL))
+      .Times(0);
+  EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
+                                             &payload, 1, NULL, NULL));
+
+  payload_router_->set_active(true);
+  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
+                                    NULL))
+      .Times(1);
+  EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
+                                            &payload, 1, NULL, NULL));
+
+  modules.clear();
+  payload_router_->SetSendingRtpModules(modules);
+  EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
+                                    NULL))
+      .Times(0);
+  EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
+                                             &payload, 1, NULL, NULL));
+}
+
+TEST_F(PayloadRouterTest, SendSimulcast) {
+  MockRtpRtcp rtp_1;
+  MockRtpRtcp rtp_2;
+  std::list modules;
+  modules.push_back(&rtp_1);
+  modules.push_back(&rtp_2);
+
+  payload_router_->SetSendingRtpModules(modules);
+
+  uint8_t payload_1 = 'a';
+  FrameType frame_type_1 = kVideoFrameKey;
+  int8_t payload_type_1 = 96;
+  RTPVideoHeader rtp_hdr_1;
+  rtp_hdr_1.simulcastIdx = 0;
+
+  payload_router_->set_active(true);
+  EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
+                                      NULL, &rtp_hdr_1))
+      .Times(1);
+  EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
+      .Times(0);
+  EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
+                                            &payload_1, 1, NULL, &rtp_hdr_1));
+
+  uint8_t payload_2 = 'b';
+  FrameType frame_type_2 = kVideoFrameDelta;
+  int8_t payload_type_2 = 97;
+  RTPVideoHeader rtp_hdr_2;
+  rtp_hdr_2.simulcastIdx = 1;
+  EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
+                                      NULL, &rtp_hdr_2))
+      .Times(1);
+  EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
+      .Times(0);
+  EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
+                                            &payload_2, 1, NULL, &rtp_hdr_2));
+
+  // Inactive.
+  payload_router_->set_active(false);
+  EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
+      .Times(0);
+  EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
+      .Times(0);
+  EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
+                                             &payload_1, 1, NULL, &rtp_hdr_1));
+  EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
+                                             &payload_2, 1, NULL, &rtp_hdr_2));
+
+  // Invalid simulcast index.
+  payload_router_->set_active(true);
+  EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
+      .Times(0);
+  EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
+      .Times(0);
+  rtp_hdr_1.simulcastIdx = 2;
+  EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
+                                             &payload_1, 1, NULL, &rtp_hdr_1));
+}
+
+TEST_F(PayloadRouterTest, MaxPayloadLength) {
+  // Without any limitations from the modules, verify we get the max payload
+  // length for IP/UDP/SRTP with a MTU of 150 bytes.
+  const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
+  EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
+  EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
+
+  MockRtpRtcp rtp_1;
+  MockRtpRtcp rtp_2;
+  std::list modules;
+  modules.push_back(&rtp_1);
+  modules.push_back(&rtp_2);
+  payload_router_->SetSendingRtpModules(modules);
+
+  // Modules return a higher length than the default value.
+  EXPECT_CALL(rtp_1, MaxDataPayloadLength())
+      .Times(1)
+      .WillOnce(Return(kDefaultMaxLength + 10));
+  EXPECT_CALL(rtp_2, MaxDataPayloadLength())
+      .Times(1)
+      .WillOnce(Return(kDefaultMaxLength + 10));
+  EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
+
+  // The modules return a value lower than default.
+  const size_t kTestMinPayloadLength = 1001;
+  EXPECT_CALL(rtp_1, MaxDataPayloadLength())
+      .Times(1)
+      .WillOnce(Return(kTestMinPayloadLength + 10));
+  EXPECT_CALL(rtp_2, MaxDataPayloadLength())
+      .Times(1)
+      .WillOnce(Return(kTestMinPayloadLength));
+  EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
+}
+
+TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
+  MockRtpRtcp rtp_1;
+  MockRtpRtcp rtp_2;
+  std::list modules;
+  modules.push_back(&rtp_1);
+  modules.push_back(&rtp_2);
+  payload_router_->SetSendingRtpModules(modules);
+
+  const uint32_t bitrate_1 = 10000;
+  const uint32_t bitrate_2 = 76543;
+  std::vector bitrates (2, bitrate_1);
+  bitrates[1] = bitrate_2;
+  EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
+      .Times(1);
+  EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
+      .Times(1);
+  payload_router_->SetTargetSendBitrates(bitrates);
+
+  bitrates.resize(1);
+  EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
+      .Times(0);
+  EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
+      .Times(0);
+  payload_router_->SetTargetSendBitrates(bitrates);
+
+  bitrates.resize(3);
+  bitrates[1] = bitrate_2;
+  bitrates[2] = bitrate_1 + bitrate_2;
+  EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
+      .Times(1);
+  EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
+      .Times(1);
+  payload_router_->SetTargetSendBitrates(bitrates);
+}
+}  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc b/media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc
new file mode 100644
index 0000000000..ea18e830e3
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video_engine/report_block_stats.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/video_engine/report_block_stats.h"
+
+namespace webrtc {
+
+namespace {
+int FractionLost(uint32_t num_lost_sequence_numbers,
+                 uint32_t num_sequence_numbers) {
+  if (num_sequence_numbers == 0) {
+    return 0;
+  }
+  return ((num_lost_sequence_numbers * 255) + (num_sequence_numbers / 2)) /
+      num_sequence_numbers;
+}
+}  // namespace
+
+
+// Helper class for rtcp statistics.
+ReportBlockStats::ReportBlockStats()
+    : num_sequence_numbers_(0),
+      num_lost_sequence_numbers_(0) {
+}
+
+void ReportBlockStats::Store(const RtcpStatistics& rtcp_stats,
+                             uint32_t remote_ssrc,
+                             uint32_t source_ssrc) {
+  RTCPReportBlock block;
+  block.cumulativeLost = rtcp_stats.cumulative_lost;
+  block.fractionLost = rtcp_stats.fraction_lost;
+  block.extendedHighSeqNum = rtcp_stats.extended_max_sequence_number;
+  block.jitter = rtcp_stats.jitter;
+  block.remoteSSRC = remote_ssrc;
+  block.sourceSSRC = source_ssrc;
+  uint32_t num_sequence_numbers = 0;
+  uint32_t num_lost_sequence_numbers = 0;
+  StoreAndAddPacketIncrement(
+      block, &num_sequence_numbers, &num_lost_sequence_numbers);
+}
+
+RTCPReportBlock ReportBlockStats::AggregateAndStore(
+    const ReportBlockVector& report_blocks) {
+  RTCPReportBlock aggregate;
+  if (report_blocks.empty()) {
+    return aggregate;
+  }
+  uint32_t num_sequence_numbers = 0;
+  uint32_t num_lost_sequence_numbers = 0;
+  ReportBlockVector::const_iterator report_block = report_blocks.begin();
+  for (; report_block != report_blocks.end(); ++report_block) {
+    aggregate.cumulativeLost += report_block->cumulativeLost;
+    aggregate.jitter += report_block->jitter;
+    StoreAndAddPacketIncrement(*report_block,
+                               &num_sequence_numbers,
+                               &num_lost_sequence_numbers);
+  }
+
+  if (report_blocks.size() == 1) {
+    // No aggregation needed.
+    return report_blocks[0];
+  }
+  // Fraction lost since previous report block.
+  aggregate.fractionLost =
+      FractionLost(num_lost_sequence_numbers, num_sequence_numbers);
+  aggregate.jitter =
+      (aggregate.jitter + report_blocks.size() / 2) / report_blocks.size();
+  return aggregate;
+}
+
+void ReportBlockStats::StoreAndAddPacketIncrement(
+    const RTCPReportBlock& report_block,
+    uint32_t* num_sequence_numbers,
+    uint32_t* num_lost_sequence_numbers) {
+  // Get diff with previous report block.
+  ReportBlockMap::iterator prev_report_block = prev_report_blocks_.find(
+        report_block.sourceSSRC);
+  if (prev_report_block != prev_report_blocks_.end()) {
+    int seq_num_diff = report_block.extendedHighSeqNum -
+                       prev_report_block->second.extendedHighSeqNum;
+    int cum_loss_diff = report_block.cumulativeLost -
+                        prev_report_block->second.cumulativeLost;
+    if (seq_num_diff >= 0 && cum_loss_diff >= 0) {
+      *num_sequence_numbers += seq_num_diff;
+      *num_lost_sequence_numbers += cum_loss_diff;
+      // Update total number of packets/lost packets.
+      num_sequence_numbers_ += seq_num_diff;
+      num_lost_sequence_numbers_ += cum_loss_diff;
+    }
+  }
+  // Store current report block.
+  prev_report_blocks_[report_block.sourceSSRC] = report_block;
+}
+
+int ReportBlockStats::FractionLostInPercent() const {
+  if (num_sequence_numbers_ == 0) {
+    return -1;
+  }
+  return FractionLost(
+      num_lost_sequence_numbers_, num_sequence_numbers_) * 100 / 255;
+}
+
+}  // namespace webrtc
+
diff --git a/media/webrtc/trunk/webrtc/video_engine/report_block_stats.h b/media/webrtc/trunk/webrtc/video_engine/report_block_stats.h
new file mode 100644
index 0000000000..dadcc9d410
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video_engine/report_block_stats.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
+#define WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
+
+#include 
+#include 
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+// Helper class for rtcp statistics.
+class ReportBlockStats {
+ public:
+  typedef std::map ReportBlockMap;
+  typedef std::vector ReportBlockVector;
+  ReportBlockStats();
+  ~ReportBlockStats() {}
+
+  // Updates stats and stores report blocks.
+  // Returns an aggregate of the |report_blocks|.
+  RTCPReportBlock AggregateAndStore(const ReportBlockVector& report_blocks);
+
+  // Updates stats and stores report block.
+  void Store(const RtcpStatistics& rtcp_stats,
+             uint32_t remote_ssrc,
+             uint32_t source_ssrc);
+
+  // Returns the total fraction of lost packets (or -1 if less than two report
+  // blocks have been stored).
+  int FractionLostInPercent() const;
+
+ private:
+  // Updates the total number of packets/lost packets.
+  // Stores the report block.
+  // Returns the number of packets/lost packets since previous report block.
+  void StoreAndAddPacketIncrement(const RTCPReportBlock& report_block,
+                                  uint32_t* num_sequence_numbers,
+                                  uint32_t* num_lost_sequence_numbers);
+
+  // The total number of packets/lost packets.
+  uint32_t num_sequence_numbers_;
+  uint32_t num_lost_sequence_numbers_;
+
+  // Map holding the last stored report block (mapped by the source SSRC).
+  ReportBlockMap prev_report_blocks_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
+
diff --git a/media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc
new file mode 100644
index 0000000000..13b7af5ba2
--- /dev/null
+++ b/media/webrtc/trunk/webrtc/video_engine/report_block_stats_unittest.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/video_engine/report_block_stats.h"
+
+namespace webrtc {
+
+class ReportBlockStatsTest : public ::testing::Test {
+ protected:
+  ReportBlockStatsTest() : kSsrc1(0x12345), kSsrc2(0x23456) {}
+
+  void SetUp() override {
+    // kSsrc1: block 1-3.
+    block1_1_.cumulativeLost = 10;
+    block1_1_.fractionLost = 123;
+    block1_1_.extendedHighSeqNum = 24000;
+    block1_1_.jitter = 777;
+    block1_1_.sourceSSRC = kSsrc1;
+    block1_2_.cumulativeLost = 15;
+    block1_2_.fractionLost = 0;
+    block1_2_.extendedHighSeqNum = 24100;
+    block1_2_.jitter = 222;
+    block1_2_.sourceSSRC = kSsrc1;
+    block1_3_.cumulativeLost = 50;
+    block1_3_.fractionLost = 0;
+    block1_3_.extendedHighSeqNum = 24200;
+    block1_3_.jitter = 333;
+    block1_3_.sourceSSRC = kSsrc1;
+    // kSsrc2: block 1,2.
+    block2_1_.cumulativeLost = 111;
+    block2_1_.fractionLost = 222;
+    block2_1_.extendedHighSeqNum = 8500;
+    block2_1_.jitter = 555;
+    block2_1_.sourceSSRC = kSsrc2;
+    block2_2_.cumulativeLost = 136;
+    block2_2_.fractionLost = 0;
+    block2_2_.extendedHighSeqNum = 8800;
+    block2_2_.jitter = 888;
+    block2_2_.sourceSSRC = kSsrc2;
+
+    ssrc1block1_.push_back(block1_1_);
+    ssrc1block2_.push_back(block1_2_);
+    ssrc12block1_.push_back(block1_1_);
+    ssrc12block1_.push_back(block2_1_);
+    ssrc12block2_.push_back(block1_2_);
+    ssrc12block2_.push_back(block2_2_);
+  }
+
+  RtcpStatistics RtcpReportBlockToRtcpStatistics(
+      const RTCPReportBlock& stats) {
+    RtcpStatistics block;
+    block.cumulative_lost = stats.cumulativeLost;
+    block.fraction_lost = stats.fractionLost;
+    block.extended_max_sequence_number = stats.extendedHighSeqNum;
+    block.jitter = stats.jitter;
+    return block;
+  }
+
+  const uint32_t kSsrc1;
+  const uint32_t kSsrc2;
+  RTCPReportBlock block1_1_;
+  RTCPReportBlock block1_2_;
+  RTCPReportBlock block1_3_;
+  RTCPReportBlock block2_1_;
+  RTCPReportBlock block2_2_;
+  std::vector ssrc1block1_;
+  std::vector ssrc1block2_;
+  std::vector ssrc12block1_;
+  std::vector ssrc12block2_;
+};
+
+TEST_F(ReportBlockStatsTest, AggregateAndStore_NoSsrc) {
+  ReportBlockStats stats;
+  std::vector empty;
+  RTCPReportBlock aggregated = stats.AggregateAndStore(empty);
+  EXPECT_EQ(0U, aggregated.fractionLost);
+  EXPECT_EQ(0U, aggregated.cumulativeLost);
+  EXPECT_EQ(0U, aggregated.jitter);
+  EXPECT_EQ(0U, aggregated.extendedHighSeqNum);
+}
+
+TEST_F(ReportBlockStatsTest, AggregateAndStore_OneSsrc) {
+  ReportBlockStats stats;
+  RTCPReportBlock aggregated = stats.AggregateAndStore(ssrc1block1_);
+  // One ssrc, no aggregation done.
+  EXPECT_EQ(123U, aggregated.fractionLost);
+  EXPECT_EQ(10U, aggregated.cumulativeLost);
+  EXPECT_EQ(777U, aggregated.jitter);
+  EXPECT_EQ(24000U, aggregated.extendedHighSeqNum);
+
+  aggregated = stats.AggregateAndStore(ssrc1block2_);
+  EXPECT_EQ(0U, aggregated.fractionLost);
+  EXPECT_EQ(15U, aggregated.cumulativeLost);
+  EXPECT_EQ(222U, aggregated.jitter);
+  EXPECT_EQ(24100U, aggregated.extendedHighSeqNum);
+
+  // fl: 100 * (15-10) / (24100-24000) = 5%
+  EXPECT_EQ(5, stats.FractionLostInPercent());
+}
+
+TEST_F(ReportBlockStatsTest, AggregateAndStore_TwoSsrcs) {
+  ReportBlockStats stats;
+  RTCPReportBlock aggregated = stats.AggregateAndStore(ssrc12block1_);
+  EXPECT_EQ(0U, aggregated.fractionLost);
+  EXPECT_EQ(10U + 111U, aggregated.cumulativeLost);
+  EXPECT_EQ((777U + 555U) / 2, aggregated.jitter);
+  EXPECT_EQ(0U, aggregated.extendedHighSeqNum);
+
+  aggregated = stats.AggregateAndStore(ssrc12block2_);
+  // fl: 255 * ((15-10) + (136-111)) / ((24100-24000) + (8800-8500)) = 19
+  EXPECT_EQ(19U, aggregated.fractionLost);
+  EXPECT_EQ(15U + 136U, aggregated.cumulativeLost);
+  EXPECT_EQ((222U + 888U) / 2, aggregated.jitter);
+  EXPECT_EQ(0U, aggregated.extendedHighSeqNum);
+
+  // fl: 100 * ((15-10) + (136-111)) / ((24100-24000) + (8800-8500)) = 7%
+  EXPECT_EQ(7, stats.FractionLostInPercent());
+}
+
+TEST_F(ReportBlockStatsTest, StoreAndGetFractionLost) {
+  const uint32_t kRemoteSsrc = 1;
+  ReportBlockStats stats;
+  EXPECT_EQ(-1, stats.FractionLostInPercent());
+
+  // First block.
+  stats.Store(RtcpReportBlockToRtcpStatistics(block1_1_), kRemoteSsrc, kSsrc1);
+  EXPECT_EQ(-1, stats.FractionLostInPercent());
+  // fl: 100 * (15-10) / (24100-24000) = 5%
+  stats.Store(RtcpReportBlockToRtcpStatistics(block1_2_), kRemoteSsrc, kSsrc1);
+  EXPECT_EQ(5, stats.FractionLostInPercent());
+  // fl: 100 * (50-10) / (24200-24000) = 20%
+  stats.Store(RtcpReportBlockToRtcpStatistics(block1_3_), kRemoteSsrc, kSsrc1);
+  EXPECT_EQ(20, stats.FractionLostInPercent());
+}
+
+}  // namespace webrtc
+
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
index e43fc751b0..4508d8e791 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
@@ -52,9 +52,4 @@ TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
   tests_->ViERenderExtendedTest();
 }
 
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
-       DISABLED_RunsRtpRtcpTestWithoutErrors) {
-  tests_->ViERtpRtcpExtendedTest();
-}
-
 }  // namespace
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc
index 206d055524..4fd2422a9c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/automated/vie_network_test.cc
@@ -25,13 +25,13 @@ class RtcpCollectorTransport : public webrtc::Transport {
   RtcpCollectorTransport() : packets_() {}
   virtual ~RtcpCollectorTransport() {}
 
-  virtual int SendPacket(int /*channel*/,
-                         const void* /*data*/,
-                         int /*len*/) OVERRIDE {
+  int SendPacket(int /*channel*/,
+                 const void* /*data*/,
+                 size_t /*len*/) override {
     EXPECT_TRUE(false);
     return 0;
   }
-  virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE {
+  int SendRTCPPacket(int channel, const void* data, size_t len) override {
     const uint8_t* buf = static_cast(data);
     webrtc::RtpUtility::RtpHeaderParser parser(buf, len);
     if (parser.RTCP()) {
@@ -39,7 +39,7 @@ class RtcpCollectorTransport : public webrtc::Transport {
       p.channel = channel;
       p.length = len;
       if (parser.ParseRtcp(&p.header)) {
-        if (p.header.payloadType == 201) {
+        if (p.header.payloadType == 201 && len >= 20) {
           buf += 20;
           len -= 20;
         } else {
@@ -68,20 +68,20 @@ class RtcpCollectorTransport : public webrtc::Transport {
   struct Packet {
     Packet() : channel(-1), length(0), header(), remb_bitrate(0), remb_ssrc() {}
     int channel;
-    int length;
+    size_t length;
     webrtc::RTPHeader header;
     double remb_bitrate;
     std::vector remb_ssrc;
   };
 
-  bool TryParseREMB(const uint8_t* buf, int length, Packet* p) {
+  bool TryParseREMB(const uint8_t* buf, size_t length, Packet* p) {
     if (length < 8) {
       return false;
     }
     if (buf[0] != 'R' || buf[1] != 'E' || buf[2] != 'M' || buf[3] != 'B') {
       return false;
     }
-    uint8_t ssrcs = buf[4];
+    size_t ssrcs = buf[4];
     uint8_t exp = buf[5] >> 2;
     uint32_t mantissa = ((buf[5] & 0x03) << 16) + (buf[6] << 8) + buf[7];
     double bitrate = mantissa * static_cast(1 << exp);
@@ -91,7 +91,7 @@ class RtcpCollectorTransport : public webrtc::Transport {
       return false;
     }
     buf += 8;
-    for (uint8_t i = 0; i < ssrcs; ++i) {
+    for (size_t i = 0; i < ssrcs; ++i) {
       uint32_t ssrc = (buf[0] << 24) + (buf[1] << 16) + (buf[2] << 8) + buf[3];
       p->remb_ssrc.push_back(ssrc);
       buf += 4;
@@ -107,13 +107,13 @@ class ViENetworkTest : public testing::Test {
   ViENetworkTest() : vie_("ViENetworkTest"), channel_(-1), transport() {}
   virtual ~ViENetworkTest() {}
 
-  virtual void SetUp() OVERRIDE {
+  void SetUp() override {
     EXPECT_EQ(0, vie_.base->CreateChannel(channel_));
     EXPECT_EQ(0, vie_.rtp_rtcp->SetRembStatus(channel_, false, true));
     EXPECT_EQ(0, vie_.network->RegisterSendTransport(channel_, transport));
   }
 
-  virtual void TearDown() OVERRIDE {
+  void TearDown() override {
     EXPECT_EQ(0, vie_.network->DeregisterSendTransport(channel_));
   }
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
index 7a83637860..0ee26a5732 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
@@ -100,7 +100,6 @@ public:
 
     // vie_autotest_rtp_rtcp.cc
     void ViERtpRtcpStandardTest();
-    void ViERtpRtcpExtendedTest();
     void ViERtpRtcpAPITest();
 
 private:
@@ -113,6 +112,13 @@ private:
                                             TbVideoChannel* video_channel,
                                             TbCaptureDevice* capture_device);
 
+    // Stops rendering into the two windows as was set up by a call to
+    // RenderCaptureDeviceAndOutputStream.
+    void StopRenderCaptureDeviceAndOutputStream(
+        TbInterfaces* video_engine,
+        TbVideoChannel* video_channel,
+        TbCaptureDevice* capture_device);
+
     void* _window1;
     void* _window2;
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h
index 5a735e53b4..fffd9484fe 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/interface/vie_autotest_windows.h
@@ -12,6 +12,7 @@
 #define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
 
 #include "webrtc/engine_configurations.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
 
 #include 
@@ -19,7 +20,6 @@
 
 // Forward declaration
 namespace webrtc {
-class ThreadWrapper;
 class CriticalSectionWrapper;
 }
 
@@ -48,7 +48,7 @@ private:
     void* _window2;
 
     bool _terminate;
-    webrtc::ThreadWrapper& _eventThread;
+    rtc::scoped_ptr _eventThread;
     webrtc::CriticalSectionWrapper& _crit;
     HWND _hwndMain;
     HWND _hwnd1;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
index e623ced297..e9ff5378e9 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
@@ -40,7 +40,7 @@ class LocalRendererEffectFilter : public webrtc::ExternalRendererEffectFilter {
                             FrameDropDetector* frame_drop_detector)
       : ExternalRendererEffectFilter(renderer),
         frame_drop_detector_(frame_drop_detector) {}
-  int Transform(int size,
+  int Transform(size_t size,
                 unsigned char* frame_buffer,
                 int64_t ntp_time_ms,
                 unsigned int timestamp,
@@ -100,7 +100,7 @@ class DecodedTimestampEffectFilter : public webrtc::ViEEffectFilter {
   explicit DecodedTimestampEffectFilter(FrameDropDetector* frame_drop_detector)
       : frame_drop_detector_(frame_drop_detector) {}
   virtual ~DecodedTimestampEffectFilter() {}
-  virtual int Transform(int size,
+  virtual int Transform(size_t size,
                         unsigned char* frame_buffer,
                         int64_t ntp_time_ms,
                         unsigned int timestamp,
@@ -593,8 +593,17 @@ int FrameDropDetector::GetNumberOfFramesDroppedAt(State state) {
 }
 
 int FrameDropMonitoringRemoteFileRenderer::DeliverFrame(
-    unsigned char *buffer, int buffer_size, uint32_t time_stamp,
+    unsigned char *buffer, size_t buffer_size, uint32_t time_stamp,
     int64_t ntp_time_ms, int64_t render_time, void* /*handle*/) {
+  ReportFrameStats(time_stamp, render_time);
+  return ViEToFileRenderer::DeliverFrame(buffer, buffer_size,
+                                         time_stamp, ntp_time_ms,
+                                         render_time, NULL);
+}
+
+void FrameDropMonitoringRemoteFileRenderer::ReportFrameStats(
+    uint32_t time_stamp,
+    int64_t render_time) {
   // |render_time| provides the ideal render time for this frame. If that time
   // has already passed we will render it immediately.
   int64_t report_render_time_us = render_time * 1000;
@@ -605,9 +614,12 @@ int FrameDropMonitoringRemoteFileRenderer::DeliverFrame(
   // Register that this frame has been rendered.
   frame_drop_detector_->ReportFrameState(FrameDropDetector::kRendered,
                                          time_stamp, report_render_time_us);
-  return ViEToFileRenderer::DeliverFrame(buffer, buffer_size,
-                                         time_stamp, ntp_time_ms,
-                                         render_time, NULL);
+}
+
+int FrameDropMonitoringRemoteFileRenderer::DeliverI420Frame(
+    const webrtc::I420VideoFrame& webrtc_frame) {
+  ReportFrameStats(webrtc_frame.timestamp(), webrtc_frame.render_time_ms());
+  return ViEToFileRenderer::DeliverI420Frame(webrtc_frame);
 }
 
 int FrameDropMonitoringRemoteFileRenderer::FrameSizeChange(
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
index 1dd2f58d48..92d7bcc33d 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
@@ -222,14 +222,20 @@ class FrameDropMonitoringRemoteFileRenderer : public ViEToFileRenderer {
   virtual ~FrameDropMonitoringRemoteFileRenderer() {}
 
   // Implementation of ExternalRenderer:
-  int FrameSizeChange(unsigned int width, unsigned int height,
-                      unsigned int number_of_streams) OVERRIDE;
-  int DeliverFrame(unsigned char* buffer, int buffer_size,
+  int FrameSizeChange(unsigned int width,
+                      unsigned int height,
+                      unsigned int number_of_streams) override;
+  int DeliverFrame(unsigned char* buffer,
+                   size_t buffer_size,
                    uint32_t time_stamp,
                    int64_t ntp_time_ms,
                    int64_t render_time,
-                   void* handle) OVERRIDE;
+                   void* handle) override;
+  int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;
+
  private:
+  void ReportFrameStats(uint32_t time_stamp, int64_t render_time);
+
   FrameDropDetector* frame_drop_detector_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc
index 5b2c6b45a9..908345d14e 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.cc
@@ -65,6 +65,12 @@ void RenderInWindow(webrtc::ViERender* video_render_interface,
   EXPECT_EQ(0, video_render_interface->StartRender(frame_provider_id));
 }
 
+void StopRenderInWindow(webrtc::ViERender* video_render_interface,
+                        int frame_provider_id) {
+  EXPECT_EQ(0, video_render_interface->StopRender(frame_provider_id));
+  EXPECT_EQ(0, video_render_interface->RemoveRenderer(frame_provider_id));
+}
+
 void RenderToFile(webrtc::ViERender* renderer_interface,
                   int frame_provider_id,
                   ViEToFileRenderer *to_file_renderer) {
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h
index 0c3ebdeeee..cb55eb610a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/primitives/general_primitives.h
@@ -48,11 +48,17 @@ void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
 // (See vie_window_manager_factory.h for more details on how to make one of
 // those). The frame provider id is a source of video frames, for instance
 // a capture device or a video channel.
+// NOTE: A call to StopRenderInWindow needs to be done in order to clear
+// up the configuration applied by this function.
 void RenderInWindow(webrtc::ViERender* video_render_interface,
-                    int  frame_provider_id,
+                    int frame_provider_id,
                     void* os_window,
                     float z_index);
 
+// Stops rendering into a window as previously set up by calling RenderInWindow.
+void StopRenderInWindow(webrtc::ViERender* video_render_interface,
+                        int frame_provider_id);
+
 // Similar in function to RenderInWindow, this function instead renders to
 // a file using a to-file-renderer. The frame provider id is a source of
 // video frames, for instance a capture device or a video channel.
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
index 41fafaef08..9ae4d4fce7 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
@@ -76,7 +76,6 @@ void ViEAutoTest::ViEExtendedTest()
     ViECodecExtendedTest();
     ViEImageProcessExtendedTest();
     ViERenderExtendedTest();
-    ViERtpRtcpExtendedTest();
 }
 
 void ViEAutoTest::ViEAPITest()
@@ -139,7 +138,7 @@ void ViEAutoTest::PrintAudioCodec(const webrtc::CodecInst audioCodec)
     ViETest::Log("\t: %u", audioCodec.pacsize);
     ViETest::Log("\t: %u", audioCodec.plfreq);
     ViETest::Log("\t: %s", audioCodec.plname);
-    ViETest::Log("\t: %u", audioCodec.pltype);
+    ViETest::Log("\t: %d", audioCodec.pltype);
     ViETest::Log("\t: %u", audioCodec.rate);
     ViETest::Log("");
 }
@@ -153,3 +152,11 @@ void ViEAutoTest::RenderCaptureDeviceAndOutputStream(
   RenderInWindow(
       video_engine->render, video_channel->videoChannel, _window2, 1);
 }
+
+void ViEAutoTest::StopRenderCaptureDeviceAndOutputStream(
+    TbInterfaces* video_engine,
+    TbVideoChannel* video_channel,
+    TbCaptureDevice* capture_device) {
+  StopRenderInWindow(video_engine->render, capture_device->captureId);
+  StopRenderInWindow(video_engine->render, video_channel->videoChannel);
+}
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
index ced235f6b5..e69a3ebc9f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
@@ -27,7 +27,7 @@ int ViEAutoTestAndroid::RunAutotest(int testSelection, int subTestSelection,
   webrtc::SetRenderAndroidVM(javaVM);
 #ifndef WEBRTC_ANDROID_OPENSLES
   // voice engine calls into ADM directly
-  webrtc::VoiceEngine::SetAndroidObjects(javaVM, env, context);
+  webrtc::VoiceEngine::SetAndroidObjects(javaVM, context);
 #endif
 
   if (subTestSelection == 0) {
@@ -150,7 +150,8 @@ int ViEAutoTestAndroid::RunAutotest(int testSelection, int subTestSelection,
           break;
 
         case 8: // RTP/RTCP
-          vieAutoTest.ViERtpRtcpExtendedTest();
+          // Note that this test is removed. It hasn't been properly cleaned up
+          // because this hopefully going away soon.
           break;
 
         default:
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc
index 4d032ef5d8..3a94e051be 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_base.cc
@@ -73,6 +73,7 @@ void ViEAutoTest::ViEBaseStandardTest() {
   // ***************************************************************
   // Testing finished. Tear down Video Engine
   // ***************************************************************
+  EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
   EXPECT_EQ(0, capture_interface->StopCapture(capture_id));
   EXPECT_EQ(0, base_interface->StopReceive(video_channel));
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc
index 4fd601264a..68b1afda02 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc
@@ -83,7 +83,7 @@ class CaptureEffectFilter : public webrtc::ViEEffectFilter {
   }
 
   // Implements video_engineEffectFilter.
-  virtual int Transform(int size,
+  virtual int Transform(size_t size,
                         unsigned char* frame_buffer,
                         int64_t ntp_time_ms,
                         unsigned int timestamp,
@@ -360,26 +360,26 @@ void ViEAutoTest::ViECaptureAPITest() {
   EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
 
   // Test GetOrientation.
-  webrtc::VideoCaptureRotation orientation;
+  webrtc::VideoRotation orientation;
   char dummy_name[5];
   EXPECT_NE(0, dev_info->GetOrientation(dummy_name, orientation));
 
   // Test SetRotation.
-  EXPECT_NE(0, video_engine.capture->SetRotateCapturedFrames(
-      capture_id, webrtc::RotateCapturedFrame_90));
+  EXPECT_NE(0, video_engine.capture->SetVideoRotation(
+                   capture_id, webrtc::kVideoRotation_90));
   EXPECT_EQ(kViECaptureDeviceDoesNotExist, video_engine.LastError());
 
   // Allocate capture device.
   EXPECT_EQ(0, video_engine.capture->AllocateCaptureDevice(*vcpm, capture_id));
 
-  EXPECT_EQ(0, video_engine.capture->SetRotateCapturedFrames(
-      capture_id, webrtc::RotateCapturedFrame_0));
-  EXPECT_EQ(0, video_engine.capture->SetRotateCapturedFrames(
-      capture_id, webrtc::RotateCapturedFrame_90));
-  EXPECT_EQ(0, video_engine.capture->SetRotateCapturedFrames(
-      capture_id, webrtc::RotateCapturedFrame_180));
-  EXPECT_EQ(0, video_engine.capture->SetRotateCapturedFrames(
-      capture_id, webrtc::RotateCapturedFrame_270));
+  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
+                   capture_id, webrtc::kVideoRotation_0));
+  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
+                   capture_id, webrtc::kVideoRotation_90));
+  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
+                   capture_id, webrtc::kVideoRotation_180));
+  EXPECT_EQ(0, video_engine.capture->SetVideoRotation(
+                   capture_id, webrtc::kVideoRotation_270));
 
   // Release the capture device
   EXPECT_EQ(0, video_engine.capture->ReleaseCaptureDevice(capture_id));
@@ -445,7 +445,7 @@ void ViEAutoTest::ViECaptureExternalCaptureTest() {
   /// **************************************************************
   //  Engine ready. Begin testing class
   /// **************************************************************
-  const unsigned int video_frame_length = (176 * 144 * 3) / 2;
+  const size_t video_frame_length = (176 * 144 * 3) / 2;
   unsigned char* video_frame = new unsigned char[video_frame_length];
   memset(video_frame, 128, 176 * 144);
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
index 93738c2a17..f5ff771a70 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/video_coding/codecs/i420/main/interface/i420.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/include/vie_capture.h"
@@ -100,7 +100,7 @@ class TestCodecObserver : public webrtc::ViEEncoderObserver,
     last_outgoing_bitrate_ += bitrate;
   }
 
-  virtual void SuspendChange(int video_channel, bool is_suspended) OVERRIDE {
+  void SuspendChange(int video_channel, bool is_suspended) override {
     suspend_change_called_++;
   }
 
@@ -122,7 +122,7 @@ class RenderFilter : public webrtc::ViEEffectFilter {
 
   virtual ~RenderFilter() {
   }
-  virtual int Transform(int size,
+  virtual int Transform(size_t size,
                         unsigned char* frame_buffer,
                         int64_t ntp_time_ms,
                         unsigned int timestamp,
@@ -190,9 +190,8 @@ void ViEAutoTest::ViECodecStandardTest() {
   const char* ip_address = "127.0.0.1";
   const uint16_t rtp_port = 6000;
 
-  webrtc::scoped_ptr
-      video_channel_transport(
-          new webrtc::test::VideoChannelTransport(network, video_channel));
+  rtc::scoped_ptr video_channel_transport(
+      new webrtc::test::VideoChannelTransport(network, video_channel));
 
   ASSERT_EQ(0, video_channel_transport->SetSendDestination(ip_address,
                                                            rtp_port));
@@ -336,7 +335,7 @@ void ViEAutoTest::ViECodecExtendedTest() {
     const char* ip_address = "127.0.0.1";
     const uint16_t rtp_port = 6000;
 
-    webrtc::scoped_ptr
+    rtc::scoped_ptr
         video_channel_transport(
             new webrtc::test::VideoChannelTransport(network, video_channel));
 
@@ -387,7 +386,7 @@ void ViEAutoTest::ViECodecExtendedTest() {
     uint16_t rtp_port_1 = 12000;
     uint16_t rtp_port_2 = 13000;
 
-    webrtc::scoped_ptr
+    rtc::scoped_ptr
         video_channel_transport_1(
             new webrtc::test::VideoChannelTransport(network, video_channel_1));
 
@@ -395,7 +394,7 @@ void ViEAutoTest::ViECodecExtendedTest() {
                                                                rtp_port_1));
     ASSERT_EQ(0, video_channel_transport_1->SetLocalReceiver(rtp_port_1));
 
-    webrtc::scoped_ptr
+    rtc::scoped_ptr
         video_channel_transport_2(
             new webrtc::test::VideoChannelTransport(network, video_channel_2));
 
@@ -515,7 +514,9 @@ void ViEAutoTest::ViECodecAPITest() {
   video_codec.startBitrate = 50;
   EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
   EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
-  EXPECT_EQ(kMinBitrate, video_codec.startBitrate);
+  // We don't allow allocated start bitrate to be decreased via SetSendCodec,
+  // and the default bitrate available in the allocator is 300.
+  EXPECT_EQ(300u, video_codec.startBitrate);
 
   memset(&video_codec, 0, sizeof(video_codec));
   EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
index e7994d3eda..1a6633232a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
@@ -15,7 +15,7 @@
 #include 
 
 #include "gflags/gflags.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
@@ -73,7 +73,7 @@ class ViEAutotestEncoderObserver : public webrtc::ViEEncoderObserver {
               << " BR: " << bitrate << std::endl;
   }
 
-  virtual void SuspendChange(int video_channel, bool is_suspended) OVERRIDE {
+  void SuspendChange(int video_channel, bool is_suspended) override {
     std::cout << "SuspendChange: " << is_suspended << std::endl;
   }
 };
@@ -283,10 +283,8 @@ int ViEAutoTest::ViECustomCall() {
   int buffer_delay_ms = 0;
   bool is_image_scale_enabled = false;
   bool remb = true;
-  webrtc::scoped_ptr
-      video_channel_transport;
-  webrtc::scoped_ptr
-      voice_channel_transport;
+  rtc::scoped_ptr video_channel_transport;
+  rtc::scoped_ptr voice_channel_transport;
 
   while (!start_call) {
     // Get the IP address to use from call.
@@ -1510,7 +1508,7 @@ void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
   int error = 0;
   int number_of_errors = 0;
   webrtc::RtcpStatistics rtcp_stats;
-  int rtt_ms = 0;
+  int64_t rtt_ms = 0;
 
   switch (stat_type) {
     case kReceivedStatistic:
@@ -1564,13 +1562,13 @@ void PrintRTPStatistics(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
                                          "ERROR: %s at line %d",
                                          __FUNCTION__, __LINE__);
   std::cout << "\tRTP bytes sent: "
-            << sent.bytes << std::endl;
+            << sent.transmitted.payload_bytes << std::endl;
   std::cout << "\tRTP packets sent: "
-            << sent.packets << std::endl;
+            << sent.transmitted.packets << std::endl;
   std::cout << "\tRTP bytes received: "
-            << received.bytes << std::endl;
+            << received.transmitted.payload_bytes << std::endl;
   std::cout << "\tRTP packets received: "
-            << received.packets << std::endl;
+            << received.transmitted.packets << std::endl;
 }
 
 void PrintBandwidthUsage(webrtc::ViERTP_RTCP* vie_rtp_rtcp,
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc
index c6f0a05981..de8d2d1dea 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc
@@ -28,7 +28,7 @@ public:
 
     ~MyEffectFilter() {}
 
-    virtual int Transform(int size,
+    virtual int Transform(size_t size,
                           unsigned char* frame_buffer,
                           int64_t ntp_time_ms,
                           unsigned int timestamp,
@@ -48,7 +48,7 @@ void ViEAutoTest::ViEImageProcessStandardTest()
     //***************************************************************
     int rtpPort = 6000;
     // Create VIE
-    TbInterfaces ViE("ViEImageProcessAPITest");
+    TbInterfaces ViE("ViEImageProcessStandardTest");
     // Create a video channel
     TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
     // Create a capture device
@@ -89,8 +89,9 @@ void ViEAutoTest::ViEImageProcessStandardTest()
     ViETest::Log("Only Window 2 should be black and white");
     AutoTestSleep(kAutoTestSleepTimeMs);
 
-    EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
-    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+    StopRenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
+
+    tbCapture.Disconnect(tbChannel.videoChannel);
 
     int rtpPort2 = rtpPort + 100;
     // Create a video channel
@@ -130,6 +131,10 @@ void ViEAutoTest::ViEImageProcessStandardTest()
     EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
         tbChannel.videoChannel));
 
+    EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel2.videoChannel));
+
+    tbCapture.Disconnect(tbChannel2.videoChannel);
+
     //***************************************************************
     //	Testing finished. Tear down Video Engine
     //***************************************************************
@@ -222,4 +227,6 @@ void ViEAutoTest::ViEImageProcessAPITest()
         tbChannel.videoChannel, false));
     EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
         tbCapture.captureId, true));
+
+    tbCapture.Disconnect(tbChannel.videoChannel);
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc
index fd9cf8156f..337bf8a656 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_linux.cc
@@ -17,7 +17,6 @@
 
 #include "webrtc/engine_configurations.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_main.h"
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc
index 03a82b4ca0..ffc9757984 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_loopback.cc
@@ -21,9 +21,9 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/include/vie_capture.h"
@@ -427,13 +427,17 @@ int VideoEngineSampleCode(void* window1, void* window2)
 
     // Set number of temporal layers.
     std::cout << std::endl;
-    std::cout << "Choose number of temporal layers (1 to 4).";
-    std::cout << "Press enter for default: \n";
+    std::cout << "Choose number of temporal layers for VP8 (1 to 4). ";
+    std::cout << "Press enter for default (=1) for other codecs: \n";
     std::getline(std::cin, str);
     int numTemporalLayers = atoi(str.c_str());
-    if(numTemporalLayers != 0)
-    {
-        videoCodec.codecSpecific.VP8.numberOfTemporalLayers = numTemporalLayers;
+    if (numTemporalLayers != 0 &&
+        videoCodec.codecType == webrtc::kVideoCodecVP8) {
+      videoCodec.codecSpecific.VP8.numberOfTemporalLayers = numTemporalLayers;
+    } else if (videoCodec.codecType == webrtc::kVideoCodecVP9) {
+      // Temporal layers for vp9 not yet supported in webrtc.
+      numTemporalLayers = 1;
+      videoCodec.codecSpecific.VP9.numberOfTemporalLayers = 1;
     }
 
     // Set start bit rate
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc
index 89575ceea9..947939309b 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_record.cc
@@ -156,9 +156,8 @@ int VideoEngineSampleRecordCode(void* window1, void* window2) {
 
   audio_channel = voe_base->CreateChannel();
 
-  webrtc::scoped_ptr
-      voice_channel_transport(
-          new webrtc::test::VoiceChannelTransport(voe_network, audio_channel));
+  rtc::scoped_ptr voice_channel_transport(
+      new webrtc::test::VoiceChannelTransport(voe_network, audio_channel));
 
   voice_channel_transport->SetSendDestination(ipAddress, audio_tx_port);
   voice_channel_transport->SetLocalReceiver(audio_rx_port);
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
index 1e9155f99f..c30fbf32a4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
@@ -12,6 +12,7 @@
 // vie_autotest_render.cc
 //
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest.h"
 #include "webrtc/video_engine/test/auto_test/interface/vie_autotest_defines.h"
@@ -57,19 +58,25 @@ public:
     }
 
     virtual int DeliverFrame(unsigned char* buffer,
-                             int bufferSize,
+                             size_t bufferSize,
                              uint32_t time_stamp,
                              int64_t ntp_time_ms,
                              int64_t render_time,
                              void* /*handle*/) {
       if (bufferSize != CalcBufferSize(webrtc::kI420, _width, _height)) {
-        ViETest::Log("Incorrect render buffer received, of length = %d\n",
-                     bufferSize);
+        ViETest::Log("Incorrect render buffer received, of length = %" PRIuS
+                     "\n", bufferSize);
         return 0;
       }
       return 0;
     }
 
+    virtual int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) {
+      EXPECT_EQ(webrtc_frame.width(), _width);
+      EXPECT_EQ(webrtc_frame.height(), _height);
+      return 0;
+    }
+
     virtual bool IsTextureSupported() { return false; }
 
 public:
@@ -213,29 +220,6 @@ void ViEAutoTest::ViERenderExtendedTest()
     AutoTestSleep(kAutoTestSleepTimeMs);
 #endif
 
-    ViETest::Log("Mirroring Local Preview (Window1) Left-Right");
-    EXPECT_EQ(0, ViE.render->MirrorRenderStream(
-        tbCapture.captureId, true, false, true));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    ViETest::Log("\nMirroring Local Preview (Window1) Left-Right and Up-Down");
-    EXPECT_EQ(0, ViE.render->MirrorRenderStream(
-        tbCapture.captureId, true, true, true));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    ViETest::Log("\nMirroring Remote Window(Window2) Up-Down");
-    EXPECT_EQ(0, ViE.render->MirrorRenderStream(
-        tbChannel.videoChannel, true, true, false));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    ViETest::Log("Disabling Mirroing on Window1 and Window2");
-    EXPECT_EQ(0, ViE.render->MirrorRenderStream(
-        tbCapture.captureId, false, false, false));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-    EXPECT_EQ(0, ViE.render->MirrorRenderStream(
-        tbChannel.videoChannel, false, false, false));
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
     ViETest::Log("\nEnabling Full Screen render in 5 sec");
 
     EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
@@ -313,6 +297,7 @@ void ViEAutoTest::ViERenderAPITest() {
   // Already started.
   EXPECT_EQ(-1, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 50));
   EXPECT_EQ(0, ViE.render->StopRender(tbChannel.videoChannel));
+
   // Invalid values.
   EXPECT_EQ(-1, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 9));
   EXPECT_EQ(-1, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel,
@@ -320,4 +305,8 @@ void ViEAutoTest::ViERenderAPITest() {
   // Valid values.
   EXPECT_EQ(0, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 11));
   EXPECT_EQ(0, ViE.render->SetExpectedRenderDelay(tbChannel.videoChannel, 499));
+
+  EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
+  EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
+  tbCapture.Disconnect(tbChannel.videoChannel);
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index b7d02156c2..47dbae4e42 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -40,52 +40,6 @@ public:
     }
 };
 
-class ViERtcpObserver: public webrtc::ViERTCPObserver
-{
-public:
-    int _channel;
-    unsigned char _subType;
-    unsigned int _name;
-    char* _data;
-    unsigned short _dataLength;
-
-    ViERtcpObserver() :
-        _channel(-1),
-        _subType(0),
-        _name(0),
-        _data(NULL),
-        _dataLength(0)
-    {
-    }
-    ~ViERtcpObserver()
-    {
-        if (_data)
-        {
-            delete[] _data;
-        }
-    }
-    virtual void OnApplicationDataReceived(
-        const int videoChannel, const unsigned char subType,
-        const unsigned int name, const char* data,
-        const unsigned short dataLengthInBytes)
-    {
-        _channel = videoChannel;
-        _subType = subType;
-        _name = name;
-        if (dataLengthInBytes > _dataLength)
-        {
-            delete[] _data;
-            _data = NULL;
-        }
-        if (_data == NULL)
-        {
-            _data = new char[dataLengthInBytes];
-        }
-        memcpy(_data, data, dataLengthInBytes);
-        _dataLength = dataLengthInBytes;
-    }
-};
-
 void ViEAutoTest::ViERtpRtcpStandardTest()
 {
     // ***************************************************************
@@ -166,7 +120,7 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     // Pacing
     //
     webrtc::RtcpStatistics received;
-    int recRttMs = 0;
+    int64_t recRttMs = 0;
     unsigned int sentTotalBitrate = 0;
     unsigned int sentVideoBitrate = 0;
     unsigned int sentFecBitrate = 0;
@@ -286,7 +240,7 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
 
     webrtc::RtcpStatistics sent;
-    int sentRttMs = 0;
+    int64_t sentRttMs = 0;
 
     // Fraction lost is a transient value that can get reset after a new rtcp
     // report block. Make regular polls to make sure it is propagated.
@@ -542,21 +496,24 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
                                                 sent_after, received_after));
     if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_after.bytes, received_before.bytes);
+      EXPECT_GT(received_after.transmitted.payload_bytes,
+                received_before.transmitted.payload_bytes);
     }
     // Simulate lost reception and verify that nothing is sent during that time.
     ViE.network->SetNetworkTransmissionState(tbChannel.videoChannel, false);
     // Allow the encoder to finish the current frame before we expect that no
     // additional packets will be sent.
     AutoTestSleep(kAutoTestSleepTimeMs);
-    received_before.bytes = received_after.bytes;
+    received_before.transmitted.payload_bytes =
+        received_after.transmitted.payload_bytes;
     ViETest::Log("Network Down...\n");
     AutoTestSleep(kAutoTestSleepTimeMs);
     EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
                                                 sent_before,
                                                 received_before));
     if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(received_before.bytes, received_after.bytes);
+      EXPECT_EQ(received_before.transmitted.payload_bytes,
+                received_after.transmitted.payload_bytes);
     }
 
     // Network reception back. Video should now be sent.
@@ -567,9 +524,11 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
                                                 sent_before,
                                                 received_before));
     if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_before.bytes, received_after.bytes);
+      EXPECT_GT(received_before.transmitted.payload_bytes,
+                received_after.transmitted.payload_bytes);
     }
-    received_after.bytes = received_before.bytes;
+    received_after.transmitted.payload_bytes =
+        received_before.transmitted.payload_bytes;
     // Buffering mode.
     EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
     EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
@@ -593,15 +552,18 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
                                                 sent_before,
                                                 received_before));
     if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_before.bytes, received_after.bytes);
+      EXPECT_GT(received_before.transmitted.payload_bytes,
+                received_after.transmitted.payload_bytes);
     }
-    received_after.bytes = received_before.bytes;
+    received_after.transmitted.payload_bytes =
+        received_before.transmitted.payload_bytes;
     AutoTestSleep(kAutoTestSleepTimeMs);
     EXPECT_EQ(0, ViE.rtp_rtcp->GetRtpStatistics(tbChannel.videoChannel,
                                                 sent_before,
                                                 received_before));
     if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_EQ(received_after.bytes, received_before.bytes);
+      EXPECT_EQ(received_after.transmitted.payload_bytes,
+                received_before.transmitted.payload_bytes);
     }
     // Network reception back. Video should now be sent.
     ViETest::Log("Network Up...\n");
@@ -611,7 +573,8 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
                                                 sent_before,
                                                 received_before));
     if (FLAGS_include_timing_dependent_tests) {
-      EXPECT_GT(received_before.bytes, received_after.bytes);
+      EXPECT_GT(received_before.transmitted.payload_bytes,
+                received_after.transmitted.payload_bytes);
     }
     // TODO(holmer): Verify that the decoded framerate doesn't decrease on an
     // outage when in buffering mode. This isn't currently possible because we
@@ -630,70 +593,6 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     //***************************************************************
 }
 
-void ViEAutoTest::ViERtpRtcpExtendedTest()
-{
-    //***************************************************************
-    //  Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-    // Create VIE
-    TbInterfaces ViE("ViERtpRtcpExtendedTest");
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-    // Create a capture device
-    TbCaptureDevice tbCapture(ViE);
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-
-    //tbChannel.StartReceive(rtpPort);
-    //tbChannel.StartSend(rtpPort);
-    TbExternalTransport myTransport(*(ViE.network), tbChannel.videoChannel,
-                                    NULL);
-
-    EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.network->RegisterSendTransport(
-        tbChannel.videoChannel, myTransport));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    //***************************************************************
-    //  Engine ready. Begin testing class
-    //***************************************************************
-
-    //
-    // Application specific RTCP
-    //
-    //
-
-    ViERtcpObserver rtcpObserver;
-    EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
-        tbChannel.videoChannel, rtcpObserver));
-
-    unsigned char subType = 3;
-    unsigned int name = static_cast (0x41424344); // 'ABCD';
-    const char* data = "ViEAutoTest Data of length 32 -\0";
-    const unsigned short numBytes = 32;
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-        tbChannel.videoChannel, subType, name, data, numBytes));
-
-    ViETest::Log("Sending RTCP application data...\n");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(subType, rtcpObserver._subType);
-    EXPECT_STRCASEEQ(data, rtcpObserver._data);
-    EXPECT_EQ(name, rtcpObserver._name);
-    EXPECT_EQ(numBytes, rtcpObserver._dataLength);
-
-    ViETest::Log("\t RTCP application data received\n");
-
-    //***************************************************************
-    //  Testing finished. Tear down Video Engine
-    //***************************************************************
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-}
-
 void ViEAutoTest::ViERtpRtcpAPITest()
 {
     //***************************************************************
@@ -853,16 +752,6 @@ void ViEAutoTest::ViERtpRtcpAPITest()
             tbChannel.videoChannel));
         EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTPObserver(
             tbChannel.videoChannel));
-
-        ViERtcpObserver rtcpObserver;
-        EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
-            tbChannel.videoChannel, rtcpObserver));
-        EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTCPObserver(
-            tbChannel.videoChannel, rtcpObserver));
-        EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
-            tbChannel.videoChannel));
-        EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
-            tbChannel.videoChannel));
     }
     //
     // PLI
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc
index 98e37a4171..6e2bac3f9f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/source/vie_autotest_win.cc
@@ -19,7 +19,6 @@
 
 #include "webrtc/engine_configurations.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 #include 
 
@@ -47,9 +46,8 @@ ViEAutoTestWindowManager::ViEAutoTestWindowManager()
     : _window1(NULL),
       _window2(NULL),
       _terminate(false),
-      _eventThread(*webrtc::ThreadWrapper::CreateThread(
-          EventProcess, this, webrtc::kNormalPriority,
-          "ViEAutotestEventThread")),
+      _eventThread(webrtc::ThreadWrapper::CreateThread(
+          EventProcess, this, "ViEAutotestEventThread")),
       _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
       _hwnd1(NULL),
       _hwnd2(NULL),
@@ -86,8 +84,7 @@ int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
   memcpy(_hwnd1Title, window1Title, TITLE_LENGTH);
   memcpy(_hwnd2Title, window2Title, TITLE_LENGTH);
 
-  unsigned int tId = 0;
-  _eventThread.Start(tId);
+  _eventThread->Start();
 
   do {
     _crit.Enter();
@@ -102,14 +99,11 @@ int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
 }
 
 int ViEAutoTestWindowManager::TerminateWindows() {
-  _eventThread.SetNotAlive();
-
   _terminate = true;
-  if (_eventThread.Stop()) {
-    _crit.Enter();
-    delete &_eventThread;
-    _crit.Leave();
-  }
+  _eventThread->Stop();
+  _crit.Enter();
+  _eventThread.reset();
+  _crit.Leave();
 
   return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi
index c8886d8f5e..e52b301651 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.gypi
@@ -12,8 +12,9 @@
       'target_name': 'vie_auto_test',
       'type': 'executable',
       'dependencies': [
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:metrics_default',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
         '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
         '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate
index da09a6e5b7..2579a20b20 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate
+++ b/media/webrtc/trunk/webrtc/video_engine/test/auto_test/vie_auto_test.isolate
@@ -7,14 +7,6 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
-      'variables': {
-        'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
-        ],
-      },
-    }],
     ['OS=="linux" or OS=="mac" or OS=="win"', {
       'variables': {
         'command': [
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc
index 8699f04aeb..1c18d498e0 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_fake_camera.cc
@@ -11,7 +11,6 @@
 
 #include 
 
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/video_engine/include/vie_capture.h"
 #include "webrtc/video_engine/test/libvietest/include/vie_file_capture_device.h"
 
@@ -34,7 +33,6 @@ bool StreamVideoFileRepeatedlyIntoCaptureDevice(void* data) {
 ViEFakeCamera::ViEFakeCamera(webrtc::ViECapture* capture_interface)
     : capture_interface_(capture_interface),
       capture_id_(-1),
-      camera_thread_(NULL),
       file_capture_device_(NULL) {
 }
 
@@ -63,9 +61,9 @@ bool ViEFakeCamera::StartCameraInNewThread(
   // Set up a thread which runs the fake camera. The capturer object is
   // thread-safe.
   camera_thread_ = webrtc::ThreadWrapper::CreateThread(
-      StreamVideoFileRepeatedlyIntoCaptureDevice, file_capture_device_);
-  unsigned int id;
-  camera_thread_->Start(id);
+      StreamVideoFileRepeatedlyIntoCaptureDevice, file_capture_device_,
+      "StreamVideoFileRepeatedlyIntoCaptureDevice");
+  camera_thread_->Start();
 
   return true;
 }
@@ -78,7 +76,7 @@ bool ViEFakeCamera::StopCamera() {
 
   int result = capture_interface_->ReleaseCaptureDevice(capture_id_);
 
-  delete camera_thread_;
+  camera_thread_.reset();
   delete file_capture_device_;
   camera_thread_ = NULL;
   file_capture_device_ = NULL;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc
index b86ff0926a..4f2db75498 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc
@@ -80,14 +80,15 @@ void ViEFileCaptureDevice::ReadFileFor(uint64_t time_slice_ms,
 
   while (elapsed_ms < time_slice_ms) {
     FramePacemaker pacemaker(max_fps);
-    int read = fread(frame_buffer, 1, frame_length_, input_file_);
+    size_t read = fread(frame_buffer, 1, frame_length_, input_file_);
 
-    if (feof(input_file_)) {
+    if (feof(input_file_) || read != frame_length_) {
       rewind(input_file_);
     }
-    input_sink_->IncomingFrame(frame_buffer, read, width_, height_,
-                               webrtc::kVideoI420,
-                               webrtc::TickTime::MillisecondTimestamp());
+    webrtc::I420VideoFrame frame;
+    frame.CreateFrame(frame_buffer, width_, height_, webrtc::kVideoRotation_0);
+    frame.set_render_time_ms(webrtc::TickTime::MillisecondTimestamp());
+    input_sink_->IncomingFrame(frame);
 
     pacemaker.SleepIfNecessary(sleeper);
     elapsed_ms = webrtc::TickTime::MillisecondTimestamp() - start_time_ms;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
index 16c73f4e6f..d0aee84bd6 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
@@ -12,6 +12,7 @@
 
 #include 
 
+#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
@@ -19,19 +20,10 @@
 namespace test {
 struct Frame {
  public:
-  Frame(unsigned char* buffer,
-        int buffer_size,
-        uint32_t timestamp,
-        int64_t render_time)
-      : buffer(new unsigned char[buffer_size]),
-        buffer_size(buffer_size),
-        timestamp(timestamp),
-        render_time(render_time) {
-    memcpy(this->buffer.get(), buffer, buffer_size);
-  }
+  Frame() : buffer(nullptr), buffer_size(0), timestamp(0), render_time(0) {}
 
-  webrtc::scoped_ptr buffer;
-  int buffer_size;
+  rtc::scoped_ptr buffer;
+  size_t buffer_size;
   uint32_t timestamp;
   int64_t render_time;
 
@@ -45,8 +37,7 @@ ViEToFileRenderer::ViEToFileRenderer()
       output_path_(),
       output_filename_(),
       thread_(webrtc::ThreadWrapper::CreateThread(
-          ViEToFileRenderer::RunRenderThread,
-          this, webrtc::kNormalPriority, "ViEToFileRendererThread")),
+          ViEToFileRenderer::RunRenderThread, this, "ViEToFileRendererThread")),
       frame_queue_cs_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
       frame_render_event_(webrtc::EventWrapper::Create()),
       render_queue_(),
@@ -73,14 +64,12 @@ bool ViEToFileRenderer::PrepareForRendering(
 
   output_filename_ = output_filename;
   output_path_ = output_path;
-  unsigned int tid;
-  return thread_->Start(tid);
+  return thread_->Start();
 }
 
 void ViEToFileRenderer::StopRendering() {
   assert(output_file_ != NULL);
   if (thread_.get() != NULL) {
-    thread_->SetNotAlive();
     // Signal that a frame is ready to be written to file.
     frame_render_event_->Set();
     // Call Stop() repeatedly, waiting for ProcessRenderQueue() to finish.
@@ -120,28 +109,54 @@ void ViEToFileRenderer::ForgetOutputFile() {
   output_path_ = "";
 }
 
+test::Frame* ViEToFileRenderer::NewFrame(size_t buffer_size) {
+  test::Frame* frame;
+  if (free_frame_queue_.empty()) {
+    frame = new test::Frame();
+  } else {
+    // Reuse an already allocated frame.
+    frame = free_frame_queue_.front();
+    free_frame_queue_.pop_front();
+  }
+  if (frame->buffer_size < buffer_size) {
+    frame->buffer.reset(new unsigned char[buffer_size]);
+    frame->buffer_size = buffer_size;
+  }
+  return frame;
+}
+
 int ViEToFileRenderer::DeliverFrame(unsigned char *buffer,
-                                    int buffer_size,
+                                    size_t buffer_size,
                                     uint32_t time_stamp,
                                     int64_t ntp_time_ms,
                                     int64_t render_time,
                                     void* /*handle*/) {
   webrtc::CriticalSectionScoped lock(frame_queue_cs_.get());
-  test::Frame* frame;
-  if (free_frame_queue_.empty()) {
-    frame = new test::Frame(buffer, buffer_size, time_stamp, render_time);
-  } else {
-    // Reuse an already allocated frame.
-    frame = free_frame_queue_.front();
-    free_frame_queue_.pop_front();
-    if (frame->buffer_size < buffer_size) {
-      frame->buffer.reset(new unsigned char[buffer_size]);
-    }
-    memcpy(frame->buffer.get(), buffer, buffer_size);
-    frame->buffer_size = buffer_size;
-    frame->timestamp = time_stamp;
-    frame->render_time = render_time;
-  }
+  test::Frame* frame = NewFrame(buffer_size);
+  memcpy(frame->buffer.get(), buffer, buffer_size);
+  frame->timestamp = time_stamp;
+  frame->render_time = render_time;
+
+  render_queue_.push_back(frame);
+  // Signal that a frame is ready to be written to file.
+  frame_render_event_->Set();
+  return 0;
+}
+
+int ViEToFileRenderer::DeliverI420Frame(
+    const webrtc::I420VideoFrame& input_frame) {
+  const size_t buffer_size =
+      CalcBufferSize(webrtc::kI420, input_frame.width(), input_frame.height());
+  webrtc::CriticalSectionScoped lock(frame_queue_cs_.get());
+  test::Frame* frame = NewFrame(buffer_size);
+  const int length =
+      ExtractBuffer(input_frame, frame->buffer_size, frame->buffer.get());
+  assert(static_cast(length) == buffer_size);
+  if (length < 0)
+    return -1;
+  frame->timestamp = input_frame.timestamp();
+  frame->render_time = input_frame.render_time_ms();
+
   render_queue_.push_back(frame);
   // Signal that a frame is ready to be written to file.
   frame_render_event_->Set();
@@ -174,8 +189,8 @@ bool ViEToFileRenderer::ProcessRenderQueue() {
     // the renderer.
     frame_queue_cs_->Leave();
     assert(output_file_);
-    int written = fwrite(frame->buffer.get(), sizeof(unsigned char),
-                              frame->buffer_size, output_file_);
+    size_t written = fwrite(frame->buffer.get(), sizeof(unsigned char),
+                            frame->buffer_size, output_file_);
     frame_queue_cs_->Enter();
     // Return the frame.
     free_frame_queue_.push_front(frame);
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h
index 2037ba910d..918edf96c3 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_I420_codec.h
@@ -24,28 +24,28 @@ public:
     TbI420Encoder();
     virtual ~TbI420Encoder();
 
-    virtual int32_t InitEncode(const webrtc::VideoCodec* codecSettings,
-                               int32_t numberOfCores,
-                               uint32_t maxPayloadSize) OVERRIDE;
+    int32_t InitEncode(const webrtc::VideoCodec* codecSettings,
+                       int32_t numberOfCores,
+                       size_t maxPayloadSize) override;
 
-    virtual int32_t Encode(
+    int32_t Encode(
         const webrtc::I420VideoFrame& inputImage,
         const webrtc::CodecSpecificInfo* codecSpecificInfo,
-        const std::vector* frameTypes) OVERRIDE;
+        const std::vector* frameTypes) override;
 
-    virtual int32_t RegisterEncodeCompleteCallback(
-        webrtc::EncodedImageCallback* callback) OVERRIDE;
+    int32_t RegisterEncodeCompleteCallback(
+        webrtc::EncodedImageCallback* callback) override;
 
-    virtual int32_t Release() OVERRIDE;
+    int32_t Release() override;
 
-    virtual int32_t SetChannelParameters(uint32_t packetLoss, int rtt) OVERRIDE;
+    int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override;
 
-    virtual int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) OVERRIDE;
+    int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override;
 
-    virtual int32_t SetPeriodicKeyFrames(bool enable) OVERRIDE;
+    int32_t SetPeriodicKeyFrames(bool enable) override;
 
-    virtual int32_t CodecConfigParameters(uint8_t* /*buffer*/,
-                                          int32_t /*size*/) OVERRIDE;
+    int32_t CodecConfigParameters(uint8_t* /*buffer*/,
+                                  int32_t /*size*/) override;
 
     struct FunctionCalls
     {
@@ -81,19 +81,18 @@ public:
     TbI420Decoder();
     virtual ~TbI420Decoder();
 
-    virtual int32_t InitDecode(const webrtc::VideoCodec* inst,
-                               int32_t numberOfCores) OVERRIDE;
-    virtual int32_t Decode(
-        const webrtc::EncodedImage& inputImage,
-        bool missingFrames,
-        const webrtc::RTPFragmentationHeader* fragmentation,
-        const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
-        int64_t renderTimeMs = -1) OVERRIDE;
+    int32_t InitDecode(const webrtc::VideoCodec* inst,
+                       int32_t numberOfCores) override;
+    int32_t Decode(const webrtc::EncodedImage& inputImage,
+                   bool missingFrames,
+                   const webrtc::RTPFragmentationHeader* fragmentation,
+                   const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
+                   int64_t renderTimeMs = -1) override;
 
-    virtual int32_t RegisterDecodeCompleteCallback(
-        webrtc::DecodedImageCallback* callback) OVERRIDE;
-    virtual int32_t Release() OVERRIDE;
-    virtual int32_t Reset() OVERRIDE;
+    int32_t RegisterDecodeCompleteCallback(
+        webrtc::DecodedImageCallback* callback) override;
+    int32_t Release() override;
+    int32_t Reset() override;
 
     struct FunctionCalls
     {
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h
index b4518a3fb0..e11c02d181 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_external_transport.h
@@ -19,12 +19,12 @@
 #include 
 
 #include "webrtc/common_types.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 namespace webrtc
 {
 class CriticalSectionWrapper;
 class EventWrapper;
-class ThreadWrapper;
 class ViENetwork;
 }
 
@@ -85,8 +85,8 @@ public:
                         TbExternalTransport::SsrcChannelMap* receive_channels);
     ~TbExternalTransport(void);
 
-    virtual int SendPacket(int channel, const void *data, int len) OVERRIDE;
-    virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE;
+    int SendPacket(int channel, const void* data, size_t len) override;
+    int SendRTCPPacket(int channel, const void* data, size_t len) override;
 
     // Should only be called before/after traffic is being processed.
     // Only one observer can be set (multiple calls will overwrite each other).
@@ -139,7 +139,7 @@ private:
     typedef struct
     {
         int8_t packetBuffer[KMaxPacketSize];
-        int32_t length;
+        size_t length;
         int32_t channel;
         int64_t receiveTime;
     } VideoPacket;
@@ -147,7 +147,7 @@ private:
     int sender_channel_;
     SsrcChannelMap* receive_channels_;
     webrtc::ViENetwork& _vieNetwork;
-    webrtc::ThreadWrapper& _thread;
+    rtc::scoped_ptr _thread;
     webrtc::EventWrapper& _event;
     webrtc::CriticalSectionWrapper& _crit;
     webrtc::CriticalSectionWrapper& _statCrit;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h
index 0409cf3e1b..7d2557e4ad 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/tb_video_channel.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
 #define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/video_engine/test/libvietest/include/tb_interfaces.h"
 
 namespace webrtc {
@@ -44,7 +44,7 @@ class TbVideoChannel {
 
  private:
   TbInterfaces& ViE;
-  webrtc::scoped_ptr channel_transport_;
+  rtc::scoped_ptr channel_transport_;
 };
 
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
index 039a1fd0fb..057f6c8011 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
@@ -24,7 +24,7 @@ class ExternalRendererEffectFilter : public webrtc::ViEEffectFilter {
   explicit ExternalRendererEffectFilter(webrtc::ExternalRenderer* renderer)
       : width_(0), height_(0), renderer_(renderer) {}
   virtual ~ExternalRendererEffectFilter() {}
-  virtual int Transform(int size,
+  virtual int Transform(size_t size,
                         unsigned char* frame_buffer,
                         int64_t ntp_time_ms,
                         unsigned int timestamp,
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h
index abc5d40cf4..afb2752a91 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_fake_camera.h
@@ -11,10 +11,10 @@
 #define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
 
 #include 
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 namespace webrtc {
 class ViECapture;
-class ThreadWrapper;
 }
 
 class ViEFileCaptureDevice;
@@ -43,7 +43,7 @@ class ViEFakeCamera {
   webrtc::ViECapture* capture_interface_;
 
   int capture_id_;
-  webrtc::ThreadWrapper* camera_thread_;
+  rtc::scoped_ptr camera_thread_;
   ViEFileCaptureDevice* file_capture_device_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
index e23dad2ad4..d873cd7a6a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
@@ -18,7 +18,7 @@
 #include 
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/video_engine/include/vie_render.h"
 
 namespace webrtc {
@@ -54,23 +54,29 @@ class ViEToFileRenderer: public webrtc::ExternalRenderer {
   bool SaveOutputFile(const std::string& prefix);
 
   // Implementation of ExternalRenderer:
-  int FrameSizeChange(unsigned int width, unsigned int height,
-                      unsigned int number_of_streams) OVERRIDE;
+  int FrameSizeChange(unsigned int width,
+                      unsigned int height,
+                      unsigned int number_of_streams) override;
 
   int DeliverFrame(unsigned char* buffer,
-                   int buffer_size,
+                   size_t buffer_size,
                    uint32_t time_stamp,
                    int64_t ntp_time_ms,
                    int64_t render_time,
-                   void* handle) OVERRIDE;
+                   void* handle) override;
 
-  bool IsTextureSupported() OVERRIDE;
+  int DeliverI420Frame(const webrtc::I420VideoFrame& webrtc_frame) override;
+
+  bool IsTextureSupported() override;
 
   const std::string GetFullOutputPath() const;
 
  private:
   typedef std::list FrameQueue;
 
+  // Returns a frame with the specified |buffer_size|. Tries to avoid allocating
+  // new frames by reusing frames from |free_frame_queue_|.
+  test::Frame* NewFrame(size_t buffer_size);
   static bool RunRenderThread(void* obj);
   void ForgetOutputFile();
   bool ProcessRenderQueue();
@@ -78,9 +84,9 @@ class ViEToFileRenderer: public webrtc::ExternalRenderer {
   FILE* output_file_;
   std::string output_path_;
   std::string output_filename_;
-  webrtc::scoped_ptr thread_;
-  webrtc::scoped_ptr frame_queue_cs_;
-  webrtc::scoped_ptr frame_render_event_;
+  rtc::scoped_ptr thread_;
+  rtc::scoped_ptr frame_queue_cs_;
+  rtc::scoped_ptr frame_render_event_;
   FrameQueue render_queue_;
   FrameQueue free_frame_queue_;
 };
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi
index 33d0b0f734..603cd4b1ea 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/libvietest.gypi
@@ -11,7 +11,8 @@
       'target_name': 'libvietest',
       'type': 'static_library',
       'dependencies': [
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/common.gyp:webrtc_common',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/test/test.gyp:test_support',
         'video_engine_core',
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc
index f807fd58a1..e9cefcf84d 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc
@@ -46,14 +46,14 @@ int32_t TbI420Encoder::Release()
     return WEBRTC_VIDEO_CODEC_OK;
 }
 
-int32_t TbI420Encoder::SetChannelParameters(uint32_t packetLoss, int rtt) {
+int32_t TbI420Encoder::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
   _functionCalls.SetChannelParameters++;
   return WEBRTC_VIDEO_CODEC_OK;
 }
 
 int32_t TbI420Encoder::InitEncode(const webrtc::VideoCodec* inst,
                                   int32_t /*numberOfCores*/,
-                                  uint32_t /*maxPayloadSize */)
+                                  size_t /*maxPayloadSize */)
 {
     _functionCalls.InitEncode++;
     if (inst == NULL)
@@ -105,9 +105,9 @@ int32_t TbI420Encoder::Encode(
     _encodedImage._timeStamp = inputImage.timestamp();
     _encodedImage._encodedHeight = inputImage.height();
     _encodedImage._encodedWidth = inputImage.width();
-    unsigned int reqSize = webrtc::CalcBufferSize(webrtc::kI420,
-                                                  _encodedImage._encodedWidth,
-                                                  _encodedImage._encodedHeight);
+    size_t reqSize = webrtc::CalcBufferSize(webrtc::kI420,
+                                            _encodedImage._encodedWidth,
+                                            _encodedImage._encodedHeight);
     if (reqSize > _encodedImage._size)
     {
 
@@ -132,7 +132,7 @@ int32_t TbI420Encoder::Encode(
     }
 
     _encodedImage._length = reqSize;
-    _encodedCompleteCallback->Encoded(_encodedImage);
+    _encodedCompleteCallback->Encoded(_encodedImage, NULL, NULL);
     return WEBRTC_VIDEO_CODEC_OK;
 }
 
@@ -232,14 +232,14 @@ int32_t TbI420Decoder::Decode(
     }
 
     // Only send complete frames.
-    if (static_cast(inputImage._length) !=
-        webrtc::CalcBufferSize(webrtc::kI420,_width,_height)) {
+    if (webrtc::CalcBufferSize(webrtc::kI420,_width,_height) !=
+        inputImage._length) {
       return WEBRTC_VIDEO_CODEC_ERROR;
     }
 
-    int ret = ConvertToI420(webrtc::kI420, inputImage._buffer, 0, 0,
-                           _width, _height,
-                           0, webrtc::kRotateNone, &_decodedImage);
+    int ret =
+        ConvertToI420(webrtc::kI420, inputImage._buffer, 0, 0, _width, _height,
+                      0, webrtc::kVideoRotation_0, &_decodedImage);
 
     if (ret < 0)
       return WEBRTC_VIDEO_CODEC_ERROR;
diff --git a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
index 7ccedc2f10..d406985ef4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
@@ -25,7 +25,6 @@
 
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/video_engine/include/vie_network.h"
 
@@ -33,9 +32,6 @@
 #pragma warning(disable: 4355) // 'this' : used in base member initializer list
 #endif
 
-const uint8_t kSenderReportPayloadType = 200;
-const uint8_t kReceiverReportPayloadType = 201;
-
 TbExternalTransport::TbExternalTransport(
     webrtc::ViENetwork& vieNetwork,
     int sender_channel,
@@ -44,9 +40,8 @@ TbExternalTransport::TbExternalTransport(
       sender_channel_(sender_channel),
       receive_channels_(receive_channels),
       _vieNetwork(vieNetwork),
-      _thread(*webrtc::ThreadWrapper::CreateThread(
-          ViEExternalTransportRun, this, webrtc::kHighPriority,
-          "AutotestTransport")),
+      _thread(webrtc::ThreadWrapper::CreateThread(
+          ViEExternalTransportRun, this, "AutotestTransport")),
       _event(*webrtc::EventWrapper::Create()),
       _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
       _statCrit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
@@ -79,20 +74,17 @@ TbExternalTransport::TbExternalTransport(
       previous_drop_(false)
 {
     srand((int) webrtc::TickTime::MicrosecondTimestamp());
-    unsigned int tId = 0;
     memset(&network_parameters_, 0, sizeof(NetworkParameters));
-    _thread.Start(tId);
+    _thread->Start();
+    _thread->SetPriority(webrtc::kHighPriority);
 }
 
 TbExternalTransport::~TbExternalTransport()
 {
-    _thread.SetNotAlive();
     _event.Set();
-    if (_thread.Stop())
-    {
-        delete &_thread;
-        delete &_event;
-    }
+    _thread->Stop();
+    delete &_event;
+
     for (std::list::iterator it = _rtpPackets.begin();
          it != _rtpPackets.end(); ++it) {
         delete *it;
@@ -107,7 +99,7 @@ TbExternalTransport::~TbExternalTransport()
     delete &_statCrit;
 }
 
-int TbExternalTransport::SendPacket(int channel, const void *data, int len)
+int TbExternalTransport::SendPacket(int channel, const void *data, size_t len)
 {
   // Parse timestamp from RTP header according to RFC 3550, section 5.1.
     uint8_t* ptr = (uint8_t*)data;
@@ -137,7 +129,7 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
         ssrc += ptr[11];
         if (ssrc != _SSRC)
         {
-            return len; // return len to avoid error in trace file
+            return static_cast(len); // avoid error in trace file
         }
     }
     if (_temporalLayers) {
@@ -182,7 +174,7 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
             }
             if (_currentRelayLayer < TID)
             {
-                return len; // return len to avoid error in trace file
+                return static_cast(len); // avoid error in trace file
             }
             if (ptr[14] & 0x80) // 2 byte PID
             {
@@ -227,10 +219,11 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
         _statCrit.Enter();
         _dropCount++;
         _statCrit.Leave();
-        return len;
+        return static_cast(len);
     }
 
     VideoPacket* newPacket = new VideoPacket();
+    assert(len <= sizeof(newPacket->packetBuffer));
     memcpy(newPacket->packetBuffer, data, len);
 
     if (_temporalLayers)
@@ -266,7 +259,7 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
     _rtpPackets.push_back(newPacket);
     _event.Set();
     _crit.Leave();
-    return len;
+    return static_cast(len);
 }
 
 void TbExternalTransport::RegisterSendFrameCallback(
@@ -285,13 +278,16 @@ void TbExternalTransport::SetTemporalToggle(unsigned char layers)
     _temporalLayers = layers;
 }
 
-int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
+int TbExternalTransport::SendRTCPPacket(int channel,
+                                        const void *data,
+                                        size_t len)
 {
     _statCrit.Enter();
     _rtcpCount++;
     _statCrit.Leave();
 
     VideoPacket* newPacket = new VideoPacket();
+    assert(len <= sizeof(newPacket->packetBuffer));
     memcpy(newPacket->packetBuffer, data, len);
     newPacket->length = len;
     newPacket->channel = channel;
@@ -304,7 +300,7 @@ int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
     _rtcpPackets.push_back(newPacket);
     _event.Set();
     _crit.Leave();
-    return len;
+    return static_cast(len);
 }
 
 void TbExternalTransport::SetNetworkParameters(
@@ -495,8 +491,10 @@ bool TbExternalTransport::ViEExternalTransportProcess()
         // Send to ViE
         if (packet)
         {
-            uint8_t pltype = static_cast(packet->packetBuffer[1]);
-            if (pltype == kSenderReportPayloadType) {
+            uint8_t packet_type = static_cast(packet->packetBuffer[1]);
+            const uint8_t kSenderReportPacketType = 200;
+            const uint8_t kReceiverReportPacketType = 201;
+            if (packet_type == kSenderReportPacketType) {
               // Sender report.
               if (receive_channels_) {
                 for (SsrcChannelMap::iterator it = receive_channels_->begin();
@@ -510,7 +508,7 @@ bool TbExternalTransport::ViEExternalTransportProcess()
                                                packet->packetBuffer,
                                                packet->length);
               }
-            } else if (pltype == kReceiverReportPayloadType) {
+            } else if (packet_type == kReceiverReportPacketType) {
               // Receiver report.
               _vieNetwork.ReceivedRTCPPacket(sender_channel_,
                                              packet->packetBuffer,
diff --git a/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi b/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi
index 2bfc6536d8..4dd8614e73 100644
--- a/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi
+++ b/media/webrtc/trunk/webrtc/video_engine/video_engine_core.gypi
@@ -12,6 +12,7 @@
       'target_name': 'video_engine_core',
       'type': 'static_library',
       'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
 
         # common_video
        '<(webrtc_root)/common_video/common_video.gyp:common_video',
@@ -31,7 +32,7 @@
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
 
         # system_wrappers
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'sources': [
         # interface
@@ -49,6 +50,8 @@
         'call_stats.h',
         'encoder_state_feedback.h',
         'overuse_frame_detector.h',
+        'payload_router.h',
+        'report_block_stats.h',
         'stream_synchronization.h',
         'vie_base_impl.h',
         'vie_capture_impl.h',
@@ -82,6 +85,8 @@
         'call_stats.cc',
         'encoder_state_feedback.cc',
         'overuse_frame_detector.cc',
+        'payload_router.cc',
+        'report_block_stats.cc',
         'stream_synchronization.cc',
         'vie_base_impl.cc',
         'vie_capture_impl.cc',
@@ -132,6 +137,8 @@
             'call_stats_unittest.cc',
             'encoder_state_feedback_unittest.cc',
             'overuse_frame_detector_unittest.cc',
+            'payload_router_unittest.cc',
+            'report_block_stats_unittest.cc',
             'stream_synchronization_unittest.cc',
             'vie_capturer_unittest.cc',
             'vie_codec_unittest.cc',
diff --git a/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate b/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate
index b95d84c45c..c8d2fc9026 100644
--- a/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate
+++ b/media/webrtc/trunk/webrtc/video_engine/video_engine_core_unittests.isolate
@@ -7,14 +7,6 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
-      'variables': {
-        'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
-        ],
-      },
-    }],
     ['OS=="linux" or OS=="mac" or OS=="win"', {
       'variables': {
         'command': [
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc
index 07546e9b6a..86b0d57a4b 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.cc
@@ -124,6 +124,23 @@ int ViEBaseImpl::SetCpuOveruseOptions(int video_channel,
   return -1;
 }
 
+void ViEBaseImpl::RegisterCpuOveruseMetricsObserver(
+    int video_channel,
+    CpuOveruseMetricsObserver* observer) {
+  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+  assert(vie_encoder);
+
+  ViEInputManagerScoped is(*(shared_data_.input_manager()));
+  ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
+  assert(provider != NULL);
+
+  ViECapturer* capturer = is.Capture(provider->Id());
+  assert(capturer);
+
+  capturer->RegisterCpuOveruseMetricsObserver(observer);
+}
+
 int ViEBaseImpl::GetCpuOveruseMetrics(int video_channel,
                                       CpuOveruseMetrics* metrics) {
   ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
@@ -173,12 +190,18 @@ int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
 
 int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
                                int original_channel) {
-  return CreateChannel(video_channel, original_channel, true);
+  return CreateChannel(video_channel, original_channel, true, false);
+}
+
+int ViEBaseImpl::CreateChannelWithoutDefaultEncoder(
+    int& video_channel,  // NOLINT
+    int original_channel) {
+  return CreateChannel(video_channel, original_channel, true, true);
 }
 
 int ViEBaseImpl::CreateReceiveChannel(int& video_channel,  // NOLINT
                                       int original_channel) {
-  return CreateChannel(video_channel, original_channel, false);
+  return CreateChannel(video_channel, original_channel, false, true);
 }
 
 int ViEBaseImpl::DeleteChannel(const int video_channel) {
@@ -192,7 +215,7 @@ int ViEBaseImpl::DeleteChannel(const int video_channel) {
 
     // Deregister the ViEEncoder if no other channel is using it.
     ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-    if (cs.ChannelUsingViEEncoder(video_channel) == false) {
+    if (!cs.ChannelUsingViEEncoder(video_channel)) {
       ViEInputManagerScoped is(*(shared_data_.input_manager()));
       ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
       if (provider) {
@@ -333,7 +356,7 @@ int ViEBaseImpl::StopReceive(const int video_channel) {
 
 int ViEBaseImpl::GetVersion(char version[1024]) {
   assert(version != NULL);
-  strcpy(version, "VideoEngine 39");
+  strcpy(version, "VideoEngine 42");
   return 0;
 }
 
@@ -342,16 +365,18 @@ int ViEBaseImpl::LastError() {
 }
 
 int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
-                               int original_channel, bool sender) {
+                               int original_channel,
+                               bool sender,
+                               bool disable_default_encoder) {
   ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
   if (!cs.Channel(original_channel)) {
     shared_data_.SetLastError(kViEBaseInvalidChannelId);
     return -1;
   }
 
-  if (shared_data_.channel_manager()->CreateChannel(&video_channel,
-                                                    original_channel,
-                                                    sender) == -1) {
+  if (shared_data_.channel_manager()->CreateChannel(
+          &video_channel, original_channel, sender, disable_default_encoder) ==
+      -1) {
     video_channel = -1;
     shared_data_.SetLastError(kViEBaseChannelCreationFailed);
     return -1;
@@ -362,4 +387,32 @@ int ViEBaseImpl::CreateChannel(int& video_channel,  // NOLINT
   return 0;
 }
 
+void ViEBaseImpl::RegisterSendStatisticsProxy(
+    int channel,
+    SendStatisticsProxy* send_statistics_proxy) {
+  LOG_F(LS_VERBOSE) << "RegisterSendStatisticsProxy on channel " << channel;
+  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(channel);
+  if (!vie_channel) {
+    shared_data_.SetLastError(kViEBaseInvalidChannelId);
+    return;
+  }
+  ViEEncoder* vie_encoder = cs.Encoder(channel);
+  assert(vie_encoder);
+
+  vie_encoder->RegisterSendStatisticsProxy(send_statistics_proxy);
+}
+
+void ViEBaseImpl::RegisterReceiveStatisticsProxy(
+    int channel,
+    ReceiveStatisticsProxy* receive_statistics_proxy) {
+  LOG_F(LS_VERBOSE) << "RegisterReceiveStatisticsProxy on channel " << channel;
+  ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(channel);
+  if (!vie_channel) {
+    shared_data_.SetLastError(kViEBaseInvalidChannelId);
+    return;
+  }
+  vie_channel->RegisterReceiveStatisticsProxy(receive_statistics_proxy);
+}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h
index 199fb24240..9f92178462 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_base_impl.h
@@ -35,16 +35,22 @@ class ViEBaseImpl
                                          CpuOveruseObserver* observer);
   virtual int SetCpuOveruseOptions(int channel,
                                    const CpuOveruseOptions& options);
+  void RegisterCpuOveruseMetricsObserver(
+      int channel,
+      CpuOveruseMetricsObserver* observer) override;
   virtual int GetCpuOveruseMetrics(int channel,
                                    CpuOveruseMetrics* metrics);
-  virtual void RegisterSendSideDelayObserver(int channel,
-      SendSideDelayObserver* observer) OVERRIDE;
+  void RegisterSendSideDelayObserver(int channel,
+                                     SendSideDelayObserver* observer) override;
   virtual void SetLoadManager(CPULoadStateCallbackInvoker* aLoadManager);
   virtual int CreateChannel(int& video_channel);  // NOLINT
   virtual int CreateChannel(int& video_channel,  // NOLINT
                             const Config* config);
   virtual int CreateChannel(int& video_channel,  // NOLINT
                             int original_channel);
+  virtual int CreateChannelWithoutDefaultEncoder(int& video_channel,  // NOLINT
+                                                 int original_channel);
+
   virtual int CreateReceiveChannel(int& video_channel,  // NOLINT
                                    int original_channel);
   virtual int DeleteChannel(const int video_channel);
@@ -66,8 +72,14 @@ class ViEBaseImpl
 
  private:
   int CreateChannel(int& video_channel, int original_channel,  // NOLINT
-                    bool sender);
+                    bool sender, bool disable_default_encoder);
 
+  void RegisterSendStatisticsProxy(
+      int channel,
+      SendStatisticsProxy* send_statistics_proxy) override;
+  void RegisterReceiveStatisticsProxy(
+      int channel,
+      ReceiveStatisticsProxy* receive_statistics_proxy) override;
   // ViEBaseImpl owns ViESharedData used by all interface implementations.
   ViESharedData shared_data_;
 };
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc
index 9581fb7f2c..aa330cc0d5 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.cc
@@ -139,13 +139,6 @@ int ViECaptureImpl::ConnectCaptureDevice(const int capture_id,
   LOG(LS_INFO) << "Connect capture id " << capture_id
                << " to channel " << video_channel;
 
-  ViEInputManagerScoped is(*(shared_data_->input_manager()));
-  ViECapturer* vie_capture = is.Capture(capture_id);
-  if (!vie_capture) {
-    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
-    return -1;
-  }
-
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEEncoder* vie_encoder = cs.Encoder(video_channel);
   if (!vie_encoder) {
@@ -158,6 +151,13 @@ int ViECaptureImpl::ConnectCaptureDevice(const int capture_id,
     shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
     return -1;
   }
+
+  ViEInputManagerScoped is(*(shared_data_->input_manager()));
+  ViECapturer* vie_capture = is.Capture(capture_id);
+  if (!vie_capture) {
+    shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
+    return -1;
+  }
   //  Check if the encoder already has a connected frame provider
   if (is.FrameProvider(vie_encoder) != NULL) {
     LOG(LS_ERROR) << "Channel already connected to capture device.";
@@ -252,26 +252,10 @@ int ViECaptureImpl::StopCapture(const int capture_id) {
   return 0;
 }
 
-int ViECaptureImpl::SetRotateCapturedFrames(
-    const int capture_id,
-    const RotateCapturedFrame rotation) {
-  int i_rotation = -1;
-  switch (rotation) {
-    case RotateCapturedFrame_0:
-      i_rotation = 0;
-      break;
-    case RotateCapturedFrame_90:
-      i_rotation = 90;
-      break;
-    case RotateCapturedFrame_180:
-      i_rotation = 180;
-      break;
-    case RotateCapturedFrame_270:
-      i_rotation = 270;
-      break;
-  }
-  LOG(LS_INFO) << "SetRotateCaptureFrames for " << capture_id
-               << ", rotation " << i_rotation;
+int ViECaptureImpl::SetVideoRotation(const int capture_id,
+                                     const VideoRotation rotation) {
+  LOG(LS_INFO) << "SetRotateCaptureFrames for " << capture_id << ", rotation "
+               << static_cast(rotation);
 
   ViEInputManagerScoped is(*(shared_data_->input_manager()));
   ViECapturer* vie_capture = is.Capture(capture_id);
@@ -279,7 +263,7 @@ int ViECaptureImpl::SetRotateCapturedFrames(
     shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
     return -1;
   }
-  if (vie_capture->SetRotateCapturedFrames(rotation) != 0) {
+  if (vie_capture->SetVideoRotation(rotation) != 0) {
     shared_data_->SetLastError(kViECaptureDeviceUnknownError);
     return -1;
   }
@@ -367,7 +351,7 @@ int ViECaptureImpl::ShowCaptureSettingsDialogBox(
 }
 
 int ViECaptureImpl::GetOrientation(const char* unique_idUTF8,
-                                   RotateCapturedFrame& orientation) {
+                                   VideoRotation& orientation) {
   if (shared_data_->input_manager()->GetOrientation(
       unique_idUTF8,
       orientation) != 0) {
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h
index 541f7a2473..99be9366b6 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_capture_impl.h
@@ -47,8 +47,8 @@ class ViECaptureImpl
       const int capture_id,
       const CaptureCapability& capture_capability = CaptureCapability());
   virtual int StopCapture(const int capture_id);
-  virtual int SetRotateCapturedFrames(const int capture_id,
-                                      const RotateCapturedFrame rotation);
+  virtual int SetVideoRotation(const int capture_id,
+                               const VideoRotation rotation);
   virtual int SetCaptureDelay(const int capture_id,
                               const unsigned int capture_delay_ms);
   virtual int NumberOfCapabilities(const char* unique_idUTF8,
@@ -62,7 +62,7 @@ class ViECaptureImpl
     const char* dialog_title, void* parent_window = NULL,
     const unsigned int x = 200, const unsigned int y = 200);
   virtual int GetOrientation(const char* unique_idUTF8,
-                             RotateCapturedFrame& orientation);
+                             VideoRotation& orientation);
   virtual int EnableBrightnessAlarm(const int capture_id, const bool enable);
   virtual int RegisterObserver(const int capture_id,
                                ViECaptureObserver& observer);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc b/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc
index e37102cd5d..7041b6b0fe 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_capturer.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/video_engine/vie_capturer.h"
 
-#include "webrtc/common_video/interface/texture_video_frame.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
@@ -21,7 +20,7 @@
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video_engine/include/vie_image_process.h"
 #include "webrtc/video_engine/overuse_frame_detector.h"
@@ -33,23 +32,52 @@ namespace webrtc {
 
 const int kThreadWaitTimeMs = 100;
 
+class RegistrableCpuOveruseMetricsObserver : public CpuOveruseMetricsObserver {
+ public:
+  void CpuOveruseMetricsUpdated(const CpuOveruseMetrics& metrics) override {
+    rtc::CritScope lock(&crit_);
+    if (observer_)
+      observer_->CpuOveruseMetricsUpdated(metrics);
+    metrics_ = metrics;
+  }
+
+  CpuOveruseMetrics GetCpuOveruseMetrics() const {
+    rtc::CritScope lock(&crit_);
+    return metrics_;
+  }
+
+  void Set(CpuOveruseMetricsObserver* observer) {
+    rtc::CritScope lock(&crit_);
+    observer_ = observer;
+  }
+
+ private:
+  mutable rtc::CriticalSection crit_;
+  CpuOveruseMetricsObserver* observer_ GUARDED_BY(crit_) = nullptr;
+  CpuOveruseMetrics metrics_ GUARDED_BY(crit_);
+};
+
 ViECapturer::ViECapturer(int capture_id,
                          int engine_id,
                          const Config& config,
                          ProcessThread& module_process_thread)
     : ViEFrameProviderBase(capture_id, engine_id),
       capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      deliver_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      effects_and_stats_cs_(CriticalSectionWrapper::CreateCriticalSection()),
       capture_module_(NULL),
-      external_capture_module_(NULL),
+      use_external_capture_(false),
       module_process_thread_(module_process_thread),
       capture_id_(capture_id),
       incoming_frame_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      capture_thread_(*ThreadWrapper::CreateThread(ViECaptureThreadFunction,
-                                                   this, kHighPriority,
-                                                   "ViECaptureThread")),
+      capture_thread_(ThreadWrapper::CreateThread(
+          ViECaptureThreadFunction, this, "ViECaptureThread")),
       capture_event_(*EventWrapper::Create()),
       deliver_event_(*EventWrapper::Create()),
+      stop_(0),
+      last_captured_timestamp_(0),
+      delta_ntp_internal_ms_(
+          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() -
+          TickTime::MillisecondTimestamp()),
       effect_filter_(NULL),
       image_proc_module_(NULL),
       image_proc_module_ref_counter_(0),
@@ -59,12 +87,13 @@ ViECapturer::ViECapturer(int capture_id,
       reported_brightness_level_(Normal),
       observer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
       observer_(NULL),
-      overuse_detector_(new OveruseFrameDetector(Clock::GetRealTimeClock())),
+      cpu_overuse_metrics_observer_(new RegistrableCpuOveruseMetricsObserver()),
+      overuse_detector_(
+          new OveruseFrameDetector(Clock::GetRealTimeClock(),
+                                   cpu_overuse_metrics_observer_.get())),
       config_(config) {
-  unsigned int t_id = 0;
-  if (!capture_thread_.Start(t_id)) {
-    assert(false);
-  }
+  capture_thread_->Start();
+  capture_thread_->SetPriority(kHighPriority);
   module_process_thread_.RegisterModule(overuse_detector_.get());
 }
 
@@ -72,12 +101,8 @@ ViECapturer::~ViECapturer() {
   module_process_thread_.DeRegisterModule(overuse_detector_.get());
 
   // Stop the thread.
-  deliver_cs_->Enter();
-  capture_cs_->Enter();
-  capture_thread_.SetNotAlive();
+  rtc::AtomicOps::Increment(&stop_);
   capture_event_.Set();
-  capture_cs_->Leave();
-  deliver_cs_->Leave();
 
   // Stop the camera input.
   if (capture_module_) {
@@ -86,14 +111,10 @@ ViECapturer::~ViECapturer() {
     capture_module_->Release();
     capture_module_ = NULL;
   }
-  if (capture_thread_.Stop()) {
-    // Thread stopped.
-    delete &capture_thread_;
-    delete &capture_event_;
-    delete &deliver_event_;
-  } else {
-    assert(false);
-  }
+
+  capture_thread_->Stop();
+  delete &capture_event_;
+  delete &deliver_event_;
 
   if (image_proc_module_) {
     VideoProcessingModule::Destroy(image_proc_module_);
@@ -125,10 +146,7 @@ int32_t ViECapturer::Init(VideoCaptureModule* capture_module) {
   capture_module_ = capture_module;
   capture_module_->RegisterCaptureDataCallback(*this);
   capture_module_->AddRef();
-  if (module_process_thread_.RegisterModule(capture_module_) != 0) {
-    return -1;
-  }
-
+  module_process_thread_.RegisterModule(capture_module_);
   return 0;
 }
 
@@ -160,8 +178,9 @@ int32_t ViECapturer::Init(const char* device_unique_idUTF8,
       ViEModuleId(engine_id_, capture_id_), device_unique_idUTF8, type);
 #endif
   } else if (device_unique_idUTF8 == NULL) {
-    capture_module_  = VideoCaptureFactory::Create(
-      ViEModuleId(engine_id_, capture_id_), external_capture_module_);
+    use_external_capture_ = true;
+    // YYY: was      ViEModuleId(engine_id_, capture_id_), external_capture_module_);
+    return 0;
   } else {
     capture_module_ = VideoCaptureFactory::Create(
       ViEModuleId(engine_id_, capture_id_), device_unique_idUTF8);
@@ -171,14 +190,14 @@ int32_t ViECapturer::Init(const char* device_unique_idUTF8,
   }
   capture_module_->AddRef();
   capture_module_->RegisterCaptureDataCallback(*this);
-  if (module_process_thread_.RegisterModule(capture_module_) != 0) {
-    return -1;
-  }
+  module_process_thread_.RegisterModule(capture_module_);
 
   return 0;
 }
 
 int ViECapturer::FrameCallbackChanged() {
+  if (use_external_capture_)
+    return -1;
   if (Started() && !CaptureCapabilityFixed()) {
     // Reconfigure the camera if a new size is required and the capture device
     // does not provide encoded frames.
@@ -202,6 +221,8 @@ int ViECapturer::FrameCallbackChanged() {
 }
 
 int32_t ViECapturer::Start(const CaptureCapability& capture_capability) {
+  if (use_external_capture_)
+    return -1;
   int width;
   int height;
   int frame_rate;
@@ -243,15 +264,21 @@ int32_t ViECapturer::Start(const CaptureCapability& capture_capability) {
 }
 
 int32_t ViECapturer::Stop() {
+  if (use_external_capture_)
+    return -1;
   requested_capability_ = CaptureCapability();
   return capture_module_->StopCapture();
 }
 
 bool ViECapturer::Started() {
+  if (use_external_capture_)
+    return false;
   return capture_module_->CaptureStarted();
 }
 
 const char* ViECapturer::CurrentDeviceName() const {
+  if (use_external_capture_)
+    return "";
   return capture_module_->CurrentDeviceName();
 }
 
@@ -263,114 +290,75 @@ void ViECapturer::SetCpuOveruseOptions(const CpuOveruseOptions& options) {
   overuse_detector_->SetOptions(options);
 }
 
+void ViECapturer::RegisterCpuOveruseMetricsObserver(
+    CpuOveruseMetricsObserver* observer) {
+  cpu_overuse_metrics_observer_->Set(observer);
+}
+
 void ViECapturer::GetCpuOveruseMetrics(CpuOveruseMetrics* metrics) const {
-  overuse_detector_->GetCpuOveruseMetrics(metrics);
+  *metrics = cpu_overuse_metrics_observer_->GetCpuOveruseMetrics();
 }
 
 int32_t ViECapturer::SetCaptureDelay(int32_t delay_ms) {
+  if (use_external_capture_)
+    return -1;
   capture_module_->SetCaptureDelay(delay_ms);
   return 0;
 }
 
-int32_t ViECapturer::SetRotateCapturedFrames(
-  const RotateCapturedFrame rotation) {
-  VideoCaptureRotation converted_rotation = kCameraRotate0;
-  switch (rotation) {
-    case RotateCapturedFrame_0:
-      converted_rotation = kCameraRotate0;
-      break;
-    case RotateCapturedFrame_90:
-      converted_rotation = kCameraRotate90;
-      break;
-    case RotateCapturedFrame_180:
-      converted_rotation = kCameraRotate180;
-      break;
-    case RotateCapturedFrame_270:
-      converted_rotation = kCameraRotate270;
-      break;
-  }
-  return capture_module_->SetCaptureRotation(converted_rotation);
+int32_t ViECapturer::SetVideoRotation(const VideoRotation rotation) {
+  if (use_external_capture_)
+    return -1;
+  return capture_module_->SetCaptureRotation(rotation);
 }
 
-int ViECapturer::IncomingFrame(unsigned char* video_frame,
-                               unsigned int video_frame_length,
-                               uint16_t width,
-                               uint16_t height,
-                               RawVideoType video_type,
-                               unsigned long long capture_time) {  // NOLINT
-  if (!external_capture_module_) {
-    return -1;
-  }
-  VideoCaptureCapability capability;
-  capability.width = width;
-  capability.height = height;
-  capability.rawType = video_type;
-  return external_capture_module_->IncomingFrame(video_frame,
-                                                 video_frame_length,
-                                                 capability, capture_time);
-}
-
-int ViECapturer::IncomingFrameI420(const ViEVideoFrameI420& video_frame,
-                                   unsigned long long capture_time) {  // NOLINT
-  if (!external_capture_module_) {
-    return -1;
-  }
-
-  int size_y = video_frame.height * video_frame.y_pitch;
-  int size_u = video_frame.u_pitch * ((video_frame.height + 1) / 2);
-  int size_v = video_frame.v_pitch * ((video_frame.height + 1) / 2);
-  CriticalSectionScoped cs(incoming_frame_cs_.get());
-  int ret = incoming_frame_.CreateFrame(size_y,
-                                       video_frame.y_plane,
-                                       size_u,
-                                       video_frame.u_plane,
-                                       size_v,
-                                       video_frame.v_plane,
-                                       video_frame.width,
-                                       video_frame.height,
-                                       video_frame.y_pitch,
-                                       video_frame.u_pitch,
-                                       video_frame.v_pitch);
-
-  if (ret < 0) {
-    LOG_F(LS_ERROR) << "Could not create I420Frame.";
-    return -1;
-  }
-
-  return external_capture_module_->IncomingI420VideoFrame(&incoming_frame_,
-                                                          capture_time);
-}
-
-void ViECapturer::SwapFrame(I420VideoFrame* frame) {
-  external_capture_module_->IncomingI420VideoFrame(frame,
-                                                   frame->render_time_ms());
-  frame->set_timestamp(0);
-  frame->set_ntp_time_ms(0);
-  frame->set_render_time_ms(0);
+void ViECapturer::IncomingFrame(const I420VideoFrame& frame) {
+  OnIncomingCapturedFrame(-1, frame);
 }
 
 void ViECapturer::OnIncomingCapturedFrame(const int32_t capture_id,
-                                          I420VideoFrame& video_frame) {
-  CriticalSectionScoped cs(capture_cs_.get());
-  // Make sure we render this frame earlier since we know the render time set
-  // is slightly off since it's being set when the frame has been received from
-  // the camera, and not when the camera actually captured the frame.
-  video_frame.set_render_time_ms(video_frame.render_time_ms() - FrameDelay());
+                                          const I420VideoFrame& video_frame) {
+  I420VideoFrame incoming_frame = video_frame;
 
-  overuse_detector_->FrameCaptured(video_frame.width(),
-                                   video_frame.height(),
-                                   video_frame.render_time_ms());
+  if (incoming_frame.ntp_time_ms() != 0) {
+    // If a NTP time stamp is set, this is the time stamp we will use.
+    incoming_frame.set_render_time_ms(
+        incoming_frame.ntp_time_ms() - delta_ntp_internal_ms_);
+  } else {  // NTP time stamp not set.
+    int64_t render_time = incoming_frame.render_time_ms() != 0 ?
+        incoming_frame.render_time_ms() : TickTime::MillisecondTimestamp();
+
+    // Make sure we render this frame earlier since we know the render time set
+    // is slightly off since it's being set when the frame was received
+    // from the camera, and not when the camera actually captured the frame.
+    render_time -= FrameDelay();
+    incoming_frame.set_render_time_ms(render_time);
+    incoming_frame.set_ntp_time_ms(
+        render_time + delta_ntp_internal_ms_);
+  }
+
+  // Convert NTP time, in ms, to RTP timestamp.
+  const int kMsToRtpTimestamp = 90;
+  incoming_frame.set_timestamp(kMsToRtpTimestamp *
+      static_cast(incoming_frame.ntp_time_ms()));
+
+  CriticalSectionScoped cs(capture_cs_.get());
+  if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
+    // We don't allow the same capture time for two frames, drop this one.
+    LOG(LS_WARNING) << "Same/old NTP timestamp for incoming frame. Dropping.";
+    return;
+  }
+
+  captured_frame_.ShallowCopy(incoming_frame);
+  last_captured_timestamp_ = incoming_frame.ntp_time_ms();
+
+  overuse_detector_->FrameCaptured(captured_frame_.width(),
+                                   captured_frame_.height(),
+                                   captured_frame_.render_time_ms());
 
   TRACE_EVENT_ASYNC_BEGIN1("webrtc", "Video", video_frame.render_time_ms(),
                            "render_time", video_frame.render_time_ms());
 
-  if (video_frame.native_handle() != NULL) {
-    captured_frame_.reset(video_frame.CloneFrame());
-  } else {
-    if (captured_frame_ == NULL || captured_frame_->native_handle() != NULL)
-      captured_frame_.reset(new I420VideoFrame());
-    captured_frame_->SwapFrame(&video_frame);
-  }
   capture_event_.Set();
 }
 
@@ -385,7 +373,7 @@ void ViECapturer::OnCaptureDelayChanged(const int32_t id,
 
 int32_t ViECapturer::RegisterEffectFilter(
     ViEEffectFilter* effect_filter) {
-  CriticalSectionScoped cs(deliver_cs_.get());
+  CriticalSectionScoped cs(effects_and_stats_cs_.get());
 
   if (effect_filter != NULL && effect_filter_ != NULL) {
     LOG_F(LS_ERROR) << "Effect filter already registered.";
@@ -420,7 +408,7 @@ int32_t ViECapturer::DecImageProcRefCount() {
 }
 
 int32_t ViECapturer::EnableDeflickering(bool enable) {
-  CriticalSectionScoped cs(deliver_cs_.get());
+  CriticalSectionScoped cs(effects_and_stats_cs_.get());
   if (enable) {
     if (deflicker_frame_stats_) {
       return -1;
@@ -441,7 +429,7 @@ int32_t ViECapturer::EnableDeflickering(bool enable) {
 }
 
 int32_t ViECapturer::EnableBrightnessAlarm(bool enable) {
-  CriticalSectionScoped cs(deliver_cs_.get());
+  CriticalSectionScoped cs(effects_and_stats_cs_.get());
   if (enable) {
     if (brightness_frame_stats_) {
       return -1;
@@ -468,17 +456,24 @@ bool ViECapturer::ViECaptureThreadFunction(void* obj) {
 bool ViECapturer::ViECaptureProcess() {
   int64_t capture_time = -1;
   if (capture_event_.Wait(kThreadWaitTimeMs) == kEventSignaled) {
+    if (rtc::AtomicOps::Load(&stop_))
+      return false;
+
     overuse_detector_->FrameProcessingStarted();
     int64_t encode_start_time = -1;
-    deliver_cs_->Enter();
-    if (SwapCapturedAndDeliverFrameIfAvailable()) {
-      capture_time = deliver_frame_->render_time_ms();
-      encode_start_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
-      DeliverI420Frame(deliver_frame_.get());
-      if (deliver_frame_->native_handle() != NULL)
-        deliver_frame_.reset();  // Release the texture so it can be reused.
+    I420VideoFrame deliver_frame;
+    {
+      CriticalSectionScoped cs(capture_cs_.get());
+      if (!captured_frame_.IsZeroSize()) {
+        deliver_frame = captured_frame_;
+        captured_frame_.Reset();
+      }
+    }
+    if (!deliver_frame.IsZeroSize()) {
+      capture_time = deliver_frame.render_time_ms();
+      encode_start_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
+      DeliverI420Frame(&deliver_frame);
     }
-    deliver_cs_->Leave();
     if (current_brightness_level_ != reported_brightness_level_) {
       CriticalSectionScoped cs(observer_cs_.get());
       if (observer_) {
@@ -501,66 +496,57 @@ bool ViECapturer::ViECaptureProcess() {
 
 void ViECapturer::DeliverI420Frame(I420VideoFrame* video_frame) {
   if (video_frame->native_handle() != NULL) {
-    ViEFrameProviderBase::DeliverFrame(video_frame);
+    ViEFrameProviderBase::DeliverFrame(video_frame, std::vector());
     return;
   }
 
   // Apply image enhancement and effect filter.
-  if (deflicker_frame_stats_) {
-    if (image_proc_module_->GetFrameStats(deflicker_frame_stats_,
-                                          *video_frame) == 0) {
-      image_proc_module_->Deflickering(video_frame, deflicker_frame_stats_);
-    } else {
-      LOG_F(LS_ERROR) << "Could not get frame stats.";
-    }
-  }
-  if (brightness_frame_stats_) {
-    if (image_proc_module_->GetFrameStats(brightness_frame_stats_,
-                                          *video_frame) == 0) {
-      int32_t brightness = image_proc_module_->BrightnessDetection(
-          *video_frame, *brightness_frame_stats_);
-
-      switch (brightness) {
-      case VideoProcessingModule::kNoWarning:
-        current_brightness_level_ = Normal;
-        break;
-      case VideoProcessingModule::kDarkWarning:
-        current_brightness_level_ = Dark;
-        break;
-      case VideoProcessingModule::kBrightWarning:
-        current_brightness_level_ = Bright;
-        break;
-      default:
-        break;
+  {
+    CriticalSectionScoped cs(effects_and_stats_cs_.get());
+    if (deflicker_frame_stats_) {
+      if (image_proc_module_->GetFrameStats(deflicker_frame_stats_,
+                                            *video_frame) == 0) {
+        image_proc_module_->Deflickering(video_frame, deflicker_frame_stats_);
+      } else {
+        LOG_F(LS_ERROR) << "Could not get frame stats.";
       }
     }
-  }
-  if (effect_filter_) {
-    unsigned int length = CalcBufferSize(kI420,
-                                         video_frame->width(),
-                                         video_frame->height());
-    scoped_ptr video_buffer(new uint8_t[length]);
-    ExtractBuffer(*video_frame, length, video_buffer.get());
-    effect_filter_->Transform(length,
-                              video_buffer.get(),
-                              video_frame->ntp_time_ms(),
-                              video_frame->timestamp(),
-                              video_frame->width(),
-                              video_frame->height());
+    if (brightness_frame_stats_) {
+      if (image_proc_module_->GetFrameStats(brightness_frame_stats_,
+                                            *video_frame) == 0) {
+        int32_t brightness = image_proc_module_->BrightnessDetection(
+            *video_frame, *brightness_frame_stats_);
+
+        switch (brightness) {
+          case VideoProcessingModule::kNoWarning:
+            current_brightness_level_ = Normal;
+            break;
+          case VideoProcessingModule::kDarkWarning:
+            current_brightness_level_ = Dark;
+            break;
+          case VideoProcessingModule::kBrightWarning:
+            current_brightness_level_ = Bright;
+            break;
+          default:
+            break;
+        }
+      }
+    }
+    if (effect_filter_) {
+      size_t length =
+          CalcBufferSize(kI420, video_frame->width(), video_frame->height());
+      rtc::scoped_ptr video_buffer(new uint8_t[length]);
+      ExtractBuffer(*video_frame, length, video_buffer.get());
+      effect_filter_->Transform(length,
+                                video_buffer.get(),
+                                video_frame->ntp_time_ms(),
+                                video_frame->timestamp(),
+                                video_frame->width(),
+                                video_frame->height());
+    }
   }
   // Deliver the captured frame to all observers (channels, renderer or file).
-  ViEFrameProviderBase::DeliverFrame(video_frame);
-}
-
-int ViECapturer::DeregisterFrameCallback(
-    const ViEFrameCallback* callbackObject) {
-  return ViEFrameProviderBase::DeregisterFrameCallback(callbackObject);
-}
-
-bool ViECapturer::IsFrameCallbackRegistered(
-    const ViEFrameCallback* callbackObject) {
-  CriticalSectionScoped cs(provider_cs_.get());
-  return ViEFrameProviderBase::IsFrameCallbackRegistered(callbackObject);
+  ViEFrameProviderBase::DeliverFrame(video_frame, std::vector());
 }
 
 bool ViECapturer::CaptureCapabilityFixed() {
@@ -614,24 +600,4 @@ void ViECapturer::OnNoPictureAlarm(const int32_t id,
   observer_->NoPictureAlarm(id, vie_alarm);
 }
 
-bool ViECapturer::SwapCapturedAndDeliverFrameIfAvailable() {
-  CriticalSectionScoped cs(capture_cs_.get());
-  if (captured_frame_ == NULL)
-    return false;
-
-  if (captured_frame_->native_handle() != NULL) {
-    deliver_frame_.reset(captured_frame_.release());
-    return true;
-  }
-
-  if (captured_frame_->IsZeroSize())
-    return false;
-
-  if (deliver_frame_ == NULL)
-    deliver_frame_.reset(new I420VideoFrame());
-  deliver_frame_->SwapFrame(captured_frame_.get());
-  captured_frame_->ResetSize();
-  return true;
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h b/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h
index 622e3e5732..ac97f56568 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_capturer.h
@@ -13,6 +13,8 @@
 
 #include 
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
@@ -20,7 +22,8 @@
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/include/vie_capture.h"
@@ -36,10 +39,10 @@ class EventWrapper;
 class CpuOveruseObserver;
 class OveruseFrameDetector;
 class ProcessThread;
-class ThreadWrapper;
 class ViEEffectFilter;
 class ViEEncoder;
 struct ViEPicture;
+class RegistrableCpuOveruseMetricsObserver;
 
 class ViECapturer
     : public ViEFrameProviderBase,
@@ -65,21 +68,9 @@ class ViECapturer
 
   // Implements ViEFrameProviderBase.
   int FrameCallbackChanged();
-  virtual int DeregisterFrameCallback(const ViEFrameCallback* callbackObject);
-  bool IsFrameCallbackRegistered(const ViEFrameCallback* callbackObject);
 
   // Implements ExternalCapture.
-  virtual int IncomingFrame(unsigned char* video_frame,
-                            unsigned int video_frame_length,
-                            uint16_t width,
-                            uint16_t height,
-                            RawVideoType video_type,
-                            unsigned long long capture_time = 0);  // NOLINT
-
-  virtual int IncomingFrameI420(const ViEVideoFrameI420& video_frame,
-                                unsigned long long capture_time = 0);  // NOLINT
-
-  virtual void SwapFrame(I420VideoFrame* frame) OVERRIDE;
+  void IncomingFrame(const I420VideoFrame& frame) override;
 
   // Start/Stop.
   int32_t Start(
@@ -91,7 +82,7 @@ class ViECapturer
   int32_t SetCaptureDelay(int32_t delay_ms);
 
   // Sets rotation of the incoming captured frame.
-  int32_t SetRotateCapturedFrames(const RotateCapturedFrame rotation);
+  int32_t SetVideoRotation(const VideoRotation rotation);
 
   // Effect filter.
   int32_t RegisterEffectFilter(ViEEffectFilter* effect_filter);
@@ -108,6 +99,7 @@ class ViECapturer
 
   void RegisterCpuOveruseObserver(CpuOveruseObserver* observer);
   void SetCpuOveruseOptions(const CpuOveruseOptions& options);
+  void RegisterCpuOveruseMetricsObserver(CpuOveruseMetricsObserver* observer);
   void GetCpuOveruseMetrics(CpuOveruseMetrics* metrics) const;
 
  protected:
@@ -122,7 +114,7 @@ class ViECapturer
 
   // Implements VideoCaptureDataCallback.
   virtual void OnIncomingCapturedFrame(const int32_t id,
-                                       I420VideoFrame& video_frame);
+                                       const I420VideoFrame& video_frame);
   virtual void OnCaptureDelayChanged(const int32_t id,
                                      const int32_t delay);
 
@@ -146,48 +138,55 @@ class ViECapturer
   static bool ViECaptureThreadFunction(void* obj);
   bool ViECaptureProcess();
 
-  void DeliverI420Frame(I420VideoFrame* video_frame);
-  void DeliverCodedFrame(VideoFrame* video_frame);
-
  private:
-  bool SwapCapturedAndDeliverFrameIfAvailable();
+  void DeliverI420Frame(I420VideoFrame* video_frame);
 
-  // Never take capture_cs_ before deliver_cs_!
-  scoped_ptr capture_cs_;
-  scoped_ptr deliver_cs_;
+  // Never take capture_cs_ before effects_and_stats_cs_!
+  rtc::scoped_ptr capture_cs_;
+  rtc::scoped_ptr effects_and_stats_cs_;
   VideoCaptureModule* capture_module_;
-  VideoCaptureExternal* external_capture_module_;
+  bool use_external_capture_;
   ProcessThread& module_process_thread_;
   const int capture_id_;
 
   // Frame used in IncomingFrameI420.
-  scoped_ptr incoming_frame_cs_;
+  rtc::scoped_ptr incoming_frame_cs_;
   I420VideoFrame incoming_frame_;
 
   // Capture thread.
-  ThreadWrapper& capture_thread_;
+  rtc::scoped_ptr capture_thread_;
   EventWrapper& capture_event_;
   EventWrapper& deliver_event_;
 
-  scoped_ptr captured_frame_;
-  scoped_ptr deliver_frame_;
+  volatile int stop_;
+
+  I420VideoFrame captured_frame_ GUARDED_BY(capture_cs_.get());
+  // Used to make sure incoming time stamp is increasing for every frame.
+  int64_t last_captured_timestamp_;
+  // Delta used for translating between NTP and internal timestamps.
+  const int64_t delta_ntp_internal_ms_;
 
   // Image processing.
-  ViEEffectFilter* effect_filter_;
+  ViEEffectFilter* effect_filter_ GUARDED_BY(effects_and_stats_cs_.get());
   VideoProcessingModule* image_proc_module_;
   int image_proc_module_ref_counter_;
-  VideoProcessingModule::FrameStats* deflicker_frame_stats_;
-  VideoProcessingModule::FrameStats* brightness_frame_stats_;
+  VideoProcessingModule::FrameStats* deflicker_frame_stats_
+      GUARDED_BY(effects_and_stats_cs_.get());
+  VideoProcessingModule::FrameStats* brightness_frame_stats_
+      GUARDED_BY(effects_and_stats_cs_.get());
   Brightness current_brightness_level_;
   Brightness reported_brightness_level_;
 
   // Statistics observer.
-  scoped_ptr observer_cs_;
+  rtc::scoped_ptr observer_cs_;
   ViECaptureObserver* observer_ GUARDED_BY(observer_cs_.get());
 
   CaptureCapability requested_capability_;
 
-  scoped_ptr overuse_detector_;
+  // Must be declared before overuse_detector_ where it's registered.
+  const rtc::scoped_ptr
+      cpu_overuse_metrics_observer_;
+  rtc::scoped_ptr overuse_detector_;
   const Config & config_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc
index edaf13b54b..18aa49665f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_capturer_unittest.cc
@@ -16,15 +16,14 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/common_video/interface/native_handle.h"
-#include "webrtc/common_video/interface/texture_video_frame.h"
 #include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
 #include "webrtc/modules/video_capture/include/mock/mock_video_capture.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/ref_count.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/scoped_vector.h"
 #include "webrtc/video_engine/mock/mock_vie_frame_provider_base.h"
 
@@ -69,7 +68,7 @@ class ViECapturerTest : public ::testing::Test {
   virtual void SetUp() {
     EXPECT_CALL(*mock_capture_module_, RegisterCaptureDataCallback(_))
         .WillRepeatedly(Invoke(this, &ViECapturerTest::SetCaptureDataCallback));
-    EXPECT_CALL(*mock_frame_callback_, DeliverFrame(_, _, _, _))
+    EXPECT_CALL(*mock_frame_callback_, DeliverFrame(_, _, _))
         .WillRepeatedly(
             WithArg<1>(Invoke(this, &ViECapturerTest::AddOutputFrame)));
 
@@ -81,6 +80,7 @@ class ViECapturerTest : public ::testing::Test {
   }
 
   virtual void TearDown() {
+    vie_capturer_->DeregisterFrameCallback(mock_frame_callback_.get());
     // ViECapturer accesses |mock_process_thread_| in destructor and should
     // be deleted first.
     vie_capturer_.reset();
@@ -94,11 +94,10 @@ class ViECapturerTest : public ::testing::Test {
     data_callback_->OnIncomingCapturedFrame(0, *frame);
   }
 
-  void AddOutputFrame(I420VideoFrame* frame) {
+  void AddOutputFrame(const I420VideoFrame* frame) {
     if (frame->native_handle() == NULL)
       output_frame_ybuffers_.push_back(frame->buffer(kYPlane));
-    // Clone the frames because ViECapturer owns the frames.
-    output_frames_.push_back(frame->CloneFrame());
+    output_frames_.push_back(new I420VideoFrame(*frame));
     output_frame_event_->Set();
   }
 
@@ -106,35 +105,87 @@ class ViECapturerTest : public ::testing::Test {
     EXPECT_EQ(kEventSignaled, output_frame_event_->Wait(FRAME_TIMEOUT_MS));
   }
 
-  scoped_ptr mock_capture_module_;
-  scoped_ptr mock_process_thread_;
-  scoped_ptr mock_frame_callback_;
+  rtc::scoped_ptr mock_capture_module_;
+  rtc::scoped_ptr mock_process_thread_;
+  rtc::scoped_ptr mock_frame_callback_;
 
   // Used to send input capture frames to ViECapturer.
   VideoCaptureDataCallback* data_callback_;
 
-  scoped_ptr vie_capturer_;
+  rtc::scoped_ptr vie_capturer_;
 
   // Input capture frames of ViECapturer.
   ScopedVector input_frames_;
 
   // Indicate an output frame has arrived.
-  scoped_ptr output_frame_event_;
+  rtc::scoped_ptr output_frame_event_;
 
   // Output delivered frames of ViECaptuer.
   ScopedVector output_frames_;
 
   // The pointers of Y plane buffers of output frames. This is used to verify
   // the frame are swapped and not copied.
-  std::vector output_frame_ybuffers_;
+  std::vector output_frame_ybuffers_;
 };
 
+TEST_F(ViECapturerTest, DoesNotRetainHandleNorCopyBuffer) {
+  // Indicate an output frame has arrived.
+  rtc::scoped_ptr frame_destroyed_event(EventWrapper::Create());
+  class TestBuffer : public webrtc::I420Buffer {
+   public:
+    TestBuffer(EventWrapper* event) : I420Buffer(5, 5), event_(event) {}
+
+   private:
+    friend class rtc::RefCountedObject;
+    ~TestBuffer() override { event_->Set(); }
+    EventWrapper* event_;
+  };
+
+  I420VideoFrame frame(
+      new rtc::RefCountedObject(frame_destroyed_event.get()), 1, 1,
+      kVideoRotation_0);
+
+  AddInputFrame(&frame);
+  WaitOutputFrame();
+
+  EXPECT_EQ(output_frames_[0]->video_frame_buffer().get(),
+            frame.video_frame_buffer().get());
+  output_frames_.clear();
+  frame.Reset();
+  EXPECT_EQ(kEventSignaled, frame_destroyed_event->Wait(FRAME_TIMEOUT_MS));
+}
+
+TEST_F(ViECapturerTest, TestNtpTimeStampSetIfRenderTimeSet) {
+  input_frames_.push_back(CreateI420VideoFrame(static_cast(0)));
+  input_frames_[0]->set_render_time_ms(5);
+  input_frames_[0]->set_ntp_time_ms(0);
+
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_GT(output_frames_[0]->ntp_time_ms(),
+            input_frames_[0]->render_time_ms());
+}
+
+TEST_F(ViECapturerTest, TestRtpTimeStampSet) {
+  input_frames_.push_back(CreateI420VideoFrame(static_cast(0)));
+  input_frames_[0]->set_render_time_ms(0);
+  input_frames_[0]->set_ntp_time_ms(1);
+  input_frames_[0]->set_timestamp(0);
+
+  AddInputFrame(input_frames_[0]);
+  WaitOutputFrame();
+  EXPECT_EQ(output_frames_[0]->timestamp(),
+            input_frames_[0]->ntp_time_ms() * 90);
+}
+
 TEST_F(ViECapturerTest, TestTextureFrames) {
   const int kNumFrame = 3;
   for (int i = 0 ; i < kNumFrame; ++i) {
     webrtc::RefCountImpl* handle =
               new webrtc::RefCountImpl();
-    input_frames_.push_back(new TextureVideoFrame(handle, i, i, i, i));
+    // Add one to |i| so that width/height > 0.
+    input_frames_.push_back(
+        new I420VideoFrame(handle, i + 1, i + 1, i + 1, i + 1));
     AddInputFrame(input_frames_[i]);
     WaitOutputFrame();
   }
@@ -144,37 +195,30 @@ TEST_F(ViECapturerTest, TestTextureFrames) {
 
 TEST_F(ViECapturerTest, TestI420Frames) {
   const int kNumFrame = 4;
-  ScopedVector copied_input_frames;
-  std::vector ybuffer_pointers;
+  std::vector ybuffer_pointers;
   for (int i = 0; i < kNumFrame; ++i) {
     input_frames_.push_back(CreateI420VideoFrame(static_cast(i + 1)));
-    ybuffer_pointers.push_back(input_frames_[i]->buffer(kYPlane));
-    // Copy input frames because the buffer data will be swapped.
-    copied_input_frames.push_back(input_frames_[i]->CloneFrame());
+    const I420VideoFrame* const_input_frame = input_frames_[i];
+    ybuffer_pointers.push_back(const_input_frame->buffer(kYPlane));
     AddInputFrame(input_frames_[i]);
     WaitOutputFrame();
   }
 
-  EXPECT_TRUE(EqualFramesVector(copied_input_frames, output_frames_));
-  // Make sure the buffer is swapped and not copied.
+  EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
+  // Make sure the buffer is not copied.
   for (int i = 0; i < kNumFrame; ++i)
     EXPECT_EQ(ybuffer_pointers[i], output_frame_ybuffers_[i]);
-  // The pipeline should be filled with frames with allocated buffers. Check
-  // the last input frame has the same allocated size after swapping.
-  EXPECT_EQ(input_frames_.back()->allocated_size(kYPlane),
-            copied_input_frames.back()->allocated_size(kYPlane));
 }
 
 TEST_F(ViECapturerTest, TestI420FrameAfterTextureFrame) {
   webrtc::RefCountImpl* handle =
       new webrtc::RefCountImpl();
-  input_frames_.push_back(new TextureVideoFrame(handle, 1, 1, 1, 1));
+  input_frames_.push_back(new I420VideoFrame(handle, 1, 1, 1, 1));
   AddInputFrame(input_frames_[0]);
   WaitOutputFrame();
 
-  input_frames_.push_back(CreateI420VideoFrame(1));
-  scoped_ptr copied_input_frame(input_frames_[1]->CloneFrame());
-  AddInputFrame(copied_input_frame.get());
+  input_frames_.push_back(CreateI420VideoFrame(2));
+  AddInputFrame(input_frames_[1]);
   WaitOutputFrame();
 
   EXPECT_TRUE(EqualFramesVector(input_frames_, output_frames_));
@@ -182,13 +226,12 @@ TEST_F(ViECapturerTest, TestI420FrameAfterTextureFrame) {
 
 TEST_F(ViECapturerTest, TestTextureFrameAfterI420Frame) {
   input_frames_.push_back(CreateI420VideoFrame(1));
-  scoped_ptr copied_input_frame(input_frames_[0]->CloneFrame());
-  AddInputFrame(copied_input_frame.get());
+  AddInputFrame(input_frames_[0]);
   WaitOutputFrame();
 
   webrtc::RefCountImpl* handle =
       new webrtc::RefCountImpl();
-  input_frames_.push_back(new TextureVideoFrame(handle, 1, 1, 1, 1));
+  input_frames_.push_back(new I420VideoFrame(handle, 1, 1, 2, 2));
   AddInputFrame(input_frames_[1]);
   WaitOutputFrame();
 
@@ -207,7 +250,6 @@ bool EqualTextureFrames(const I420VideoFrame& frame1,
   return ((frame1.native_handle() == frame2.native_handle()) &&
           (frame1.width() == frame2.width()) &&
           (frame1.height() == frame2.height()) &&
-          (frame1.timestamp() == frame2.timestamp()) &&
           (frame1.render_time_ms() == frame2.render_time_ms()));
 }
 
@@ -218,8 +260,6 @@ bool EqualBufferFrames(const I420VideoFrame& frame1,
           (frame1.stride(kYPlane) == frame2.stride(kYPlane)) &&
           (frame1.stride(kUPlane) == frame2.stride(kUPlane)) &&
           (frame1.stride(kVPlane) == frame2.stride(kVPlane)) &&
-          (frame1.timestamp() == frame2.timestamp()) &&
-          (frame1.ntp_time_ms() == frame2.ntp_time_ms()) &&
           (frame1.render_time_ms() == frame2.render_time_ms()) &&
           (frame1.allocated_size(kYPlane) == frame2.allocated_size(kYPlane)) &&
           (frame1.allocated_size(kUPlane) == frame2.allocated_size(kUPlane)) &&
@@ -248,14 +288,11 @@ I420VideoFrame* CreateI420VideoFrame(uint8_t data) {
   const int width = 36;
   const int height = 24;
   const int kSizeY = width * height * 2;
-  const int kSizeUV = width * height;
   uint8_t buffer[kSizeY];
   memset(buffer, data, kSizeY);
   frame->CreateFrame(
-      kSizeY, buffer, kSizeUV, buffer, kSizeUV, buffer, width, height, width,
+      buffer, buffer, buffer, width, height, width,
       width / 2, width / 2);
-  frame->set_timestamp(data);
-  frame->set_ntp_time_ms(data);
   frame->set_render_time_ms(data);
   return frame;
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc b/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc
index f311fe929a..514b793db2 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_channel.cc
@@ -13,10 +13,13 @@
 #include 
 #include 
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/experiments.h"
+#include "webrtc/frame_callback.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
@@ -27,12 +30,14 @@
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_engine/include/vie_errors.h"
 #include "webrtc/video_engine/include/vie_image_process.h"
 #include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/video_engine/payload_router.h"
+#include "webrtc/video_engine/report_block_stats.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" // for webrtc::SenderInfo
 
@@ -43,50 +48,6 @@ const int kInvalidRtpExtensionId = 0;
 static const int kMaxTargetDelayMs = 10000;
 static const float kMaxIncompleteTimeMultiplier = 3.5f;
 
-namespace {
-
-RTCPReportBlock AggregateReportBlocks(
-    const std::vector& report_blocks,
-    std::map* prev_report_blocks) {
-  int fraction_lost_sum = 0;
-  int fl_seq_num_sum = 0;
-  int jitter_sum = 0;
-  int number_of_report_blocks = 0;
-  RTCPReportBlock aggregate;
-  std::vector::const_iterator report_block =
-      report_blocks.begin();
-  for (; report_block != report_blocks.end(); ++report_block) {
-    aggregate.cumulativeLost += report_block->cumulativeLost;
-    std::map::iterator prev_report_block =
-        prev_report_blocks->find(report_block->sourceSSRC);
-    if (prev_report_block != prev_report_blocks->end()) {
-      // Skip the first report block since we won't be able to get a correct
-      // weight for it.
-      int seq_num_diff = report_block->extendedHighSeqNum -
-                         prev_report_block->second.extendedHighSeqNum;
-      if (seq_num_diff > 0) {
-        fraction_lost_sum += report_block->fractionLost * seq_num_diff;
-        fl_seq_num_sum += seq_num_diff;
-      }
-    }
-    jitter_sum += report_block->jitter;
-    ++number_of_report_blocks;
-    (*prev_report_blocks)[report_block->sourceSSRC] = *report_block;
-  }
-  if (fl_seq_num_sum > 0) {
-    aggregate.fractionLost =
-        (fraction_lost_sum + fl_seq_num_sum / 2) / fl_seq_num_sum;
-  }
-  if (number_of_report_blocks > 0) {
-    aggregate.jitter =
-        (jitter_sum + number_of_report_blocks / 2) / number_of_report_blocks;
-  }
-  // Not well defined for aggregated report blocks.
-  aggregate.extendedHighSeqNum = 0;
-  return aggregate;
-}
-}  // namespace
-
 // Helper class receiving statistics callbacks.
 class ChannelStatsObserver : public CallStatsObserver {
  public:
@@ -94,10 +55,30 @@ class ChannelStatsObserver : public CallStatsObserver {
   virtual ~ChannelStatsObserver() {}
 
   // Implements StatsObserver.
-  virtual void OnRttUpdate(uint32_t rtt) {
+  virtual void OnRttUpdate(int64_t rtt) {
     owner_->OnRttUpdate(rtt);
   }
 
+ private:
+  ViEChannel* const owner_;
+};
+
+class ViEChannelProtectionCallback : public VCMProtectionCallback {
+ public:
+  ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {}
+  ~ViEChannelProtectionCallback() {}
+
+
+  int ProtectionRequest(
+      const FecProtectionParams* delta_fec_params,
+      const FecProtectionParams* key_fec_params,
+      uint32_t* sent_video_rate_bps,
+      uint32_t* sent_nack_rate_bps,
+      uint32_t* sent_fec_rate_bps) override {
+    return owner_->ProtectionRequest(delta_fec_params, key_fec_params,
+                                     sent_video_rate_bps, sent_nack_rate_bps,
+                                     sent_fec_rate_bps);
+  }
  private:
   ViEChannel* owner_;
 };
@@ -112,8 +93,9 @@ ViEChannel::ViEChannel(int32_t channel_id,
                        RemoteBitrateEstimator* remote_bitrate_estimator,
                        RtcpRttStats* rtt_stats,
                        PacedSender* paced_sender,
-                       RtpRtcp* default_rtp_rtcp,
-                       bool sender)
+                       PacketRouter* packet_router,
+                       bool sender,
+                       bool disable_default_encoder)
     : ViEFrameProviderBase(channel_id, engine_id),
       channel_id_(channel_id),
       engine_id_(engine_id),
@@ -121,74 +103,65 @@ ViEChannel::ViEChannel(int32_t channel_id,
       num_socket_threads_(kViESocketThreads),
       callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_rtcp_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-      default_rtp_rtcp_(default_rtp_rtcp),
-      vcm_(VideoCodingModule::Create()),
+      send_payload_router_(new PayloadRouter()),
+      vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
+      vcm_(VideoCodingModule::Create(nullptr)),
       vie_receiver_(channel_id, vcm_, remote_bitrate_estimator, this),
       vie_sender_(channel_id),
       vie_sync_(vcm_, this),
       stats_observer_(new ChannelStatsObserver(this)),
+      vcm_receive_stats_callback_(NULL),
       module_process_thread_(module_process_thread),
       codec_observer_(NULL),
       do_key_frame_callbackRequest_(false),
       rtp_observer_(NULL),
-      rtcp_observer_(NULL),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
-      pad_with_redundant_payloads_(false),
+      packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
       send_timestamp_extension_id_(kInvalidRtpExtensionId),
       absolute_send_time_extension_id_(kInvalidRtpExtensionId),
+      video_rotation_extension_id_(kInvalidRtpExtensionId),
       external_transport_(NULL),
       decoder_reset_(true),
       wait_for_key_frame_(false),
-      decode_thread_(NULL),
       effect_filter_(NULL),
       color_enhancement_(false),
       mtu_(0),
       sender_(sender),
+      disable_default_encoder_(disable_default_encoder),
       nack_history_size_sender_(kSendSidePacketHistorySize),
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
-      start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
-  RtpRtcp::Configuration configuration;
-  configuration.id = ViEModuleId(engine_id, channel_id);
-  configuration.audio = false;
-  configuration.default_module = default_rtp_rtcp;
-  configuration.outgoing_transport = &vie_sender_;
-  configuration.rtcp_feedback = this;
-  configuration.intra_frame_callback = intra_frame_observer;
-  configuration.bandwidth_callback = bandwidth_observer;
-  configuration.rtt_stats = rtt_stats;
+      report_block_stats_sender_(new ReportBlockStats()),
+      report_block_stats_receiver_(new ReportBlockStats()) {
+  RtpRtcp::Configuration configuration = CreateRtpRtcpConfiguration();
   configuration.remote_bitrate_estimator = remote_bitrate_estimator;
-  configuration.paced_sender = paced_sender;
   configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
-  configuration.send_bitrate_observer = &send_bitrate_observer_;
-  configuration.send_frame_count_observer = &send_frame_count_observer_;
-  configuration.send_side_delay_observer = &send_side_delay_observer_;
-
   rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
   vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
   vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
 }
 
 int32_t ViEChannel::Init() {
-  if (module_process_thread_.RegisterModule(
-      vie_receiver_.GetReceiveStatistics()) != 0) {
-    return -1;
-  }
+  module_process_thread_.RegisterModule(vie_receiver_.GetReceiveStatistics());
+
   // RTP/RTCP initialization.
-  if (rtp_rtcp_->SetSendingMediaStatus(false) != 0) {
-    return -1;
-  }
-  if (module_process_thread_.RegisterModule(rtp_rtcp_.get()) != 0) {
-    return -1;
-  }
+  rtp_rtcp_->SetSendingMediaStatus(false);
+  module_process_thread_.RegisterModule(rtp_rtcp_.get());
+
   rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqFirRtp);
   rtp_rtcp_->SetRTCPStatus(kRtcpCompound);
   if (paced_sender_) {
     rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
   }
+  if (sender_) {
+    packet_router_->AddRtpModule(rtp_rtcp_.get());
+    std::list send_rtp_modules(1, rtp_rtcp_.get());
+    send_payload_router_->SetSendingRtpModules(send_rtp_modules);
+    DCHECK(!send_payload_router_->active());
+  }
   if (vcm_->InitializeReceiver() != 0) {
     return -1;
   }
@@ -203,22 +176,25 @@ int32_t ViEChannel::Init() {
   vcm_->RegisterReceiveStatisticsCallback(this);
   vcm_->RegisterDecoderTimingCallback(this);
   vcm_->SetRenderDelay(kViEDefaultRenderDelayMs);
-  if (module_process_thread_.RegisterModule(vcm_) != 0) {
-    return -1;
-  }
+
+  module_process_thread_.RegisterModule(vcm_);
+  module_process_thread_.RegisterModule(&vie_sync_);
+
 #ifdef VIDEOCODEC_VP8
-  VideoCodec video_codec;
-  if (vcm_->Codec(kVideoCodecVP8, &video_codec) == VCM_OK) {
-    rtp_rtcp_->RegisterSendPayload(video_codec);
-    // TODO(holmer): Can we call SetReceiveCodec() here instead?
-    if (!vie_receiver_.RegisterPayload(video_codec)) {
-      return -1;
+  if (!disable_default_encoder_) {
+    VideoCodec video_codec;
+    if (vcm_->Codec(kVideoCodecVP8, &video_codec) == VCM_OK) {
+      rtp_rtcp_->RegisterSendPayload(video_codec);
+      // TODO(holmer): Can we call SetReceiveCodec() here instead?
+      if (!vie_receiver_.RegisterPayload(video_codec)) {
+        return -1;
+      }
+      vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_);
+      vcm_->RegisterSendCodec(&video_codec, number_of_cores_,
+                              rtp_rtcp_->MaxDataPayloadLength());
+    } else {
+      assert(false);
     }
-    vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_);
-    vcm_->RegisterSendCodec(&video_codec, number_of_cores_,
-                           rtp_rtcp_->MaxDataPayloadLength());
-  } else {
-    assert(false);
   }
 #endif
 
@@ -232,9 +208,12 @@ ViEChannel::~ViEChannel() {
   module_process_thread_.DeRegisterModule(rtp_rtcp_.get());
   module_process_thread_.DeRegisterModule(vcm_);
   module_process_thread_.DeRegisterModule(&vie_sync_);
+  send_payload_router_->SetSendingRtpModules(std::list());
+  packet_router_->RemoveRtpModule(rtp_rtcp_.get());
   while (simulcast_rtp_rtcp_.size() > 0) {
     std::list::iterator it = simulcast_rtp_rtcp_.begin();
     RtpRtcp* rtp_rtcp = *it;
+    packet_router_->RemoveRtpModule(rtp_rtcp);
     module_process_thread_.DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
     simulcast_rtp_rtcp_.erase(it);
@@ -252,51 +231,114 @@ ViEChannel::~ViEChannel() {
 }
 
 void ViEChannel::UpdateHistograms() {
-  const float kMinCallLengthInMinutes = 0.5f;
-  float elapsed_minutes =
-      (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
-  if (elapsed_minutes < kMinCallLengthInMinutes) {
-    return;
-  }
-  RtcpPacketTypeCounter rtcp_sent;
-  RtcpPacketTypeCounter rtcp_received;
-  GetRtcpPacketTypeCounters(&rtcp_sent, &rtcp_received);
+  int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
 
   if (sender_) {
-    if (rtcp_received.nack_requests > 0) {
-      RTC_HISTOGRAM_PERCENTAGE(
-          "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
-              rtcp_received.UniqueNackRequestsInPercent());
-    }
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
-        rtcp_received.nack_packets / elapsed_minutes);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
-        rtcp_received.fir_packets / elapsed_minutes);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
-        rtcp_received.pli_packets / elapsed_minutes);
-  } else if (vie_receiver_.GetRemoteSsrc() > 0)  {
-    // Get receive stats if we are receiving packets, i.e. there is a remote
-    // ssrc.
-    if (rtcp_sent.nack_requests > 0) {
-      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
-          rtcp_sent.UniqueNackRequestsInPercent());
-    }
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
-        rtcp_sent.nack_packets / elapsed_minutes);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
-        rtcp_sent.fir_packets / elapsed_minutes);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
-        rtcp_sent.pli_packets / elapsed_minutes);
-
-    webrtc::VCMFrameCount frames;
-    if (vcm_->ReceivedFrameCount(frames) == VCM_OK) {
-      uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
-      if (total_frames > 0) {
-        RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
-            static_cast((frames.numKeyFrames * 1000.0f / total_frames) +
-                0.5f));
+    RtcpPacketTypeCounter rtcp_counter;
+    GetSendRtcpPacketTypeCounter(&rtcp_counter);
+    int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
+          rtcp_counter.nack_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
+          rtcp_counter.fir_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
+          rtcp_counter.pli_packets * 60 / elapsed_sec);
+      if (rtcp_counter.nack_requests > 0) {
+        RTC_HISTOGRAM_PERCENTAGE(
+            "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
+                rtcp_counter.UniqueNackRequestsInPercent());
+      }
+      int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
+      if (fraction_lost != -1) {
+        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
+            fraction_lost);
       }
     }
+  } else if (vie_receiver_.GetRemoteSsrc() > 0) {
+    // Get receive stats if we are receiving packets, i.e. there is a remote
+    // ssrc.
+    RtcpPacketTypeCounter rtcp_counter;
+    GetReceiveRtcpPacketTypeCounter(&rtcp_counter);
+    int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+          rtcp_counter.nack_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+          rtcp_counter.fir_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+          rtcp_counter.pli_packets * 60 / elapsed_sec);
+      if (rtcp_counter.nack_requests > 0) {
+        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+            rtcp_counter.UniqueNackRequestsInPercent());
+      }
+      int fraction_lost = report_block_stats_receiver_->FractionLostInPercent();
+      if (fraction_lost != -1) {
+        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
+            fraction_lost);
+      }
+    }
+
+    StreamDataCounters rtp;
+    StreamDataCounters rtx;
+    GetReceiveStreamDataCounters(&rtp, &rtx);
+    StreamDataCounters rtp_rtx = rtp;
+    rtp_rtx.Add(rtx);
+    elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.BitrateReceivedInKbps",
+          rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
+          rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateReceivedInKbps",
+          rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000);
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
+              rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+      uint32_t ssrc = 0;
+      if (vie_receiver_.GetRtxSsrc(&ssrc)) {
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
+            rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+      }
+      if (vie_receiver_.IsFecEnabled()) {
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps",
+            rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000);
+      }
+    }
+  }
+}
+
+void ViEChannel::UpdateHistogramsAtStopSend() {
+  StreamDataCounters rtp;
+  StreamDataCounters rtx;
+  GetSendStreamDataCounters(&rtp, &rtx);
+  StreamDataCounters rtp_rtx = rtp;
+  rtp_rtx.Add(rtx);
+
+  int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
+      Clock::GetRealTimeClock()->TimeInMilliseconds()) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
+    return;
+  }
+  RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.BitrateSentInKbps",
+      rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateSentInKbps",
+      rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
+  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateSentInKbps",
+      rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000);
+  RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RetransmittedBitrateSentInKbps",
+      rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+  if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateSentInKbps",
+        rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000);
+  }
+  bool fec_enabled = false;
+  uint8_t pltype_red;
+  uint8_t pltype_fec;
+  rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+  if (fec_enabled) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
+        rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000);
   }
 }
 
@@ -319,6 +361,12 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
   // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
   // set explicitly.
   bool restart_rtp = false;
+  bool router_was_active = send_payload_router_->active();
+  send_payload_router_->set_active(false);
+  send_payload_router_->SetSendingRtpModules(std::list());
+  packet_router_->RemoveRtpModule(rtp_rtcp_.get());
+  for (RtpRtcp* module : simulcast_rtp_rtcp_)
+    packet_router_->RemoveRtpModule(module);
   if (rtp_rtcp_->Sending() && new_stream) {
     restart_rtp = true;
     rtp_rtcp_->SetSendingStatus(false);
@@ -334,154 +382,183 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
   uint8_t payload_type_fec;
   rtp_rtcp_->GenericFECStatus(fec_enabled, payload_type_red, payload_type_fec);
 
-  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  std::vector registered_modules;
+  std::vector deregistered_modules;
+  {
+    CriticalSectionScoped cs(rtp_rtcp_cs_.get());
 
-  if (video_codec.numberOfSimulcastStreams > 0) {
-    // Set correct bitrate to base layer.
-    // Create our simulcast RTP modules.
-    int num_modules_to_add = video_codec.numberOfSimulcastStreams -
-        simulcast_rtp_rtcp_.size() - 1;
-    if (num_modules_to_add < 0) {
-      num_modules_to_add = 0;
-    }
-
-    // Add back removed rtp modules. Order is important (allocate from front of
-    // removed modules) to preserve RTP settings such as SSRCs for simulcast
-    // streams.
-    std::list new_rtp_modules;
-    for (; removed_rtp_rtcp_.size() > 0 && num_modules_to_add > 0;
-         --num_modules_to_add) {
-      new_rtp_modules.push_back(removed_rtp_rtcp_.front());
-      removed_rtp_rtcp_.pop_front();
-    }
-
-    for (int i = 0; i < num_modules_to_add; ++i)
-      new_rtp_modules.push_back(CreateRtpRtcpModule());
-
-    // Initialize newly added modules.
-    for (std::list::iterator it = new_rtp_modules.begin();
-         it != new_rtp_modules.end();
-         ++it) {
-      RtpRtcp* rtp_rtcp = *it;
-
-      rtp_rtcp->SetRTCPStatus(rtp_rtcp_->RTCP());
-
-      if (rtp_rtcp_->StorePackets()) {
-        rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
-      } else if (paced_sender_) {
-        rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
+    if (video_codec.numberOfSimulcastStreams > 0) {
+      // Set correct bitrate to base layer.
+      // Create our simulcast RTP modules.
+      int num_modules_to_add =
+          video_codec.numberOfSimulcastStreams - simulcast_rtp_rtcp_.size() - 1;
+      if (num_modules_to_add < 0) {
+        num_modules_to_add = 0;
       }
 
-      if (fec_enabled) {
-        rtp_rtcp->SetGenericFECStatus(
-            fec_enabled, payload_type_red, payload_type_fec);
+      // Add back removed rtp modules. Order is important (allocate from front
+      // of removed modules) to preserve RTP settings such as SSRCs for
+      // simulcast streams.
+      std::list new_rtp_modules;
+      for (; removed_rtp_rtcp_.size() > 0 && num_modules_to_add > 0;
+           --num_modules_to_add) {
+        new_rtp_modules.push_back(removed_rtp_rtcp_.front());
+        removed_rtp_rtcp_.pop_front();
       }
-      rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending());
-      rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia());
 
-      int mode;
-      uint32_t ssrc;
-      int payload_type;
-      rtp_rtcp_->RTXSendStatus(&mode, &ssrc, &payload_type);
-      rtp_rtcp->SetRTXSendStatus(mode);
-      simulcast_rtp_rtcp_.push_back(rtp_rtcp);
+      for (int i = 0; i < num_modules_to_add; ++i)
+        new_rtp_modules.push_back(CreateRtpRtcpModule());
 
-      // Silently ignore error.
-      module_process_thread_.RegisterModule(rtp_rtcp);
-    }
+      // Initialize newly added modules.
+      for (std::list::iterator it = new_rtp_modules.begin();
+           it != new_rtp_modules.end(); ++it) {
+        RtpRtcp* rtp_rtcp = *it;
 
-    // Remove last in list if we have too many.
-    for (int j = simulcast_rtp_rtcp_.size();
-         j > (video_codec.numberOfSimulcastStreams - 1);
-         j--) {
-      RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
-      module_process_thread_.DeRegisterModule(rtp_rtcp);
-      rtp_rtcp->SetSendingStatus(false);
-      rtp_rtcp->SetSendingMediaStatus(false);
-      rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
-      rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
-      simulcast_rtp_rtcp_.pop_back();
-      removed_rtp_rtcp_.push_front(rtp_rtcp);
-    }
-    uint8_t idx = 0;
-    // Configure all simulcast modules.
-    for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end();
-         it++) {
-      idx++;
-      RtpRtcp* rtp_rtcp = *it;
-      rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
-      if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
-        return -1;
-      }
-      if (mtu_ != 0) {
-        rtp_rtcp->SetMaxTransferUnit(mtu_);
-      }
-      if (restart_rtp) {
-        rtp_rtcp->SetSendingStatus(true);
-        rtp_rtcp->SetSendingMediaStatus(true);
-      }
-      if (send_timestamp_extension_id_ != kInvalidRtpExtensionId) {
-        // Deregister in case the extension was previously enabled.
-        rtp_rtcp->DeregisterSendRtpHeaderExtension(
-            kRtpExtensionTransmissionTimeOffset);
-        if (rtp_rtcp->RegisterSendRtpHeaderExtension(
-            kRtpExtensionTransmissionTimeOffset,
-            send_timestamp_extension_id_) != 0) {
+        rtp_rtcp->SetRTCPStatus(rtp_rtcp_->RTCP());
+
+        if (rtp_rtcp_->StorePackets()) {
+          rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
+        } else if (paced_sender_) {
+          rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
         }
-      } else {
-        rtp_rtcp->DeregisterSendRtpHeaderExtension(
-            kRtpExtensionTransmissionTimeOffset);
-      }
-      if (absolute_send_time_extension_id_ != kInvalidRtpExtensionId) {
-        // Deregister in case the extension was previously enabled.
-        rtp_rtcp->DeregisterSendRtpHeaderExtension(
-            kRtpExtensionAbsoluteSendTime);
-        if (rtp_rtcp->RegisterSendRtpHeaderExtension(
-            kRtpExtensionAbsoluteSendTime,
-            absolute_send_time_extension_id_) != 0) {
-        }
-      } else {
-        rtp_rtcp->DeregisterSendRtpHeaderExtension(
-            kRtpExtensionAbsoluteSendTime);
-      }
-      rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(
-          rtp_rtcp_->GetSendChannelRtcpStatisticsCallback());
-      rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
-          rtp_rtcp_->GetSendChannelRtpStatisticsCallback());
-    }
-    // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
-    // modules can be deleted after this step.
-    vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
-  } else {
-    while (!simulcast_rtp_rtcp_.empty()) {
-      RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
-      module_process_thread_.DeRegisterModule(rtp_rtcp);
-      rtp_rtcp->SetSendingStatus(false);
-      rtp_rtcp->SetSendingMediaStatus(false);
-      rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
-      rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
-      simulcast_rtp_rtcp_.pop_back();
-      removed_rtp_rtcp_.push_front(rtp_rtcp);
-    }
-    // Clear any previous modules.
-    vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
-  }
 
-  // Don't log this error, no way to check in advance if this pl_type is
-  // registered or not...
-  rtp_rtcp_->DeRegisterSendPayload(video_codec.plType);
-  if (rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
-    return -1;
-  }
-  if (restart_rtp) {
-    rtp_rtcp_->SetSendingStatus(true);
-    for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
-         it != simulcast_rtp_rtcp_.end(); ++it) {
-      (*it)->SetSendingStatus(true);
-      (*it)->SetSendingMediaStatus(true);
+        if (fec_enabled) {
+          rtp_rtcp->SetGenericFECStatus(fec_enabled, payload_type_red,
+                                        payload_type_fec);
+        }
+        rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending());
+        rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia());
+        rtp_rtcp->SetRtxSendStatus(rtp_rtcp_->RtxSendStatus());
+        simulcast_rtp_rtcp_.push_back(rtp_rtcp);
+
+        // Silently ignore error.
+        registered_modules.push_back(rtp_rtcp);
+      }
+
+      // Remove last in list if we have too many.
+      for (int j = simulcast_rtp_rtcp_.size();
+           j > (video_codec.numberOfSimulcastStreams - 1); j--) {
+        RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
+        deregistered_modules.push_back(rtp_rtcp);
+        rtp_rtcp->SetSendingStatus(false);
+        rtp_rtcp->SetSendingMediaStatus(false);
+        rtp_rtcp->RegisterRtcpStatisticsCallback(NULL);
+        rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
+        simulcast_rtp_rtcp_.pop_back();
+        removed_rtp_rtcp_.push_front(rtp_rtcp);
+      }
+      uint8_t idx = 0;
+      // Configure all simulcast modules.
+      for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
+           it != simulcast_rtp_rtcp_.end(); it++) {
+        idx++;
+        RtpRtcp* rtp_rtcp = *it;
+        rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
+        if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
+          return -1;
+        }
+        if (mtu_ != 0) {
+          rtp_rtcp->SetMaxTransferUnit(mtu_);
+        }
+        if (restart_rtp) {
+          rtp_rtcp->SetSendingStatus(true);
+          rtp_rtcp->SetSendingMediaStatus(true);
+        }
+        if (send_timestamp_extension_id_ != kInvalidRtpExtensionId) {
+          // Deregister in case the extension was previously enabled.
+          rtp_rtcp->DeregisterSendRtpHeaderExtension(
+              kRtpExtensionTransmissionTimeOffset);
+          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
+                  kRtpExtensionTransmissionTimeOffset,
+                  send_timestamp_extension_id_) != 0) {
+            LOG(LS_WARNING) << "Register Transmission Time Offset failed";
+          }
+        } else {
+          rtp_rtcp->DeregisterSendRtpHeaderExtension(
+              kRtpExtensionTransmissionTimeOffset);
+        }
+        if (absolute_send_time_extension_id_ != kInvalidRtpExtensionId) {
+          // Deregister in case the extension was previously enabled.
+          rtp_rtcp->DeregisterSendRtpHeaderExtension(
+              kRtpExtensionAbsoluteSendTime);
+          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
+                  kRtpExtensionAbsoluteSendTime,
+                  absolute_send_time_extension_id_) != 0) {
+            LOG(LS_WARNING) << "Register Absolute Send Time failed";
+          }
+        } else {
+          rtp_rtcp->DeregisterSendRtpHeaderExtension(
+              kRtpExtensionAbsoluteSendTime);
+        }
+        if (video_rotation_extension_id_ != kInvalidRtpExtensionId) {
+          // Deregister in case the extension was previously enabled.
+          rtp_rtcp->DeregisterSendRtpHeaderExtension(
+              kRtpExtensionVideoRotation);
+          if (rtp_rtcp->RegisterSendRtpHeaderExtension(
+                  kRtpExtensionVideoRotation, video_rotation_extension_id_) !=
+              0) {
+            LOG(LS_WARNING) << "Register VideoRotation extension failed";
+          }
+        } else {
+          rtp_rtcp->DeregisterSendRtpHeaderExtension(
+              kRtpExtensionVideoRotation);
+        }
+        rtp_rtcp->RegisterRtcpStatisticsCallback(
+            rtp_rtcp_->GetRtcpStatisticsCallback());
+        rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
+            rtp_rtcp_->GetSendChannelRtpStatisticsCallback());
+      }
+      // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
+      // modules can be deleted after this step.
+      vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
+    } else {
+      while (!simulcast_rtp_rtcp_.empty()) {
+        RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
+        deregistered_modules.push_back(rtp_rtcp);
+        rtp_rtcp->SetSendingStatus(false);
+        rtp_rtcp->SetSendingMediaStatus(false);
+        rtp_rtcp->RegisterRtcpStatisticsCallback(NULL);
+        rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
+        simulcast_rtp_rtcp_.pop_back();
+        removed_rtp_rtcp_.push_front(rtp_rtcp);
+      }
+      // Clear any previous modules.
+      vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
     }
+
+    // Don't log this error, no way to check in advance if this pl_type is
+    // registered or not...
+    rtp_rtcp_->DeRegisterSendPayload(video_codec.plType);
+    if (rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
+      return -1;
+    }
+    if (restart_rtp) {
+      rtp_rtcp_->SetSendingStatus(true);
+      for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
+           it != simulcast_rtp_rtcp_.end(); ++it) {
+        (*it)->SetSendingStatus(true);
+        (*it)->SetSendingMediaStatus(true);
+      }
+    }
+    // Update the packet and payload routers with the sending RTP RTCP modules.
+    packet_router_->AddRtpModule(rtp_rtcp_.get());
+    for (RtpRtcp* module : simulcast_rtp_rtcp_)
+      packet_router_->AddRtpModule(module);
+
+    std::list active_send_modules;
+    active_send_modules.push_back(rtp_rtcp_.get());
+    for (std::list::const_iterator cit = simulcast_rtp_rtcp_.begin();
+         cit != simulcast_rtp_rtcp_.end(); ++cit) {
+      active_send_modules.push_back(*cit);
+    }
+    send_payload_router_->SetSendingRtpModules(active_send_modules);
+    if (router_was_active)
+      send_payload_router_->set_active(true);
   }
+  for (RtpRtcp* rtp_rtcp : registered_modules)
+    module_process_thread_.RegisterModule(rtp_rtcp);
+  for (RtpRtcp* rtp_rtcp : deregistered_modules)
+    module_process_thread_.DeRegisterModule(rtp_rtcp);
   return 0;
 }
 
@@ -551,12 +628,9 @@ int32_t ViEChannel::DeRegisterExternalDecoder(const uint8_t pl_type) {
 
 int32_t ViEChannel::ReceiveCodecStatistics(uint32_t* num_key_frames,
                                            uint32_t* num_delta_frames) {
-  VCMFrameCount received_frames;
-  if (vcm_->ReceivedFrameCount(received_frames) != VCM_OK) {
-    return -1;
-  }
-  *num_key_frames = received_frames.numKeyFrames;
-  *num_delta_frames = received_frames.numDeltaFrames;
+  CriticalSectionScoped cs(callback_cs_.get());
+  *num_key_frames = receive_frame_counts_.key_frames;
+  *num_delta_frames = receive_frame_counts_.delta_frames;
   return 0;
 }
 
@@ -594,7 +668,7 @@ int32_t ViEChannel::SetSignalPacketLossStatus(bool enable,
   return 0;
 }
 
-int32_t ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
+void ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end();
@@ -602,12 +676,11 @@ int32_t ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
     RtpRtcp* rtp_rtcp = *it;
     rtp_rtcp->SetRTCPStatus(rtcp_mode);
   }
-  return rtp_rtcp_->SetRTCPStatus(rtcp_mode);
+  rtp_rtcp_->SetRTCPStatus(rtcp_mode);
 }
 
-int32_t ViEChannel::GetRTCPMode(RTCPMethod* rtcp_mode) {
-  *rtcp_mode = rtp_rtcp_->RTCP();
-  return 0;
+RTCPMethod ViEChannel::GetRTCPMode() const {
+  return rtp_rtcp_->RTCP();
 }
 
 int32_t ViEChannel::SetNACKStatus(const bool enable) {
@@ -679,6 +752,23 @@ int32_t ViEChannel::SetFECStatus(const bool enable,
   return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
 }
 
+bool ViEChannel::IsSendingFecEnabled() {
+  bool fec_enabled = false;
+  uint8_t pltype_red = 0;
+  uint8_t pltype_fec = 0;
+  rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+  if (fec_enabled)
+    return true;
+
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  for (auto* module : simulcast_rtp_rtcp_) {
+    module->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+    if (fec_enabled)
+      return true;
+  }
+  return false;
+}
+
 int32_t ViEChannel::ProcessFECRequest(
     const bool enable,
     const unsigned char payload_typeRED,
@@ -728,9 +818,7 @@ int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
       nack_history_size_sender_ = kSendSidePacketHistorySize;
     }
   }
-  if (rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_) != 0) {
-    return -1;
-  }
+  rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
   return 0;
 }
 
@@ -773,10 +861,8 @@ int32_t ViEChannel::SetKeyFrameRequestMethod(
   return rtp_rtcp_->SetKeyFrameRequestMethod(method);
 }
 
-bool ViEChannel::EnableRemb(bool enable) {
-  if (rtp_rtcp_->SetREMBStatus(enable) != 0)
-    return false;
-  return true;
+void ViEChannel::EnableRemb(bool enable) {
+  rtp_rtcp_->SetREMBStatus(enable);
 }
 
 int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) {
@@ -849,6 +935,37 @@ int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
   return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1;
 }
 
+int ViEChannel::SetSendVideoRotationStatus(bool enable, int id) {
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  int error = 0;
+  if (enable) {
+    // Enable the extension, but disable possible old id to avoid errors.
+    video_rotation_extension_id_ = id;
+    rtp_rtcp_->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
+    error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
+        kRtpExtensionVideoRotation, id);
+    for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
+         it != simulcast_rtp_rtcp_.end(); it++) {
+      (*it)->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
+      error |=
+          (*it)->RegisterSendRtpHeaderExtension(kRtpExtensionVideoRotation, id);
+    }
+  } else {
+    // Disable the extension.
+    video_rotation_extension_id_ = kInvalidRtpExtensionId;
+    rtp_rtcp_->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
+    for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
+         it != simulcast_rtp_rtcp_.end(); it++) {
+      (*it)->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation);
+    }
+  }
+  return error;
+}
+
+int ViEChannel::SetReceiveVideoRotationStatus(bool enable, int id) {
+  return vie_receiver_.SetReceiveVideoRotationStatus(enable, id) ? 0 : -1;
+}
+
 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   rtp_rtcp_->SetRtcpXrRrtrStatus(enable);
@@ -859,12 +976,11 @@ void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
   paced_sender_->SetStatus(enable);
 }
 
-int32_t ViEChannel::EnableTMMBR(const bool enable) {
-  return rtp_rtcp_->SetTMMBRStatus(enable);
+void ViEChannel::EnableTMMBR(bool enable) {
+  rtp_rtcp_->SetTMMBRStatus(enable);
 }
 
 int32_t ViEChannel::EnableKeyFrameRequestCallback(const bool enable) {
-
   CriticalSectionScoped cs(callback_cs_.get());
   if (enable && !codec_observer_) {
     LOG(LS_ERROR) << "No ViECodecObserver set.";
@@ -921,22 +1037,6 @@ int32_t ViEChannel::GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]) {
   return 0;
 }
 
-void ViEChannel::SetPadWithRedundantPayloads(bool enable) {
-  {
-    CriticalSectionScoped cs(callback_cs_.get());
-    pad_with_redundant_payloads_ = enable;
-  }
-  int mode;
-  uint32_t ssrc;
-  int payload_type;
-  rtp_rtcp_->RTXSendStatus(&mode, &ssrc, &payload_type);
-  if (mode != kRtxOff) {
-    // Since RTX was already enabled we have to reset it with payload-based
-    // padding on.
-    SetRtxSendStatus(true);
-  }
-}
-
 int ViEChannel::SetRtxSendPayloadType(int payload_type) {
   rtp_rtcp_->SetRtxSendPayloadType(payload_type);
   for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
@@ -948,18 +1048,13 @@ int ViEChannel::SetRtxSendPayloadType(int payload_type) {
 }
 
 void ViEChannel::SetRtxSendStatus(bool enable) {
-  int rtx_settings = kRtxOff;
-  if (enable) {
-    CriticalSectionScoped cs(callback_cs_.get());
-    rtx_settings = kRtxRetransmitted;
-    if (pad_with_redundant_payloads_)
-      rtx_settings |= kRtxRedundantPayloads;
-  }
-  rtp_rtcp_->SetRTXSendStatus(rtx_settings);
+  int rtx_settings =
+      enable ? kRtxRetransmitted | kRtxRedundantPayloads : kRtxOff;
+  rtp_rtcp_->SetRtxSendStatus(rtx_settings);
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end(); it++) {
-    (*it)->SetRTXSendStatus(rtx_settings);
+    (*it)->SetRtxSendStatus(rtx_settings);
   }
 }
 
@@ -971,21 +1066,42 @@ int32_t ViEChannel::SetStartSequenceNumber(uint16_t sequence_number) {
   if (rtp_rtcp_->Sending()) {
     return -1;
   }
-  return rtp_rtcp_->SetSequenceNumber(sequence_number);
+  rtp_rtcp_->SetSequenceNumber(sequence_number);
+  return 0;
 }
 
 void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) {
   assert(!rtp_rtcp_->Sending());
-  default_rtp_rtcp_->SetRtpStateForSsrc(ssrc, rtp_state);
+  if (rtp_rtcp_->SetRtpStateForSsrc(ssrc, rtp_state))
+    return;
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  for (auto* module : simulcast_rtp_rtcp_) {
+    if (module->SetRtpStateForSsrc(ssrc, rtp_state))
+      return;
+  }
+  for (auto* module : removed_rtp_rtcp_) {
+    if (module->SetRtpStateForSsrc(ssrc, rtp_state))
+      return;
+  }
 }
 
 RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) {
   assert(!rtp_rtcp_->Sending());
 
   RtpState rtp_state;
-  if (!default_rtp_rtcp_->GetRtpStateForSsrc(ssrc, &rtp_state)) {
-    LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
+  if (rtp_rtcp_->GetRtpStateForSsrc(ssrc, &rtp_state))
+    return rtp_state;
+
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  for (auto* module : simulcast_rtp_rtcp_) {
+    if (module->GetRtpStateForSsrc(ssrc, &rtp_state))
+      return rtp_state;
   }
+  for (auto* module : removed_rtp_rtcp_) {
+    if (module->GetRtpStateForSsrc(ssrc, &rtp_state))
+      return rtp_state;
+  }
+  LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc;
   return rtp_state;
 }
 
@@ -1015,20 +1131,6 @@ int32_t ViEChannel::RegisterRtpObserver(ViERTPObserver* observer) {
   return 0;
 }
 
-int32_t ViEChannel::RegisterRtcpObserver(ViERTCPObserver* observer) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (observer) {
-    if (rtcp_observer_) {
-      LOG_F(LS_ERROR) << "Observer already registered.";
-      return -1;
-    }
-    rtcp_observer_ = observer;
-  } else {
-    rtcp_observer_ = NULL;
-  }
-  return 0;
-}
-
 int32_t ViEChannel::SendApplicationDefinedRTCPPacket(
     const uint8_t sub_type,
     uint32_t name,
@@ -1065,7 +1167,7 @@ int32_t ViEChannel::GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh,
                                               uint32_t* jitterSamples,
                                               uint16_t* fractionLost,
                                               uint32_t* cumulativeLost,
-                                              int32_t* rttMs) {
+                                             int32_t* rttMs) {
   // TODO: how do we do this for simulcast ? average for all
   // except cumulative_lost that is the sum ?
   // CriticalSectionScoped cs(rtp_rtcp_cs_.get());
@@ -1117,8 +1219,8 @@ int32_t ViEChannel::GetRemoteRTCPReceiverInfo(uint32_t& NTPHigh,
   *cumulativeLost = statistics->cumulativeLost;
   *jitterSamples = statistics->jitter;
 
-  uint16_t dummy;
-  uint16_t rtt = 0;
+  int64_t dummy;
+  int64_t rtt = 0;
   if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
     LOG_F(LS_ERROR) << "failed to get RTT";
     return -1;
@@ -1131,7 +1233,7 @@ int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost,
                                           uint32_t* cumulative_lost,
                                           uint32_t* extended_max,
                                           uint32_t* jitter_samples,
-                                          int32_t* rtt_ms) {
+                                          int64_t* rtt_ms) {
   // Aggregate the report blocks associated with streams sent on this channel.
   std::vector report_blocks;
   rtp_rtcp_->RemoteRTCPStat(&report_blocks);
@@ -1162,19 +1264,17 @@ int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost,
     remote_ssrc = report_blocks[0].remoteSSRC;
   }
 
-  RTCPReportBlock report;
-  if (report_blocks.size() > 1)
-    report = AggregateReportBlocks(report_blocks, &prev_report_blocks_);
-  else
-    report = report_blocks[0];
-
+  // TODO(asapersson): Change report_block_stats to not rely on
+  // GetSendRtcpStatistics to be called.
+  RTCPReportBlock report =
+      report_block_stats_sender_->AggregateAndStore(report_blocks);
   *fraction_lost = report.fractionLost;
   *cumulative_lost = report.cumulativeLost;
   *extended_max = report.extendedHighSeqNum;
   *jitter_samples = report.jitter;
 
-  uint16_t dummy;
-  uint16_t rtt = 0;
+  int64_t dummy;
+  int64_t rtt = 0;
   if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
     return -1;
   }
@@ -1184,12 +1284,12 @@ int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost,
 
 void ViEChannel::RegisterSendChannelRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
-  rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(callback);
+  rtp_rtcp_->RegisterRtcpStatisticsCallback(callback);
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   for (std::list::const_iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end();
        ++it) {
-    (*it)->RegisterSendChannelRtcpStatisticsCallback(callback);
+    (*it)->RegisterRtcpStatisticsCallback(callback);
   }
 }
 
@@ -1200,7 +1300,7 @@ int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost,
                                               uint32_t* cumulative_lost,
                                               uint32_t* extended_max,
                                               uint32_t* jitter_samples,
-                                              int32_t* rtt_ms) {
+                                              int64_t* rtt_ms) {
   uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
   StreamStatistician* statistician =
       vie_receiver_.GetReceiveStatistics()->GetStatistician(remote_ssrc);
@@ -1214,8 +1314,12 @@ int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost,
   *extended_max = receive_stats.extended_max_sequence_number;
   *jitter_samples = receive_stats.jitter;
 
-  uint16_t dummy = 0;
-  uint16_t rtt = 0;
+  // TODO(asapersson): Change report_block_stats to not rely on
+  // GetReceivedRtcpStatistics to be called.
+  report_block_stats_receiver_->Store(receive_stats, remote_ssrc, 0);
+
+  int64_t dummy = 0;
+  int64_t rtt = 0;
   rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy);
   *rtt_ms = rtt;
   return 0;
@@ -1225,11 +1329,17 @@ void ViEChannel::RegisterReceiveChannelRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
   vie_receiver_.GetReceiveStatistics()->RegisterRtcpStatisticsCallback(
       callback);
+  rtp_rtcp_->RegisterRtcpStatisticsCallback(callback);
 }
 
-int32_t ViEChannel::GetRtpStatistics(uint32_t* bytes_sent,
+void ViEChannel::RegisterRtcpPacketTypeCounterObserver(
+    RtcpPacketTypeCounterObserver* observer) {
+  rtcp_packet_type_counter_observer_.Set(observer);
+}
+
+int32_t ViEChannel::GetRtpStatistics(size_t* bytes_sent,
                                      uint32_t* packets_sent,
-                                     uint32_t* bytes_received,
+                                     size_t* bytes_received,
                                      uint32_t* packets_received) const {
   StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
       GetStatistician(vie_receiver_.GetRemoteSsrc());
@@ -1244,7 +1354,7 @@ int32_t ViEChannel::GetRtpStatistics(uint32_t* bytes_sent,
   for (std::list::const_iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end();
        it++) {
-    uint32_t bytes_sent_temp = 0;
+    size_t bytes_sent_temp = 0;
     uint32_t packets_sent_temp = 0;
     RtpRtcp* rtp_rtcp = *it;
     rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
@@ -1253,7 +1363,7 @@ int32_t ViEChannel::GetRtpStatistics(uint32_t* bytes_sent,
   }
   for (std::list::const_iterator it = removed_rtp_rtcp_.begin();
        it != removed_rtp_rtcp_.end(); ++it) {
-    uint32_t bytes_sent_temp = 0;
+    size_t bytes_sent_temp = 0;
     uint32_t packets_sent_temp = 0;
     RtpRtcp* rtp_rtcp = *it;
     rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
@@ -1263,6 +1373,48 @@ int32_t ViEChannel::GetRtpStatistics(uint32_t* bytes_sent,
   return 0;
 }
 
+void ViEChannel::GetSendStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  rtp_rtcp_->GetSendStreamDataCounters(rtp_counters, rtx_counters);
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  for (std::list::const_iterator it = simulcast_rtp_rtcp_.begin();
+       it != simulcast_rtp_rtcp_.end();
+       it++) {
+    StreamDataCounters rtp_data;
+    StreamDataCounters rtx_data;
+    (*it)->GetSendStreamDataCounters(&rtp_data, &rtx_data);
+    rtp_counters->Add(rtp_data);
+    rtx_counters->Add(rtx_data);
+  }
+  for (std::list::const_iterator it = removed_rtp_rtcp_.begin();
+       it != removed_rtp_rtcp_.end(); ++it) {
+    StreamDataCounters rtp_data;
+    StreamDataCounters rtx_data;
+    (*it)->GetSendStreamDataCounters(&rtp_data, &rtx_data);
+    rtp_counters->Add(rtp_data);
+    rtx_counters->Add(rtx_data);
+  }
+}
+
+void ViEChannel::GetReceiveStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
+      GetStatistician(vie_receiver_.GetRemoteSsrc());
+  if (statistician) {
+    statistician->GetReceiveStreamDataCounters(rtp_counters);
+  }
+  uint32_t rtx_ssrc = 0;
+  if (vie_receiver_.GetRtxSsrc(&rtx_ssrc)) {
+    StreamStatistician* statistician =
+        vie_receiver_.GetReceiveStatistics()->GetStatistician(rtx_ssrc);
+    if (statistician) {
+      statistician->GetReceiveStreamDataCounters(rtx_counters);
+    }
+  }
+}
+
 void ViEChannel::RegisterSendChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback) {
   rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(callback);
@@ -1281,28 +1433,35 @@ void ViEChannel::RegisterReceiveChannelRtpStatisticsCallback(
   vie_receiver_.GetReceiveStatistics()->RegisterRtpStatisticsCallback(callback);
 }
 
-void ViEChannel::GetRtcpPacketTypeCounters(
-    RtcpPacketTypeCounter* packets_sent,
-    RtcpPacketTypeCounter* packets_received) const {
-  rtp_rtcp_->GetRtcpPacketTypeCounters(packets_sent, packets_received);
+void ViEChannel::GetSendRtcpPacketTypeCounter(
+    RtcpPacketTypeCounter* packet_counter) const {
+  std::map counter_map =
+      rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap();
+
+  RtcpPacketTypeCounter counter;
+  counter.Add(counter_map[rtp_rtcp_->SSRC()]);
 
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   for (std::list::const_iterator it = simulcast_rtp_rtcp_.begin();
        it != simulcast_rtp_rtcp_.end(); ++it) {
-    RtcpPacketTypeCounter sent;
-    RtcpPacketTypeCounter received;
-    (*it)->GetRtcpPacketTypeCounters(&sent, &received);
-    packets_sent->Add(sent);
-    packets_received->Add(received);
+    counter.Add(counter_map[(*it)->SSRC()]);
   }
   for (std::list::const_iterator it = removed_rtp_rtcp_.begin();
        it != removed_rtp_rtcp_.end(); ++it) {
-    RtcpPacketTypeCounter sent;
-    RtcpPacketTypeCounter received;
-    (*it)->GetRtcpPacketTypeCounters(&sent, &received);
-    packets_sent->Add(sent);
-    packets_received->Add(received);
+    counter.Add(counter_map[(*it)->SSRC()]);
   }
+  *packet_counter = counter;
+}
+
+void ViEChannel::GetReceiveRtcpPacketTypeCounter(
+    RtcpPacketTypeCounter* packet_counter) const {
+  std::map counter_map =
+      rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap();
+
+  RtcpPacketTypeCounter counter;
+  counter.Add(counter_map[vie_receiver_.GetRemoteSsrc()]);
+
+  *packet_counter = counter;
 }
 
 int32_t ViEChannel::GetRemoteRTCPSenderInfo(SenderInfo* sender_info) const {
@@ -1384,11 +1543,6 @@ void ViEChannel::RegisterSendBitrateObserver(
   send_bitrate_observer_.Set(observer);
 }
 
-void ViEChannel::GetReceiveBandwidthEstimatorStats(
-    ReceiveBandwidthEstimatorStats* output) const {
-  vie_receiver_.GetReceiveBandwidthEstimatorStats(output);
-}
-
 int32_t ViEChannel::StartRTPDump(const char file_nameUTF8[1024],
                                  RTPDirections direction) {
   if (direction == kRtpIncoming) {
@@ -1428,11 +1582,14 @@ int32_t ViEChannel::StartSend() {
     rtp_rtcp->SetSendingMediaStatus(true);
     rtp_rtcp->SetSendingStatus(true);
   }
+  send_payload_router_->set_active(true);
   vie_receiver_.StartRTCPReceive();
   return 0;
 }
 
 int32_t ViEChannel::StopSend() {
+  UpdateHistogramsAtStopSend();
+  send_payload_router_->set_active(false);
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   rtp_rtcp_->SetSendingMediaStatus(false);
   for (std::list::iterator it = simulcast_rtp_rtcp_.begin();
@@ -1466,7 +1623,6 @@ bool ViEChannel::Sending() {
 }
 
 int32_t ViEChannel::StartReceive() {
-  CriticalSectionScoped cs(callback_cs_.get());
   if (StartDecodeThread() != 0) {
     vie_receiver_.StopReceive();
     return -1;
@@ -1514,7 +1670,7 @@ int32_t ViEChannel::DeregisterSendTransport() {
 }
 
 int32_t ViEChannel::ReceivedRTPPacket(
-    const void* rtp_packet, const int32_t rtp_packet_length,
+    const void* rtp_packet, const size_t rtp_packet_length,
     const PacketTime& packet_time) {
   {
     CriticalSectionScoped cs(callback_cs_.get());
@@ -1527,7 +1683,7 @@ int32_t ViEChannel::ReceivedRTPPacket(
 }
 
 int32_t ViEChannel::ReceivedRTCPPacket(
-  const void* rtcp_packet, const int32_t rtcp_packet_length) {
+  const void* rtcp_packet, const size_t rtcp_packet_length) {
   {
     CriticalSectionScoped cs(callback_cs_.get());
     if (!external_transport_) {
@@ -1566,6 +1722,14 @@ RtpRtcp* ViEChannel::rtp_rtcp() {
   return rtp_rtcp_.get();
 }
 
+scoped_refptr ViEChannel::send_payload_router() {
+  return send_payload_router_;
+}
+
+VCMProtectionCallback* ViEChannel::vcm_protection_callback() {
+  return vcm_protection_callback_.get();
+}
+
 CallStatsObserver* ViEChannel::GetStatsObserver() {
   return stats_observer_.get();
 }
@@ -1594,10 +1758,9 @@ int32_t ViEChannel::FrameToRender(
     if (pre_render_callback_ != NULL)
       pre_render_callback_->FrameCallback(&video_frame);
     if (effect_filter_) {
-      unsigned int length = CalcBufferSize(kI420,
-                                           video_frame.width(),
-                                           video_frame.height());
-      scoped_ptr video_buffer(new uint8_t[length]);
+      size_t length =
+          CalcBufferSize(kI420, video_frame.width(), video_frame.height());
+      rtc::scoped_ptr video_buffer(new uint8_t[length]);
       ExtractBuffer(video_frame, length, video_buffer.get());
       effect_filter_->Transform(length,
                                 video_buffer.get(),
@@ -1617,7 +1780,9 @@ int32_t ViEChannel::FrameToRender(
     arr_ofCSRC[0] = vie_receiver_.GetRemoteSsrc();
     no_of_csrcs = 1;
   }
-  DeliverFrame(&video_frame, no_of_csrcs, arr_ofCSRC);
+  std::vector csrcs(arr_ofCSRC, arr_ofCSRC + no_of_csrcs);
+  DeliverFrame(&video_frame, csrcs);
+
   return 0;
 }
 
@@ -1631,13 +1796,23 @@ void ViEChannel::IncomingCodecChanged(const VideoCodec& codec) {
   receive_codec_ = codec;
 }
 
-int32_t ViEChannel::OnReceiveStatisticsUpdate(const uint32_t bit_rate,
-                                              const uint32_t frame_rate) {
+void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) {
   CriticalSectionScoped cs(callback_cs_.get());
-  if (codec_observer_) {
+  if (codec_observer_)
     codec_observer_->IncomingRate(channel_id_, frame_rate, bit_rate);
-  }
-  return 0;
+}
+
+void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) {
+  CriticalSectionScoped cs(callback_cs_.get());
+  if (vcm_receive_stats_callback_ != NULL)
+    vcm_receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets);
+}
+
+void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) {
+  CriticalSectionScoped cs(callback_cs_.get());
+  receive_frame_counts_ = frame_counts;
+  if (vcm_receive_stats_callback_ != NULL)
+    vcm_receive_stats_callback_->OnFrameCountsUpdated(frame_counts);
 }
 
 void ViEChannel::OnDecoderTiming(int decode_ms,
@@ -1694,14 +1869,40 @@ bool ViEChannel::ChannelDecodeThreadFunction(void* obj) {
 }
 
 bool ViEChannel::ChannelDecodeProcess() {
+  // TODO(pbos): Make sure the decoder thread doesn't run for send-only
+  // channels.
   vcm_->Decode(kMaxDecodeWaitTimeMs);
   return true;
 }
 
-void ViEChannel::OnRttUpdate(uint32_t rtt) {
+void ViEChannel::OnRttUpdate(int64_t rtt) {
   vcm_->SetReceiveChannelParameters(rtt);
 }
 
+int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params,
+                                  const FecProtectionParams* key_fec_params,
+                                  uint32_t* video_rate_bps,
+                                  uint32_t* nack_rate_bps,
+                                  uint32_t* fec_rate_bps) {
+  uint32_t not_used = 0;
+  rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
+  rtp_rtcp_->BitrateSent(¬_used, video_rate_bps, fec_rate_bps,
+                         nack_rate_bps);
+  CriticalSectionScoped cs(rtp_rtcp_cs_.get());
+  for (auto* module : simulcast_rtp_rtcp_) {
+    uint32_t child_video_rate = 0;
+    uint32_t child_fec_rate = 0;
+    uint32_t child_nack_rate = 0;
+    module->SetFecParameters(delta_fec_params, key_fec_params);
+    module->BitrateSent(¬_used, &child_video_rate, &child_fec_rate,
+                        &child_nack_rate);
+    *video_rate_bps += child_video_rate;
+    *nack_rate_bps += child_nack_rate;
+    *fec_rate_bps += child_fec_rate;
+  }
+  return 0;
+}
+
 void ViEChannel::ReserveRtpRtcpModules(size_t num_modules) {
   for (size_t total_modules =
            1 + simulcast_rtp_rtcp_.size() + removed_rtp_rtcp_.size();
@@ -1710,7 +1911,7 @@ void ViEChannel::ReserveRtpRtcpModules(size_t num_modules) {
     RtpRtcp* rtp_rtcp = CreateRtpRtcpModule();
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
-    rtp_rtcp->RegisterSendChannelRtcpStatisticsCallback(NULL);
+    rtp_rtcp->RegisterRtcpStatisticsCallback(NULL);
     rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(NULL);
     removed_rtp_rtcp_.push_back(rtp_rtcp);
   }
@@ -1741,19 +1942,26 @@ RtpRtcp* ViEChannel::GetRtpRtcpModule(size_t index) const {
   return *it;
 }
 
-RtpRtcp* ViEChannel::CreateRtpRtcpModule() {
+RtpRtcp::Configuration ViEChannel::CreateRtpRtcpConfiguration() {
   RtpRtcp::Configuration configuration;
   configuration.id = ViEModuleId(engine_id_, channel_id_);
-  configuration.audio = false;  // Video.
-  configuration.default_module = default_rtp_rtcp_;
+  configuration.audio = false;
   configuration.outgoing_transport = &vie_sender_;
   configuration.intra_frame_callback = intra_frame_observer_;
   configuration.bandwidth_callback = bandwidth_observer_.get();
   configuration.rtt_stats = rtt_stats_;
+  configuration.rtcp_packet_type_counter_observer =
+      &rtcp_packet_type_counter_observer_;
   configuration.paced_sender = paced_sender_;
+  configuration.send_bitrate_observer = &send_bitrate_observer_;
+  configuration.send_frame_count_observer = &send_frame_count_observer_;
   configuration.send_side_delay_observer = &send_side_delay_observer_;
 
-  return RtpRtcp::CreateRtpRtcp(configuration);
+  return configuration;
+}
+
+RtpRtcp* ViEChannel::CreateRtpRtcpModule() {
+  return RtpRtcp::CreateRtpRtcp(CreateRtpRtcpConfiguration());
 }
 
 int32_t ViEChannel::StartDecodeThread() {
@@ -1763,19 +1971,9 @@ int32_t ViEChannel::StartDecodeThread() {
     return 0;
   }
   decode_thread_ = ThreadWrapper::CreateThread(ChannelDecodeThreadFunction,
-                                                   this, kHighestPriority,
-                                                   "DecodingThread");
-  if (!decode_thread_) {
-    return -1;
-  }
-
-  unsigned int thread_id;
-  if (decode_thread_->Start(thread_id) == false) {
-    delete decode_thread_;
-    decode_thread_ = NULL;
-    LOG(LS_ERROR) << "Could not start decode thread.";
-    return -1;
-  }
+                                               this, "DecodingThread");
+  decode_thread_->Start();
+  decode_thread_->SetPriority(kHighestPriority);
   return 0;
 }
 
@@ -1784,24 +1982,17 @@ int32_t ViEChannel::StopDecodeThread() {
     return 0;
   }
 
-  decode_thread_->SetNotAlive();
-  if (decode_thread_->Stop()) {
-    delete decode_thread_;
-  } else {
-    assert(false && "could not stop decode thread");
-  }
-  decode_thread_ = NULL;
+  vcm_->TriggerDecoderShutdown();
+
+  decode_thread_->Stop();
+  decode_thread_.reset();
+  frame_delivery_thread_checker_.DetachFromThread();
+
   return 0;
 }
 
 int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id,
-                                          VoEVideoSync* ve_sync_interface) {
-  if (ve_sync_interface) {
-    // Register lip sync
-    module_process_thread_.RegisterModule(&vie_sync_);
-  } else {
-    module_process_thread_.DeRegisterModule(&vie_sync_);
-  }
+                                    VoEVideoSync* ve_sync_interface) {
   return vie_sync_.ConfigureSync(ve_channel_id,
                                  ve_sync_interface,
                                  rtp_rtcp_.get(),
@@ -1833,24 +2024,6 @@ void ViEChannel::RegisterPreDecodeImageCallback(
   vcm_->RegisterPreDecodeImageCallback(pre_decode_callback);
 }
 
-void ViEChannel::OnApplicationDataReceived(const int32_t id,
-                                           const uint8_t sub_type,
-                                           const uint32_t name,
-                                           const uint16_t length,
-                                           const uint8_t* data) {
-  if (channel_id_ != ChannelId(id)) {
-    return;
-  }
-  CriticalSectionScoped cs(callback_cs_.get());
-  {
-    if (rtcp_observer_) {
-      rtcp_observer_->OnApplicationDataReceived(
-          channel_id_, sub_type, name, reinterpret_cast(data),
-          length);
-    }
-  }
-}
-
 int32_t ViEChannel::OnInitializeDecoder(
     const int32_t id,
     const int8_t payload_type,
@@ -1858,8 +2031,8 @@ int32_t ViEChannel::OnInitializeDecoder(
     const int frequency,
     const uint8_t channels,
     const uint32_t rate) {
-  LOG(LS_INFO) << "OnInitializeDecoder " << payload_type << " "
-               << payload_name;
+  LOG(LS_INFO) << "OnInitializeDecoder " << static_cast(payload_type)
+               << " " << payload_name;
   vcm_->ResetDecoder();
 
   CriticalSectionScoped cs(callback_cs_.get());
@@ -1903,8 +2076,15 @@ void ViEChannel::RegisterSendFrameCountObserver(
   send_frame_count_observer_.Set(observer);
 }
 
+void ViEChannel::RegisterReceiveStatisticsProxy(
+    ReceiveStatisticsProxy* receive_statistics_proxy) {
+  CriticalSectionScoped cs(callback_cs_.get());
+  vcm_receive_stats_callback_ = receive_statistics_proxy;
+}
+
 void ViEChannel::ReceivedBWEPacket(int64_t arrival_time_ms,
-    int payload_size, const RTPHeader& header) {
+                                   size_t payload_size,
+                                   const RTPHeader& header) {
   vie_receiver_.ReceivedBWEPacket(arrival_time_ms, payload_size, header);
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel.h b/media/webrtc/trunk/webrtc/video_engine/vie_channel.h
index caab4c4808..1197115e28 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_channel.h
@@ -13,11 +13,13 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/scoped_refptr.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/include/vie_network.h"
@@ -37,13 +39,16 @@ class CriticalSectionWrapper;
 class EncodedImageCallback;
 class I420FrameCallback;
 class PacedSender;
+class PacketRouter;
+class PayloadRouter;
 class ProcessThread;
+class ReceiveStatisticsProxy;
+class ReportBlockStats;
 class RtcpRttStats;
-class RtpRtcp;
 class ThreadWrapper;
+class ViEChannelProtectionCallback;
 class ViEDecoderObserver;
 class ViEEffectFilter;
-class ViERTCPObserver;
 class ViERTPObserver;
 class VideoCodingModule;
 class VideoDecoder;
@@ -59,11 +64,11 @@ class ViEChannel
       public VCMDecoderTimingCallback,
       public VCMPacketRequestCallback,
       public VCMReceiveStateCallback,
-      public RtcpFeedback,
       public RtpFeedback,
       public ViEFrameProviderBase {
  public:
   friend class ChannelStatsObserver;
+  friend class ViEChannelProtectionCallback;
 
   ViEChannel(int32_t channel_id,
              int32_t engine_id,
@@ -75,8 +80,9 @@ class ViEChannel
              RemoteBitrateEstimator* remote_bitrate_estimator,
              RtcpRttStats* rtt_stats,
              PacedSender* paced_sender,
-             RtpRtcp* default_rtp_rtcp,
-             bool sender);
+             PacketRouter* packet_router,
+             bool sender,
+             bool disable_default_encoder);
   ~ViEChannel();
 
   int32_t Init();
@@ -111,8 +117,8 @@ class ViEChannel
   // key frames.
   int32_t SetSignalPacketLossStatus(bool enable, bool only_key_frames);
 
-  int32_t SetRTCPMode(const RTCPMethod rtcp_mode);
-  int32_t GetRTCPMode(RTCPMethod* rtcp_mode);
+  void SetRTCPMode(const RTCPMethod rtcp_mode);
+  RTCPMethod GetRTCPMode() const;
   int32_t SetNACKStatus(const bool enable);
   int32_t SetFECStatus(const bool enable,
                        const unsigned char payload_typeRED,
@@ -120,18 +126,21 @@ class ViEChannel
   int32_t SetHybridNACKFECStatus(const bool enable,
                                  const unsigned char payload_typeRED,
                                  const unsigned char payload_typeFEC);
+  bool IsSendingFecEnabled();
   int SetSenderBufferingMode(int target_delay_ms);
   int SetReceiverBufferingMode(int target_delay_ms);
   int32_t SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
-  bool EnableRemb(bool enable);
+  void EnableRemb(bool enable);
   int SetSendTimestampOffsetStatus(bool enable, int id);
   int SetReceiveTimestampOffsetStatus(bool enable, int id);
   int SetSendAbsoluteSendTimeStatus(bool enable, int id);
   int SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
   bool GetReceiveAbsoluteSendTimeStatus() const;
+  int SetSendVideoRotationStatus(bool enable, int id);
+  int SetReceiveVideoRotationStatus(bool enable, int id);
   void SetRtcpXrRrtrStatus(bool enable);
   void SetTransmissionSmoothingStatus(bool enable);
-  int32_t EnableTMMBR(const bool enable);
+  void EnableTMMBR(bool enable);
   int32_t EnableKeyFrameRequestCallback(const bool enable);
 
   // Sets SSRC for outgoing stream.
@@ -149,8 +158,6 @@ class ViEChannel
   int32_t GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]);
 
   int SetRtxSendPayloadType(int payload_type);
-  // Only has an effect once RTX is enabled.
-  void SetPadWithRedundantPayloads(bool enable);
   void SetRtxReceivePayloadType(int payload_type);
 
   // Sets the starting sequence number, must be called before StartSend.
@@ -165,7 +172,6 @@ class ViEChannel
   // Gets the CName of the incoming stream.
   int32_t GetRemoteRTCPCName(char rtcp_cname[]);
   int32_t RegisterRtpObserver(ViERTPObserver* observer);
-  int32_t RegisterRtcpObserver(ViERTCPObserver* observer);
   int32_t SendApplicationDefinedRTCPPacket(
       const uint8_t sub_type,
       uint32_t name,
@@ -187,7 +193,7 @@ class ViEChannel
                                 uint32_t* cumulative_lost,
                                 uint32_t* extended_max,
                                 uint32_t* jitter_samples,
-                                int32_t* rtt_ms);
+                                int64_t* rtt_ms);
 
   // Called on receipt of RTCP report block from remote side.
   void RegisterSendChannelRtcpStatisticsCallback(
@@ -198,18 +204,26 @@ class ViEChannel
                                     uint32_t* cumulative_lost,
                                     uint32_t* extended_max,
                                     uint32_t* jitter_samples,
-                                    int32_t* rtt_ms);
+                                    int64_t* rtt_ms);
 
   // Called on generation of RTCP stats
   void RegisterReceiveChannelRtcpStatisticsCallback(
       RtcpStatisticsCallback* callback);
 
   // Gets sent/received packets statistics.
-  int32_t GetRtpStatistics(uint32_t* bytes_sent,
+  int32_t GetRtpStatistics(size_t* bytes_sent,
                            uint32_t* packets_sent,
-                           uint32_t* bytes_received,
+                           size_t* bytes_received,
                            uint32_t* packets_received) const;
 
+  // Gets send statistics for the rtp and rtx stream.
+  void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
+                                 StreamDataCounters* rtx_counters) const;
+
+  // Gets received stream data counters.
+  void GetReceiveStreamDataCounters(StreamDataCounters* rtp_counters,
+                                    StreamDataCounters* rtx_counters) const;
+
   // Called on update of RTP statistics.
   void RegisterSendChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback);
@@ -218,8 +232,11 @@ class ViEChannel
   void RegisterReceiveChannelRtpStatisticsCallback(
       StreamDataCountersCallback* callback);
 
-  void GetRtcpPacketTypeCounters(RtcpPacketTypeCounter* packets_sent,
-                                 RtcpPacketTypeCounter* packets_received) const;
+  void GetSendRtcpPacketTypeCounter(
+      RtcpPacketTypeCounter* packet_counter) const;
+
+  void GetReceiveRtcpPacketTypeCounter(
+      RtcpPacketTypeCounter* packet_counter) const;
 
 
   int32_t GetRemoteRTCPSenderInfo(SenderInfo* sender_info) const;
@@ -232,8 +249,6 @@ class ViEChannel
   // to avoid deadlocks.
   bool GetSendSideDelay(int* avg_send_delay, int* max_send_delay) const;
   void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
-  void GetReceiveBandwidthEstimatorStats(
-      ReceiveBandwidthEstimatorStats* output) const;
 
   // Called on any new send bitrate estimate.
   void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
@@ -242,13 +257,6 @@ class ViEChannel
                        RTPDirections direction);
   int32_t StopRTPDump(RTPDirections direction);
 
-  // Implements RtcpFeedback.
-  // TODO(pwestin) Depricate this functionality.
-  virtual void OnApplicationDataReceived(const int32_t id,
-                                         const uint8_t sub_type,
-                                         const uint32_t name,
-                                         const uint16_t length,
-                                         const uint8_t* data);
   // Implements RtpFeedback.
   virtual int32_t OnInitializeDecoder(
       const int32_t id,
@@ -298,12 +306,12 @@ class ViEChannel
 
   // Incoming packet from external transport.
   int32_t ReceivedRTPPacket(const void* rtp_packet,
-                            const int32_t rtp_packet_length,
+                            const size_t rtp_packet_length,
                             const PacketTime& packet_time);
 
   // Incoming packet from external transport.
   int32_t ReceivedRTCPPacket(const void* rtcp_packet,
-                             const int32_t rtcp_packet_length);
+                             const size_t rtcp_packet_length);
 
   // Sets the maximum transfer unit size for the network link, i.e. including
   // IP, UDP and RTP headers.
@@ -319,6 +327,9 @@ class ViEChannel
 
   // Gets the modules used by the channel.
   RtpRtcp* rtp_rtcp();
+  scoped_refptr send_payload_router();
+  VCMProtectionCallback* vcm_protection_callback();
+
 
   CallStatsObserver* GetStatsObserver();
 
@@ -333,8 +344,9 @@ class ViEChannel
   virtual void IncomingCodecChanged(const VideoCodec& codec);
 
   // Implements VCMReceiveStatisticsCallback.
-  virtual int32_t OnReceiveStatisticsUpdate(const uint32_t bit_rate,
-                                    const uint32_t frame_rate);
+  void OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) override;
+  void OnDiscardedPacketsUpdated(int discarded_packets) override;
+  void OnFrameCountsUpdated(const FrameCounts& frame_counts) override;
 
   // Implements VCMDecoderTimingCallback.
   virtual void OnDecoderTiming(int decode_ms,
@@ -374,21 +386,31 @@ class ViEChannel
       EncodedImageCallback* pre_decode_callback);
 
   void RegisterSendFrameCountObserver(FrameCountObserver* observer);
-
-  void ReceivedBWEPacket(int64_t arrival_time_ms, int payload_size,
+  void RegisterRtcpPacketTypeCounterObserver(
+      RtcpPacketTypeCounterObserver* observer);
+  void RegisterReceiveStatisticsProxy(
+      ReceiveStatisticsProxy* receive_statistics_proxy);
+  void ReceivedBWEPacket(int64_t arrival_time_ms, size_t payload_size,
                          const RTPHeader& header);
 
  protected:
   static bool ChannelDecodeThreadFunction(void* obj);
   bool ChannelDecodeProcess();
 
-  void OnRttUpdate(uint32_t rtt);
+  void OnRttUpdate(int64_t rtt);
+
+  int ProtectionRequest(const FecProtectionParams* delta_fec_params,
+                        const FecProtectionParams* key_fec_params,
+                        uint32_t* sent_video_rate_bps,
+                        uint32_t* sent_nack_rate_bps,
+                        uint32_t* sent_fec_rate_bps);
 
  private:
   void ReserveRtpRtcpModules(size_t total_modules)
       EXCLUSIVE_LOCKS_REQUIRED(rtp_rtcp_cs_);
   RtpRtcp* GetRtpRtcpModule(size_t simulcast_idx) const
       EXCLUSIVE_LOCKS_REQUIRED(rtp_rtcp_cs_);
+  RtpRtcp::Configuration CreateRtpRtcpConfiguration();
   RtpRtcp* CreateRtpRtcpModule();
   // Assumed to be protected.
   int32_t StartDecodeThread();
@@ -403,6 +425,7 @@ class ViEChannel
   void SetRtxSendStatus(bool enable);
 
   void UpdateHistograms();
+  void UpdateHistogramsAtStopSend();
 
   // ViEChannel exposes methods that allow to modify observers and callbacks
   // to be modified. Such an API-style is cumbersome to implement and maintain
@@ -425,7 +448,7 @@ class ViEChannel
     // Note: this should be implemented with a RW-lock to allow simultaneous
     // calls into the callback. However that doesn't seem to be needed for the
     // current type of callbacks covered by this class.
-    scoped_ptr critsect_;
+    rtc::scoped_ptr critsect_;
     T* callback_ GUARDED_BY(critsect_);
 
    private:
@@ -441,68 +464,96 @@ class ViEChannel
       if (callback_)
         callback_->Notify(total_stats, retransmit_stats, ssrc);
     }
-  }
-  send_bitrate_observer_;
+  } send_bitrate_observer_;
 
   class RegisterableFrameCountObserver
       : public RegisterableCallback {
-    virtual void FrameCountUpdated(FrameType frame_type,
-                                   uint32_t frame_count,
-                                   const unsigned int ssrc) {
+   public:
+    virtual void FrameCountUpdated(const FrameCounts& frame_counts,
+                                   uint32_t ssrc) {
       CriticalSectionScoped cs(critsect_.get());
       if (callback_)
-        callback_->FrameCountUpdated(frame_type, frame_count, ssrc);
+        callback_->FrameCountUpdated(frame_counts, ssrc);
     }
+
+   private:
   } send_frame_count_observer_;
 
   class RegisterableSendSideDelayObserver :
       public RegisterableCallback {
-    virtual void SendSideDelayUpdated(int avg_delay_ms,
-                                      int max_delay_ms,
-                                      uint32_t ssrc) OVERRIDE {
+    void SendSideDelayUpdated(int avg_delay_ms,
+                              int max_delay_ms,
+                              uint32_t ssrc) override {
       CriticalSectionScoped cs(critsect_.get());
       if (callback_)
         callback_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
     }
   } send_side_delay_observer_;
 
+  class RegisterableRtcpPacketTypeCounterObserver
+      : public RegisterableCallback {
+   public:
+    void RtcpPacketTypesCounterUpdated(
+        uint32_t ssrc,
+        const RtcpPacketTypeCounter& packet_counter) override {
+      CriticalSectionScoped cs(critsect_.get());
+      if (callback_)
+        callback_->RtcpPacketTypesCounterUpdated(ssrc, packet_counter);
+      counter_map_[ssrc] = packet_counter;
+    }
+
+    virtual std::map GetPacketTypeCounterMap()
+        const {
+      CriticalSectionScoped cs(critsect_.get());
+      return counter_map_;
+    }
+
+   private:
+    std::map counter_map_
+        GUARDED_BY(critsect_);
+  } rtcp_packet_type_counter_observer_;
+
   int32_t channel_id_;
   int32_t engine_id_;
   uint32_t number_of_cores_;
   uint8_t num_socket_threads_;
 
   // Used for all registered callbacks except rendering.
-  scoped_ptr callback_cs_;
-  scoped_ptr rtp_rtcp_cs_;
-
-  RtpRtcp* default_rtp_rtcp_;
+  rtc::scoped_ptr callback_cs_;
+  rtc::scoped_ptr rtp_rtcp_cs_;
 
   // Owned modules/classes.
-  scoped_ptr rtp_rtcp_;
+  rtc::scoped_ptr rtp_rtcp_;
   std::list simulcast_rtp_rtcp_;
   std::list removed_rtp_rtcp_;
+  scoped_refptr send_payload_router_;
+  rtc::scoped_ptr vcm_protection_callback_;
+
   VideoCodingModule* const vcm_;
   ViEReceiver vie_receiver_;
   ViESender vie_sender_;
   ViESyncModule vie_sync_;
 
   // Helper to report call statistics.
-  scoped_ptr stats_observer_;
+  rtc::scoped_ptr stats_observer_;
 
   // Not owned.
+  VCMReceiveStatisticsCallback* vcm_receive_stats_callback_
+      GUARDED_BY(callback_cs_);
+  FrameCounts receive_frame_counts_ GUARDED_BY(callback_cs_);
   ProcessThread& module_process_thread_;
   ViEDecoderObserver* codec_observer_;
   bool do_key_frame_callbackRequest_;
   ViERTPObserver* rtp_observer_;
-  ViERTCPObserver* rtcp_observer_;
   RtcpIntraFrameObserver* intra_frame_observer_;
   RtcpRttStats* rtt_stats_;
   PacedSender* paced_sender_;
-  bool pad_with_redundant_payloads_;
+  PacketRouter* packet_router_;
 
-  scoped_ptr bandwidth_observer_;
+  rtc::scoped_ptr bandwidth_observer_;
   int send_timestamp_extension_id_;
   int absolute_send_time_extension_id_;
+  int video_rotation_extension_id_;
 
   Transport* external_transport_;
 
@@ -510,7 +561,7 @@ class ViEChannel
   // Current receive codec used for codec change callback.
   VideoCodec receive_codec_;
   bool wait_for_key_frame_;
-  ThreadWrapper* decode_thread_;
+  rtc::scoped_ptr decode_thread_;
 
   ViEEffectFilter* effect_filter_;
   bool color_enhancement_;
@@ -518,13 +569,15 @@ class ViEChannel
   // User set MTU, -1 if not set.
   uint16_t mtu_;
   const bool sender_;
+  // Used to skip default encoder in the new API.
+  const bool disable_default_encoder_;
 
   int nack_history_size_sender_;
   int max_nack_reordering_threshold_;
   I420FrameCallback* pre_render_callback_;
-  const int64_t start_ms_;
 
-  std::map prev_report_blocks_;
+  rtc::scoped_ptr report_block_stats_sender_;
+  rtc::scoped_ptr report_block_stats_receiver_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc b/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc
index 9c2d59f014..3b752cc722 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.cc
@@ -10,10 +10,12 @@
 
 #include "webrtc/video_engine/vie_channel_group.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common.h"
 #include "webrtc/experiments.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/include/paced_sender.h"
+#include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
@@ -21,9 +23,11 @@
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/encoder_state_feedback.h"
+#include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/vie_channel.h"
 #include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_engine/vie_remb.h"
+#include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 namespace {
@@ -32,19 +36,16 @@ static const uint32_t kTimeOffsetSwitchThreshold = 30;
 
 class WrappingBitrateEstimator : public RemoteBitrateEstimator {
  public:
-  WrappingBitrateEstimator(int engine_id,
-                           RemoteBitrateObserver* observer,
+  WrappingBitrateEstimator(RemoteBitrateObserver* observer,
                            Clock* clock,
                            const Config& config)
       : observer_(observer),
         clock_(clock),
         crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-        engine_id_(engine_id),
         min_bitrate_bps_(config.Get().min_rate),
-        rate_control_type_(kAimdControl),
         rbe_(RemoteBitrateEstimatorFactory().Create(observer_,
                                                     clock_,
-                                                    rate_control_type_,
+                                                    kAimdControl,
                                                     min_bitrate_bps_)),
         using_absolute_send_time_(false),
         packets_since_absolute_send_time_(0) {
@@ -52,56 +53,45 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
 
   virtual ~WrappingBitrateEstimator() {}
 
-  virtual void IncomingPacket(int64_t arrival_time_ms,
-                              int payload_size,
-                              const RTPHeader& header) OVERRIDE {
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header) override {
     CriticalSectionScoped cs(crit_sect_.get());
     PickEstimatorFromHeader(header);
     rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
   }
 
-  virtual int32_t Process() OVERRIDE {
+  int32_t Process() override {
     CriticalSectionScoped cs(crit_sect_.get());
     return rbe_->Process();
   }
 
-  virtual int32_t TimeUntilNextProcess() OVERRIDE {
+  int64_t TimeUntilNextProcess() override {
     CriticalSectionScoped cs(crit_sect_.get());
     return rbe_->TimeUntilNextProcess();
   }
 
-  virtual void OnRttUpdate(uint32_t rtt) OVERRIDE {
+  void OnRttUpdate(int64_t rtt) override {
     CriticalSectionScoped cs(crit_sect_.get());
     rbe_->OnRttUpdate(rtt);
   }
 
-  virtual void RemoveStream(unsigned int ssrc) OVERRIDE {
+  void RemoveStream(unsigned int ssrc) override {
     CriticalSectionScoped cs(crit_sect_.get());
     rbe_->RemoveStream(ssrc);
   }
 
-  virtual bool LatestEstimate(std::vector* ssrcs,
-                              unsigned int* bitrate_bps) const OVERRIDE {
+  bool LatestEstimate(std::vector* ssrcs,
+                      unsigned int* bitrate_bps) const override {
     CriticalSectionScoped cs(crit_sect_.get());
     return rbe_->LatestEstimate(ssrcs, bitrate_bps);
   }
 
-  virtual bool GetStats(ReceiveBandwidthEstimatorStats* output) const OVERRIDE {
+  bool GetStats(ReceiveBandwidthEstimatorStats* output) const override {
     CriticalSectionScoped cs(crit_sect_.get());
     return rbe_->GetStats(output);
   }
 
-  void SetConfig(const webrtc::Config& config) {
-    CriticalSectionScoped cs(crit_sect_.get());
-    RateControlType new_control_type =
-        config.Get().enabled ? kAimdControl :
-                                                      kMimdControl;
-    if (new_control_type != rate_control_type_) {
-      rate_control_type_ = new_control_type;
-      PickEstimator();
-    }
-  }
-
  private:
   void PickEstimatorFromHeader(const RTPHeader& header)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
@@ -132,20 +122,18 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
   void PickEstimator() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_.get()) {
     if (using_absolute_send_time_) {
       rbe_.reset(AbsoluteSendTimeRemoteBitrateEstimatorFactory().Create(
-          observer_, clock_, rate_control_type_, min_bitrate_bps_));
+          observer_, clock_, kAimdControl, min_bitrate_bps_));
     } else {
       rbe_.reset(RemoteBitrateEstimatorFactory().Create(
-          observer_, clock_, rate_control_type_, min_bitrate_bps_));
+          observer_, clock_, kAimdControl, min_bitrate_bps_));
     }
   }
 
   RemoteBitrateObserver* observer_;
   Clock* clock_;
-  scoped_ptr crit_sect_;
-  const int engine_id_;
+  rtc::scoped_ptr crit_sect_;
   const uint32_t min_bitrate_bps_;
-  RateControlType rate_control_type_;
-  scoped_ptr rbe_;
+  rtc::scoped_ptr rbe_;
   bool using_absolute_send_time_;
   uint32_t packets_since_absolute_send_time_;
 
@@ -153,89 +141,331 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
 };
 }  // namespace
 
-ChannelGroup::ChannelGroup(int engine_id,
-                           ProcessThread* process_thread,
-                           const Config* config)
+ChannelGroup::ChannelGroup(ProcessThread* process_thread, const Config* config)
     : remb_(new VieRemb()),
-      bitrate_controller_(
-          BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
-                                                     true)),
+      bitrate_allocator_(new BitrateAllocator()),
       call_stats_(new CallStats()),
       encoder_state_feedback_(new EncoderStateFeedback()),
+      packet_router_(new PacketRouter()),
+      pacer_(new PacedSender(Clock::GetRealTimeClock(),
+                             packet_router_.get(),
+                             BitrateController::kDefaultStartBitrateKbps,
+                             PacedSender::kDefaultPaceMultiplier *
+                                 BitrateController::kDefaultStartBitrateKbps,
+                             0)),
+      encoder_map_cs_(CriticalSectionWrapper::CreateCriticalSection()),
       config_(config),
       own_config_(),
-      process_thread_(process_thread) {
+      process_thread_(process_thread),
+      pacer_thread_(ProcessThread::Create()),
+      // Constructed last as this object calls the provided callback on
+      // construction.
+      bitrate_controller_(
+          BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
+                                                     this)) {
   if (!config) {
     own_config_.reset(new Config);
     config_ = own_config_.get();
   }
-  assert(config_);  // Must have a valid config pointer here.
+  DCHECK(config_);  // Must have a valid config pointer here.
 
   remote_bitrate_estimator_.reset(
-      new WrappingBitrateEstimator(engine_id,
-                                   remb_.get(),
+      new WrappingBitrateEstimator(remb_.get(),
                                    Clock::GetRealTimeClock(),
                                    *config_));
 
   call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
 
+  pacer_thread_->RegisterModule(pacer_.get());
+  pacer_thread_->Start();
+
   process_thread->RegisterModule(remote_bitrate_estimator_.get());
   process_thread->RegisterModule(call_stats_.get());
   process_thread->RegisterModule(bitrate_controller_.get());
 }
 
 ChannelGroup::~ChannelGroup() {
+  pacer_thread_->Stop();
+  pacer_thread_->DeRegisterModule(pacer_.get());
   process_thread_->DeRegisterModule(bitrate_controller_.get());
   process_thread_->DeRegisterModule(call_stats_.get());
   process_thread_->DeRegisterModule(remote_bitrate_estimator_.get());
   call_stats_->DeregisterStatsObserver(remote_bitrate_estimator_.get());
-  assert(channels_.empty());
-  assert(!remb_->InUse());
+  DCHECK(channels_.empty());
+  DCHECK(channel_map_.empty());
+  DCHECK(!remb_->InUse());
+  DCHECK(vie_encoder_map_.empty());
+  DCHECK(send_encoders_.empty());
+}
+
+bool ChannelGroup::CreateSendChannel(int channel_id,
+                                     int engine_id,
+                                     int number_of_cores,
+                                     bool disable_default_encoder) {
+  rtc::scoped_ptr vie_encoder(new ViEEncoder(
+      channel_id, number_of_cores, *config_, *process_thread_, pacer_.get(),
+      bitrate_allocator_.get(), bitrate_controller_.get(), false));
+  if (!vie_encoder->Init()) {
+    return false;
+  }
+  ViEEncoder* encoder = vie_encoder.get();
+  if (!CreateChannel(channel_id, engine_id, number_of_cores,
+                     vie_encoder.release(), true, disable_default_encoder)) {
+    return false;
+  }
+  ViEChannel* channel = channel_map_[channel_id];
+  // Connect the encoder with the send packet router, to enable sending.
+  encoder->StartThreadsAndSetSharedMembers(channel->send_payload_router(),
+                                           channel->vcm_protection_callback());
+
+  // Register the ViEEncoder to get key frame requests for this channel.
+  unsigned int ssrc = 0;
+  int stream_idx = 0;
+  channel->GetLocalSSRC(stream_idx, &ssrc);
+  encoder_state_feedback_->AddEncoder(ssrc, encoder);
+  std::list ssrcs;
+  ssrcs.push_back(ssrc);
+  encoder->SetSsrcs(ssrcs);
+  return true;
+}
+
+bool ChannelGroup::CreateReceiveChannel(int channel_id,
+                                        int engine_id,
+                                        int base_channel_id,
+                                        int number_of_cores,
+                                        bool disable_default_encoder) {
+  ViEEncoder* encoder = GetEncoder(base_channel_id);
+  return CreateChannel(channel_id, engine_id, number_of_cores, encoder, false,
+                       disable_default_encoder);
+}
+
+bool ChannelGroup::CreateChannel(int channel_id,
+                                 int engine_id,
+                                 int number_of_cores,
+                                 ViEEncoder* vie_encoder,
+                                 bool sender,
+                                 bool disable_default_encoder) {
+  DCHECK(vie_encoder);
+
+  rtc::scoped_ptr channel(new ViEChannel(
+      channel_id, engine_id, number_of_cores, *config_, *process_thread_,
+      encoder_state_feedback_->GetRtcpIntraFrameObserver(),
+      bitrate_controller_->CreateRtcpBandwidthObserver(),
+      remote_bitrate_estimator_.get(), call_stats_->rtcp_rtt_stats(),
+      pacer_.get(), packet_router_.get(), sender, disable_default_encoder));
+  if (channel->Init() != 0) {
+    return false;
+  }
+  if (!disable_default_encoder) {
+    VideoCodec encoder;
+    if (vie_encoder->GetEncoder(&encoder) != 0) {
+      return false;
+    }
+    if (sender && channel->SetSendCodec(encoder) != 0) {
+      return false;
+    }
+  }
+
+  // Register the channel to receive stats updates.
+  call_stats_->RegisterStatsObserver(channel->GetStatsObserver());
+
+  // Store the channel, add it to the channel group and save the vie_encoder.
+  channel_map_[channel_id] = channel.release();
+  {
+    CriticalSectionScoped lock(encoder_map_cs_.get());
+    vie_encoder_map_[channel_id] = vie_encoder;
+    if (sender)
+      send_encoders_[channel_id] = vie_encoder;
+  }
+
+  return true;
+}
+
+void ChannelGroup::DeleteChannel(int channel_id) {
+  ViEChannel* vie_channel = PopChannel(channel_id);
+
+  ViEEncoder* vie_encoder = GetEncoder(channel_id);
+  DCHECK(vie_encoder != NULL);
+
+  call_stats_->DeregisterStatsObserver(vie_channel->GetStatsObserver());
+  SetChannelRembStatus(channel_id, false, false, vie_channel);
+
+  // If we're owning the encoder, remove the feedback and stop all encoding
+  // threads and processing. This must be done before deleting the channel.
+  if (vie_encoder->channel_id() == channel_id) {
+    encoder_state_feedback_->RemoveEncoder(vie_encoder);
+    vie_encoder->StopThreadsAndRemoveSharedMembers();
+  }
+
+  unsigned int remote_ssrc = 0;
+  vie_channel->GetRemoteSSRC(&remote_ssrc);
+  RemoveChannel(channel_id);
+  remote_bitrate_estimator_->RemoveStream(remote_ssrc);
+
+  // Check if other channels are using the same encoder.
+  if (OtherChannelsUsingEncoder(channel_id)) {
+    vie_encoder = NULL;
+  } else {
+    // Delete later when we've released the critsect.
+  }
+
+  // We can't erase the item before we've checked for other channels using
+  // same ViEEncoder.
+  PopEncoder(channel_id);
+
+  delete vie_channel;
+  // Leave the write critsect before deleting the objects.
+  // Deleting a channel can cause other objects, such as renderers, to be
+  // deleted, which might take time.
+  // If statment just to show that this object is not always deleted.
+  if (vie_encoder) {
+    LOG(LS_VERBOSE) << "ViEEncoder deleted for channel " << channel_id;
+    delete vie_encoder;
+  }
+
+  LOG(LS_VERBOSE) << "Channel deleted " << channel_id;
 }
 
 void ChannelGroup::AddChannel(int channel_id) {
   channels_.insert(channel_id);
 }
 
-void ChannelGroup::RemoveChannel(int channel_id, unsigned int ssrc) {
+void ChannelGroup::RemoveChannel(int channel_id) {
   channels_.erase(channel_id);
-  remote_bitrate_estimator_->RemoveStream(ssrc);
 }
 
-bool ChannelGroup::HasChannel(int channel_id) {
+bool ChannelGroup::HasChannel(int channel_id) const {
   return channels_.find(channel_id) != channels_.end();
 }
 
-bool ChannelGroup::Empty() {
+bool ChannelGroup::Empty() const {
   return channels_.empty();
 }
 
-BitrateController* ChannelGroup::GetBitrateController() {
+ViEChannel* ChannelGroup::GetChannel(int channel_id) const {
+  ChannelMap::const_iterator it = channel_map_.find(channel_id);
+  if (it == channel_map_.end()) {
+    LOG(LS_ERROR) << "Channel doesn't exist " << channel_id;
+    return NULL;
+  }
+  return it->second;
+}
+
+ViEEncoder* ChannelGroup::GetEncoder(int channel_id) const {
+  CriticalSectionScoped lock(encoder_map_cs_.get());
+  EncoderMap::const_iterator it = vie_encoder_map_.find(channel_id);
+  if (it == vie_encoder_map_.end()) {
+    return NULL;
+  }
+  return it->second;
+}
+
+ViEChannel* ChannelGroup::PopChannel(int channel_id) {
+  ChannelMap::iterator c_it = channel_map_.find(channel_id);
+  DCHECK(c_it != channel_map_.end());
+  ViEChannel* channel = c_it->second;
+  channel_map_.erase(c_it);
+
+  return channel;
+}
+
+ViEEncoder* ChannelGroup::PopEncoder(int channel_id) {
+  CriticalSectionScoped lock(encoder_map_cs_.get());
+  auto it = vie_encoder_map_.find(channel_id);
+  DCHECK(it != vie_encoder_map_.end());
+  ViEEncoder* encoder = it->second;
+  vie_encoder_map_.erase(it);
+
+  it = send_encoders_.find(channel_id);
+  if (it != send_encoders_.end())
+    send_encoders_.erase(it);
+
+  return encoder;
+}
+
+std::vector ChannelGroup::GetChannelIds() const {
+  std::vector ids;
+  for (auto channel : channel_map_)
+    ids.push_back(channel.first);
+  return ids;
+}
+
+bool ChannelGroup::OtherChannelsUsingEncoder(int channel_id) const {
+  CriticalSectionScoped lock(encoder_map_cs_.get());
+  EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
+  if (orig_it == vie_encoder_map_.end()) {
+    // No ViEEncoder for this channel.
+    return false;
+  }
+
+  // Loop through all other channels to see if anyone points at the same
+  // ViEEncoder.
+  for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
+       comp_it != vie_encoder_map_.end(); ++comp_it) {
+    // Make sure we're not comparing the same channel with itself.
+    if (comp_it->first != channel_id) {
+      if (comp_it->second == orig_it->second) {
+        return true;
+      }
+    }
+  }
+  return false;
+}
+
+void ChannelGroup::SetLoadManager(CPULoadStateCallbackInvoker* load_manager) {
+  for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
+       comp_it != vie_encoder_map_.end(); ++comp_it) {
+    comp_it->second->SetLoadManager(load_manager);
+  }
+}
+
+void ChannelGroup::SetSyncInterface(VoEVideoSync* sync_interface) {
+  for (auto channel : channel_map_) {
+    channel.second->SetVoiceChannel(-1, sync_interface);
+  }
+}
+
+void ChannelGroup::GetChannelsUsingEncoder(int channel_id,
+                                           ChannelList* channels) const {
+  CriticalSectionScoped lock(encoder_map_cs_.get());
+  EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
+
+  for (ChannelMap::const_iterator c_it = channel_map_.begin();
+       c_it != channel_map_.end(); ++c_it) {
+    EncoderMap::const_iterator comp_it = vie_encoder_map_.find(c_it->first);
+    DCHECK(comp_it != vie_encoder_map_.end());
+    if (comp_it->second == orig_it->second) {
+      channels->push_back(c_it->second);
+    }
+  }
+}
+
+BitrateController* ChannelGroup::GetBitrateController() const {
   return bitrate_controller_.get();
 }
 
-RemoteBitrateEstimator* ChannelGroup::GetRemoteBitrateEstimator() {
+RemoteBitrateEstimator* ChannelGroup::GetRemoteBitrateEstimator() const {
   return remote_bitrate_estimator_.get();
 }
 
-CallStats* ChannelGroup::GetCallStats() {
+CallStats* ChannelGroup::GetCallStats() const {
   return call_stats_.get();
 }
 
-EncoderStateFeedback* ChannelGroup::GetEncoderStateFeedback() {
+EncoderStateFeedback* ChannelGroup::GetEncoderStateFeedback() const {
   return encoder_state_feedback_.get();
 }
 
-bool ChannelGroup::SetChannelRembStatus(int channel_id, bool sender,
-                                        bool receiver, ViEChannel* channel) {
+int64_t ChannelGroup::GetPacerQueuingDelayMs() const {
+  return pacer_->QueueInMs();
+}
+
+void ChannelGroup::SetChannelRembStatus(int channel_id,
+                                        bool sender,
+                                        bool receiver,
+                                        ViEChannel* channel) {
   // Update the channel state.
-  if (sender || receiver) {
-    if (!channel->EnableRemb(true)) {
-      return false;
-    }
-  } else {
-    channel->EnableRemb(false);
-  }
+  channel->EnableRemb(sender || receiver);
   // Update the REMB instance with necessary RTP modules.
   RtpRtcp* rtp_module = channel->rtp_rtcp();
   if (sender) {
@@ -248,12 +478,23 @@ bool ChannelGroup::SetChannelRembStatus(int channel_id, bool sender,
   } else {
     remb_->RemoveReceiveChannel(rtp_module);
   }
-  return true;
 }
 
-void ChannelGroup::SetBandwidthEstimationConfig(const webrtc::Config& config) {
-  WrappingBitrateEstimator* estimator =
-      static_cast(remote_bitrate_estimator_.get());
-  estimator->SetConfig(config);
+void ChannelGroup::OnNetworkChanged(uint32_t target_bitrate_bps,
+                                    uint8_t fraction_loss,
+                                    int64_t rtt) {
+  bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt);
+  int pad_up_to_bitrate_bps = 0;
+  {
+    CriticalSectionScoped lock(encoder_map_cs_.get());
+    for (const auto& encoder : send_encoders_) {
+      pad_up_to_bitrate_bps +=
+          encoder.second->GetPaddingNeededBps(target_bitrate_bps);
+    }
+  }
+  pacer_->UpdateBitrate(
+      target_bitrate_bps / 1000,
+      PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000,
+      pad_up_to_bitrate_bps / 1000);
 }
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h b/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h
index 40ce71ef19..6db1a98bc9 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_channel_group.h
@@ -11,59 +11,114 @@
 #ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_GROUP_H_
 #define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_GROUP_H_
 
+#include 
+#include 
 #include 
+#include 
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 
 namespace webrtc {
 
-class BitrateController;
+class BitrateAllocator;
 class CallStats;
 class Config;
 class EncoderStateFeedback;
+class PacedSender;
+class PacketRouter;
 class ProcessThread;
 class RemoteBitrateEstimator;
 class ViEChannel;
 class ViEEncoder;
 class VieRemb;
+class VoEVideoSync;
+class CPULoadStateCallbackInvoker;
+
+typedef std::list ChannelList;
 
 // Channel group contains data common for several channels. All channels in the
 // group are assumed to send/receive data to the same end-point.
-class ChannelGroup {
+class ChannelGroup : public BitrateObserver {
  public:
-  ChannelGroup(int engine_id, ProcessThread* process_thread,
-               const Config* config);
+  ChannelGroup(ProcessThread* process_thread, const Config* config);
   ~ChannelGroup();
-
+  bool CreateSendChannel(int channel_id,
+                         int engine_id,
+                         int number_of_cores,
+                         bool disable_default_encoder);
+  bool CreateReceiveChannel(int channel_id,
+                            int engine_id,
+                            int base_channel_id,
+                            int number_of_cores,
+                            bool disable_default_encoder);
+  void DeleteChannel(int channel_id);
   void AddChannel(int channel_id);
-  void RemoveChannel(int channel_id, unsigned int ssrc);
-  bool HasChannel(int channel_id);
-  bool Empty();
+  void RemoveChannel(int channel_id);
+  bool HasChannel(int channel_id) const;
+  bool Empty() const;
+  ViEChannel* GetChannel(int channel_id) const;
+  ViEEncoder* GetEncoder(int channel_id) const;
+  std::vector GetChannelIds() const;
+  bool OtherChannelsUsingEncoder(int channel_id) const;
+  void SetLoadManager(CPULoadStateCallbackInvoker* load_manager);
+  void GetChannelsUsingEncoder(int channel_id, ChannelList* channels) const;
 
-  bool SetChannelRembStatus(int channel_id, bool sender, bool receiver,
+  void SetSyncInterface(VoEVideoSync* sync_interface);
+
+  void SetChannelRembStatus(int channel_id,
+                            bool sender,
+                            bool receiver,
                             ViEChannel* channel);
-  void SetBandwidthEstimationConfig(const webrtc::Config& config);
 
-  BitrateController* GetBitrateController();
-  CallStats* GetCallStats();
-  RemoteBitrateEstimator* GetRemoteBitrateEstimator();
-  EncoderStateFeedback* GetEncoderStateFeedback();
+  BitrateController* GetBitrateController() const;
+  CallStats* GetCallStats() const;
+  RemoteBitrateEstimator* GetRemoteBitrateEstimator() const;
+  EncoderStateFeedback* GetEncoderStateFeedback() const;
+  int64_t GetPacerQueuingDelayMs() const;
+
+  // Implements BitrateObserver.
+  void OnNetworkChanged(uint32_t target_bitrate_bps,
+                        uint8_t fraction_loss,
+                        int64_t rtt) override;
 
  private:
+  typedef std::map ChannelMap;
   typedef std::set ChannelSet;
+  typedef std::map EncoderMap;
 
-  scoped_ptr remb_;
-  scoped_ptr bitrate_controller_;
-  scoped_ptr call_stats_;
-  scoped_ptr remote_bitrate_estimator_;
-  scoped_ptr encoder_state_feedback_;
+  bool CreateChannel(int channel_id,
+                     int engine_id,
+                     int number_of_cores,
+                     ViEEncoder* vie_encoder,
+                     bool sender,
+                     bool disable_default_encoder);
+  ViEChannel* PopChannel(int channel_id);
+  ViEEncoder* PopEncoder(int channel_id);
+
+  rtc::scoped_ptr remb_;
+  rtc::scoped_ptr bitrate_allocator_;
+  rtc::scoped_ptr call_stats_;
+  rtc::scoped_ptr remote_bitrate_estimator_;
+  rtc::scoped_ptr encoder_state_feedback_;
+  rtc::scoped_ptr packet_router_;
+  rtc::scoped_ptr pacer_;
   ChannelSet channels_;
+  ChannelMap channel_map_;
+  // Maps Channel id -> ViEEncoder.
+  EncoderMap vie_encoder_map_;
+  EncoderMap send_encoders_;
+  rtc::scoped_ptr encoder_map_cs_;
+
   const Config* config_;
   // Placeholder for the case where this owns the config.
-  scoped_ptr own_config_;
+  rtc::scoped_ptr own_config_;
 
   // Registered at construct time and assumed to outlive this class.
   ProcessThread* process_thread_;
+  rtc::scoped_ptr pacer_thread_;
+
+  rtc::scoped_ptr bitrate_controller_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc b/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc
index 11d77c64ae..669f9213cd 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.cc
@@ -10,6 +10,8 @@
 
 #include "webrtc/video_engine/vie_channel_manager.h"
 
+#include 
+
 #include "webrtc/common.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
@@ -19,6 +21,7 @@
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/encoder_state_feedback.h"
 #include "webrtc/video_engine/vie_channel.h"
+#include "webrtc/video_engine/vie_channel_group.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_engine/vie_remb.h"
@@ -26,19 +29,16 @@
 
 namespace webrtc {
 
-ViEChannelManager::ViEChannelManager(
-    int engine_id,
-    int number_of_cores,
-    const Config& config)
+ViEChannelManager::ViEChannelManager(int engine_id,
+                                     int number_of_cores,
+                                     const Config& config)
     : channel_id_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       engine_id_(engine_id),
       number_of_cores_(number_of_cores),
       free_channel_ids_(new bool[kViEMaxNumberOfChannels]),
       free_channel_ids_size_(kViEMaxNumberOfChannels),
       voice_sync_interface_(NULL),
-      voice_engine_(NULL),
       module_process_thread_(NULL),
-      engine_config_(config),
       load_manager_(NULL)
 {
   for (int idx = 0; idx < free_channel_ids_size_; idx++) {
@@ -47,10 +47,12 @@ ViEChannelManager::ViEChannelManager(
 }
 
 ViEChannelManager::~ViEChannelManager() {
-  while (channel_map_.size() > 0) {
-    ChannelMap::iterator it = channel_map_.begin();
-    // DeleteChannel will erase this channel from the map and invalidate |it|.
-    DeleteChannel(it->first);
+  while (!channel_groups_.empty()) {
+    // The channel group is deleted by DeleteChannel when all its channels have
+    // been deleted.
+    for (int channel_id : channel_groups_.front()->GetChannelIds()) {
+      DeleteChannel(channel_id);
+    }
   }
 
   if (voice_sync_interface_) {
@@ -66,8 +68,6 @@ ViEChannelManager::~ViEChannelManager() {
     free_channel_ids_size_ = 0;
   }
   assert(channel_groups_.empty());
-  assert(channel_map_.empty());
-  assert(vie_encoder_map_.empty());
 }
 
 void ViEChannelManager::SetModuleProcessThread(
@@ -79,9 +79,9 @@ void ViEChannelManager::SetModuleProcessThread(
 void ViEChannelManager::SetLoadManager(
     CPULoadStateCallbackInvoker* load_manager) {
   load_manager_ = load_manager;
-  for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
-       comp_it != vie_encoder_map_.end(); ++comp_it) {
-    comp_it->second->SetLoadManager(load_manager);
+  for (ChannelGroups::const_iterator it = channel_groups_.begin();
+       it != channel_groups_.end(); ++it) {
+    (*it)->SetLoadManager(load_manager);
   }
 }
 
@@ -96,56 +96,25 @@ int ViEChannelManager::CreateChannel(int* channel_id,
   }
 
   // Create a new channel group and add this channel.
-  ChannelGroup* group = new ChannelGroup(engine_id_, module_process_thread_,
-                                         channel_group_config);
-  BitrateController* bitrate_controller = group->GetBitrateController();
-  ViEEncoder* vie_encoder = new ViEEncoder(engine_id_, new_channel_id,
-                                           number_of_cores_,
-                                           engine_config_,
-                                           *module_process_thread_,
-                                           bitrate_controller);
+  rtc::scoped_ptr group(
+      new ChannelGroup(module_process_thread_, channel_group_config));
 
-  RtcpBandwidthObserver* bandwidth_observer =
-      bitrate_controller->CreateRtcpBandwidthObserver();
-  RemoteBitrateEstimator* remote_bitrate_estimator =
-      group->GetRemoteBitrateEstimator();
-  EncoderStateFeedback* encoder_state_feedback =
-      group->GetEncoderStateFeedback();
-  RtcpRttStats* rtcp_rtt_stats =
-      group->GetCallStats()->rtcp_rtt_stats();
-
-  if (!(vie_encoder->Init() &&
-        CreateChannelObject(new_channel_id, vie_encoder, bandwidth_observer,
-                            remote_bitrate_estimator, rtcp_rtt_stats,
-                            encoder_state_feedback->GetRtcpIntraFrameObserver(),
-                            true))) {
-    delete vie_encoder;
-    vie_encoder = NULL;
+  if (!group->CreateSendChannel(new_channel_id, engine_id_, number_of_cores_,
+                                false)) {
     ReturnChannelId(new_channel_id);
-    delete group;
     return -1;
   }
 
-  // Add ViEEncoder to EncoderFeedBackObserver.
-  unsigned int ssrc = 0;
-  int idx = 0;
-  channel_map_[new_channel_id]->GetLocalSSRC(idx, &ssrc);
-  encoder_state_feedback->AddEncoder(ssrc, vie_encoder);
-  std::list ssrcs;
-  ssrcs.push_back(ssrc);
-  vie_encoder->SetSsrcs(ssrcs);
   *channel_id = new_channel_id;
   group->AddChannel(*channel_id);
-  channel_groups_.push_back(group);
-  // Register the channel to receive stats updates.
-  group->GetCallStats()->RegisterStatsObserver(
-      channel_map_[new_channel_id]->GetStatsObserver());
+  channel_groups_.push_back(group.release());
   return 0;
 }
 
 int ViEChannelManager::CreateChannel(int* channel_id,
                                      int original_channel,
-                                     bool sender) {
+                                     bool sender,
+                                     bool disable_default_encoder) {
   CriticalSectionScoped cs(channel_id_critsect_);
 
   ChannelGroup* channel_group = FindGroup(original_channel);
@@ -156,69 +125,27 @@ int ViEChannelManager::CreateChannel(int* channel_id,
   if (new_channel_id == -1) {
     return -1;
   }
-  BitrateController* bitrate_controller = channel_group->GetBitrateController();
-  RtcpBandwidthObserver* bandwidth_observer =
-      bitrate_controller->CreateRtcpBandwidthObserver();
-  RemoteBitrateEstimator* remote_bitrate_estimator =
-      channel_group->GetRemoteBitrateEstimator();
-  EncoderStateFeedback* encoder_state_feedback =
-      channel_group->GetEncoderStateFeedback();
-    RtcpRttStats* rtcp_rtt_stats =
-        channel_group->GetCallStats()->rtcp_rtt_stats();
-
-  ViEEncoder* vie_encoder = NULL;
   if (sender) {
-    // We need to create a new ViEEncoder.
-    vie_encoder = new ViEEncoder(engine_id_, new_channel_id, number_of_cores_,
-                                 engine_config_,
-                                 *module_process_thread_,
-                                 bitrate_controller);
-    if (!(vie_encoder->Init() &&
-        CreateChannelObject(
-            new_channel_id,
-            vie_encoder,
-            bandwidth_observer,
-            remote_bitrate_estimator,
-            rtcp_rtt_stats,
-            encoder_state_feedback->GetRtcpIntraFrameObserver(),
-            sender))) {
-      delete vie_encoder;
-      vie_encoder = NULL;
+    if (!channel_group->CreateSendChannel(new_channel_id, engine_id_,
+                                          number_of_cores_,
+                                          disable_default_encoder)) {
+      ReturnChannelId(new_channel_id);
+      return -1;
     }
-    // Register the ViEEncoder to get key frame requests for this channel.
-    unsigned int ssrc = 0;
-    int stream_idx = 0;
-    channel_map_[new_channel_id]->GetLocalSSRC(stream_idx, &ssrc);
-    encoder_state_feedback->AddEncoder(ssrc, vie_encoder);
   } else {
-    vie_encoder = ViEEncoderPtr(original_channel);
-    assert(vie_encoder);
-    if (!CreateChannelObject(
-        new_channel_id,
-        vie_encoder,
-        bandwidth_observer,
-        remote_bitrate_estimator,
-        rtcp_rtt_stats,
-        encoder_state_feedback->GetRtcpIntraFrameObserver(),
-        sender)) {
-      vie_encoder = NULL;
+    if (!channel_group->CreateReceiveChannel(new_channel_id, engine_id_,
+                                             original_channel, number_of_cores_,
+                                             disable_default_encoder)) {
+      ReturnChannelId(new_channel_id);
+      return -1;
     }
   }
-  if (!vie_encoder) {
-    ReturnChannelId(new_channel_id);
-    return -1;
-  }
   *channel_id = new_channel_id;
   channel_group->AddChannel(*channel_id);
-  // Register the channel to receive stats updates.
-  channel_group->GetCallStats()->RegisterStatsObserver(
-      channel_map_[new_channel_id]->GetStatsObserver());
   return 0;
 }
 
 int ViEChannelManager::DeleteChannel(int channel_id) {
-  ViEChannel* vie_channel = NULL;
-  ViEEncoder* vie_encoder = NULL;
   ChannelGroup* group = NULL;
   {
     // Write lock to make sure no one is using the channel.
@@ -227,45 +154,11 @@ int ViEChannelManager::DeleteChannel(int channel_id) {
     // Protect the maps.
     CriticalSectionScoped cs(channel_id_critsect_);
 
-    ChannelMap::iterator c_it = channel_map_.find(channel_id);
-    if (c_it == channel_map_.end()) {
-      // No such channel.
-      return -1;
-    }
-    vie_channel = c_it->second;
-    channel_map_.erase(c_it);
-
-    ReturnChannelId(channel_id);
-
-    // Find the encoder object.
-    EncoderMap::iterator e_it = vie_encoder_map_.find(channel_id);
-    assert(e_it != vie_encoder_map_.end());
-    vie_encoder = e_it->second;
-
     group = FindGroup(channel_id);
-    group->GetCallStats()->DeregisterStatsObserver(
-        vie_channel->GetStatsObserver());
-    group->SetChannelRembStatus(channel_id, false, false, vie_channel);
-
-    // Remove the feedback if we're owning the encoder.
-    if (vie_encoder->channel_id() == channel_id) {
-      group->GetEncoderStateFeedback()->RemoveEncoder(vie_encoder);
-    }
-
-    unsigned int remote_ssrc = 0;
-    vie_channel->GetRemoteSSRC(&remote_ssrc);
-    group->RemoveChannel(channel_id, remote_ssrc);
-
-    // Check if other channels are using the same encoder.
-    if (ChannelUsingViEEncoder(channel_id)) {
-      vie_encoder = NULL;
-    } else {
-      // Delete later when we've released the critsect.
-    }
-
-    // We can't erase the item before we've checked for other channels using
-    // same ViEEncoder.
-    vie_encoder_map_.erase(e_it);
+    if (group == NULL)
+      return -1;
+    ReturnChannelId(channel_id);
+    group->DeleteChannel(channel_id);
 
     if (group->Empty()) {
       channel_groups_.remove(group);
@@ -273,15 +166,6 @@ int ViEChannelManager::DeleteChannel(int channel_id) {
       group = NULL;  // Prevent group from being deleted.
     }
   }
-  delete vie_channel;
-  // Leave the write critsect before deleting the objects.
-  // Deleting a channel can cause other objects, such as renderers, to be
-  // deleted, which might take time.
-  // If statment just to show that this object is not always deleted.
-  if (vie_encoder) {
-    LOG(LS_VERBOSE) << "ViEEncoder deleted for channel " << channel_id;
-    delete vie_encoder;
-  }
   // If statment just to show that this object is not always deleted.
   if (group) {
     // Delete the group if empty last since the encoder holds a pointer to the
@@ -289,7 +173,6 @@ int ViEChannelManager::DeleteChannel(int channel_id) {
     LOG(LS_VERBOSE) << "Channel group deleted for channel " << channel_id;
     delete group;
   }
-  LOG(LS_VERBOSE) << "Channel deleted " << channel_id;
   return 0;
 }
 
@@ -308,14 +191,12 @@ int ViEChannelManager::SetVoiceEngine(VoiceEngine* voice_engine) {
     }
   }
 
-  for (ChannelMap::iterator it = channel_map_.begin(); it != channel_map_.end();
-       ++it) {
-    it->second->SetVoiceChannel(-1, sync_interface);
+  for (ChannelGroup* group : channel_groups_) {
+    group->SetSyncInterface(sync_interface);
   }
   if (voice_sync_interface_) {
     voice_sync_interface_->Release();
   }
-  voice_engine_ = voice_engine;
   voice_sync_interface_ = sync_interface;
   return 0;
 }
@@ -344,11 +225,6 @@ int ViEChannelManager::DisconnectVoiceChannel(int channel_id) {
   return -1;
 }
 
-VoiceEngine* ViEChannelManager::GetVoiceEngine() {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  return voice_engine_;
-}
-
 bool ViEChannelManager::SetRembStatus(int channel_id, bool sender,
                                       bool receiver) {
   CriticalSectionScoped cs(channel_id_critsect_);
@@ -359,7 +235,8 @@ bool ViEChannelManager::SetRembStatus(int channel_id, bool sender,
   ViEChannel* channel = ViEChannelPtr(channel_id);
   assert(channel);
 
-  return group->SetChannelRembStatus(channel_id, sender, receiver, channel);
+  group->SetChannelRembStatus(channel_id, sender, receiver, channel);
+  return true;
 }
 
 bool ViEChannelManager::SetReservedTransmitBitrate(
@@ -396,17 +273,6 @@ void ViEChannelManager::UpdateSsrcs(int channel_id,
   }
 }
 
-bool ViEChannelManager::SetBandwidthEstimationConfig(
-    int channel_id, const webrtc::Config& config) {
-  CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelGroup* group = FindGroup(channel_id);
-  if (!group) {
-    return false;
-  }
-  group->SetBandwidthEstimationConfig(config);
-  return true;
-}
-
 bool ViEChannelManager::GetEstimatedSendBandwidth(
     int channel_id, uint32_t* estimated_bandwidth) const {
   CriticalSectionScoped cs(channel_id_critsect_);
@@ -433,66 +299,46 @@ bool ViEChannelManager::GetEstimatedReceiveBandwidth(
   return true;
 }
 
-bool ViEChannelManager::CreateChannelObject(
-    int channel_id,
-    ViEEncoder* vie_encoder,
-    RtcpBandwidthObserver* bandwidth_observer,
-    RemoteBitrateEstimator* remote_bitrate_estimator,
-    RtcpRttStats* rtcp_rtt_stats,
-    RtcpIntraFrameObserver* intra_frame_observer,
-    bool sender) {
-  PacedSender* paced_sender = vie_encoder->GetPacedSender();
+bool ViEChannelManager::GetPacerQueuingDelayMs(int channel_id,
+                                               int64_t* delay_ms) const {
+  CriticalSectionScoped cs(channel_id_critsect_);
+  ChannelGroup* group = FindGroup(channel_id);
+  if (!group)
+    return false;
+  *delay_ms = group->GetPacerQueuingDelayMs();
+  return true;
+}
 
-  // Register the channel at the encoder.
-  RtpRtcp* send_rtp_rtcp_module = vie_encoder->SendRtpRtcpModule();
-
-  ViEChannel* vie_channel = new ViEChannel(channel_id, engine_id_,
-                                           number_of_cores_,
-                                           engine_config_,
-                                           *module_process_thread_,
-                                           intra_frame_observer,
-                                           bandwidth_observer,
-                                           remote_bitrate_estimator,
-                                           rtcp_rtt_stats,
-                                           paced_sender,
-                                           send_rtp_rtcp_module,
-                                           sender);
-  if (vie_channel->Init() != 0) {
-    delete vie_channel;
+bool ViEChannelManager::SetBitrateConfig(int channel_id,
+                                         int min_bitrate_bps,
+                                         int start_bitrate_bps,
+                                         int max_bitrate_bps) {
+  CriticalSectionScoped cs(channel_id_critsect_);
+  ChannelGroup* group = FindGroup(channel_id);
+  if (!group)
     return false;
-  }
-  VideoCodec encoder;
-  if (vie_encoder->GetEncoder(&encoder) != 0) {
-    delete vie_channel;
-    return false;
-  }
-  if (sender && vie_channel->SetSendCodec(encoder) != 0) {
-    delete vie_channel;
-    return false;
-  }
-  // Store the channel, add it to the channel group and save the vie_encoder.
-  channel_map_[channel_id] = vie_channel;
-  vie_encoder_map_[channel_id] = vie_encoder;
+  BitrateController* bitrate_controller = group->GetBitrateController();
+  if (start_bitrate_bps > 0)
+    bitrate_controller->SetStartBitrate(start_bitrate_bps);
+  bitrate_controller->SetMinMaxBitrate(min_bitrate_bps, max_bitrate_bps);
   return true;
 }
 
 ViEChannel* ViEChannelManager::ViEChannelPtr(int channel_id) const {
   CriticalSectionScoped cs(channel_id_critsect_);
-  ChannelMap::const_iterator it = channel_map_.find(channel_id);
-  if (it == channel_map_.end()) {
-    LOG(LS_ERROR) << "Channel doesn't exist " << channel_id;
+  ChannelGroup* group = FindGroup(channel_id);
+  if (group == NULL)
     return NULL;
-  }
-  return it->second;
+  return group->GetChannel(channel_id);
 }
 
 ViEEncoder* ViEChannelManager::ViEEncoderPtr(int video_channel_id) const {
   CriticalSectionScoped cs(channel_id_critsect_);
-  EncoderMap::const_iterator it = vie_encoder_map_.find(video_channel_id);
-  if (it == vie_encoder_map_.end()) {
+  ChannelGroup* group = FindGroup(video_channel_id);
+  if (group == NULL) {
     return NULL;
   }
-  return it->second;
+  return group->GetEncoder(video_channel_id);
 }
 
 int ViEChannelManager::FreeChannelId() {
@@ -528,39 +374,20 @@ ChannelGroup* ViEChannelManager::FindGroup(int channel_id) const {
 
 bool ViEChannelManager::ChannelUsingViEEncoder(int channel_id) const {
   CriticalSectionScoped cs(channel_id_critsect_);
-  EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
-  if (orig_it == vie_encoder_map_.end()) {
-    // No ViEEncoder for this channel.
+  ChannelGroup* group = FindGroup(channel_id);
+  if (group == NULL) {
     return false;
   }
-
-  // Loop through all other channels to see if anyone points at the same
-  // ViEEncoder.
-  for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
-       comp_it != vie_encoder_map_.end(); ++comp_it) {
-    // Make sure we're not comparing the same channel with itself.
-    if (comp_it->first != channel_id) {
-      if (comp_it->second == orig_it->second) {
-        return true;
-      }
-    }
-  }
-  return false;
+  return group->OtherChannelsUsingEncoder(channel_id);
 }
 
 void ViEChannelManager::ChannelsUsingViEEncoder(int channel_id,
                                                 ChannelList* channels) const {
   CriticalSectionScoped cs(channel_id_critsect_);
-  EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
-
-  for (ChannelMap::const_iterator c_it = channel_map_.begin();
-       c_it != channel_map_.end(); ++c_it) {
-    EncoderMap::const_iterator comp_it = vie_encoder_map_.find(c_it->first);
-    assert(comp_it != vie_encoder_map_.end());
-    if (comp_it->second == orig_it->second) {
-      channels->push_back(c_it->second);
-    }
-  }
+  ChannelGroup* group = FindGroup(channel_id);
+  if (group == NULL)
+    return;
+  group->GetChannelsUsingEncoder(channel_id, channels);
 }
 
 ViEChannelManagerScoped::ViEChannelManagerScoped(
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h b/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h
index f2a64bd340..f6eb9dab4c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_channel_manager.h
@@ -15,16 +15,15 @@
 #include 
 
 #include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/include/vie_rtp_rtcp.h"
-#include "webrtc/video_engine/vie_channel_group.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/video_engine/vie_manager_base.h"
 #include "webrtc/video_engine/vie_remb.h"
 
 namespace webrtc {
 
+class ChannelGroup;
 class Config;
 class CriticalSectionWrapper;
 class ProcessThread;
@@ -36,8 +35,6 @@ class VoiceEngine;
 
 typedef std::list ChannelGroups;
 typedef std::list ChannelList;
-typedef std::map ChannelMap;
-typedef std::map EncoderMap;
 
 class ViEChannelManager: private ViEManagerBase {
   friend class ViEChannelManagerScoped;
@@ -58,7 +55,11 @@ class ViEChannelManager: private ViEManagerBase {
   // Creates a new channel grouped with |original_channel|. The new channel
   // will get its own |ViEEncoder| if |sender| is set to true. It will be a
   // receive only channel, without an own |ViEEncoder| if |sender| is false.
-  int CreateChannel(int* channel_id, int original_channel, bool sender);
+  // Doesn't internally allocate an encoder if |disable_default_encoder|.
+  int CreateChannel(int* channel_id,
+                    int original_channel,
+                    bool sender,
+                    bool disable_default_encoder);
 
   // Deletes a channel.
   int DeleteChannel(int channel_id);
@@ -72,8 +73,6 @@ class ViEChannelManager: private ViEManagerBase {
   // Disables lip sync of the channel.
   int DisconnectVoiceChannel(int channel_id);
 
-  VoiceEngine* GetVoiceEngine();
-
   // Adds a channel to include when sending REMB.
   bool SetRembStatus(int channel_id, bool sender, bool receiver);
 
@@ -84,26 +83,21 @@ class ViEChannelManager: private ViEManagerBase {
   // it will simply be ignored and no error is returned.
   void UpdateSsrcs(int channel_id, const std::list& ssrcs);
 
-  // Sets bandwidth estimation related configurations.
-  bool SetBandwidthEstimationConfig(int channel_id,
-                                    const webrtc::Config& config);
-
   bool GetEstimatedSendBandwidth(int channel_id,
                                  uint32_t* estimated_bandwidth) const;
   bool GetEstimatedReceiveBandwidth(int channel_id,
                                     uint32_t* estimated_bandwidth) const;
 
- private:
-  // Creates a channel object connected to |vie_encoder|. Assumed to be called
-  // protected.
-  bool CreateChannelObject(int channel_id,
-                           ViEEncoder* vie_encoder,
-                           RtcpBandwidthObserver* bandwidth_observer,
-                           RemoteBitrateEstimator* remote_bitrate_estimator,
-                           RtcpRttStats* rtcp_rtt_stats,
-                           RtcpIntraFrameObserver* intra_frame_observer,
-                           bool sender);
+  bool GetPacerQueuingDelayMs(int channel_id, int64_t* delay_ms) const;
 
+  bool SetBitrateConfig(int channel_id,
+                        int min_bitrate_bps,
+                        int start_bitrate_bps,
+                        int max_bitrate_bps);
+
+  bool ReAllocateBitrates(int channel_id);
+
+ private:
   // Used by ViEChannelScoped, forcing a manager user to use scoped.
   // Returns a pointer to the channel with id 'channel_id'.
   ViEChannel* ViEChannelPtr(int channel_id) const;
@@ -131,8 +125,6 @@ class ViEChannelManager: private ViEManagerBase {
   int engine_id_;
   int number_of_cores_;
 
-  // TODO(mflodman) Make part of channel group.
-  ChannelMap channel_map_;
   bool* free_channel_ids_;
   int free_channel_ids_size_;
 
@@ -140,13 +132,9 @@ class ViEChannelManager: private ViEManagerBase {
   std::list channel_groups_;
 
   // TODO(mflodman) Make part of channel group.
-  // Maps Channel id -> ViEEncoder.
-  EncoderMap vie_encoder_map_;
   VoEVideoSync* voice_sync_interface_;
 
-  VoiceEngine* voice_engine_;
   ProcessThread* module_process_thread_;
-  const Config& engine_config_;
   CPULoadStateCallbackInvoker* load_manager_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc
index 3b1bce5019..b4c1b8c9d4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.cc
@@ -189,22 +189,21 @@ int ViECodecImpl::SetSendCodec(const int video_channel,
     LOG(LS_INFO) << "New max bitrate set " << video_codec_internal.maxBitrate;
   }
 
-  if (video_codec_internal.startBitrate < video_codec_internal.minBitrate) {
-    video_codec_internal.startBitrate = video_codec_internal.minBitrate;
+  if (video_codec_internal.startBitrate > 0) {
+    if (video_codec_internal.startBitrate < video_codec_internal.minBitrate) {
+      video_codec_internal.startBitrate = video_codec_internal.minBitrate;
+    }
+    if (video_codec_internal.startBitrate > video_codec_internal.maxBitrate) {
+      video_codec_internal.startBitrate = video_codec_internal.maxBitrate;
+    }
   }
-  if (video_codec_internal.startBitrate > video_codec_internal.maxBitrate) {
-    video_codec_internal.startBitrate = video_codec_internal.maxBitrate;
-  }
-
-  VideoCodec encoder;
-  vie_encoder->GetEncoder(&encoder);
 
   // Make sure to generate a new SSRC if the codec type and/or resolution has
   // changed. This won't have any effect if the user has set an SSRC.
-  bool new_rtp_stream = false;
-  if (encoder.codecType != video_codec_internal.codecType) {
-    new_rtp_stream = true;
-  }
+  bool new_rtp_stream = true;
+  VideoCodec encoder;
+  if (vie_encoder->GetEncoder(&encoder) == 0)
+    new_rtp_stream = encoder.codecType != video_codec_internal.codecType;
 
   ViEInputManagerScoped is(*(shared_data_->input_manager()));
 
@@ -254,7 +253,8 @@ int ViECodecImpl::SetSendCodec(const int video_channel,
   shared_data_->channel_manager()->UpdateSsrcs(video_channel, ssrcs);
 
   // Update the protection mode, we might be switching NACK/FEC.
-  vie_encoder->UpdateProtectionMethod(vie_encoder->nack_enabled());
+  vie_encoder->UpdateProtectionMethod(vie_encoder->nack_enabled(),
+                                      vie_channel->IsSendingFecEnabled());
 
   // Get new best format for frame provider.
   ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
@@ -285,7 +285,7 @@ int ViECodecImpl::SetReceiveCodec(const int video_channel,
                                   const VideoCodec& video_codec) {
   LOG(LS_INFO) << "SetReceiveCodec for channel " << video_channel;
   LOG(LS_INFO) << "Codec type " << video_codec.codecType
-               << ", payload type " << video_codec.plType;
+               << ", payload type " << static_cast(video_codec.plType);
 
   if (CodecValid(video_codec) == false) {
     shared_data_->SetLastError(kViECodecInvalidCodec);
@@ -412,6 +412,12 @@ int ViECodecImpl::GetReceiveSideDelay(const int video_channel,
   return 0;
 }
 
+uint32_t ViECodecImpl::GetLastObservedBitrateBps(int video_channel) const {
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
+  assert(vie_encoder != nullptr);
+  return vie_encoder->LastObservedBitrateBps();
+}
 
 int ViECodecImpl::GetCodecTargetBitrate(const int video_channel,
                                         unsigned int* bitrate) const {
@@ -661,7 +667,8 @@ bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
   }
 
   if (video_codec.plType == 0 || video_codec.plType > 127) {
-    LOG(LS_ERROR) << "Invalif payload type: " << video_codec.plType;
+    LOG(LS_ERROR) << "Invalid payload type: "
+                  << static_cast(video_codec.plType);
     return false;
   }
 
@@ -672,7 +679,8 @@ bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
     return false;
   }
 
-  if (video_codec.startBitrate < kViEMinCodecBitrate) {
+  if (video_codec.startBitrate > 0 &&
+      video_codec.startBitrate < kViEMinCodecBitrate) {
     LOG(LS_ERROR) << "Invalid start bitrate.";
     return false;
   }
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h
index 93b9a29a17..51537b4685 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_codec_impl.h
@@ -51,6 +51,7 @@ class ViECodecImpl
                                         unsigned int& delta_frames) const;
   virtual int GetReceiveSideDelay(const int video_channel,
                                   int* delay_ms) const;
+  uint32_t GetLastObservedBitrateBps(int video_channel) const override;
   virtual int GetCodecTargetBitrate(const int video_channel,
                                     unsigned int* bitrate) const;
   virtual int GetNumDiscardedPackets(int video_channel) const;
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc b/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc
index ba089e6b61..84b68fc160 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_encoder.cc
@@ -14,10 +14,11 @@
 
 #include 
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_video/interface/video_image.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/frame_callback.h"
 #include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
@@ -29,9 +30,10 @@
 #include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
+#include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_engine/include/vie_image_process.h"
-#include "webrtc/frame_callback.h"
+#include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/vie_defines.h"
 
 namespace webrtc {
@@ -43,12 +45,6 @@ static const float kEncoderPausePacerMargin = 2.0f;
 // Don't stop the encoder unless the delay is above this configured value.
 static const int kMinPacingDelayMs = 200;
 
-// Allow packets to be transmitted in up to 2 times max video bitrate if the
-// bandwidth estimate allows it.
-// TODO(holmer): Expose transmission start, min and max bitrates in the
-// VideoEngine API and remove the kTransmissionMaxBitrateMultiplier.
-static const int kTransmissionMaxBitrateMultiplier = 2;
-
 static const float kStopPaddingThresholdMs = 2000;
 
 std::vector AllocateStreamBitrates(
@@ -95,35 +91,15 @@ class ViEBitrateObserver : public BitrateObserver {
   }
   virtual ~ViEBitrateObserver() {}
   // Implements BitrateObserver.
-  virtual void OnNetworkChanged(const uint32_t bitrate_bps,
-                                const uint8_t fraction_lost,
-                                const uint32_t rtt) {
+  virtual void OnNetworkChanged(uint32_t bitrate_bps,
+                                uint8_t fraction_lost,
+                                int64_t rtt) {
     owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
   }
  private:
   ViEEncoder* owner_;
 };
 
-class ViEPacedSenderCallback : public PacedSender::Callback {
- public:
-  explicit ViEPacedSenderCallback(ViEEncoder* owner)
-      : owner_(owner) {
-  }
-  virtual ~ViEPacedSenderCallback() {}
-  virtual bool TimeToSendPacket(uint32_t ssrc,
-                                uint16_t sequence_number,
-                                int64_t capture_time_ms,
-                                bool retransmission) {
-    return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
-                                    retransmission);
-  }
-  virtual int TimeToSendPadding(int bytes) {
-    return owner_->TimeToSendPadding(bytes);
-  }
- private:
-  ViEEncoder* owner_;
-};
-
 class ViECPULoadStateObserver : public CPULoadStateObserver {
  public:
   explicit ViECPULoadStateObserver(ViEEncoder* owner)
@@ -139,56 +115,51 @@ class ViECPULoadStateObserver : public CPULoadStateObserver {
   ViEEncoder* owner_;
 };
 
-ViEEncoder::ViEEncoder(int32_t engine_id,
-                       int32_t channel_id,
+ViEEncoder::ViEEncoder(int32_t channel_id,
                        uint32_t number_of_cores,
                        const Config& config,
                        ProcessThread& module_process_thread,
-                       BitrateController* bitrate_controller)
-  : engine_id_(engine_id),
-    channel_id_(channel_id),
-    number_of_cores_(number_of_cores),
-    vcm_(*webrtc::VideoCodingModule::Create()),
-    vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
-                                                            channel_id))),
-    callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-    data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
-    bitrate_controller_(bitrate_controller),
-    time_of_last_incoming_frame_ms_(0),
-    load_manager_(NULL),
-    send_padding_(false),
-    min_transmit_bitrate_kbps_(0),
-    target_delay_ms_(0),
-    network_is_transmitting_(true),
-    encoder_paused_(false),
-    encoder_paused_and_dropped_frame_(false),
-    fec_enabled_(false),
-    nack_enabled_(false),
-    codec_observer_(NULL),
-    effect_filter_(NULL),
-    module_process_thread_(module_process_thread),
-    has_received_sli_(false),
-    picture_id_sli_(0),
-    has_received_rpsi_(false),
-    picture_id_rpsi_(0),
-    qm_callback_(NULL),
-    video_suspended_(false),
-    pre_encode_callback_(NULL),
-    start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
-  RtpRtcp::Configuration configuration;
-  configuration.id = ViEModuleId(engine_id_, channel_id_);
-  configuration.audio = false;  // Video.
-
-  default_rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
+                       PacedSender* pacer,
+                       BitrateAllocator* bitrate_allocator,
+                       BitrateController* bitrate_controller,
+                       bool disable_default_encoder)
+    : channel_id_(channel_id),
+      number_of_cores_(number_of_cores),
+      disable_default_encoder_(disable_default_encoder),
+      vcm_(*webrtc::VideoCodingModule::Create(this)),
+      vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(-1, channel_id))),
+      send_payload_router_(NULL),
+      vcm_protection_callback_(NULL),
+      callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
+      pacer_(pacer),
+      bitrate_allocator_(bitrate_allocator),
+      bitrate_controller_(bitrate_controller),
+      time_of_last_incoming_frame_ms_(0),
+      load_manager_(NULL),
+      send_padding_(false),
+      min_transmit_bitrate_kbps_(0),
+      last_observed_bitrate_bps_(0),
+      target_delay_ms_(0),
+      network_is_transmitting_(true),
+      encoder_paused_(false),
+      encoder_paused_and_dropped_frame_(false),
+      fec_enabled_(false),
+      nack_enabled_(false),
+      codec_observer_(NULL),
+      effect_filter_(NULL),
+      module_process_thread_(module_process_thread),
+      has_received_sli_(false),
+      picture_id_sli_(0),
+      has_received_rpsi_(false),
+      picture_id_rpsi_(0),
+      qm_callback_(NULL),
+      video_suspended_(false),
+      pre_encode_callback_(NULL),
+      start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()),
+      send_statistics_proxy_(NULL) {
   bitrate_observer_.reset(new ViEBitrateObserver(this));
-  pacing_callback_.reset(new ViEPacedSenderCallback(this));
   loadstate_observer_.reset(new ViECPULoadStateObserver(this));
-  paced_sender_.reset(new PacedSender(
-      Clock::GetRealTimeClock(),
-      pacing_callback_.get(),
-      kDefaultStartBitrateKbps,
-      PacedSender::kDefaultPaceMultiplier * kDefaultStartBitrateKbps,
-      0));
 }
 
 bool ViEEncoder::Init() {
@@ -200,36 +171,29 @@ bool ViEEncoder::Init() {
   // Enable content analysis if load management enabled
   vpm_.EnableContentAnalysis(load_manager_ != NULL);
 
-  if (module_process_thread_.RegisterModule(&vcm_) != 0 ||
-      module_process_thread_.RegisterModule(default_rtp_rtcp_.get()) != 0 ||
-      module_process_thread_.RegisterModule(paced_sender_.get()) != 0) {
-    return false;
-  }
   if (qm_callback_) {
     delete qm_callback_;
   }
   qm_callback_ = new QMVideoSettingsCallback(&vpm_);
 
+  if (!disable_default_encoder_) {
 #ifdef VIDEOCODEC_VP8
-  VideoCodecType codec_type = webrtc::kVideoCodecVP8;
+    VideoCodecType codec_type = webrtc::kVideoCodecVP8;
 #else
-  VideoCodecType codec_type = webrtc::kVideoCodecI420;
+    VideoCodecType codec_type = webrtc::kVideoCodecI420;
 #endif
-
-  VideoCodec video_codec;
-  if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
-    return false;
-  }
-  {
-    CriticalSectionScoped cs(data_cs_.get());
-    send_padding_ = video_codec.numberOfSimulcastStreams > 1;
-  }
-  if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
-                             default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
-    return false;
-  }
-  if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
-    return false;
+    VideoCodec video_codec;
+    if (vcm_.Codec(codec_type, &video_codec) != VCM_OK) {
+      return false;
+    }
+    {
+      CriticalSectionScoped cs(data_cs_.get());
+      send_padding_ = video_codec.numberOfSimulcastStreams > 1;
+    }
+    if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
+                               PayloadRouter::DefaultMaxPayloadLength()) != 0) {
+      return false;
+    }
   }
   if (vcm_.RegisterTransportCallback(this) != 0) {
     return false;
@@ -243,6 +207,25 @@ bool ViEEncoder::Init() {
   return true;
 }
 
+void ViEEncoder::StartThreadsAndSetSharedMembers(
+    scoped_refptr send_payload_router,
+    VCMProtectionCallback* vcm_protection_callback) {
+  DCHECK(send_payload_router_ == NULL);
+  DCHECK(vcm_protection_callback_ == NULL);
+
+  send_payload_router_ = send_payload_router;
+  vcm_protection_callback_ = vcm_protection_callback;
+
+  module_process_thread_.RegisterModule(&vcm_);
+}
+
+void ViEEncoder::StopThreadsAndRemoveSharedMembers() {
+  vcm_.RegisterProtectionCallback(NULL);
+  vcm_protection_callback_ = NULL;
+  module_process_thread_.DeRegisterModule(&vcm_);
+  module_process_thread_.DeRegisterModule(&vpm_);
+}
+
 void ViEEncoder::SetLoadManager(CPULoadStateCallbackInvoker* load_manager) {
   load_manager_ = load_manager;
   if (load_manager_) {
@@ -253,26 +236,20 @@ void ViEEncoder::SetLoadManager(CPULoadStateCallbackInvoker* load_manager) {
 
 ViEEncoder::~ViEEncoder() {
   UpdateHistograms();
-  if (bitrate_controller_) {
-    bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
-  }
+  if (bitrate_allocator_)
+    bitrate_allocator_->RemoveBitrateObserver(bitrate_observer_.get());
   if (load_manager_) {
     load_manager_->RemoveObserver(loadstate_observer_.get());
   }
-  module_process_thread_.DeRegisterModule(&vcm_);
-  module_process_thread_.DeRegisterModule(&vpm_);
-  module_process_thread_.DeRegisterModule(default_rtp_rtcp_.get());
-  module_process_thread_.DeRegisterModule(paced_sender_.get());
   VideoCodingModule::Destroy(&vcm_);
   VideoProcessingModule::Destroy(&vpm_);
   delete qm_callback_;
 }
 
 void ViEEncoder::UpdateHistograms() {
-  const float kMinCallLengthInMinutes = 0.5f;
-  float elapsed_minutes =
-      (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
-  if (elapsed_minutes < kMinCallLengthInMinutes) {
+  int64_t elapsed_sec =
+      (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
     return;
   }
   webrtc::VCMFrameCount frames;
@@ -297,9 +274,9 @@ void ViEEncoder::SetNetworkTransmissionState(bool is_transmitting) {
     network_is_transmitting_ = is_transmitting;
   }
   if (is_transmitting) {
-    paced_sender_->Resume();
+    pacer_->Resume();
   } else {
-    paced_sender_->Pause();
+    pacer_->Pause();
   }
 }
 
@@ -338,6 +315,7 @@ int32_t ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
 }
 
 int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
+  DCHECK(send_payload_router_ != NULL);
   webrtc::VideoCodec current_send_codec;
   if (vcm_.SendCodec(¤t_send_codec) == VCM_OK) {
     uint32_t current_bitrate_bps = 0;
@@ -351,11 +329,12 @@ int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
     return -1;
   }
 
+  if (disable_default_encoder_)
+    return 0;
+
   // If the external encoder is the current send codec, use vcm internal
   // encoder.
   if (current_send_codec.plType == pl_type) {
-    uint16_t max_data_payload_length =
-        default_rtp_rtcp_->MaxDataPayloadLength();
     {
       CriticalSectionScoped cs(data_cs_.get());
       send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
@@ -365,6 +344,7 @@ int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
     // a hack to prevent the following code from crashing.  This should be fixed
     // for realz.  https://code.google.com/p/chromium/issues/detail?id=348222
     current_send_codec.extra_options = NULL;
+    size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
     if (vcm_.RegisterSendCodec(¤t_send_codec, number_of_cores_,
                                max_data_payload_length) != VCM_OK) {
       LOG(LS_INFO) << "De-registered the currently used external encoder ("
@@ -377,63 +357,58 @@ int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
 }
 
 int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
+  DCHECK(send_payload_router_ != NULL);
   // Setting target width and height for VPM.
   if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
                                video_codec.maxFramerate) != VPM_OK) {
     return -1;
   }
 
-  if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
-    return -1;
-  }
-  // Convert from kbps to bps.
-  std::vector stream_bitrates = AllocateStreamBitrates(
-      video_codec.startBitrate * 1000,
-      video_codec.simulcastStream,
-      video_codec.numberOfSimulcastStreams);
-  default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
-
-  uint16_t max_data_payload_length =
-      default_rtp_rtcp_->MaxDataPayloadLength();
-
   {
     CriticalSectionScoped cs(data_cs_.get());
     send_padding_ = video_codec.numberOfSimulcastStreams > 1;
   }
-  if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
+
+  // Add a bitrate observer to the allocator and update the start, max and
+  // min bitrates of the bitrate controller as needed.
+  int allocated_bitrate_bps;
+  int new_bwe_candidate_bps = bitrate_allocator_->AddBitrateObserver(
+      bitrate_observer_.get(), video_codec.startBitrate * 1000,
+      video_codec.minBitrate * 1000, video_codec.maxBitrate * 1000,
+      &allocated_bitrate_bps);
+
+  // Only set the start/min/max bitrate of the bitrate controller if the start
+  // bitrate is greater than zero. The new API sets these via the channel group
+  // and passes a zero start bitrate to SetSendCodec.
+  // TODO(holmer): Remove this when the new API has been launched.
+  if (video_codec.startBitrate > 0) {
+    if (new_bwe_candidate_bps > 0) {
+      uint32_t current_bwe_bps = 0;
+      bitrate_controller_->AvailableBandwidth(¤t_bwe_bps);
+      bitrate_controller_->SetStartBitrate(std::max(
+          static_cast(new_bwe_candidate_bps), current_bwe_bps));
+    }
+
+    int new_bwe_min_bps = 0;
+    int new_bwe_max_bps = 0;
+    bitrate_allocator_->GetMinMaxBitrateSumBps(&new_bwe_min_bps,
+                                               &new_bwe_max_bps);
+    bitrate_controller_->SetMinMaxBitrate(new_bwe_min_bps, new_bwe_max_bps);
+  }
+
+  webrtc::VideoCodec modified_video_codec = video_codec;
+  modified_video_codec.startBitrate = allocated_bitrate_bps / 1000;
+
+  size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
+  if (vcm_.RegisterSendCodec(&modified_video_codec, number_of_cores_,
                              max_data_payload_length) != VCM_OK) {
     return -1;
   }
-
-  // Set this module as sending right away, let the slave module in the channel
-  // start and stop sending.
-  if (default_rtp_rtcp_->SetSendingStatus(true) != 0) {
-    return -1;
-  }
-
-  bitrate_controller_->SetBitrateObserver(bitrate_observer_.get(),
-                                          video_codec.startBitrate * 1000,
-                                          video_codec.minBitrate * 1000,
-                                          kTransmissionMaxBitrateMultiplier *
-                                          video_codec.maxBitrate * 1000);
-
-  CriticalSectionScoped crit(data_cs_.get());
-  int pad_up_to_bitrate_kbps = video_codec.startBitrate;
-  if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
-    pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
-
-  paced_sender_->UpdateBitrate(
-      video_codec.startBitrate,
-      PacedSender::kDefaultPaceMultiplier * video_codec.startBitrate,
-      pad_up_to_bitrate_kbps);
-
   return 0;
 }
 
 int32_t ViEEncoder::GetEncoder(VideoCodec* video_codec) {
-  if (vcm_.SendCodec(video_codec) != 0) {
-    return -1;
-  }
+  *video_codec = vcm_.GetSendCodec();
   return 0;
 }
 
@@ -463,25 +438,64 @@ int32_t ViEEncoder::ScaleInputImage(bool enable) {
   return 0;
 }
 
-bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
-                                  uint16_t sequence_number,
-                                  int64_t capture_time_ms,
-                                  bool retransmission) {
-  return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
-                                             capture_time_ms, retransmission);
-}
-
-int ViEEncoder::TimeToSendPadding(int bytes) {
-  bool send_padding;
+int ViEEncoder::GetPaddingNeededBps(int bitrate_bps) const {
+  int64_t time_of_last_incoming_frame_ms;
+  int min_transmit_bitrate_bps;
   {
     CriticalSectionScoped cs(data_cs_.get());
-    send_padding =
+    bool send_padding =
         send_padding_ || video_suspended_ || min_transmit_bitrate_kbps_ > 0;
+    if (!send_padding)
+      return 0;
+    time_of_last_incoming_frame_ms = time_of_last_incoming_frame_ms_;
+    min_transmit_bitrate_bps = 1000 * min_transmit_bitrate_kbps_;
   }
-  if (send_padding) {
-    return default_rtp_rtcp_->TimeToSendPadding(bytes);
+
+  VideoCodec send_codec;
+  if (vcm_.SendCodec(&send_codec) != 0)
+    return 0;
+  SimulcastStream* stream_configs = send_codec.simulcastStream;
+  // Allocate the bandwidth between the streams.
+  std::vector stream_bitrates = AllocateStreamBitrates(
+      bitrate_bps, stream_configs, send_codec.numberOfSimulcastStreams);
+
+  bool video_is_suspended = vcm_.VideoSuspended();
+
+  // Find the max amount of padding we can allow ourselves to send at this
+  // point, based on which streams are currently active and what our current
+  // available bandwidth is.
+  int pad_up_to_bitrate_bps = 0;
+  if (send_codec.numberOfSimulcastStreams == 0) {
+    pad_up_to_bitrate_bps = send_codec.minBitrate * 1000;
+  } else {
+    pad_up_to_bitrate_bps =
+        stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate *
+        1000;
+    for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
+      pad_up_to_bitrate_bps += stream_configs[i].targetBitrate * 1000;
+    }
   }
-  return 0;
+
+  // Disable padding if only sending one stream and video isn't suspended and
+  // min-transmit bitrate isn't used (applied later).
+  if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
+    pad_up_to_bitrate_bps = 0;
+
+  // The amount of padding should decay to zero if no frames are being
+  // captured unless a min-transmit bitrate is used.
+  int64_t now_ms = TickTime::MillisecondTimestamp();
+  if (now_ms - time_of_last_incoming_frame_ms > kStopPaddingThresholdMs)
+    pad_up_to_bitrate_bps = 0;
+
+  // Pad up to min bitrate.
+  if (pad_up_to_bitrate_bps < min_transmit_bitrate_bps)
+    pad_up_to_bitrate_bps = min_transmit_bitrate_bps;
+
+  // Padding may never exceed bitrate estimate.
+  if (pad_up_to_bitrate_bps > bitrate_bps)
+    pad_up_to_bitrate_bps = bitrate_bps;
+
+  return pad_up_to_bitrate_bps;
 }
 
 bool ViEEncoder::EncoderPaused() const {
@@ -494,12 +508,12 @@ bool ViEEncoder::EncoderPaused() const {
     // Buffered mode.
     // TODO(pwestin): Workaround until nack is configured as a time and not
     // number of packets.
-    return paced_sender_->QueueInMs() >=
-        std::max(static_cast(target_delay_ms_ * kEncoderPausePacerMargin),
-                 kMinPacingDelayMs);
+    return pacer_->QueueInMs() >=
+           std::max(
+               static_cast(target_delay_ms_ * kEncoderPausePacerMargin),
+               kMinPacingDelayMs);
   }
-  if (paced_sender_->ExpectedQueueTimeMs() >
-      PacedSender::kDefaultMaxQueueLengthMs) {
+  if (pacer_->ExpectedQueueTimeMs() > PacedSender::kDefaultMaxQueueLengthMs) {
     // Too much data in pacer queue, drop frame.
     return true;
   }
@@ -523,16 +537,14 @@ void ViEEncoder::TraceFrameDropEnd() {
   encoder_paused_and_dropped_frame_ = false;
 }
 
-RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
-  return default_rtp_rtcp_.get();
-}
-
 void ViEEncoder::DeliverFrame(int id,
                               I420VideoFrame* video_frame,
-                              int num_csrcs,
-                              const uint32_t CSRC[kRtpCsrcSize]) {
-  if (default_rtp_rtcp_->SendingMedia() == false) {
-    // We've paused or we have no channels attached, don't encode.
+                              const std::vector& csrcs) {
+  DCHECK(send_payload_router_ != NULL);
+  DCHECK(csrcs.empty());
+  if (!send_payload_router_->active()) {
+    // We've paused or we have no channels attached, don't waste resources on
+    // encoding.
     return;
   }
   {
@@ -545,38 +557,17 @@ void ViEEncoder::DeliverFrame(int id,
     TraceFrameDropEnd();
   }
 
-  // Convert render time, in ms, to RTP timestamp.
-  const int kMsToRtpTimestamp = 90;
-  const uint32_t time_stamp =
-      kMsToRtpTimestamp *
-      static_cast(video_frame->render_time_ms());
-
   TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame->render_time_ms(),
                           "Encode");
-  video_frame->set_timestamp(time_stamp);
-
-  // Make sure the CSRC list is correct.
-  if (num_csrcs > 0) {
-    uint32_t tempCSRC[kRtpCsrcSize];
-    for (int i = 0; i < num_csrcs; i++) {
-      if (CSRC[i] == 1) {
-        tempCSRC[i] = default_rtp_rtcp_->SSRC();
-      } else {
-        tempCSRC[i] = CSRC[i];
-      }
-    }
-    default_rtp_rtcp_->SetCSRCs(tempCSRC, (uint8_t) num_csrcs);
-  }
-
   I420VideoFrame* decimated_frame = NULL;
   // TODO(wuchengli): support texture frames.
   if (video_frame->native_handle() == NULL) {
     {
       CriticalSectionScoped cs(callback_cs_.get());
       if (effect_filter_) {
-        unsigned int length =
+        size_t length =
             CalcBufferSize(kI420, video_frame->width(), video_frame->height());
-        scoped_ptr video_buffer(new uint8_t[length]);
+        rtc::scoped_ptr video_buffer(new uint8_t[length]);
         ExtractBuffer(*video_frame, length, video_buffer.get());
         effect_filter_->Transform(length,
                                   video_buffer.get(),
@@ -654,7 +645,6 @@ void ViEEncoder::DeliverFrame(int id,
 }
 
 void ViEEncoder::DelayChanged(int id, int frame_delay) {
-  default_rtp_rtcp_->SetCameraDelay(frame_delay);
 }
 
 int ViEEncoder::GetPreferedFrameSettings(int* width,
@@ -687,8 +677,9 @@ int32_t ViEEncoder::SendCodecStatistics(
   return 0;
 }
 
-int32_t ViEEncoder::PacerQueuingDelayMs() const {
-  return paced_sender_->QueueInMs();
+uint32_t ViEEncoder::LastObservedBitrateBps() const {
+  CriticalSectionScoped cs(data_cs_.get());
+  return last_observed_bitrate_bps_;
 }
 
 int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
@@ -697,28 +688,19 @@ int ViEEncoder::CodecTargetBitrate(uint32_t* bitrate) const {
   return 0;
 }
 
-int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
-  bool fec_enabled = false;
-  uint8_t dummy_ptype_red = 0;
-  uint8_t dummy_ptypeFEC = 0;
+int32_t ViEEncoder::UpdateProtectionMethod(bool nack, bool fec) {
+  DCHECK(send_payload_router_ != NULL);
+  DCHECK(vcm_protection_callback_ != NULL);
 
-  // Updated protection method to VCM to get correct packetization sizes.
-  // FEC has larger overhead than NACK -> set FEC if used.
-  int32_t error = default_rtp_rtcp_->GenericFECStatus(fec_enabled,
-                                                      dummy_ptype_red,
-                                                      dummy_ptypeFEC);
-  if (error) {
-    return -1;
-  }
-  if (fec_enabled_ == fec_enabled && nack_enabled_ == enable_nack) {
+  if (fec_enabled_ == fec && nack_enabled_ == nack) {
     // No change needed, we're already in correct state.
     return 0;
   }
-  fec_enabled_ = fec_enabled;
-  nack_enabled_ = enable_nack;
+  fec_enabled_ = fec;
+  nack_enabled_ = nack;
 
   // Set Video Protection for VCM.
-  if (fec_enabled && nack_enabled_) {
+  if (fec_enabled_ && nack_enabled_) {
     vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
   } else {
     vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
@@ -727,11 +709,10 @@ int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
   }
 
   if (fec_enabled_ || nack_enabled_) {
-    vcm_.RegisterProtectionCallback(this);
+    vcm_.RegisterProtectionCallback(vcm_protection_callback_);
     // The send codec must be registered to set correct MTU.
     webrtc::VideoCodec codec;
     if (vcm_.SendCodec(&codec) == 0) {
-      uint16_t max_pay_load = default_rtp_rtcp_->MaxDataPayloadLength();
       uint32_t current_bitrate_bps = 0;
       if (vcm_.Bitrate(¤t_bitrate_bps) != 0) {
         LOG_F(LS_WARNING) <<
@@ -739,7 +720,9 @@ int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
       }
       // Convert to start bitrate in kbps.
       codec.startBitrate = (current_bitrate_bps + 500) / 1000;
-      if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
+      size_t max_payload_length = send_payload_router_->MaxPayloadLength();
+      if (vcm_.RegisterSendCodec(&codec, number_of_cores_,
+                                 max_payload_length) != 0) {
         return -1;
       }
     }
@@ -767,36 +750,30 @@ void ViEEncoder::SetSenderBufferingMode(int target_delay_ms) {
   }
 }
 
-int32_t ViEEncoder::SendData(
-    const FrameType frame_type,
-    const uint8_t payload_type,
-    const uint32_t time_stamp,
-    int64_t capture_time_ms,
-    const uint8_t* payload_data,
-    const uint32_t payload_size,
-    const webrtc::RTPFragmentationHeader& fragmentation_header,
-    const RTPVideoHeader* rtp_video_hdr) {
-  // New encoded data, hand over to the rtp module.
-  return default_rtp_rtcp_->SendOutgoingData(frame_type,
-                                             payload_type,
-                                             time_stamp,
-                                             capture_time_ms,
-                                             payload_data,
-                                             payload_size,
-                                             &fragmentation_header,
-                                             rtp_video_hdr);
+void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) {
+  CriticalSectionScoped cs(callback_cs_.get());
+  if (send_statistics_proxy_ != nullptr)
+    send_statistics_proxy_->OnSetRates(bitrate_bps, framerate);
 }
 
-int32_t ViEEncoder::ProtectionRequest(
-    const FecProtectionParams* delta_fec_params,
-    const FecProtectionParams* key_fec_params,
-    uint32_t* sent_video_rate_bps,
-    uint32_t* sent_nack_rate_bps,
-    uint32_t* sent_fec_rate_bps) {
-  default_rtp_rtcp_->SetFecParameters(delta_fec_params, key_fec_params);
-  default_rtp_rtcp_->BitrateSent(NULL, sent_video_rate_bps, sent_fec_rate_bps,
-                                sent_nack_rate_bps);
-  return 0;
+int32_t ViEEncoder::SendData(
+    const uint8_t payload_type,
+    const EncodedImage& encoded_image,
+    const webrtc::RTPFragmentationHeader& fragmentation_header,
+    const RTPVideoHeader* rtp_video_hdr) {
+  DCHECK(send_payload_router_ != NULL);
+
+  {
+    CriticalSectionScoped cs(callback_cs_.get());
+    if (send_statistics_proxy_ != NULL)
+      send_statistics_proxy_->OnSendEncodedImage(encoded_image, rtp_video_hdr);
+  }
+
+  return send_payload_router_->RoutePayload(
+      VCMEncodedFrame::ConvertFrameType(encoded_image._frameType), payload_type,
+      encoded_image._timeStamp, encoded_image.capture_time_ms_,
+      encoded_image._buffer, encoded_image._length, &fragmentation_header,
+      rtp_video_hdr) ? 0 : -1;
 }
 
 int32_t ViEEncoder::SendStatistics(const uint32_t bit_rate,
@@ -911,15 +888,16 @@ void ViEEncoder::SetMinTransmitBitrate(int min_transmit_bitrate_kbps) {
 }
 
 // Called from ViEBitrateObserver.
-void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
-                                  const uint8_t fraction_lost,
-                                  const uint32_t round_trip_time_ms) {
+void ViEEncoder::OnNetworkChanged(uint32_t bitrate_bps,
+                                  uint8_t fraction_lost,
+                                  int64_t round_trip_time_ms) {
   LOG(LS_VERBOSE) << "OnNetworkChanged, bitrate" << bitrate_bps
                   << " packet loss " << fraction_lost
                   << " rtt " << round_trip_time_ms;
+  DCHECK(send_payload_router_ != NULL);
   vcm_.SetChannelParameters(bitrate_bps, fraction_lost, round_trip_time_ms);
   bool video_is_suspended = vcm_.VideoSuspended();
-  int bitrate_kbps = bitrate_bps / 1000;
+
   VideoCodec send_codec;
   if (vcm_.SendCodec(&send_codec) != 0) {
     return;
@@ -927,54 +905,16 @@ void ViEEncoder::OnNetworkChanged(const uint32_t bitrate_bps,
   SimulcastStream* stream_configs = send_codec.simulcastStream;
   // Allocate the bandwidth between the streams.
   std::vector stream_bitrates = AllocateStreamBitrates(
-      bitrate_bps,
-      stream_configs,
-      send_codec.numberOfSimulcastStreams);
-  // Find the max amount of padding we can allow ourselves to send at this
-  // point, based on which streams are currently active and what our current
-  // available bandwidth is.
-  int pad_up_to_bitrate_kbps = 0;
-  if (send_codec.numberOfSimulcastStreams == 0) {
-    pad_up_to_bitrate_kbps = send_codec.minBitrate;
-  } else {
-    pad_up_to_bitrate_kbps =
-        stream_configs[send_codec.numberOfSimulcastStreams - 1].minBitrate;
-    for (int i = 0; i < send_codec.numberOfSimulcastStreams - 1; ++i) {
-      pad_up_to_bitrate_kbps += stream_configs[i].targetBitrate;
-    }
-  }
-
-  // Disable padding if only sending one stream and video isn't suspended and
-  // min-transmit bitrate isn't used (applied later).
-  if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
-    pad_up_to_bitrate_kbps = 0;
+      bitrate_bps, stream_configs, send_codec.numberOfSimulcastStreams);
+  send_payload_router_->SetTargetSendBitrates(stream_bitrates);
 
   {
     CriticalSectionScoped cs(data_cs_.get());
-    // The amount of padding should decay to zero if no frames are being
-    // captured unless a min-transmit bitrate is used.
-    int64_t now_ms = TickTime::MillisecondTimestamp();
-    if (now_ms - time_of_last_incoming_frame_ms_ > kStopPaddingThresholdMs)
-      pad_up_to_bitrate_kbps = 0;
-
-    // Pad up to min bitrate.
-    if (pad_up_to_bitrate_kbps < min_transmit_bitrate_kbps_)
-      pad_up_to_bitrate_kbps = min_transmit_bitrate_kbps_;
-
-    // Padding may never exceed bitrate estimate.
-    if (pad_up_to_bitrate_kbps > bitrate_kbps)
-      pad_up_to_bitrate_kbps = bitrate_kbps;
-
-    paced_sender_->UpdateBitrate(
-        bitrate_kbps,
-        PacedSender::kDefaultPaceMultiplier * bitrate_kbps,
-        pad_up_to_bitrate_kbps);
-    default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
+    last_observed_bitrate_bps_ = bitrate_bps;
     if (video_suspended_ == video_is_suspended)
       return;
     video_suspended_ = video_is_suspended;
   }
-
   // Video suspend-state changed, inform codec observer.
   CriticalSectionScoped crit(callback_cs_.get());
   if (codec_observer_) {
@@ -989,10 +929,6 @@ void ViEEncoder::onLoadStateChanged(CPULoadState load_state) {
   vcm_.SetCPULoadState(load_state);
 }
 
-PacedSender* ViEEncoder::GetPacedSender() {
-  return paced_sender_.get();
-}
-
 int32_t ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
   CriticalSectionScoped cs(callback_cs_.get());
   if (effect_filter != NULL && effect_filter_ != NULL) {
@@ -1013,7 +949,7 @@ int ViEEncoder::StopDebugRecording() {
 
 void ViEEncoder::SuspendBelowMinBitrate() {
   vcm_.SuspendBelowMinBitrate();
-  bitrate_controller_->EnforceMinBitrate(false);
+  bitrate_allocator_->EnforceMinBitrate(false);
 }
 
 void ViEEncoder::RegisterPreEncodeCallback(
@@ -1036,6 +972,12 @@ void ViEEncoder::DeRegisterPostEncodeImageCallback() {
   vcm_.RegisterPostEncodeImageCallback(NULL);
 }
 
+void ViEEncoder::RegisterSendStatisticsProxy(
+    SendStatisticsProxy* send_statistics_proxy) {
+  CriticalSectionScoped cs(callback_cs_.get());
+  send_statistics_proxy_ = send_statistics_proxy;
+}
+
 QMVideoSettingsCallback::QMVideoSettingsCallback(VideoProcessingModule* vpm)
     : vpm_(vpm) {
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.h b/media/webrtc/trunk/webrtc/video_engine/vie_encoder.h
index 5f6c7f3f30..44bfa4968f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_encoder.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_encoder.h
@@ -13,16 +13,19 @@
 
 #include 
 #include 
+#include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/modules/video_processing/main/interface/video_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/frame_callback.h"
+#include "webrtc/system_wrappers/interface/scoped_refptr.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/video_engine/vie_frame_provider_base.h"
 
@@ -32,37 +35,50 @@ class Config;
 class CriticalSectionWrapper;
 class EncodedImageCallback;
 class PacedSender;
+class PayloadRouter;
 class ProcessThread;
 class QMVideoSettingsCallback;
-class RtpRtcp;
+class SendStatisticsProxy;
 class ViEBitrateObserver;
 class ViEEffectFilter;
 class ViEEncoderObserver;
 class VideoCodingModule;
-class ViEPacedSenderCallback;
 class ViECPULoadStateObserver;
 
 class ViEEncoder
     : public RtcpIntraFrameObserver,
+      public VideoEncoderRateObserver,
       public VCMPacketizationCallback,
-      public VCMProtectionCallback,
       public VCMSendStatisticsCallback,
       public ViEFrameCallback {
  public:
   friend class ViEBitrateObserver;
-  friend class ViEPacedSenderCallback;
   friend class ViECPULoadStateObserver;
 
-  ViEEncoder(int32_t engine_id,
-             int32_t channel_id,
+  ViEEncoder(int32_t channel_id,
              uint32_t number_of_cores,
              const Config& config,
              ProcessThread& module_process_thread,
-             BitrateController* bitrate_controller);
+             PacedSender* pacer,
+             BitrateAllocator* bitrate_allocator,
+             BitrateController* bitrate_controller,
+             bool disable_default_encoder);
   ~ViEEncoder();
 
   bool Init();
 
+  // This function is assumed to be called before any frames are delivered and
+  // only once.
+  // Ideally this would be done in Init, but the dependencies between ViEEncoder
+  // and ViEChannel makes it really hard to do in a good way.
+  void StartThreadsAndSetSharedMembers(
+      scoped_refptr send_payload_router,
+      VCMProtectionCallback* vcm_protection_callback);
+
+  // This function must be called before the corresponding ViEChannel is
+  // deleted.
+  void StopThreadsAndRemoveSharedMembers();
+
   void SetNetworkTransmissionState(bool is_transmitting);
 
   // Returns the id of the owning channel.
@@ -86,73 +102,53 @@ class ViEEncoder
     unsigned char config_parameters[kConfigParameterSize],
     unsigned char& config_parameters_size);
 
-  PacedSender* GetPacedSender();
-
   // Scale or crop/pad image.
   int32_t ScaleInputImage(bool enable);
 
-  // RTP settings.
-  RtpRtcp* SendRtpRtcpModule();
-
   // Implementing ViEFrameCallback.
-  virtual void DeliverFrame(int id,
-                            I420VideoFrame* video_frame,
-                            int num_csrcs = 0,
-                            const uint32_t CSRC[kRtpCsrcSize] = NULL) OVERRIDE;
-  virtual void DelayChanged(int id, int frame_delay) OVERRIDE;
-  virtual int GetPreferedFrameSettings(int* width,
-                                       int* height,
-                                       int* frame_rate) OVERRIDE;
+  void DeliverFrame(int id,
+                    I420VideoFrame* video_frame,
+                    const std::vector& csrcs) override;
+  void DelayChanged(int id, int frame_delay) override;
+  int GetPreferedFrameSettings(int* width,
+                               int* height,
+                               int* frame_rate) override;
 
-  virtual void ProviderDestroyed(int id) OVERRIDE {
-    return;
-  }
+  void ProviderDestroyed(int id) override { return; }
 
   int32_t SendKeyFrame();
   int32_t SendCodecStatistics(uint32_t* num_key_frames,
                               uint32_t* num_delta_frames);
 
-  int PacerQueuingDelayMs() const;
-
+  uint32_t LastObservedBitrateBps() const;
   int CodecTargetBitrate(uint32_t* bitrate) const;
   // Loss protection.
-  int32_t UpdateProtectionMethod(bool enable_nack);
+  int32_t UpdateProtectionMethod(bool nack, bool fec);
   bool nack_enabled() const { return nack_enabled_; }
 
   // Buffering mode.
   void SetSenderBufferingMode(int target_delay_ms);
 
-  // Implements VCMPacketizationCallback.
-  virtual int32_t SendData(
-    FrameType frame_type,
-    uint8_t payload_type,
-    uint32_t time_stamp,
-    int64_t capture_time_ms,
-    const uint8_t* payload_data,
-    uint32_t payload_size,
-    const RTPFragmentationHeader& fragmentation_header,
-    const RTPVideoHeader* rtp_video_hdr) OVERRIDE;
+  // Implements VideoEncoderRateObserver.
+  void OnSetRates(uint32_t bitrate_bps, int framerate) override;
 
-  // Implements VideoProtectionCallback.
-  virtual int ProtectionRequest(
-      const FecProtectionParams* delta_fec_params,
-      const FecProtectionParams* key_fec_params,
-      uint32_t* sent_video_rate_bps,
-      uint32_t* sent_nack_rate_bps,
-      uint32_t* sent_fec_rate_bps) OVERRIDE;
+  // Implements VCMPacketizationCallback.
+  int32_t SendData(uint8_t payload_type,
+                   const EncodedImage& encoded_image,
+                   const RTPFragmentationHeader& fragmentation_header,
+                   const RTPVideoHeader* rtp_video_hdr) override;
 
   // Implements VideoSendStatisticsCallback.
-  virtual int32_t SendStatistics(const uint32_t bit_rate,
-                                 const uint32_t frame_rate) OVERRIDE;
+  int32_t SendStatistics(const uint32_t bit_rate,
+                         const uint32_t frame_rate) override;
 
   int32_t RegisterCodecObserver(ViEEncoderObserver* observer);
 
   // Implements RtcpIntraFrameObserver.
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) OVERRIDE;
-  virtual void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) OVERRIDE;
-  virtual void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) OVERRIDE;
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc,
-                                  uint32_t new_ssrc) OVERRIDE;
+  void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
+  void OnReceivedSLI(uint32_t ssrc, uint8_t picture_id) override;
+  void OnReceivedRPSI(uint32_t ssrc, uint64_t picture_id) override;
+  void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) override;
 
   // Sets SSRCs for all streams.
   bool SetSsrcs(const std::list& ssrcs);
@@ -183,21 +179,21 @@ class ViEEncoder
         EncodedImageCallback* post_encode_callback);
   void DeRegisterPostEncodeImageCallback();
 
+  void RegisterSendStatisticsProxy(SendStatisticsProxy* send_statistics_proxy);
+
   int channel_id() const { return channel_id_; }
 
+  int GetPaddingNeededBps(int bitrate_bps) const;
+
  protected:
   // Called by BitrateObserver.
-  void OnNetworkChanged(const uint32_t bitrate_bps,
-                        const uint8_t fraction_lost,
-                        const uint32_t round_trip_time_ms);
+  void OnNetworkChanged(uint32_t bitrate_bps,
+                        uint8_t fraction_lost,
+                        int64_t round_trip_time_ms);
 
   // Called by CPULoadStateObserver
   void onLoadStateChanged(CPULoadState load_state);
 
-  // Called by PacedSender.
-  bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
-                        int64_t capture_time_ms, bool retransmission);
-  int TimeToSendPadding(int bytes);
  private:
   bool EncoderPaused() const EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
   void TraceFrameDropStart() EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
@@ -205,27 +201,30 @@ class ViEEncoder
 
   void UpdateHistograms();
 
-  int32_t engine_id_;
   const int channel_id_;
   const uint32_t number_of_cores_;
+  const bool disable_default_encoder_;
 
   VideoCodingModule& vcm_;
   VideoProcessingModule& vpm_;
-  scoped_ptr default_rtp_rtcp_;
-  scoped_ptr callback_cs_;
-  scoped_ptr data_cs_;
-  scoped_ptr bitrate_observer_;
-  scoped_ptr paced_sender_;
-  scoped_ptr loadstate_observer_;
-  scoped_ptr pacing_callback_;
+  scoped_refptr send_payload_router_;
+  VCMProtectionCallback* vcm_protection_callback_;
 
-  BitrateController* bitrate_controller_;
+  rtc::scoped_ptr callback_cs_;
+  rtc::scoped_ptr data_cs_;
+  rtc::scoped_ptr bitrate_observer_;
+  rtc::scoped_ptr loadstate_observer_;
+
+  PacedSender* const pacer_;
+  BitrateAllocator* const bitrate_allocator_;
+  BitrateController* const bitrate_controller_;
   // Owned by PeerConnection, not ViEEncoder
   CPULoadStateCallbackInvoker* load_manager_;
 
   int64_t time_of_last_incoming_frame_ms_ GUARDED_BY(data_cs_);
   bool send_padding_ GUARDED_BY(data_cs_);
   int min_transmit_bitrate_kbps_ GUARDED_BY(data_cs_);
+  uint32_t last_observed_bitrate_bps_ GUARDED_BY(data_cs_);
   int target_delay_ms_ GUARDED_BY(data_cs_);
   bool network_is_transmitting_ GUARDED_BY(data_cs_);
   bool encoder_paused_ GUARDED_BY(data_cs_);
@@ -251,6 +250,8 @@ class ViEEncoder
   bool video_suspended_ GUARDED_BY(data_cs_);
   I420FrameCallback* pre_encode_callback_ GUARDED_BY(callback_cs_);
   const int64_t start_ms_;
+
+  SendStatisticsProxy* send_statistics_proxy_ GUARDED_BY(callback_cs_);
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc
index 7eb015c4b6..1be2d31618 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.cc
@@ -99,12 +99,12 @@ int ViEExternalCodecImpl::DeRegisterExternalSendCodec(
 
 int ViEExternalCodecImpl::RegisterExternalReceiveCodec(
     const int video_channel,
-    const unsigned int pl_type,
+    const unsigned char pl_type,
     VideoDecoder* decoder,
     bool decoder_render,
     int render_delay) {
-  LOG(LS_INFO) << "Register exrernal decoder for channel " << video_channel
-               << ", pl_type " << pl_type
+  LOG(LS_INFO) << "Register external decoder for channel " << video_channel
+               << ", pl_type " << static_cast(pl_type)
                << ", decoder_render " << decoder_render
                << ", render_delay " << render_delay;
   assert(decoder != NULL);
@@ -127,7 +127,7 @@ int ViEExternalCodecImpl::RegisterExternalReceiveCodec(
 int ViEExternalCodecImpl::DeRegisterExternalReceiveCodec(
     const int video_channel, const unsigned char pl_type) {
   LOG(LS_INFO) << "DeRegisterExternalReceiveCodec for channel " << video_channel
-               << ", pl_type " << pl_type;
+               << ", pl_type " << static_cast(pl_type);
 
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h
index 7c8f4a20fd..da5a445ddf 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_external_codec_impl.h
@@ -31,7 +31,7 @@ class ViEExternalCodecImpl
   virtual int DeRegisterExternalSendCodec(const int video_channel,
                                           const unsigned char pl_type);
   virtual int RegisterExternalReceiveCodec(const int video_channel,
-                                           const unsigned int pl_type,
+                                           const unsigned char pl_type,
                                            VideoDecoder* decoder,
                                            bool decoder_render = false,
                                            int render_delay = 0);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc b/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc
index 919da9fa1b..cb6e206190 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_file_image.cc
@@ -74,9 +74,8 @@ int ViEFileImage::ConvertPictureToI420VideoFrame(int engine_id,
   int half_width = (picture.width + 1) / 2;
   video_frame->CreateEmptyFrame(picture.width, picture.height,
                                 picture.width, half_width, half_width);
-  return ConvertToI420(kI420, picture.data, 0, 0,
-                       picture.width, picture.height,
-                       0, kRotateNone, video_frame);
+  return ConvertToI420(kI420, picture.data, 0, 0, picture.width, picture.height,
+                       0, kVideoRotation_0, video_frame);
 }
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc b/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc
index a5cd838e2b..d7b453d155 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.cc
@@ -12,6 +12,7 @@
 
 #include 
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_video/interface/i420_video_frame.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
@@ -25,52 +26,50 @@ ViEFrameProviderBase::ViEFrameProviderBase(int Id, int engine_id)
       engine_id_(engine_id),
       provider_cs_(CriticalSectionWrapper::CreateCriticalSection()),
       frame_delay_(0) {
+  frame_delivery_thread_checker_.DetachFromThread();
 }
 
 ViEFrameProviderBase::~ViEFrameProviderBase() {
-  if (frame_callbacks_.size() > 0) {
-    LOG_F(LS_WARNING) << "FrameCallbacks still exist when Provider deleted: "
-                      << frame_callbacks_.size();
-  }
+  DCHECK(thread_checker_.CalledOnValidThread());
+  // XXX!!! FIX THIS - remove our callback
+  // DCHECK(frame_callbacks_.empty());
 
-  for (FrameCallbacks::iterator it = frame_callbacks_.begin();
-       it != frame_callbacks_.end(); ++it) {
-    (*it)->ProviderDestroyed(id_);
+  // TODO(tommi): Remove this when we're confident we've fixed the places where
+  // cleanup wasn't being done.
+  for (ViEFrameCallback* callback : frame_callbacks_) {
+    LOG_F(LS_WARNING) << "FrameCallback still registered.";
+    callback->ProviderDestroyed(id_);
   }
-  frame_callbacks_.clear();
 }
 
-int ViEFrameProviderBase::Id() {
+int ViEFrameProviderBase::Id() const {
   return id_;
 }
 
-void ViEFrameProviderBase::DeliverFrame(
-    I420VideoFrame* video_frame,
-    int num_csrcs,
-    const uint32_t CSRC[kRtpCsrcSize]) {
+void ViEFrameProviderBase::DeliverFrame(I420VideoFrame* video_frame,
+                                        const std::vector& csrcs) {
+  DCHECK(frame_delivery_thread_checker_.CalledOnValidThread());
 #ifdef DEBUG_
   const TickTime start_process_time = TickTime::Now();
 #endif
   CriticalSectionScoped cs(provider_cs_.get());
 
   // Deliver the frame to all registered callbacks.
-  if (frame_callbacks_.size() > 0) {
-    if (frame_callbacks_.size() == 1) {
-      // We don't have to copy the frame.
-      frame_callbacks_.front()->DeliverFrame(id_, video_frame, num_csrcs, CSRC);
-    } else {
-      for (FrameCallbacks::iterator it = frame_callbacks_.begin();
-           it != frame_callbacks_.end(); ++it) {
-        if (video_frame->native_handle() != NULL) {
-          (*it)->DeliverFrame(id_, video_frame, num_csrcs, CSRC);
-        } else {
-          // Make a copy of the frame for all callbacks.
-          if (!extra_frame_.get()) {
-            extra_frame_.reset(new I420VideoFrame());
-          }
-          extra_frame_->CopyFrame(*video_frame);
-          (*it)->DeliverFrame(id_, extra_frame_.get(), num_csrcs, CSRC);
+  if (frame_callbacks_.size() == 1) {
+    // We don't have to copy the frame.
+    frame_callbacks_.front()->DeliverFrame(id_, video_frame, csrcs);
+  } else {
+    for (ViEFrameCallback* callback : frame_callbacks_) {
+      if (video_frame->native_handle() != NULL) {
+        callback->DeliverFrame(id_, video_frame, csrcs);
+      } else {
+        // Make a copy of the frame for all callbacks.
+        if (!extra_frame_.get()) {
+          extra_frame_.reset(new I420VideoFrame());
         }
+        // TODO(mflodman): We can get rid of this frame copy.
+        extra_frame_->CopyFrame(*video_frame);
+        callback->DeliverFrame(id_, extra_frame_.get(), csrcs);
       }
     }
   }
@@ -85,34 +84,50 @@ void ViEFrameProviderBase::DeliverFrame(
 }
 
 void ViEFrameProviderBase::SetFrameDelay(int frame_delay) {
+  // Called on the capture thread (see OnIncomingCapturedFrame).
+  // To test, run ViEStandardIntegrationTest.RunsBaseTestWithoutErrors
+  // in vie_auto_tests.
+  // In the same test, it appears that it's also called on a thread that's
+  // neither the ctor thread nor the capture thread.
   CriticalSectionScoped cs(provider_cs_.get());
   frame_delay_ = frame_delay;
 
-  for (FrameCallbacks::iterator it = frame_callbacks_.begin();
-       it != frame_callbacks_.end(); ++it) {
-    (*it)->DelayChanged(id_, frame_delay);
+  for (ViEFrameCallback* callback : frame_callbacks_) {
+    callback->DelayChanged(id_, frame_delay);
   }
 }
 
 int ViEFrameProviderBase::FrameDelay() {
+  // Called on the default thread in WebRtcVideoMediaChannelTest.SetSend
+  // (libjingle_media_unittest).
+
+  // Called on neither the ctor thread nor the capture thread in
+  // BitrateEstimatorTest.ImmediatelySwitchToAST (video_engine_tests).
+
+  // Most of the time Called on the capture thread (see OnCaptureDelayChanged).
+  // To test, run ViEStandardIntegrationTest.RunsBaseTestWithoutErrors
+  // in vie_auto_tests.
   return frame_delay_;
 }
 
 int ViEFrameProviderBase::GetBestFormat(int* best_width,
                                         int* best_height,
                                         int* best_frame_rate) {
+  DCHECK(thread_checker_.CalledOnValidThread());
   int largest_width = 0;
   int largest_height = 0;
   int highest_frame_rate = 0;
 
-  CriticalSectionScoped cs(provider_cs_.get());
-  for (FrameCallbacks::iterator it = frame_callbacks_.begin();
-       it != frame_callbacks_.end(); ++it) {
+  // Here we don't need to grab the provider_cs_ lock to run through the list
+  // of callbacks.  The reason is that we know that we're currently on the same
+  // thread that is the only thread that will modify the callback list and
+  // we can be sure that the thread won't race with itself.
+  for (ViEFrameCallback* callback : frame_callbacks_) {
     int prefered_width = 0;
     int prefered_height = 0;
     int prefered_frame_rate = 0;
-    if ((*it)->GetPreferedFrameSettings(&prefered_width, &prefered_height,
-                                        &prefered_frame_rate) == 0) {
+    if (callback->GetPreferedFrameSettings(&prefered_width, &prefered_height,
+                                           &prefered_frame_rate) == 0) {
       if (prefered_width > largest_width) {
         largest_width = prefered_width;
       }
@@ -132,12 +147,13 @@ int ViEFrameProviderBase::GetBestFormat(int* best_width,
 
 int ViEFrameProviderBase::RegisterFrameCallback(
     int observer_id, ViEFrameCallback* callback_object) {
-  assert(callback_object);
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(callback_object);
   {
     CriticalSectionScoped cs(provider_cs_.get());
     if (std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
                   callback_object) != frame_callbacks_.end()) {
-      assert(false && "frameObserver already registered");
+      DCHECK(false && "frameObserver already registered");
       return -1;
     }
     frame_callbacks_.push_back(callback_object);
@@ -152,33 +168,35 @@ int ViEFrameProviderBase::RegisterFrameCallback(
 
 int ViEFrameProviderBase::DeregisterFrameCallback(
     const ViEFrameCallback* callback_object) {
-  assert(callback_object);
-  CriticalSectionScoped cs(provider_cs_.get());
-
-  FrameCallbacks::iterator it = std::find(frame_callbacks_.begin(),
-                                          frame_callbacks_.end(),
-                                          callback_object);
-  if (it == frame_callbacks_.end()) {
-    return -1;
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(callback_object);
+  {
+    CriticalSectionScoped cs(provider_cs_.get());
+    FrameCallbacks::iterator it = std::find(frame_callbacks_.begin(),
+                                            frame_callbacks_.end(),
+                                            callback_object);
+    if (it == frame_callbacks_.end()) {
+      return -1;
+    }
+    frame_callbacks_.erase(it);
   }
-  frame_callbacks_.erase(it);
 
   // Notify implementer of this class that the callback list have changed.
   FrameCallbackChanged();
+
   return 0;
 }
 
 bool ViEFrameProviderBase::IsFrameCallbackRegistered(
     const ViEFrameCallback* callback_object) {
-  assert(callback_object);
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(callback_object);
 
-  CriticalSectionScoped cs(provider_cs_.get());
+  // Here we don't need to grab the lock to do this lookup.
+  // The reason is that we know that we're currently on the same thread that
+  // is the only thread that will modify the callback list and subsequently the
+  // thread doesn't race with itself.
   return std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
                    callback_object) != frame_callbacks_.end();
 }
-
-int ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks() {
-  CriticalSectionScoped cs(provider_cs_.get());
-  return frame_callbacks_.size();
-}
 }  // namespac webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h b/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h
index e5a137b8bd..b3ace3bdfc 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_frame_provider_base.h
@@ -13,8 +13,9 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/common_types.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -29,8 +30,7 @@ class ViEFrameCallback {
  public:
   virtual void DeliverFrame(int id,
                             I420VideoFrame* video_frame,
-                            int num_csrcs = 0,
-                            const uint32_t CSRC[kRtpCsrcSize] = NULL) = 0;
+                            const std::vector& csrcs) = 0;
 
   // The capture delay has changed from the provider. |frame_delay| is given in
   // ms.
@@ -56,18 +56,20 @@ class ViEFrameProviderBase {
   virtual ~ViEFrameProviderBase();
 
   // Returns the frame provider id.
-  int Id();
+  int Id() const;
 
   // Register frame callbacks, i.e. a receiver of the captured frame.
-  virtual int RegisterFrameCallback(int observer_id,
-                                    ViEFrameCallback* callback_object);
+  // Must be called on the same thread as the provider was constructed on.
+  int RegisterFrameCallback(int observer_id, ViEFrameCallback* callback);
 
-  virtual int DeregisterFrameCallback(const ViEFrameCallback* callback_object);
+  // Unregisters a previously registered callback.  Returns -1 if the callback
+  // object hasn't been registered.
+  // Must be called on the same thread as the provider was constructed on.
+  int DeregisterFrameCallback(const ViEFrameCallback* callback);
 
-  virtual bool IsFrameCallbackRegistered(
-      const ViEFrameCallback* callback_object);
-
-  int NumberOfRegisteredFrameCallbacks();
+  // Determines if a callback is currently registered.
+  // Must be called on the same thread as the provider was constructed on.
+  bool IsFrameCallbackRegistered(const ViEFrameCallback* callback);
 
   // FrameCallbackChanged
   // Inherited classes should check for new frame_settings and reconfigure
@@ -76,24 +78,26 @@ class ViEFrameProviderBase {
 
  protected:
   void DeliverFrame(I420VideoFrame* video_frame,
-                    int num_csrcs = 0,
-                    const uint32_t CSRC[kRtpCsrcSize] = NULL);
+                    const std::vector& csrcs);
   void SetFrameDelay(int frame_delay);
   int FrameDelay();
   int GetBestFormat(int* best_width,
                     int* best_height,
                     int* best_frame_rate);
 
-  int id_;
-  int engine_id_;
+  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker frame_delivery_thread_checker_;
+
+  const int id_;
+  const int engine_id_;
 
   // Frame callbacks.
   typedef std::vector FrameCallbacks;
   FrameCallbacks frame_callbacks_;
-  scoped_ptr provider_cs_;
+  const rtc::scoped_ptr provider_cs_;
 
  private:
-  scoped_ptr extra_frame_;
+  rtc::scoped_ptr extra_frame_;
   int frame_delay_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h
index 38a6a08cf8..64775b1e19 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_image_process_impl.h
@@ -37,25 +37,25 @@ class ViEImageProcessImpl
   virtual int EnableDeflickering(const int capture_id, const bool enable);
   virtual int EnableColorEnhancement(const int video_channel,
                                      const bool enable);
-  virtual void RegisterPreEncodeCallback(
+  void RegisterPreEncodeCallback(
       int video_channel,
-      I420FrameCallback* pre_encode_callback) OVERRIDE;
-  virtual void DeRegisterPreEncodeCallback(int video_channel) OVERRIDE;
+      I420FrameCallback* pre_encode_callback) override;
+  void DeRegisterPreEncodeCallback(int video_channel) override;
 
-  virtual void RegisterPostEncodeImageCallback(
+  void RegisterPostEncodeImageCallback(
       int video_channel,
-      EncodedImageCallback* post_encode_callback) OVERRIDE;
-  virtual void DeRegisterPostEncodeCallback(int video_channel) OVERRIDE;
+      EncodedImageCallback* post_encode_callback) override;
+  void DeRegisterPostEncodeCallback(int video_channel) override;
 
-  virtual void RegisterPreDecodeImageCallback(
-        int video_channel,
-        EncodedImageCallback* post_encode_callback) OVERRIDE;
-  virtual void DeRegisterPreDecodeCallback(int video_channel) OVERRIDE;
-
-  virtual void RegisterPreRenderCallback(
+  void RegisterPreDecodeImageCallback(
       int video_channel,
-      I420FrameCallback* pre_render_callback) OVERRIDE;
-  virtual void DeRegisterPreRenderCallback(int video_channel) OVERRIDE;
+      EncodedImageCallback* post_encode_callback) override;
+  void DeRegisterPreDecodeCallback(int video_channel) override;
+
+  void RegisterPreRenderCallback(
+      int video_channel,
+      I420FrameCallback* pre_render_callback) override;
+  void DeRegisterPreRenderCallback(int video_channel) override;
 
  protected:
   explicit ViEImageProcessImpl(ViESharedData* shared_data);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_impl.h
index 086282d187..34fd7f7f3b 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_impl.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
 #define WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/video_engine/vie_defines.h"
 
 #include "webrtc/video_engine/vie_base_impl.h"
@@ -100,7 +100,7 @@ class VideoEngineImpl
 
  private:
   // Placeholder for the case where this owns the config.
-  scoped_ptr own_config_;
+  rtc::scoped_ptr own_config_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc b/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc
index 7874aa8d7d..e3067e0191 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.cc
@@ -113,29 +113,12 @@ int ViEInputManager::GetCaptureCapability(
 }
 
 int ViEInputManager::GetOrientation(const char* device_unique_idUTF8,
-                                    RotateCapturedFrame& orientation) {
+                                    VideoRotation& orientation) {
   CriticalSectionScoped cs(device_info_cs_.get());
   GetDeviceInfo();
   assert(capture_device_info_);
-  VideoCaptureRotation module_orientation;
-  int result = capture_device_info_->GetOrientation(device_unique_idUTF8,
-                                                    module_orientation);
-  // Copy from module type to public type.
-  switch (module_orientation) {
-    case kCameraRotate0:
-      orientation = RotateCapturedFrame_0;
-      break;
-    case kCameraRotate90:
-      orientation = RotateCapturedFrame_90;
-      break;
-    case kCameraRotate180:
-      orientation = RotateCapturedFrame_180;
-      break;
-    case kCameraRotate270:
-      orientation = RotateCapturedFrame_270;
-      break;
-  }
-  return result;
+  return capture_device_info_->GetOrientation(device_unique_idUTF8,
+                                              orientation);
 }
 
 int ViEInputManager::DisplayCaptureSettingsDialogBox(
@@ -260,12 +243,6 @@ int ViEInputManager::DestroyCaptureDevice(const int capture_id) {
       LOG(LS_ERROR) << "No such capture device id: " << capture_id;
       return -1;
     }
-    uint32_t num_callbacks =
-        vie_capture->NumberOfRegisteredFrameCallbacks();
-    if (num_callbacks > 0) {
-      LOG(LS_WARNING) << num_callbacks << " still registered to capture id "
-                      << capture_id << " when destroying capture device.";
-    }
     vie_frame_provider_map_.erase(capture_id);
     ReturnCaptureId(capture_id);
     // Leave cs before deleting the capture object. This is because deleting the
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h b/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h
index 26252069f0..8a2b441639 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_input_manager.h
@@ -13,9 +13,10 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/video_capture/include/video_capture.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/common_video/rotation.h"
 #include "webrtc/video_engine/include/vie_capture.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/video_engine/vie_frame_provider_base.h"
@@ -64,7 +65,7 @@ class ViEInputManager : private ViEManagerBase {
                                       uint32_t positionX,
                                       uint32_t positionY);
   int GetOrientation(const char* device_unique_idUTF8,
-                     RotateCapturedFrame& orientation);
+                     VideoRotation& orientation);
 
   // Creates a capture module for the specified capture device and assigns
   // a capture device id for the device.
@@ -98,8 +99,8 @@ class ViEInputManager : private ViEManagerBase {
 
   const Config& config_;
   int engine_id_;
-  scoped_ptr map_cs_;
-  scoped_ptr device_info_cs_;
+  rtc::scoped_ptr map_cs_;
+  rtc::scoped_ptr device_info_cs_;
 
   typedef std::map FrameProviderMap;
   FrameProviderMap vie_frame_provider_map_;
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc
index 4a9d866585..5d45187a61 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.cc
@@ -15,6 +15,7 @@
 #include 
 #endif
 
+#include "webrtc/base/checks.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/video_engine/include/vie_errors.h"
@@ -51,6 +52,26 @@ int ViENetworkImpl::Release() {
   return ref_count;
 }
 
+ViENetworkImpl::ViENetworkImpl(ViESharedData* shared_data)
+    : shared_data_(shared_data) {
+}
+
+ViENetworkImpl::~ViENetworkImpl() {
+}
+
+void ViENetworkImpl::SetBitrateConfig(int video_channel,
+                                      int min_bitrate_bps,
+                                      int start_bitrate_bps,
+                                      int max_bitrate_bps) {
+  LOG_F(LS_INFO) << "channel: " << video_channel
+                 << " new bitrate config: min=" << min_bitrate_bps
+                 << ", start=" << start_bitrate_bps
+                 << ", max=" << max_bitrate_bps;
+  bool success = shared_data_->channel_manager()->SetBitrateConfig(
+      video_channel, min_bitrate_bps, start_bitrate_bps, max_bitrate_bps);
+  DCHECK(success);
+}
+
 void ViENetworkImpl::SetNetworkTransmissionState(const int video_channel,
                                                  const bool is_transmitting) {
   LOG_F(LS_INFO) << "channel: " << video_channel
@@ -64,11 +85,6 @@ void ViENetworkImpl::SetNetworkTransmissionState(const int video_channel,
   vie_encoder->SetNetworkTransmissionState(is_transmitting);
 }
 
-ViENetworkImpl::ViENetworkImpl(ViESharedData* shared_data)
-    : shared_data_(shared_data) {}
-
-ViENetworkImpl::~ViENetworkImpl() {}
-
 int ViENetworkImpl::RegisterSendTransport(const int video_channel,
                                           Transport& transport) {
   LOG_F(LS_INFO) << "channel: " << video_channel;
@@ -111,7 +127,7 @@ int ViENetworkImpl::DeregisterSendTransport(const int video_channel) {
 }
 
 int ViENetworkImpl::ReceivedRTPPacket(const int video_channel, const void* data,
-                                      const int length,
+                                      const size_t length,
                                       const PacketTime& packet_time) {
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
@@ -123,7 +139,7 @@ int ViENetworkImpl::ReceivedRTPPacket(const int video_channel, const void* data,
 }
 
 int ViENetworkImpl::ReceivedRTCPPacket(const int video_channel,
-                                       const void* data, const int length) {
+                                       const void* data, const size_t length) {
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -149,7 +165,9 @@ int ViENetworkImpl::SetMTU(int video_channel, unsigned int mtu) {
 }
 
 int ViENetworkImpl::ReceivedBWEPacket(const int video_channel,
-    int64_t arrival_time_ms, int payload_size, const RTPHeader& header) {
+                                      int64_t arrival_time_ms,
+                                      size_t payload_size,
+                                      const RTPHeader& header) {
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -160,11 +178,4 @@ int ViENetworkImpl::ReceivedBWEPacket(const int video_channel,
   vie_channel->ReceivedBWEPacket(arrival_time_ms, payload_size, header);
   return 0;
 }
-
-bool ViENetworkImpl::SetBandwidthEstimationConfig(
-    int video_channel, const webrtc::Config& config) {
-  LOG_F(LS_INFO) << "channel: " << video_channel;
-  return shared_data_->channel_manager()->SetBandwidthEstimationConfig(
-      video_channel, config);
-}
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h
index e9ca75469d..1354f8cf4a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_network_impl.h
@@ -24,29 +24,29 @@ class ViENetworkImpl
       public ViERefCount {
  public:
   // Implements ViENetwork.
-  virtual int Release() OVERRIDE;
-  virtual void SetNetworkTransmissionState(const int video_channel,
-                                           const bool is_transmitting) OVERRIDE;
-  virtual int RegisterSendTransport(const int video_channel,
-                                    Transport& transport) OVERRIDE;
-  virtual int DeregisterSendTransport(const int video_channel) OVERRIDE;
-  virtual int ReceivedRTPPacket(const int video_channel,
-                                const void* data,
-                                const int length,
-                                const PacketTime& packet_time) OVERRIDE;
-  virtual int ReceivedRTCPPacket(const int video_channel,
-                                 const void* data,
-                                 const int length) OVERRIDE;
-  virtual int SetMTU(int video_channel, unsigned int mtu) OVERRIDE;
+  int Release() override;
+  void SetBitrateConfig(int video_channel,
+                        int min_bitrate_bps,
+                        int start_bitrate_bps,
+                        int max_bitrate_bps) override;
+  void SetNetworkTransmissionState(const int video_channel,
+                                   const bool is_transmitting) override;
+  int RegisterSendTransport(const int video_channel,
+                            Transport& transport) override;
+  int DeregisterSendTransport(const int video_channel) override;
+  int ReceivedRTPPacket(const int video_channel,
+                        const void* data,
+                        const size_t length,
+                        const PacketTime& packet_time) override;
+  int ReceivedRTCPPacket(const int video_channel,
+                         const void* data,
+                         const size_t length) override;
+  int SetMTU(int video_channel, unsigned int mtu) override;
 
-  virtual int ReceivedBWEPacket(const int video_channel,
-                                int64_t arrival_time_ms,
-                                int payload_size,
-                                const RTPHeader& header) OVERRIDE;
-
-  virtual bool SetBandwidthEstimationConfig(
-      int video_channel,
-      const webrtc::Config& config) OVERRIDE;
+  int ReceivedBWEPacket(const int video_channel,
+                        int64_t arrival_time_ms,
+                        size_t payload_size,
+                        const RTPHeader& header) override;
 
  protected:
   explicit ViENetworkImpl(ViESharedData* shared_data);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc b/media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc
index 284109d925..daac6b93c4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_receiver.cc
@@ -16,6 +16,7 @@
 #include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
@@ -24,6 +25,7 @@
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
 #include "webrtc/system_wrappers/interface/trace.h"
@@ -58,11 +60,13 @@ ViEReceiver::ViEReceiver(const int32_t channel_id,
       receiving_rtcp_(false),
       restored_packet_in_use_(false),
       receiving_ast_enabled_(false),
+      receiving_cvo_enabled_(false),
       last_packet_log_ms_(-1) {
   assert(remote_bitrate_estimator);
 }
 
 ViEReceiver::~ViEReceiver() {
+  UpdateHistograms();
   if (rtp_dump_) {
     rtp_dump_->Stop();
     RtpDump::DestroyRtpDump(rtp_dump_);
@@ -70,6 +74,19 @@ ViEReceiver::~ViEReceiver() {
   }
 }
 
+void ViEReceiver::UpdateHistograms() {
+  FecPacketCounter counter = fec_receiver_->GetPacketCounter();
+  if (counter.num_packets > 0) {
+    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedFecPacketsInPercent",
+        counter.num_fec_packets * 100 / counter.num_packets);
+  }
+  if (counter.num_fec_packets > 0) {
+    RTC_HISTOGRAM_PERCENTAGE(
+        "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+            counter.num_recovered_packets * 100 / counter.num_fec_packets);
+  }
+}
+
 bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
   int8_t old_pltype = -1;
   if (rtp_payload_registry_->ReceivePayloadType(video_codec.plName,
@@ -111,6 +128,14 @@ void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
   rtp_payload_registry_->SetRtxSsrc(ssrc);
 }
 
+bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
+  return rtp_payload_registry_->GetRtxSsrc(ssrc);
+}
+
+bool ViEReceiver::IsFecEnabled() const {
+  return rtp_payload_registry_->ulpfec_payload_type() > -1;
+}
+
 uint32_t ViEReceiver::GetRemoteSsrc() const {
   return rtp_receiver_->SSRC();
 }
@@ -165,22 +190,38 @@ bool ViEReceiver::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
   }
 }
 
+bool ViEReceiver::SetReceiveVideoRotationStatus(bool enable, int id) {
+  if (enable) {
+    if (rtp_header_parser_->RegisterRtpHeaderExtension(
+            kRtpExtensionVideoRotation, id)) {
+      receiving_cvo_enabled_ = true;
+      return true;
+    } else {
+      return false;
+    }
+  } else {
+    receiving_cvo_enabled_ = false;
+    return rtp_header_parser_->DeregisterRtpHeaderExtension(
+        kRtpExtensionVideoRotation);
+  }
+}
+
 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
-                                   int rtp_packet_length,
+                                   size_t rtp_packet_length,
                                    const PacketTime& packet_time) {
   return InsertRTPPacket(static_cast(rtp_packet),
                          rtp_packet_length, packet_time);
 }
 
 int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet,
-                                    int rtcp_packet_length) {
+                                    size_t rtcp_packet_length) {
   return InsertRTCPPacket(static_cast(rtcp_packet),
                           rtcp_packet_length);
 }
 
-int32_t ViEReceiver::OnReceivedPayloadData(
-    const uint8_t* payload_data, const uint16_t payload_size,
-    const WebRtcRTPHeader* rtp_header) {
+int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
+                                           const size_t payload_size,
+                                           const WebRtcRTPHeader* rtp_header) {
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
   rtp_header_with_ntp.ntp_time_ms =
       ntp_estimator_->Estimate(rtp_header->header.timestamp);
@@ -194,7 +235,7 @@ int32_t ViEReceiver::OnReceivedPayloadData(
 }
 
 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
-                                    int rtp_packet_length) {
+                                    size_t rtp_packet_length) {
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
     return false;
@@ -205,7 +246,7 @@ bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
 }
 
 void ViEReceiver::ReceivedBWEPacket(
-    int64_t arrival_time_ms, int payload_size, const RTPHeader& header) {
+    int64_t arrival_time_ms, size_t payload_size, const RTPHeader& header) {
   // Only forward if the incoming packet *and* the channel are both configured
   // to receive absolute sender time. RTP time stamps may have different rates
   // for audio and video and shouldn't be mixed.
@@ -216,7 +257,7 @@ void ViEReceiver::ReceivedBWEPacket(
 }
 
 int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
-                                 int rtp_packet_length,
+                                 size_t rtp_packet_length,
                                  const PacketTime& packet_time) {
   {
     CriticalSectionScoped cs(receive_cs_.get());
@@ -224,8 +265,7 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
       return -1;
     }
     if (rtp_dump_) {
-      rtp_dump_->DumpPacket(rtp_packet,
-                            static_cast(rtp_packet_length));
+      rtp_dump_->DumpPacket(rtp_packet, rtp_packet_length);
     }
   }
 
@@ -234,7 +274,7 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
                                  &header)) {
     return -1;
   }
-  int payload_length = rtp_packet_length - header.headerLength;
+  size_t payload_length = rtp_packet_length - header.headerLength;
   int64_t arrival_time_ms;
   int64_t now_ms = clock_->TimeInMilliseconds();
   if (packet_time.timestamp != -1)
@@ -278,15 +318,15 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
 }
 
 bool ViEReceiver::ReceivePacket(const uint8_t* packet,
-                                int packet_length,
+                                size_t packet_length,
                                 const RTPHeader& header,
                                 bool in_order) {
   if (rtp_payload_registry_->IsEncapsulated(header)) {
     return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
   }
   const uint8_t* payload = packet + header.headerLength;
-  int payload_length = packet_length - header.headerLength;
-  assert(payload_length >= 0);
+  assert(packet_length >= header.headerLength);
+  size_t payload_length = packet_length - header.headerLength;
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
@@ -297,12 +337,15 @@ bool ViEReceiver::ReceivePacket(const uint8_t* packet,
 }
 
 bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
-                                                    int packet_length,
+                                                    size_t packet_length,
                                                     const RTPHeader& header) {
   if (rtp_payload_registry_->IsRed(header)) {
     int8_t ulpfec_pt = rtp_payload_registry_->ulpfec_payload_type();
-    if (packet[header.headerLength] == ulpfec_pt)
-      rtp_receive_statistics_->FecPacketReceived(header.ssrc);
+    if (packet[header.headerLength] == ulpfec_pt) {
+      rtp_receive_statistics_->FecPacketReceived(header, packet_length);
+      // Notify vcm about received FEC packets to avoid NACKing these packets.
+      NotifyReceiverOfFecPacket(header);
+    }
     if (fec_receiver_->AddReceivedRedPacket(
             header, packet, packet_length, ulpfec_pt) != 0) {
       return false;
@@ -317,7 +360,7 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
     // Remove the RTX header and parse the original RTP header.
     if (packet_length < header.headerLength)
       return false;
-    if (packet_length > static_cast(sizeof(restored_packet_)))
+    if (packet_length > sizeof(restored_packet_))
       return false;
     CriticalSectionScoped cs(receive_cs_.get());
     if (restored_packet_in_use_) {
@@ -339,8 +382,35 @@ bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
   return false;
 }
 
+void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
+  int8_t last_media_payload_type =
+      rtp_payload_registry_->last_received_media_payload_type();
+  if (last_media_payload_type < 0) {
+    LOG(LS_WARNING) << "Failed to get last media payload type.";
+    return;
+  }
+  // Fake an empty media packet.
+  WebRtcRTPHeader rtp_header = {};
+  rtp_header.header = header;
+  rtp_header.header.payloadType = last_media_payload_type;
+  rtp_header.header.paddingLength = 0;
+  PayloadUnion payload_specific;
+  if (!rtp_payload_registry_->GetPayloadSpecifics(last_media_payload_type,
+                                                  &payload_specific)) {
+    LOG(LS_WARNING) << "Failed to get payload specifics.";
+    return;
+  }
+  rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
+  rtp_header.type.Video.rotation = kVideoRotation_0;
+  if (header.extension.hasVideoRotation) {
+    rtp_header.type.Video.rotation =
+        ConvertCVOByteToVideoRotation(header.extension.videoRotation);
+  }
+  OnReceivedPayloadData(NULL, 0, &rtp_header);
+}
+
 int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
-                                  int rtcp_packet_length) {
+                                  size_t rtcp_packet_length) {
   {
     CriticalSectionScoped cs(receive_cs_.get());
     if (!receiving_rtcp_) {
@@ -348,8 +418,7 @@ int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
     }
 
     if (rtp_dump_) {
-      rtp_dump_->DumpPacket(
-          rtcp_packet, static_cast(rtcp_packet_length));
+      rtp_dump_->DumpPacket(rtcp_packet, rtcp_packet_length);
     }
 
     std::list::iterator it = rtp_rtcp_simulcast_.begin();
@@ -364,7 +433,7 @@ int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
     return ret;
   }
 
-  uint16_t rtt = 0;
+  int64_t rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
   if (rtt == 0) {
     // Waiting for valid rtt.
@@ -436,11 +505,6 @@ int ViEReceiver::StopRTPDump() {
   return 0;
 }
 
-void ViEReceiver::GetReceiveBandwidthEstimatorStats(
-    ReceiveBandwidthEstimatorStats* output) const {
-  remote_bitrate_estimator_->GetStats(output);
-}
-
 ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
   return rtp_receive_statistics_.get();
 }
@@ -463,7 +527,7 @@ bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header,
   if (!statistician)
     return false;
   // Check if this is a retransmission.
-  uint16_t min_rtt = 0;
+  int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.h b/media/webrtc/trunk/webrtc/video_engine/vie_receiver.h
index 5bbc9becbc..1002f64165 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_receiver.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_receiver.h
@@ -13,10 +13,10 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/include/vie_network.h"
 #include "webrtc/video_engine/vie_defines.h"
@@ -49,6 +49,9 @@ class ViEReceiver : public RtpData {
   void SetNackStatus(bool enable, int max_nack_reordering_threshold);
   void SetRtxPayloadType(int payload_type);
   void SetRtxSsrc(uint32_t ssrc);
+  bool GetRtxSsrc(uint32_t* ssrc) const;
+
+  bool IsFecEnabled() const;
 
   uint32_t GetRemoteSsrc() const;
   int GetCsrcs(uint32_t* csrcs) const;
@@ -61,6 +64,7 @@ class ViEReceiver : public RtpData {
 
   bool SetReceiveTimestampOffsetStatus(bool enable, int id);
   bool SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
+  bool SetReceiveVideoRotationStatus(bool enable, int id);
 
   void StartReceive();
   void StopReceive();
@@ -72,54 +76,51 @@ class ViEReceiver : public RtpData {
   int StopRTPDump();
 
   // Receives packets from external transport.
-  int ReceivedRTPPacket(const void* rtp_packet, int rtp_packet_length,
+  int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length,
                         const PacketTime& packet_time);
-  int ReceivedRTCPPacket(const void* rtcp_packet, int rtcp_packet_length);
+  int ReceivedRTCPPacket(const void* rtcp_packet, size_t rtcp_packet_length);
 
   // Implements RtpData.
-  virtual int32_t OnReceivedPayloadData(
-      const uint8_t* payload_data,
-      const uint16_t payload_size,
-      const WebRtcRTPHeader* rtp_header) OVERRIDE;
-  virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                 int packet_length) OVERRIDE;
-
-  void GetReceiveBandwidthEstimatorStats(
-      ReceiveBandwidthEstimatorStats* output) const;
+  int32_t OnReceivedPayloadData(const uint8_t* payload_data,
+                                const size_t payload_size,
+                                const WebRtcRTPHeader* rtp_header) override;
+  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   ReceiveStatistics* GetReceiveStatistics() const;
 
-  void ReceivedBWEPacket(int64_t arrival_time_ms, int payload_size,
+  void ReceivedBWEPacket(int64_t arrival_time_ms, size_t payload_size,
                          const RTPHeader& header);
  private:
-  int InsertRTPPacket(const uint8_t* rtp_packet, int rtp_packet_length,
+  int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length,
                       const PacketTime& packet_time);
   bool ReceivePacket(const uint8_t* packet,
-                     int packet_length,
+                     size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   // Parses and handles for instance RTX and RED headers.
   // This function assumes that it's being called from only one thread.
   bool ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
-                                         int packet_length,
+                                         size_t packet_length,
                                          const RTPHeader& header);
-  int InsertRTCPPacket(const uint8_t* rtcp_packet, int rtcp_packet_length);
+  void NotifyReceiverOfFecPacket(const RTPHeader& header);
+  int InsertRTCPPacket(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
+  void UpdateHistograms();
 
-  scoped_ptr receive_cs_;
+  rtc::scoped_ptr receive_cs_;
   Clock* clock_;
-  scoped_ptr rtp_header_parser_;
-  scoped_ptr rtp_payload_registry_;
-  scoped_ptr rtp_receiver_;
-  scoped_ptr rtp_receive_statistics_;
-  scoped_ptr fec_receiver_;
+  rtc::scoped_ptr rtp_header_parser_;
+  rtc::scoped_ptr rtp_payload_registry_;
+  rtc::scoped_ptr rtp_receiver_;
+  rtc::scoped_ptr rtp_receive_statistics_;
+  rtc::scoped_ptr fec_receiver_;
   RtpRtcp* rtp_rtcp_;
   std::list rtp_rtcp_simulcast_;
   VideoCodingModule* vcm_;
   RemoteBitrateEstimator* remote_bitrate_estimator_;
 
-  scoped_ptr ntp_estimator_;
+  rtc::scoped_ptr ntp_estimator_;
 
   RtpDump* rtp_dump_;
   bool receiving_;
@@ -127,6 +128,7 @@ class ViEReceiver : public RtpData {
   uint8_t restored_packet_[kViEMaxMtu];
   bool restored_packet_in_use_;
   bool receiving_ast_enabled_;
+  bool receiving_cvo_enabled_;
   int64_t last_packet_log_ms_;
 };
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h b/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h
index 95451c4dde..61533e10e2 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_ref_count.h
@@ -13,7 +13,7 @@
 #ifndef WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
 #define WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -32,7 +32,7 @@ class ViERefCount {
 
  private:
   volatile int count_;
-  scoped_ptr crit_;
+  rtc::scoped_ptr crit_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_remb.cc b/media/webrtc/trunk/webrtc/video_engine/vie_remb.cc
index d04f0a30ad..2cf794c5d4 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_remb.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_remb.cc
@@ -136,8 +136,7 @@ void VieRemb::OnReceiveBitrateChanged(const std::vector& ssrcs,
   list_crit_->Leave();
 
   if (sender) {
-    // TODO(holmer): Change RTP module API to take a const vector reference.
-    sender->SetREMBData(bitrate_, ssrcs.size(), &ssrcs[0]);
+    sender->SetREMBData(bitrate_, ssrcs);
   }
 }
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_remb.h b/media/webrtc/trunk/webrtc/video_engine/vie_remb.h
index 66e1c8e689..9f38259ca8 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_remb.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_remb.h
@@ -15,10 +15,10 @@
 #include 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/interface/module.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -57,7 +57,7 @@ class VieRemb : public RemoteBitrateObserver {
  private:
   typedef std::list RtpModules;
 
-  scoped_ptr list_crit_;
+  rtc::scoped_ptr list_crit_;
 
   // The last time a REMB was sent.
   int64_t last_remb_time_;
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc b/media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc
index 1f0b70c516..3949d4482c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_remb_unittest.cc
@@ -16,10 +16,10 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/video_engine/vie_remb.h"
 
@@ -37,8 +37,8 @@ class ViERembTest : public ::testing::Test {
     process_thread_.reset(new NiceMock);
     vie_remb_.reset(new VieRemb());
   }
-  scoped_ptr process_thread_;
-  scoped_ptr vie_remb_;
+  rtc::scoped_ptr process_thread_;
+  rtc::scoped_ptr vie_remb_;
 };
 
 TEST_F(ViERembTest, OneModuleTestForSendingRemb) {
@@ -53,12 +53,12 @@ TEST_F(ViERembTest, OneModuleTestForSendingRemb) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Lower bitrate to send another REMB packet.
-  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, 1, _))
+  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, ssrcs))
         .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate - 100);
 
@@ -78,14 +78,14 @@ TEST_F(ViERembTest, LowerEstimateToSendRemb) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
         .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Lower the estimate with more than 3% to trigger a call to SetREMBData right
   // away.
   bitrate_estimate = bitrate_estimate - 100;
-  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 }
@@ -104,7 +104,7 @@ TEST_F(ViERembTest, VerifyIncreasingAndDecreasing) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
 
   // Call OnReceiveBitrateChanged twice to get a first estimate.
-  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0], 2, _))
+  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0], ssrcs))
         .Times(1);
   TickTime::AdvanceFakeClock(1000);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
@@ -112,7 +112,7 @@ TEST_F(ViERembTest, VerifyIncreasingAndDecreasing) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1] + 100);
 
   // Lower the estimate to trigger a callback.
-  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[1], 2, _))
+  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[1], ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1]);
 
@@ -135,17 +135,17 @@ TEST_F(ViERembTest, NoRembForIncreasedBitrate) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, 2, _))
+  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Increased estimate shouldn't trigger a callback right away.
-  EXPECT_CALL(rtp_0, SetREMBData(_, _, _))
+  EXPECT_CALL(rtp_0, SetREMBData(_, _))
       .Times(0);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate + 1);
 
   // Decreasing the estimate less than 3% shouldn't trigger a new callback.
-  EXPECT_CALL(rtp_0, SetREMBData(_, _, _))
+  EXPECT_CALL(rtp_0, SetREMBData(_, _))
       .Times(0);
   int lower_estimate = bitrate_estimate * 98 / 100;
   vie_remb_->OnReceiveBitrateChanged(ssrcs, lower_estimate);
@@ -169,13 +169,13 @@ TEST_F(ViERembTest, ChangeSendRtpModule) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, 2, _))
+  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Decrease estimate to trigger a REMB.
   bitrate_estimate = bitrate_estimate - 100;
-  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, 2, _))
+  EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
@@ -186,7 +186,7 @@ TEST_F(ViERembTest, ChangeSendRtpModule) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   bitrate_estimate = bitrate_estimate - 100;
-  EXPECT_CALL(rtp_1, SetREMBData(bitrate_estimate, 2, _))
+  EXPECT_CALL(rtp_1, SetREMBData(bitrate_estimate, ssrcs))
         .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
@@ -205,18 +205,18 @@ TEST_F(ViERembTest, OnlyOneRembForDoubleProcess) {
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   // Call OnReceiveBitrateChanged twice to get a first estimate.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(rtp, SetREMBData(_, _, _))
+  EXPECT_CALL(rtp, SetREMBData(_, _))
         .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Lower the estimate, should trigger a call to SetREMBData right away.
   bitrate_estimate = bitrate_estimate - 100;
-  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
+  EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, ssrcs))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Call OnReceiveBitrateChanged again, this should not trigger a new callback.
-  EXPECT_CALL(rtp, SetREMBData(_, _, _))
+  EXPECT_CALL(rtp, SetREMBData(_, _))
       .Times(0);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
   vie_remb_->RemoveReceiveChannel(&rtp);
@@ -237,13 +237,13 @@ TEST_F(ViERembTest, NoSendingRtpModule) {
 
   // Call OnReceiveBitrateChanged twice to get a first estimate.
   TickTime::AdvanceFakeClock(1000);
-  EXPECT_CALL(rtp, SetREMBData(_, _, _))
+  EXPECT_CALL(rtp, SetREMBData(_, _))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 
   // Lower the estimate to trigger a new packet REMB packet.
   bitrate_estimate = bitrate_estimate - 100;
-  EXPECT_CALL(rtp, SetREMBData(_, _, _))
+  EXPECT_CALL(rtp, SetREMBData(_, _))
       .Times(1);
   vie_remb_->OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
 }
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc
index 00c0186a26..27920602b6 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.cc
@@ -236,23 +236,6 @@ int ViERenderImpl::ConfigureRender(int render_id, const unsigned int z_order,
   return 0;
 }
 
-int ViERenderImpl::MirrorRenderStream(const int render_id, const bool enable,
-                                      const bool mirror_xaxis,
-                                      const bool mirror_yaxis) {
-  ViERenderManagerScoped rs(*(shared_data_->render_manager()));
-  ViERenderer* renderer = rs.Renderer(render_id);
-  if (!renderer) {
-    shared_data_->SetLastError(kViERenderInvalidRenderId);
-    return -1;
-  }
-  if (renderer->EnableMirroring(render_id, enable, mirror_xaxis, mirror_yaxis)
-      != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  return 0;
-}
-
 int ViERenderImpl::AddRenderer(const int render_id,
                                RawVideoType video_input_format,
                                ExternalRenderer* external_renderer) {
@@ -323,29 +306,4 @@ int ViERenderImpl::AddRenderer(const int render_id,
   }
 }
 
-int ViERenderImpl::AddRenderCallback(int render_id,
-                                     VideoRenderCallback* callback) {
-  if (render_id < kViEChannelIdBase || render_id > kViEChannelIdMax)
-    return -1;
-  // This is a channel.
-  ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
-  ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
-  if (!frame_provider) {
-    shared_data_->SetLastError(kViERenderInvalidRenderId);
-    return -1;
-  }
-  ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
-      render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
-  if (!renderer) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-  if (renderer->SetVideoRenderCallback(render_id, callback) != 0) {
-    shared_data_->SetLastError(kViERenderUnknownError);
-    return -1;
-  }
-
-  return frame_provider->RegisterFrameCallback(render_id, renderer);
-}
-
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h
index 370a7a345c..5b36d1686f 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_render_impl.h
@@ -40,13 +40,8 @@ class ViERenderImpl
   virtual int ConfigureRender(int render_id, const unsigned int z_order,
                               const float left, const float top,
                               const float right, const float bottom);
-  virtual int MirrorRenderStream(const int render_id, const bool enable,
-                                 const bool mirror_xaxis,
-                                 const bool mirror_yaxis);
   virtual int AddRenderer(const int render_id, RawVideoType video_input_format,
                           ExternalRenderer* renderer);
-  virtual int AddRenderCallback(int render_id,
-                                VideoRenderCallback* callback) OVERRIDE;
 
  protected:
   explicit ViERenderImpl(ViESharedData* shared_data);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h b/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h
index c1314cdedc..db1626a1e0 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_render_manager.h
@@ -14,7 +14,7 @@
 #include 
 #include 
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/vie_manager_base.h"
 
@@ -54,7 +54,7 @@ class ViERenderManager : private ViEManagerBase {
   // Methods used by ViERenderScoped.
   ViERenderer* ViERenderPtr(int32_t render_id) const;
 
-  scoped_ptr list_cs_;
+  rtc::scoped_ptr list_cs_;
   int32_t engine_id_;
   // Protected by ViEManagerBase.
   typedef std::map RendererMap;
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc b/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc
index b4ec73828d..bfeeba216a 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_renderer.cc
@@ -39,8 +39,7 @@ ViERenderer::~ViERenderer(void) {
   if (render_callback_)
     render_module_.DeleteIncomingRenderStream(render_id_);
 
-  if (incoming_external_callback_)
-    delete incoming_external_callback_;
+  delete incoming_external_callback_;
 }
 
 int32_t ViERenderer::StartRender() {
@@ -50,11 +49,6 @@ int32_t ViERenderer::StopRender() {
   return render_module_.StopRender(render_id_);
 }
 
-int32_t ViERenderer::GetLastRenderedFrame(const int32_t renderID,
-                                          I420VideoFrame& video_frame) {
-  return render_module_.GetLastRenderedFrame(renderID, video_frame);
-}
-
 int ViERenderer::SetExpectedRenderDelay(int render_delay) {
   return render_module_.SetExpectedRenderDelay(render_id_, render_delay);
 }
@@ -72,14 +66,6 @@ VideoRender& ViERenderer::RenderModule() {
   return render_module_;
 }
 
-int32_t ViERenderer::EnableMirroring(const int32_t render_id,
-                                     const bool enable,
-                                     const bool mirror_xaxis,
-                                     const bool mirror_yaxis) {
-  return render_module_.MirrorRenderStream(render_id, enable, mirror_xaxis,
-                                           mirror_yaxis);
-}
-
 int32_t ViERenderer::SetTimeoutImage(const I420VideoFrame& timeout_image,
                                      const int32_t timeout_value) {
   return render_module_.SetTimeoutImage(render_id_, timeout_image,
@@ -137,8 +123,7 @@ int32_t ViERenderer::Init(const uint32_t z_order,
 
 void ViERenderer::DeliverFrame(int id,
                                I420VideoFrame* video_frame,
-                               int num_csrcs,
-                               const uint32_t CSRC[kRtpCsrcSize]) {
+                               const std::vector& csrcs) {
   render_callback_->RenderFrame(render_id_, *video_frame);
 }
 
@@ -159,8 +144,7 @@ ViEExternalRendererImpl::ViEExternalRendererImpl()
     : external_renderer_(NULL),
       external_renderer_format_(kVideoUnknown),
       external_renderer_width_(0),
-      external_renderer_height_(0),
-      converted_frame_(new VideoFrame()) {
+      external_renderer_height_(0) {
 }
 
 int ViEExternalRendererImpl::SetViEExternalRenderer(
@@ -173,7 +157,24 @@ int ViEExternalRendererImpl::SetViEExternalRenderer(
 
 int32_t ViEExternalRendererImpl::RenderFrame(
     const uint32_t stream_id,
-    I420VideoFrame&   video_frame) {
+    const I420VideoFrame& video_frame) {
+  if (external_renderer_format_ != kVideoI420)
+    return ConvertAndRenderFrame(stream_id, video_frame);
+
+  // Fast path for I420 without frame copy.
+  NotifyFrameSizeChange(stream_id, video_frame);
+  if (video_frame.native_handle() == NULL ||
+      external_renderer_->IsTextureSupported()) {
+    external_renderer_->DeliverI420Frame(video_frame);
+  } else {
+    // TODO(wuchengli): readback the pixels and deliver the frame.
+  }
+  return 0;
+}
+
+int32_t ViEExternalRendererImpl::ConvertAndRenderFrame(
+    uint32_t stream_id,
+    const I420VideoFrame& video_frame) {
   if (video_frame.native_handle() != NULL) {
     NotifyFrameSizeChange(stream_id, video_frame);
 
@@ -190,31 +191,20 @@ int32_t ViEExternalRendererImpl::RenderFrame(
     return 0;
   }
 
-  VideoFrame* out_frame = converted_frame_.get();
-
   // Convert to requested format.
   VideoType type =
       RawVideoTypeToCommonVideoVideoType(external_renderer_format_);
-  int buffer_size = CalcBufferSize(type, video_frame.width(),
-                                   video_frame.height());
-  if (buffer_size <= 0) {
+  size_t buffer_size = CalcBufferSize(type, video_frame.width(),
+                                      video_frame.height());
+  if (buffer_size == 0) {
     // Unsupported video format.
     assert(false);
     return -1;
   }
-  converted_frame_->VerifyAndAllocate(buffer_size);
+  converted_frame_.resize(buffer_size);
+  uint8_t* out_frame = &converted_frame_[0];
 
   switch (external_renderer_format_) {
-    case kVideoI420: {
-      // TODO(mikhal): need to copy the buffer as is.
-      // can the output here be a I420 frame?
-      int length = ExtractBuffer(video_frame, out_frame->Size(),
-                                 out_frame->Buffer());
-      if (length < 0)
-        return -1;
-      out_frame->SetLength(length);
-      break;
-    }
     case kVideoYV12:
     case kVideoYUY2:
     case kVideoUYVY:
@@ -222,13 +212,9 @@ int32_t ViEExternalRendererImpl::RenderFrame(
     case kVideoRGB24:
     case kVideoRGB565:
     case kVideoARGB4444:
-    case kVideoARGB1555 :
-      {
-        if (ConvertFromI420(video_frame, type, 0,
-                            converted_frame_->Buffer()) < 0)
-          return -1;
-        converted_frame_->SetLength(buffer_size);
-      }
+    case kVideoARGB1555:
+      if (ConvertFromI420(video_frame, type, 0, out_frame) < 0)
+        return -1;
       break;
     case kVideoIYUV:
       // no conversion available
@@ -242,8 +228,8 @@ int32_t ViEExternalRendererImpl::RenderFrame(
   NotifyFrameSizeChange(stream_id, video_frame);
 
   if (out_frame) {
-    external_renderer_->DeliverFrame(out_frame->Buffer(),
-                                     out_frame->Length(),
+    external_renderer_->DeliverFrame(out_frame,
+                                     converted_frame_.size(),
                                      video_frame.timestamp(),
                                      video_frame.ntp_time_ms(),
                                      video_frame.render_time_ms(),
@@ -254,7 +240,7 @@ int32_t ViEExternalRendererImpl::RenderFrame(
 
 void ViEExternalRendererImpl::NotifyFrameSizeChange(
     const uint32_t stream_id,
-    I420VideoFrame& video_frame) {
+    const I420VideoFrame& video_frame) {
   if (external_renderer_width_ != video_frame.width() ||
       external_renderer_height_ != video_frame.height()) {
     external_renderer_width_ = video_frame.width();
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h b/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h
index 907735cad7..5a7fdc19f1 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_renderer.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
 #define WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
 
+#include 
+
 #include "webrtc/modules/video_render/include/video_render_defines.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/video_engine/include/vie_render.h"
 #include "webrtc/video_engine/vie_frame_provider_base.h"
 
@@ -32,17 +33,19 @@ class ViEExternalRendererImpl : public VideoRenderCallback {
 
   // Implements VideoRenderCallback.
   virtual int32_t RenderFrame(const uint32_t stream_id,
-                              I420VideoFrame& video_frame);
+                              const I420VideoFrame& video_frame);
 
  private:
   void NotifyFrameSizeChange(const uint32_t stream_id,
-                             I420VideoFrame& video_frame);
+                             const I420VideoFrame& video_frame);
+  int32_t ConvertAndRenderFrame(uint32_t stream_id,
+                                const I420VideoFrame& video_frame);
   ExternalRenderer* external_renderer_;
   RawVideoType external_renderer_format_;
   int external_renderer_width_;
   int external_renderer_height_;
   // Converted_frame_ in color format specified by render_format_.
-  scoped_ptr converted_frame_;
+  std::vector converted_frame_;
 };
 
 class ViERenderer: public ViEFrameCallback {
@@ -61,9 +64,6 @@ class ViERenderer: public ViEFrameCallback {
   int32_t StartRender();
   int32_t StopRender();
 
-  int32_t GetLastRenderedFrame(const int32_t renderID,
-                               I420VideoFrame& video_frame);
-
   int SetExpectedRenderDelay(int render_delay);
 
   int32_t ConfigureRenderer(const unsigned int z_order,
@@ -74,11 +74,6 @@ class ViERenderer: public ViEFrameCallback {
 
   VideoRender& RenderModule();
 
-  int32_t EnableMirroring(const int32_t render_id,
-                          const bool enable,
-                          const bool mirror_xaxis,
-                          const bool mirror_yaxis);
-
   int32_t SetTimeoutImage(const I420VideoFrame& timeout_image,
                           const int32_t timeout_value);
   int32_t SetRenderStartImage(const I420VideoFrame& start_image);
@@ -103,8 +98,7 @@ class ViERenderer: public ViEFrameCallback {
   // Implement ViEFrameCallback
   virtual void DeliverFrame(int id,
                             I420VideoFrame* video_frame,
-                            int num_csrcs = 0,
-                            const uint32_t CSRC[kRtpCsrcSize] = NULL);
+                            const std::vector& csrcs);
   virtual void DelayChanged(int id, int frame_delay);
   virtual int GetPreferedFrameSettings(int* width,
                                        int* height,
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc b/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc
index 0a6eb4dc51..b8f8bd33a2 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.cc
@@ -207,25 +207,10 @@ int ViERTP_RTCPImpl::SetRtxSendPayloadType(const int video_channel,
   return 0;
 }
 
-int ViERTP_RTCPImpl::SetPadWithRedundantPayloads(int video_channel,
-                                                 bool enable) {
-  LOG_F(LS_INFO) << "channel: " << video_channel
-                 << " pad with redundant payloads: " << (enable ? "enable" :
-                 "disable");
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->SetPadWithRedundantPayloads(enable);
-  return 0;
-}
-
 int ViERTP_RTCPImpl::SetRtxReceivePayloadType(const int video_channel,
                                               const uint8_t payload_type) {
   LOG_F(LS_INFO) << "channel: " << video_channel
-               << " payload_type: " << static_cast(payload_type);
+                 << " payload_type: " << static_cast(payload_type);
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -292,10 +277,7 @@ int ViERTP_RTCPImpl::SetRTCPStatus(const int video_channel,
   }
 
   RTCPMethod module_mode = ViERTCPModeToRTCPMethod(rtcp_mode);
-  if (vie_channel->SetRTCPMode(module_mode) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
+  vie_channel->SetRTCPMode(module_mode);
   return 0;
 }
 
@@ -307,11 +289,7 @@ int ViERTP_RTCPImpl::GetRTCPStatus(const int video_channel,
     shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
     return -1;
   }
-  RTCPMethod module_mode = kRtcpOff;
-  if (vie_channel->GetRTCPMode(&module_mode) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
+  RTCPMethod module_mode = vie_channel->GetRTCPMode();
   rtcp_mode = RTCPMethodToViERTCPMode(module_mode);
   return 0;
 }
@@ -401,8 +379,8 @@ int ViERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
     shared_data_->SetLastError(kViERtpRtcpNotSending);
     return -1;
   }
-  RTCPMethod method;
-  if (vie_channel->GetRTCPMode(&method) != 0 || method == kRtcpOff) {
+  RTCPMethod method = vie_channel->GetRTCPMode();
+  if (method == kRtcpOff) {
     shared_data_->SetLastError(kViERtpRtcpRtcpDisabled);
     return -1;
   }
@@ -435,7 +413,8 @@ int ViERTP_RTCPImpl::SetNACKStatus(const int video_channel, const bool enable) {
     shared_data_->SetLastError(kViERtpRtcpUnknownError);
     return -1;
   }
-  vie_encoder->UpdateProtectionMethod(enable);
+  vie_encoder->UpdateProtectionMethod(enable,
+                                      vie_channel->IsSendingFecEnabled());
   return 0;
 }
 
@@ -463,7 +442,7 @@ int ViERTP_RTCPImpl::SetFECStatus(const int video_channel, const bool enable,
     shared_data_->SetLastError(kViERtpRtcpUnknownError);
     return -1;
   }
-  vie_encoder->UpdateProtectionMethod(false);
+  vie_encoder->UpdateProtectionMethod(false, true);
   return 0;
 }
 
@@ -496,7 +475,7 @@ int ViERTP_RTCPImpl::SetHybridNACKFECStatus(
     shared_data_->SetLastError(kViERtpRtcpUnknownError);
     return -1;
   }
-  vie_encoder->UpdateProtectionMethod(enable);
+  vie_encoder->UpdateProtectionMethod(enable, enable);
   return 0;
 }
 
@@ -576,10 +555,7 @@ int ViERTP_RTCPImpl::SetTMMBRStatus(const int video_channel,
     shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
     return -1;
   }
-  if (vie_channel->EnableTMMBR(enable) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpUnknownError);
-    return -1;
-  }
+  vie_channel->EnableTMMBR(enable);
   return 0;
 }
 
@@ -619,7 +595,7 @@ int ViERTP_RTCPImpl::SetReceiveTimestampOffsetStatus(int video_channel,
                                                      bool enable,
                                                      int id) {
   LOG_F(LS_INFO) << "channel: " << video_channel
-                 << "enable: " << (enable ? "on" : "off") << " id: " << id;
+                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -637,7 +613,7 @@ int ViERTP_RTCPImpl::SetSendAbsoluteSendTimeStatus(int video_channel,
                                                    bool enable,
                                                    int id) {
   LOG_F(LS_INFO) << "channel: " << video_channel
-                 << "enable: " << (enable ? "on" : "off") << " id: " << id;
+                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
 
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
@@ -656,7 +632,7 @@ int ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(int video_channel,
                                                       bool enable,
                                                       int id) {
   LOG_F(LS_INFO) << "channel: " << video_channel
-                 << "enable: " << (enable ? "on" : "off") << " id: " << id;
+                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -670,6 +646,43 @@ int ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(int video_channel,
   return 0;
 }
 
+int ViERTP_RTCPImpl::SetSendVideoRotationStatus(int video_channel,
+                                                bool enable,
+                                                int id) {
+  LOG_F(LS_INFO) << "channel: " << video_channel
+                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
+
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(video_channel);
+  if (!vie_channel) {
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  if (vie_channel->SetSendVideoRotationStatus(enable, id) != 0) {
+    shared_data_->SetLastError(kViERtpRtcpUnknownError);
+    return -1;
+  }
+  return 0;
+}
+
+int ViERTP_RTCPImpl::SetReceiveVideoRotationStatus(int video_channel,
+                                                   bool enable,
+                                                   int id) {
+  LOG_F(LS_INFO) << "channel: " << video_channel
+                 << " enable: " << (enable ? "on" : "off") << " id: " << id;
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(video_channel);
+  if (!vie_channel) {
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  if (vie_channel->SetReceiveVideoRotationStatus(enable, id) != 0) {
+    shared_data_->SetLastError(kViERtpRtcpUnknownError);
+    return -1;
+  }
+  return 0;
+}
+
 int ViERTP_RTCPImpl::SetRtcpXrRrtrStatus(int video_channel, bool enable) {
   LOG_F(LS_INFO) << "channel: " << video_channel
                  << " enable: " << (enable ? "on" : "off");
@@ -725,7 +738,7 @@ int ViERTP_RTCPImpl::SetReservedTransmitBitrate(
 int ViERTP_RTCPImpl::GetReceiveChannelRtcpStatistics(
     const int video_channel,
     RtcpStatistics& basic_stats,
-    int& rtt_ms) const {
+    int64_t& rtt_ms) const {
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -750,7 +763,7 @@ int ViERTP_RTCPImpl::GetReceiveChannelRtcpStatistics(
 
 int ViERTP_RTCPImpl::GetSendChannelRtcpStatistics(const int video_channel,
                                                   RtcpStatistics& basic_stats,
-                                                  int& rtt_ms) const {
+                                                  int64_t& rtt_ms) const {
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
@@ -782,27 +795,39 @@ int ViERTP_RTCPImpl::GetRtpStatistics(const int video_channel,
     shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
     return -1;
   }
-  if (vie_channel->GetRtpStatistics(&sent.bytes,
-                                    &sent.packets,
-                                    &received.bytes,
-                                    &received.packets) != 0) {
+  if (vie_channel->GetRtpStatistics(&sent.transmitted.payload_bytes,
+                                    &sent.transmitted.packets,
+                                    &received.transmitted.payload_bytes,
+                                    &received.transmitted.packets) != 0) {
     shared_data_->SetLastError(kViERtpRtcpUnknownError);
     return -1;
   }
   return 0;
 }
 
-int ViERTP_RTCPImpl::GetRtcpPacketTypeCounters(
+int ViERTP_RTCPImpl::GetSendRtcpPacketTypeCounter(
     int video_channel,
-    RtcpPacketTypeCounter* packets_sent,
-    RtcpPacketTypeCounter* packets_received) const {
+    RtcpPacketTypeCounter* packet_counter) const {
   ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
     shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
     return -1;
   }
-  vie_channel->GetRtcpPacketTypeCounters(packets_sent, packets_received);
+  vie_channel->GetSendRtcpPacketTypeCounter(packet_counter);
+  return 0;
+}
+
+int ViERTP_RTCPImpl::GetReceiveRtcpPacketTypeCounter(
+    int video_channel,
+    RtcpPacketTypeCounter* packet_counter) const {
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(video_channel);
+  if (!vie_channel) {
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  vie_channel->GetReceiveRtcpPacketTypeCounter(packet_counter);
   return 0;
 }
 
@@ -861,28 +886,12 @@ int ViERTP_RTCPImpl::GetEstimatedReceiveBandwidth(
   return 0;
 }
 
-int ViERTP_RTCPImpl::GetReceiveBandwidthEstimatorStats(
-    const int video_channel,
-    ReceiveBandwidthEstimatorStats* output) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  vie_channel->GetReceiveBandwidthEstimatorStats(output);
-  return 0;
-}
-
 int ViERTP_RTCPImpl::GetPacerQueuingDelayMs(
-    const int video_channel, int* delay_ms) const {
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEEncoder* vie_encoder = cs.Encoder(video_channel);
-  if (!vie_encoder) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    const int video_channel, int64_t* delay_ms) const {
+  if (!shared_data_->channel_manager()->GetPacerQueuingDelayMs(video_channel,
+                                                               delay_ms)) {
     return -1;
   }
-  *delay_ms = vie_encoder->PacerQueuingDelayMs();
   return 0;
 }
 
@@ -954,37 +963,6 @@ int ViERTP_RTCPImpl::DeregisterRTPObserver(const int video_channel) {
   return 0;
 }
 
-int ViERTP_RTCPImpl::RegisterRTCPObserver(const int video_channel,
-                                          ViERTCPObserver& observer) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterRtcpObserver(&observer) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterRTCPObserver(const int video_channel) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterRtcpObserver(NULL) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
-    return -1;
-  }
-  return 0;
-}
-
 int ViERTP_RTCPImpl::RegisterSendChannelRtcpStatisticsCallback(
   int video_channel, RtcpStatisticsCallback* callback) {
   LOG_F(LS_INFO) << "channel " << video_channel;
@@ -1123,4 +1101,18 @@ int ViERTP_RTCPImpl::DeregisterSendFrameCountObserver(
   vie_channel->RegisterSendFrameCountObserver(NULL);
   return 0;
 }
+
+int ViERTP_RTCPImpl::RegisterRtcpPacketTypeCounterObserver(
+    int video_channel,
+    RtcpPacketTypeCounterObserver* observer) {
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  ViEChannel* vie_channel = cs.Channel(video_channel);
+  if (!vie_channel) {
+    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
+    return -1;
+  }
+  vie_channel->RegisterRtcpPacketTypeCounterObserver(observer);
+  return 0;
+}
+
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h b/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h
index 5953da2f5e..8f92a72525 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_rtp_rtcp_impl.h
@@ -41,16 +41,14 @@ class ViERTP_RTCPImpl
                              unsigned int CSRCs[kRtpCsrcSize]) const;
   virtual int SetRtxSendPayloadType(const int video_channel,
                                     const uint8_t payload_type);
-  virtual int SetPadWithRedundantPayloads(int video_channel, bool enable);
   virtual int SetRtxReceivePayloadType(const int video_channel,
                                        const uint8_t payload_type);
   virtual int SetStartSequenceNumber(const int video_channel,
                                      uint16_t sequence_number);
-  virtual void SetRtpStateForSsrc(int video_channel,
-                                  uint32_t ssrc,
-                                  const RtpState& rtp_state) OVERRIDE;
-  virtual RtpState GetRtpStateForSsrc(int video_channel,
-                                      uint32_t ssrc) OVERRIDE;
+  void SetRtpStateForSsrc(int video_channel,
+                          uint32_t ssrc,
+                          const RtpState& rtp_state) override;
+  RtpState GetRtpStateForSsrc(int video_channel, uint32_t ssrc) override;
   virtual int SetRTCPStatus(const int video_channel,
                             const ViERTCPMode rtcp_mode);
   virtual int GetRTCPStatus(const int video_channel,
@@ -101,6 +99,12 @@ class ViERTP_RTCPImpl
   virtual int SetReceiveAbsoluteSendTimeStatus(int video_channel,
                                                bool enable,
                                                int id);
+  virtual int SetSendVideoRotationStatus(int video_channel,
+                                         bool enable,
+                                         int id);
+  virtual int SetReceiveVideoRotationStatus(int video_channel,
+                                            bool enable,
+                                            int id);
   virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable);
   virtual int SetTransmissionSmoothingStatus(int video_channel, bool enable);
   virtual int SetMinTransmitBitrate(int video_channel,
@@ -109,17 +113,19 @@ class ViERTP_RTCPImpl
       int video_channel, unsigned int reserved_transmit_bitrate_bps);
   virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
                                               RtcpStatistics& basic_stats,
-                                              int& rtt_ms) const;
+                                              int64_t& rtt_ms) const;
   virtual int GetSendChannelRtcpStatistics(const int video_channel,
                                            RtcpStatistics& basic_stats,
-                                           int& rtt_ms) const;
+                                           int64_t& rtt_ms) const;
   virtual int GetRtpStatistics(const int video_channel,
                                StreamDataCounters& sent,
                                StreamDataCounters& received) const;
-  virtual int GetRtcpPacketTypeCounters(
+  virtual int GetSendRtcpPacketTypeCounter(
       int video_channel,
-      RtcpPacketTypeCounter* packets_sent,
-      RtcpPacketTypeCounter* packets_received) const;
+      RtcpPacketTypeCounter* packet_counter) const;
+  virtual int GetReceiveRtcpPacketTypeCounter(
+      int video_channel,
+      RtcpPacketTypeCounter* packet_counter) const;
   virtual int GetRemoteRTCPSenderInfo(const int video_channel,
                                       SenderInfo* sender_info) const;
   virtual int GetBandwidthUsage(const int video_channel,
@@ -133,10 +139,8 @@ class ViERTP_RTCPImpl
   virtual int GetEstimatedReceiveBandwidth(
       const int video_channel,
       unsigned int* estimated_bandwidth) const;
-  virtual int GetReceiveBandwidthEstimatorStats(
-      const int video_channel, ReceiveBandwidthEstimatorStats* output) const;
   virtual int GetPacerQueuingDelayMs(const int video_channel,
-                                     int* delay_ms) const;
+                                     int64_t* delay_ms) const;
   virtual int StartRTPDump(const int video_channel,
                            const char file_nameUTF8[1024],
                            RTPDirections direction);
@@ -144,18 +148,15 @@ class ViERTP_RTCPImpl
   virtual int RegisterRTPObserver(const int video_channel,
                                   ViERTPObserver& observer);
   virtual int DeregisterRTPObserver(const int video_channel);
-  virtual int RegisterRTCPObserver(const int video_channel,
-                                   ViERTCPObserver& observer);
-  virtual int DeregisterRTCPObserver(const int video_channel);
 
   virtual int RegisterSendChannelRtcpStatisticsCallback(
       int channel, RtcpStatisticsCallback* callback);
   virtual int DeregisterSendChannelRtcpStatisticsCallback(
       int channel, RtcpStatisticsCallback* callback);
   virtual int RegisterReceiveChannelRtcpStatisticsCallback(
-        int channel, RtcpStatisticsCallback* callback);
-    virtual int DeregisterReceiveChannelRtcpStatisticsCallback(
-        int channel, RtcpStatisticsCallback* callback);
+      int channel, RtcpStatisticsCallback* callback);
+  virtual int DeregisterReceiveChannelRtcpStatisticsCallback(
+      int channel, RtcpStatisticsCallback* callback);
   virtual int RegisterSendChannelRtpStatisticsCallback(
       int channel, StreamDataCountersCallback* callback);
   virtual int DeregisterSendChannelRtpStatisticsCallback(
@@ -172,6 +173,9 @@ class ViERTP_RTCPImpl
       int channel, FrameCountObserver* callback);
   virtual int DeregisterSendFrameCountObserver(
       int channel, FrameCountObserver* callback);
+  int RegisterRtcpPacketTypeCounterObserver(
+      int video_channel,
+      RtcpPacketTypeCounterObserver* observer) override;
 
  protected:
   explicit ViERTP_RTCPImpl(ViESharedData* shared_data);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc b/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc
index 28bf390334..db7a5b116d 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_sender.cc
@@ -85,7 +85,7 @@ int ViESender::StopRTPDump() {
   return 0;
 }
 
-int ViESender::SendPacket(int vie_id, const void* data, int len) {
+int ViESender::SendPacket(int vie_id, const void* data, size_t len) {
   CriticalSectionScoped cs(critsect_.get());
   if (!transport_) {
     // No transport
@@ -94,14 +94,13 @@ int ViESender::SendPacket(int vie_id, const void* data, int len) {
   assert(ChannelId(vie_id) == channel_id_);
 
   if (rtp_dump_) {
-    rtp_dump_->DumpPacket(static_cast(data),
-                          static_cast(len));
+    rtp_dump_->DumpPacket(static_cast(data), len);
   }
 
   return transport_->SendPacket(channel_id_, data, len);
 }
 
-int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
+int ViESender::SendRTCPPacket(int vie_id, const void* data, size_t len) {
   CriticalSectionScoped cs(critsect_.get());
   if (!transport_) {
     return -1;
@@ -109,8 +108,7 @@ int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
   assert(ChannelId(vie_id) == channel_id_);
 
   if (rtp_dump_) {
-    rtp_dump_->DumpPacket(static_cast(data),
-                          static_cast(len));
+    rtp_dump_->DumpPacket(static_cast(data), len);
   }
 
   return transport_->SendRTCPPacket(channel_id_, data, len);
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sender.h b/media/webrtc/trunk/webrtc/video_engine/vie_sender.h
index f910cb1af6..3e6bd95549 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sender.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_sender.h
@@ -13,9 +13,9 @@
 #ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
 #define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_engine/vie_defines.h"
 
@@ -40,13 +40,13 @@ class ViESender: public Transport {
   int StopRTPDump();
 
   // Implements Transport.
-  virtual int SendPacket(int vie_id, const void* data, int len) OVERRIDE;
-  virtual int SendRTCPPacket(int vie_id, const void* data, int len) OVERRIDE;
+  int SendPacket(int vie_id, const void* data, size_t len) override;
+  int SendRTCPPacket(int vie_id, const void* data, size_t len) override;
 
  private:
   const int32_t channel_id_;
 
-  scoped_ptr critsect_;
+  rtc::scoped_ptr critsect_;
 
   Transport* transport_;
   RtpDump* rtp_dump_;
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc b/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc
index 6a6989430e..39ad2843dc 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.cc
@@ -24,12 +24,12 @@ ViESharedData::ViESharedData(const Config& config)
       channel_manager_(new ViEChannelManager(0, number_cores_, config)),
       input_manager_(new ViEInputManager(0, config)),
       render_manager_(new ViERenderManager(0)),
-      module_process_thread_(ProcessThread::CreateProcessThread()),
+      module_process_thread_(ProcessThread::Create()),
       load_manager_(NULL),
       last_error_(0) {
   Trace::CreateTrace();
-  channel_manager_->SetModuleProcessThread(module_process_thread_);
-  input_manager_->SetModuleProcessThread(module_process_thread_);
+  channel_manager_->SetModuleProcessThread(module_process_thread_.get());
+  input_manager_->SetModuleProcessThread(module_process_thread_.get());
   module_process_thread_->Start();
 }
 
@@ -40,7 +40,6 @@ ViESharedData::~ViESharedData() {
   render_manager_.reset();
 
   module_process_thread_->Stop();
-  ProcessThread::DestroyProcessThread(module_process_thread_);
   Trace::ReturnTrace();
 }
 
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h b/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h
index 1eac5c8daa..53d4f2d96c 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_shared_data.h
@@ -16,7 +16,7 @@
 
 #include 
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -51,10 +51,10 @@ class ViESharedData {
  private:
   const int number_cores_;
 
-  scoped_ptr channel_manager_;
-  scoped_ptr input_manager_;
-  scoped_ptr render_manager_;
-  ProcessThread* module_process_thread_;
+  rtc::scoped_ptr channel_manager_;
+  rtc::scoped_ptr input_manager_;
+  rtc::scoped_ptr render_manager_;
+  rtc::scoped_ptr module_process_thread_;
   // Owned by PeerConnection, not ViEEngine
   CPULoadStateCallbackInvoker* load_manager_;
   mutable int last_error_;
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc b/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc
index c24087139f..cf592927a8 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.cc
@@ -22,8 +22,6 @@
 
 namespace webrtc {
 
-enum { kSyncInterval = 1000};
-
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
   if (!receiver.Timestamp(&stream->latest_timestamp))
@@ -93,9 +91,9 @@ int ViESyncModule::VoiceChannel() {
   return voe_channel_id_;
 }
 
-int32_t ViESyncModule::TimeUntilNextProcess() {
-  return static_cast(kSyncInterval -
-      (TickTime::Now() - last_sync_time_).Milliseconds());
+int64_t ViESyncModule::TimeUntilNextProcess() {
+  const int64_t kSyncIntervalMs = 1000;
+  return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
 }
 
 int32_t ViESyncModule::Process() {
diff --git a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h b/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h
index e6ca5fd275..a75d716c08 100644
--- a/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h
+++ b/media/webrtc/trunk/webrtc/video_engine/vie_sync_module.h
@@ -14,8 +14,8 @@
 #ifndef WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
 #define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/interface/module.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/video_engine/stream_synchronization.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
@@ -45,11 +45,11 @@ class ViESyncModule : public Module {
   int SetTargetBufferingDelay(int target_delay_ms);
 
   // Implements Module.
-  virtual int32_t TimeUntilNextProcess() OVERRIDE;
-  virtual int32_t Process() OVERRIDE;
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
 
  private:
-  scoped_ptr data_cs_;
+  rtc::scoped_ptr data_cs_;
   VideoCodingModule* vcm_;
   ViEChannel* vie_channel_;
   RtpReceiver* video_receiver_;
@@ -57,7 +57,7 @@ class ViESyncModule : public Module {
   int voe_channel_id_;
   VoEVideoSync* voe_sync_interface_;
   TickTime last_sync_time_;
-  scoped_ptr sync_;
+  rtc::scoped_ptr sync_;
   StreamSynchronization::Measurements audio_measurement_;
   StreamSynchronization::Measurements video_measurement_;
 };
diff --git a/media/webrtc/trunk/webrtc/video_engine_tests.isolate b/media/webrtc/trunk/webrtc/video_engine_tests.isolate
index fc840829b9..5aa962323d 100644
--- a/media/webrtc/trunk/webrtc/video_engine_tests.isolate
+++ b/media/webrtc/trunk/webrtc/video_engine_tests.isolate
@@ -7,11 +7,10 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
+    ['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', {
       'variables': {
         'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
+          '<(DEPTH)/resources/foreman_cif_short.yuv',
         ],
       },
     }],
@@ -22,7 +21,6 @@
         ],
         'files': [
           '<(DEPTH)/DEPS',
-          '<(DEPTH)/resources/foreman_cif_short.yuv',
           '<(PRODUCT_DIR)/video_engine_tests<(EXECUTABLE_SUFFIX)',
         ],
       },
diff --git a/media/webrtc/trunk/webrtc/video_frame.h b/media/webrtc/trunk/webrtc/video_frame.h
index f76b9af373..bd3e2b8a63 100644
--- a/media/webrtc/trunk/webrtc/video_frame.h
+++ b/media/webrtc/trunk/webrtc/video_frame.h
@@ -11,160 +11,161 @@
 #ifndef WEBRTC_VIDEO_FRAME_H_
 #define WEBRTC_VIDEO_FRAME_H_
 
-#include 
-
-#include "webrtc/common_video/plane.h"
-// TODO(pbos): Remove scoped_refptr include (and AddRef/Release if they're not
-// used).
-#include "webrtc/system_wrappers/interface/scoped_refptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/common_video/interface/native_handle.h"
+#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/rotation.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-enum PlaneType {
-  kYPlane = 0,
-  kUPlane = 1,
-  kVPlane = 2,
-  kNumOfPlanes = 3
-};
-
 class I420VideoFrame {
  public:
   I420VideoFrame();
-  virtual ~I420VideoFrame();
-  // Infrastructure for refCount implementation.
-  // Implements dummy functions for reference counting so that non reference
-  // counted instantiation can be done. These functions should not be called
-  // when creating the frame with new I420VideoFrame().
-  // Note: do not pass a I420VideoFrame created with new I420VideoFrame() or
-  // equivalent to a scoped_refptr or memory leak will occur.
-  virtual int32_t AddRef() {
-    assert(false);
-    return -1;
-  }
-  virtual int32_t Release() {
-    assert(false);
-    return -1;
-  }
+  I420VideoFrame(const rtc::scoped_refptr& buffer,
+                 uint32_t timestamp,
+                 int64_t render_time_ms,
+                 VideoRotation rotation);
+  I420VideoFrame(NativeHandle* handle,
+                 int width,
+                 int height,
+                 uint32_t timestamp,
+                 int64_t render_time_ms);
+
+  // TODO(pbos): Make all create/copy functions void, they should not be able to
+  // fail (which should be DCHECK/CHECKed instead).
 
   // CreateEmptyFrame: Sets frame dimensions and allocates buffers based
   // on set dimensions - height and plane stride.
   // If required size is bigger than the allocated one, new buffers of adequate
   // size will be allocated.
   // Return value: 0 on success, -1 on error.
-  virtual int CreateEmptyFrame(int width,
-                               int height,
-                               int stride_y,
-                               int stride_u,
-                               int stride_v);
+  int CreateEmptyFrame(int width,
+                       int height,
+                       int stride_y,
+                       int stride_u,
+                       int stride_v);
 
   // CreateFrame: Sets the frame's members and buffers. If required size is
   // bigger than allocated one, new buffers of adequate size will be allocated.
   // Return value: 0 on success, -1 on error.
-  virtual int CreateFrame(int size_y,
-                          const uint8_t* buffer_y,
-                          int size_u,
-                          const uint8_t* buffer_u,
-                          int size_v,
-                          const uint8_t* buffer_v,
-                          int width,
-                          int height,
-                          int stride_y,
-                          int stride_u,
-                          int stride_v);
+  int CreateFrame(const uint8_t* buffer_y,
+                  const uint8_t* buffer_u,
+                  const uint8_t* buffer_v,
+                  int width,
+                  int height,
+                  int stride_y,
+                  int stride_u,
+                  int stride_v);
 
-  // Copy frame: If required size is bigger than allocated one, new buffers of
-  // adequate size will be allocated.
+  // TODO(guoweis): remove the previous CreateFrame when chromium has this code.
+  int CreateFrame(const uint8_t* buffer_y,
+                  const uint8_t* buffer_u,
+                  const uint8_t* buffer_v,
+                  int width,
+                  int height,
+                  int stride_y,
+                  int stride_u,
+                  int stride_v,
+                  VideoRotation rotation);
+
+  // CreateFrame: Sets the frame's members and buffers. If required size is
+  // bigger than allocated one, new buffers of adequate size will be allocated.
+  // |buffer| must be a packed I420 buffer.
   // Return value: 0 on success, -1 on error.
-  virtual int CopyFrame(const I420VideoFrame& videoFrame);
+  int CreateFrame(const uint8_t* buffer,
+                  int width,
+                  int height,
+                  VideoRotation rotation);
 
-  // Make a copy of |this|. The caller owns the returned frame.
-  // Return value: a new frame on success, NULL on error.
-  virtual I420VideoFrame* CloneFrame() const;
+  // Deep copy frame: If required size is bigger than allocated one, new
+  // buffers of adequate size will be allocated.
+  // Return value: 0 on success, -1 on error.
+  int CopyFrame(const I420VideoFrame& videoFrame);
 
-  // Swap Frame.
-  virtual void SwapFrame(I420VideoFrame* videoFrame);
+  // Creates a shallow copy of |videoFrame|, i.e, the this object will retain a
+  // reference to the video buffer also retained by |videoFrame|.
+  void ShallowCopy(const I420VideoFrame& videoFrame);
+
+  // Release frame buffer and reset time stamps.
+  void Reset();
 
   // Get pointer to buffer per plane.
-  virtual uint8_t* buffer(PlaneType type);
+  uint8_t* buffer(PlaneType type);
   // Overloading with const.
-  virtual const uint8_t* buffer(PlaneType type) const;
+  const uint8_t* buffer(PlaneType type) const;
 
   // Get allocated size per plane.
-  virtual int allocated_size(PlaneType type) const;
+  int allocated_size(PlaneType type) const;
 
   // Get allocated stride per plane.
-  virtual int stride(PlaneType type) const;
-
-  // Set frame width.
-  virtual int set_width(int width);
-
-  // Set frame height.
-  virtual int set_height(int height);
+  int stride(PlaneType type) const;
 
   // Get frame width.
-  virtual int width() const { return width_; }
+  int width() const;
 
   // Get frame height.
-  virtual int height() const { return height_; }
+  int height() const;
 
   // Set frame timestamp (90kHz).
-  virtual void set_timestamp(uint32_t timestamp) { timestamp_ = timestamp; }
+  void set_timestamp(uint32_t timestamp) { timestamp_ = timestamp; }
 
   // Get frame timestamp (90kHz).
-  virtual uint32_t timestamp() const { return timestamp_; }
+  uint32_t timestamp() const { return timestamp_; }
 
   // Set capture ntp time in miliseconds.
-  virtual void set_ntp_time_ms(int64_t ntp_time_ms) {
+  void set_ntp_time_ms(int64_t ntp_time_ms) {
     ntp_time_ms_ = ntp_time_ms;
   }
 
   // Get capture ntp time in miliseconds.
-  virtual int64_t ntp_time_ms() const { return ntp_time_ms_; }
+  int64_t ntp_time_ms() const { return ntp_time_ms_; }
+
+  // Naming convention for Coordination of Video Orientation. Please see
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
+  //
+  // "pending rotation" or "pending" = a frame that has a VideoRotation > 0.
+  //
+  // "not pending" = a frame that has a VideoRotation == 0.
+  //
+  // "apply rotation" = modify a frame from being "pending" to being "not
+  //                    pending" rotation (a no-op for "unrotated").
+  //
+  VideoRotation rotation() const { return rotation_; }
+  void set_rotation(VideoRotation rotation) {
+    rotation_ = rotation;
+  }
 
   // Set render time in miliseconds.
-  virtual void set_render_time_ms(int64_t render_time_ms) {
+  void set_render_time_ms(int64_t render_time_ms) {
     render_time_ms_ = render_time_ms;
   }
 
   // Get render time in miliseconds.
-  virtual int64_t render_time_ms() const { return render_time_ms_; }
+  int64_t render_time_ms() const { return render_time_ms_; }
 
   // Return true if underlying plane buffers are of zero size, false if not.
-  virtual bool IsZeroSize() const;
-
-  // Reset underlying plane buffers sizes to 0. This function doesn't
-  // clear memory.
-  virtual void ResetSize();
+  bool IsZeroSize() const;
 
   // Return the handle of the underlying video frame. This is used when the
   // frame is backed by a texture. The object should be destroyed when it is no
   // longer in use, so the underlying resource can be freed.
-  virtual void* native_handle() const;
+  void* native_handle() const;
 
- protected:
-  // Verifies legality of parameters.
-  // Return value: 0 on success, -1 on error.
-  virtual int CheckDimensions(int width,
-                              int height,
-                              int stride_y,
-                              int stride_u,
-                              int stride_v);
+  // Return the underlying buffer.
+  rtc::scoped_refptr video_frame_buffer() const;
+
+  // Set the underlying buffer.
+  void set_video_frame_buffer(
+      const rtc::scoped_refptr& buffer);
 
  private:
-  // Get the pointer to a specific plane.
-  const Plane* GetPlane(PlaneType type) const;
-  // Overloading with non-const.
-  Plane* GetPlane(PlaneType type);
-
-  Plane y_plane_;
-  Plane u_plane_;
-  Plane v_plane_;
-  int width_;
-  int height_;
+  // An opaque reference counted handle that stores the pixel data.
+  rtc::scoped_refptr video_frame_buffer_;
   uint32_t timestamp_;
   int64_t ntp_time_ms_;
   int64_t render_time_ms_;
+  VideoRotation rotation_;
 };
 
 enum VideoFrameType {
@@ -178,42 +179,23 @@ enum VideoFrameType {
 // TODO(pbos): Rename EncodedFrame and reformat this class' members.
 class EncodedImage {
  public:
-  EncodedImage()
-      : _encodedWidth(0),
-        _encodedHeight(0),
-        _timeStamp(0),
-        capture_time_ms_(0),
-        _frameType(kDeltaFrame),
-        _buffer(NULL),
-        _length(0),
-        _size(0),
-        _completeFrame(false) {}
+  EncodedImage() : EncodedImage(nullptr, 0, 0) {}
+  EncodedImage(uint8_t* buffer, size_t length, size_t size)
+      : _buffer(buffer), _length(length), _size(size) {}
 
-  EncodedImage(uint8_t* buffer, uint32_t length, uint32_t size)
-      : _encodedWidth(0),
-        _encodedHeight(0),
-        _timeStamp(0),
-        ntp_time_ms_(0),
-        capture_time_ms_(0),
-        _frameType(kDeltaFrame),
-        _buffer(buffer),
-        _length(length),
-        _size(size),
-        _completeFrame(false) {}
-
-  uint32_t _encodedWidth;
-  uint32_t _encodedHeight;
-  uint32_t _timeStamp;
+  uint32_t _encodedWidth = 0;
+  uint32_t _encodedHeight = 0;
+  uint32_t _timeStamp = 0;
   // NTP time of the capture time in local timebase in milliseconds.
-  int64_t ntp_time_ms_;
-  int64_t capture_time_ms_;
-  VideoFrameType _frameType;
+  int64_t ntp_time_ms_ = 0;
+  int64_t capture_time_ms_ = 0;
+  // TODO(pbos): Use webrtc::FrameType directly (and remove VideoFrameType).
+  VideoFrameType _frameType = kDeltaFrame;
   uint8_t* _buffer;
-  uint32_t _length;
-  uint32_t _size;
-  bool _completeFrame;
+  size_t _length;
+  size_t _size;
+  bool _completeFrame = false;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_FRAME_H_
-
diff --git a/media/webrtc/trunk/webrtc/video_receive_stream.h b/media/webrtc/trunk/webrtc/video_receive_stream.h
index a8620d98c3..0f2151df03 100644
--- a/media/webrtc/trunk/webrtc/video_receive_stream.h
+++ b/media/webrtc/trunk/webrtc/video_receive_stream.h
@@ -39,8 +39,9 @@ class VideoReceiveStream {
     Decoder()
         : decoder(NULL),
           payload_type(0),
-          renderer(false),
+          is_renderer(false),
           expected_delay_ms(0) {}
+    std::string ToString() const;
 
     // The actual decoder instance.
     VideoDecoder* decoder;
@@ -54,7 +55,7 @@ class VideoReceiveStream {
     std::string payload_name;
 
     // 'true' if the decoder handles rendering as well.
-    bool renderer;
+    bool is_renderer;
 
     // The expected delay for decoding and rendering, i.e. the frame will be
     // delivered this many milliseconds, if possible, earlier than the ideal
@@ -63,22 +64,29 @@ class VideoReceiveStream {
     int expected_delay_ms;
   };
 
-  struct Stats : public SsrcStats {
-    Stats()
-        : network_frame_rate(0),
-          decode_frame_rate(0),
-          render_frame_rate(0),
-          avg_delay_ms(0),
-          discarded_packets(0),
-          ssrc(0) {}
+  struct Stats {
+    int network_frame_rate = 0;
+    int decode_frame_rate = 0;
+    int render_frame_rate = 0;
 
-    int network_frame_rate;
-    int decode_frame_rate;
-    int render_frame_rate;
-    int avg_delay_ms;
-    int discarded_packets;
-    uint32_t ssrc;
+    // Decoder stats.
+    FrameCounts frame_counts;
+    int decode_ms = 0;
+    int max_decode_ms = 0;
+    int current_delay_ms = 0;
+    int target_delay_ms = 0;
+    int jitter_buffer_ms = 0;
+    int min_playout_delay_ms = 0;
+    int render_delay_ms = 0;
+
+    int total_bitrate_bps = 0;
+    int discarded_packets = 0;
+
+    uint32_t ssrc = 0;
     std::string c_name;
+    StreamDataCounters rtp_stats;
+    RtcpPacketTypeCounter rtcp_packet_type_counts;
+    RtcpStatistics rtcp_stats;
   };
 
   struct Config {
@@ -89,6 +97,7 @@ class VideoReceiveStream {
           pre_decode_callback(NULL),
           pre_render_callback(NULL),
           target_delay_ms(0) {}
+    std::string ToString() const;
 
     // Decoders for every payload that we can receive.
     std::vector decoders;
@@ -100,6 +109,7 @@ class VideoReceiveStream {
             local_ssrc(0),
             rtcp_mode(newapi::kRtcpReducedSize),
             remb(true) {}
+      std::string ToString() const;
 
       // Synchronization source (stream identifier) to be received.
       uint32_t remote_ssrc;
diff --git a/media/webrtc/trunk/webrtc/video_renderer.h b/media/webrtc/trunk/webrtc/video_renderer.h
index 3946499475..34d339a508 100644
--- a/media/webrtc/trunk/webrtc/video_renderer.h
+++ b/media/webrtc/trunk/webrtc/video_renderer.h
@@ -23,6 +23,8 @@ class VideoRenderer {
   virtual void RenderFrame(const I420VideoFrame& video_frame,
                            int time_to_render_ms) = 0;
 
+  virtual bool IsTextureSupported() const = 0;
+
  protected:
   virtual ~VideoRenderer() {}
 };
diff --git a/media/webrtc/trunk/webrtc/video_send_stream.h b/media/webrtc/trunk/webrtc/video_send_stream.h
index a9aba94526..4ae0b6b58a 100644
--- a/media/webrtc/trunk/webrtc/video_send_stream.h
+++ b/media/webrtc/trunk/webrtc/video_send_stream.h
@@ -29,7 +29,7 @@ class VideoSendStreamInput {
   // These methods do not lock internally and must be called sequentially.
   // If your application switches input sources synchronization must be done
   // externally to make sure that any old frames are not delivered concurrently.
-  virtual void SwapFrame(I420VideoFrame* video_frame) = 0;
+  virtual void IncomingCapturedFrame(const I420VideoFrame& video_frame) = 0;
 
  protected:
   virtual ~VideoSendStreamInput() {}
@@ -37,17 +37,37 @@ class VideoSendStreamInput {
 
 class VideoSendStream {
  public:
+  struct StreamStats {
+    FrameCounts frame_counts;
+    int width = 0;
+    int height = 0;
+    // TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
+    int total_bitrate_bps = 0;
+    int retransmit_bitrate_bps = 0;
+    int avg_delay_ms = 0;
+    int max_delay_ms = 0;
+    StreamDataCounters rtp_stats;
+    RtcpPacketTypeCounter rtcp_packet_type_counts;
+    RtcpStatistics rtcp_stats;
+  };
+
   struct Stats {
     Stats()
         : input_frame_rate(0),
           encode_frame_rate(0),
+          avg_encode_time_ms(0),
+          encode_usage_percent(0),
+          target_media_bitrate_bps(0),
           media_bitrate_bps(0),
           suspended(false) {}
     int input_frame_rate;
     int encode_frame_rate;
+    int avg_encode_time_ms;
+    int encode_usage_percent;
+    int target_media_bitrate_bps;
     int media_bitrate_bps;
     bool suspended;
-    std::map substreams;
+    std::map substreams;
   };
 
   struct Config {
@@ -70,7 +90,7 @@ class VideoSendStream {
 
       // Uninitialized VideoEncoder instance to be used for encoding. Will be
       // initialized from inside the VideoSendStream.
-      webrtc::VideoEncoder* encoder;
+      VideoEncoder* encoder;
     } encoder_settings;
 
     static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
@@ -95,17 +115,13 @@ class VideoSendStream {
       // Settings for RTP retransmission payload format, see RFC 4588 for
       // details.
       struct Rtx {
-        Rtx() : payload_type(-1), pad_with_redundant_payloads(false) {}
+        Rtx() : payload_type(-1) {}
         std::string ToString() const;
         // SSRCs to use for the RTX streams.
         std::vector ssrcs;
 
         // Payload type to use for the RTX stream.
         int payload_type;
-        // Use redundant payloads to pad the bitrate. Instead of padding with
-        // randomized packets, we will preemptively retransmit media packets on
-        // the RTX stream.
-        bool pad_with_redundant_payloads;
       } rtx;
 
       // RTCP CNAME, see RFC 3550.
@@ -151,7 +167,7 @@ class VideoSendStream {
   // with the VideoStream settings.
   virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
-  virtual Stats GetStats() const = 0;
+  virtual Stats GetStats() = 0;
 
  protected:
   virtual ~VideoSendStream() {}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn b/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn
index 126d2c811e..c55dc8ed07 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/voice_engine/BUILD.gn
@@ -11,9 +11,6 @@ import("../build/webrtc.gni")
 source_set("voice_engine") {
 
   sources = [
-    "../common_types.h",
-    "../engine_configurations.h",
-    "../typedefs.h",
     "include/voe_audio_processing.h",
     "include/voe_base.h",
     "include/voe_codec.h",
@@ -82,6 +79,11 @@ source_set("voice_engine") {
 
   if (is_win) {
     defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
+
+    cflags = [
+      # TODO(kjellander): Bug 261: fix this warning.
+      "/wd4373",  # virtual function override.
+    ]
   }
 
   configs += [ "..:common_config" ]
@@ -94,6 +96,7 @@ source_set("voice_engine") {
   }
 
   deps = [
+    "..:webrtc_common",
     "../common_audio",
     "../modules/audio_coding",
     "../modules/audio_conference_mixer",
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel.cc b/media/webrtc/trunk/webrtc/voice_engine/channel.cc
index d97bad3880..c4ddd2fe94 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/voice_engine/channel.h"
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
@@ -58,8 +59,8 @@ class StatisticsProxy : public RtcpStatisticsCallback {
      ssrc_(ssrc) {}
   virtual ~StatisticsProxy() {}
 
-  virtual void StatisticsUpdated(const RtcpStatistics& statistics,
-                                 uint32_t ssrc) OVERRIDE {
+  void StatisticsUpdated(const RtcpStatistics& statistics,
+                         uint32_t ssrc) override {
     if (ssrc != ssrc_)
       return;
 
@@ -70,6 +71,8 @@ class StatisticsProxy : public RtcpStatisticsCallback {
     }
   }
 
+  void CNameChanged(const char* cname, uint32_t ssrc) override {}
+
   void ResetStatistics() {
     CriticalSectionScoped cs(stats_lock_.get());
     stats_ = ChannelStatistics();
@@ -84,27 +87,63 @@ class StatisticsProxy : public RtcpStatisticsCallback {
   // StatisticsUpdated calls are triggered from threads in the RTP module,
   // while GetStats calls can be triggered from the public voice engine API,
   // hence synchronization is needed.
-  scoped_ptr stats_lock_;
+  rtc::scoped_ptr stats_lock_;
   const uint32_t ssrc_;
   ChannelStatistics stats_;
 };
 
-class VoEBitrateObserver : public BitrateObserver {
+class VoERtcpObserver : public RtcpBandwidthObserver {
  public:
-  explicit VoEBitrateObserver(Channel* owner)
-      : owner_(owner) {}
-  virtual ~VoEBitrateObserver() {}
+  explicit VoERtcpObserver(Channel* owner) : owner_(owner) {}
+  virtual ~VoERtcpObserver() {}
 
-  // Implements BitrateObserver.
-  virtual void OnNetworkChanged(const uint32_t bitrate_bps,
-                                const uint8_t fraction_lost,
-                                const uint32_t rtt) OVERRIDE {
-    // |fraction_lost| has a scale of 0 - 255.
-    owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
+  void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
+    // Not used for Voice Engine.
+  }
+
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                    int64_t rtt,
+                                    int64_t now_ms) override {
+    // TODO(mflodman): Do we need to aggregate reports here or can we jut send
+    // what we get? I.e. do we ever get multiple reports bundled into one RTCP
+    // report for VoiceEngine?
+    if (report_blocks.empty())
+      return;
+
+    int fraction_lost_aggregate = 0;
+    int total_number_of_packets = 0;
+
+    // If receiving multiple report blocks, calculate the weighted average based
+    // on the number of packets a report refers to.
+    for (ReportBlockList::const_iterator block_it = report_blocks.begin();
+         block_it != report_blocks.end(); ++block_it) {
+      // Find the previous extended high sequence number for this remote SSRC,
+      // to calculate the number of RTP packets this report refers to. Ignore if
+      // we haven't seen this SSRC before.
+      std::map::iterator seq_num_it =
+          extended_max_sequence_number_.find(block_it->sourceSSRC);
+      int number_of_packets = 0;
+      if (seq_num_it != extended_max_sequence_number_.end()) {
+        number_of_packets = block_it->extendedHighSeqNum - seq_num_it->second;
+      }
+      fraction_lost_aggregate += number_of_packets * block_it->fractionLost;
+      total_number_of_packets += number_of_packets;
+
+      extended_max_sequence_number_[block_it->sourceSSRC] =
+          block_it->extendedHighSeqNum;
+    }
+    int weighted_fraction_lost = 0;
+    if (total_number_of_packets > 0) {
+      weighted_fraction_lost = (fraction_lost_aggregate +
+          total_number_of_packets / 2) / total_number_of_packets;
+    }
+    owner_->OnIncomingFractionLoss(weighted_fraction_lost);
   }
 
  private:
   Channel* owner_;
+  // Maps remote side ssrc to extended highest sequence number received.
+  std::map extended_max_sequence_number_;
 };
 
 int32_t
@@ -112,13 +151,14 @@ Channel::SendData(FrameType frameType,
                   uint8_t   payloadType,
                   uint32_t  timeStamp,
                   const uint8_t*  payloadData,
-                  uint16_t  payloadSize,
+                  size_t    payloadSize,
                   const RTPFragmentationHeader* fragmentation)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
-                 " payloadSize=%u, fragmentation=0x%x)",
-                 frameType, payloadType, timeStamp, payloadSize, fragmentation);
+                 " payloadSize=%" PRIuS ", fragmentation=0x%x)",
+                 frameType, payloadType, timeStamp,
+                 payloadSize, fragmentation);
 
     if (_includeAudioLevelIndication)
     {
@@ -155,14 +195,13 @@ Channel::SendData(FrameType frameType,
 }
 
 int32_t
-Channel::InFrameType(int16_t frameType)
+Channel::InFrameType(FrameType frame_type)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::InFrameType(frameType=%d)", frameType);
+                 "Channel::InFrameType(frame_type=%d)", frame_type);
 
     CriticalSectionScoped cs(&_callbackCritSect);
-    // 1 indicates speech
-    _sendFrameType = (frameType == 1) ? 1 : 0;
+    _sendFrameType = (frame_type == kAudioFrameSpeech);
     return 0;
 }
 
@@ -182,13 +221,14 @@ Channel::OnRxVadDetected(int vadDecision)
 }
 
 int
-Channel::SendPacket(int channel, const void *data, int len)
+Channel::SendPacket(int channel, const void *data, size_t len)
 {
     channel = VoEChannelId(channel);
     assert(channel == _channelId);
 
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::SendPacket(channel=%d, len=%d)", channel, len);
+                 "Channel::SendPacket(channel=%d, len=%" PRIuS ")", channel,
+                 len);
 
     CriticalSectionScoped cs(&_callbackCritSect);
 
@@ -201,7 +241,7 @@ Channel::SendPacket(int channel, const void *data, int len)
     }
 
     uint8_t* bufferToSendPtr = (uint8_t*)data;
-    int32_t bufferLength = len;
+    size_t bufferLength = len;
 
     // Dump the RTP packet to a file (if RTP dump is enabled).
     if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
@@ -226,13 +266,14 @@ Channel::SendPacket(int channel, const void *data, int len)
 }
 
 int
-Channel::SendRTCPPacket(int channel, const void *data, int len)
+Channel::SendRTCPPacket(int channel, const void *data, size_t len)
 {
     channel = VoEChannelId(channel);
     assert(channel == _channelId);
 
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len);
+                 "Channel::SendRTCPPacket(channel=%d, len=%" PRIuS ")", channel,
+                 len);
 
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_transportPtr == NULL)
@@ -245,7 +286,7 @@ Channel::SendRTCPPacket(int channel, const void *data, int len)
     }
 
     uint8_t* bufferToSendPtr = (uint8_t*)data;
-    int32_t bufferLength = len;
+    size_t bufferLength = len;
 
     // Dump the RTCP packet to a file (if RTP dump is enabled).
     if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1)
@@ -323,36 +364,6 @@ void Channel::ResetStatistics(uint32_t ssrc) {
   statistics_proxy_->ResetStatistics();
 }
 
-void
-Channel::OnApplicationDataReceived(int32_t id,
-                                   uint8_t subType,
-                                   uint32_t name,
-                                   uint16_t length,
-                                   const uint8_t* data)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnApplicationDataReceived(id=%d, subType=%u,"
-                 " name=%u, length=%u)",
-                 id, subType, name, length);
-
-    int32_t channel = VoEChannelId(id);
-    assert(channel == _channelId);
-
-    if (_rtcpObserver)
-    {
-        CriticalSectionScoped cs(&_callbackCritSect);
-
-        if (_rtcpObserverPtr)
-        {
-            _rtcpObserverPtr->OnApplicationDataReceived(channel,
-                                                        subType,
-                                                        name,
-                                                        data,
-                                                        length);
-        }
-    }
-}
-
 int32_t
 Channel::OnInitializeDecoder(
     int32_t id,
@@ -397,11 +408,11 @@ Channel::OnInitializeDecoder(
 
 int32_t
 Channel::OnReceivedPayloadData(const uint8_t* payloadData,
-                               uint16_t payloadSize,
+                               size_t payloadSize,
                                const WebRtcRTPHeader* rtpHeader)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnReceivedPayloadData(payloadSize=%d,"
+                 "Channel::OnReceivedPayloadData(payloadSize=%" PRIuS ","
                  " payloadType=%u, audioChannel=%u)",
                  payloadSize,
                  rtpHeader->header.payloadType,
@@ -434,7 +445,7 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
     UpdatePacketDelay(rtpHeader->header.timestamp,
                       rtpHeader->header.sequenceNumber);
 
-    uint16_t round_trip_time = 0;
+    int64_t round_trip_time = 0;
     _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time,
                         NULL, NULL, NULL);
 
@@ -449,7 +460,7 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
 }
 
 bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
-                                int rtp_packet_length) {
+                                size_t rtp_packet_length) {
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
     WEBRTC_TRACE(kTraceDebug, webrtc::kTraceVoice, _channelId,
@@ -790,10 +801,8 @@ Channel::Channel(int32_t channelId,
     _rxVadObserverPtr(NULL),
     _oldVadDecision(-1),
     _sendFrameType(0),
-    _rtcpObserverPtr(NULL),
     _externalMixing(false),
     _mixFileWithMicrophone(false),
-    _rtcpObserver(false),
     _mute(false),
     _panLeft(1.0f),
     _panRight(1.0f),
@@ -815,12 +824,7 @@ Channel::Channel(int32_t channelId,
     _rxAgcIsEnabled(false),
     _rxNsIsEnabled(false),
     restored_packet_in_use_(false),
-    bitrate_controller_(
-        BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
-                                                   true)),
-    rtcp_bandwidth_observer_(
-        bitrate_controller_->CreateRtcpBandwidthObserver()),
-    send_bitrate_observer_(new VoEBitrateObserver(this)),
+    rtcp_observer_(new VoERtcpObserver(this)),
     network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock()))
 {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
@@ -833,10 +837,9 @@ Channel::Channel(int32_t channelId,
     configuration.id = VoEModuleId(instanceId, channelId);
     configuration.audio = true;
     configuration.outgoing_transport = this;
-    configuration.rtcp_feedback = this;
     configuration.audio_messages = this;
     configuration.receive_statistics = rtp_receive_statistics_.get();
-    configuration.bandwidth_callback = rtcp_bandwidth_observer_.get();
+    configuration.bandwidth_callback = rtcp_observer_.get();
 
     _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
 
@@ -846,6 +849,10 @@ Channel::Channel(int32_t channelId,
 
     Config audioproc_config;
     audioproc_config.Set(new ExperimentalAgc(false));
+    audioproc_config.Set(
+      new ExtendedFilter(config.Get().enabled));
+    audioproc_config.Set(
+      new DelayAgnostic(config.Get().enabled));
     rx_audioproc_.reset(AudioProcessing::Create(audioproc_config));
 }
 
@@ -910,12 +917,8 @@ Channel::~Channel()
                      " (Audio coding module)");
     }
     // De-register modules in process thread
-    if (_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()) == -1)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceVoice,
-                     VoEId(_instanceId,_channelId),
-                     "~Channel() failed to deregister RTP/RTCP module");
-    }
+    _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
+
     // End of modules shutdown
 
     // Delete other objects
@@ -951,24 +954,16 @@ Channel::Init()
 
     // --- Add modules to process thread (for periodic schedulation)
 
-    const bool processThreadFail =
-        ((_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get()) != 0) ||
-        false);
-    if (processThreadFail)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_CANNOT_INIT_CHANNEL, kTraceError,
-            "Channel::Init() modules not registered");
-        return -1;
-    }
+    _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get());
+
     // --- ACM initialization
 
-    if ((audio_coding_->InitializeReceiver() == -1) ||
+    if ((audio_coding_->InitializeReceiver() == -1)
 #ifdef WEBRTC_CODEC_AVT
         // out-of-band Dtmf tones are played out by default
-        (audio_coding_->SetDtmfPlayoutStatus(true) == -1) ||
+        || (audio_coding_->SetDtmfPlayoutStatus(true) == -1)
 #endif
-        (audio_coding_->InitializeSender() == -1))
+       )
     {
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
@@ -985,15 +980,8 @@ Channel::Init()
     // be transmitted since the Transport object will then be invalid.
     telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
     // RTCP is enabled by default.
-    if (_rtpRtcpModule->SetRTCPStatus(kRtcpCompound) == -1)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-            "Channel::Init() RTP/RTCP module not initialized");
-        return -1;
-    }
-
-     // --- Register all permanent callbacks
+    _rtpRtcpModule->SetRTCPStatus(kRtcpCompound);
+    // --- Register all permanent callbacks
     const bool fail =
         (audio_coding_->RegisterTransportCallback(this) == -1) ||
         (audio_coding_->RegisterVADCallback(this) == -1);
@@ -1356,28 +1344,16 @@ Channel::SetSendCodec(const CodecInst& codec)
         return -1;
     }
 
-    bitrate_controller_->SetBitrateObserver(send_bitrate_observer_.get(),
-                                            codec.rate, 0, 0);
-
     return 0;
 }
 
-void
-Channel::OnNetworkChanged(const uint32_t bitrate_bps,
-                          const uint8_t fraction_lost,  // 0 - 255.
-                          const uint32_t rtt) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-      "Channel::OnNetworkChanged(bitrate_bps=%d, fration_lost=%d, rtt=%d)",
-      bitrate_bps, fraction_lost, rtt);
-  // |fraction_lost| from BitrateObserver is short time observation of packet
-  // loss rate from past. We use network predictor to make a more reasonable
-  // loss rate estimation.
+void Channel::OnIncomingFractionLoss(int fraction_lost) {
   network_predictor_->UpdatePacketLossRate(fraction_lost);
-  uint8_t loss_rate = network_predictor_->GetLossRate();
+  uint8_t average_fraction_loss = network_predictor_->GetLossRate();
+
   // Normalizes rate to 0 - 100.
-  if (audio_coding_->SetPacketLossRate(100 * loss_rate / 255) != 0) {
-    _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR,
-        kTraceError, "OnNetworkChanged() failed to set packet loss rate");
+  if (audio_coding_->SetPacketLossRate(
+      100 * average_fraction_loss / 255) != 0) {
     assert(false);  // This should not happen.
   }
 }
@@ -1387,6 +1363,7 @@ Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetVADStatus(mode=%d)", mode);
+    assert(!(disableDTX && enableVAD));  // disableDTX mode is deprecated.
     // To disable VAD, DTX must be disabled too
     disableDTX = ((enableVAD == false) ? true : disableDTX);
     if (audio_coding_->SetVAD(!disableDTX, enableVAD, mode) != 0)
@@ -1527,7 +1504,7 @@ Channel::GetRecPayloadType(CodecInst& codec)
     }
     codec.pltype = payloadType;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::GetRecPayloadType() => pltype=%u", codec.pltype);
+                 "Channel::GetRecPayloadType() => pltype=%d", codec.pltype);
     return 0;
 }
 
@@ -1593,6 +1570,19 @@ int Channel::SetOpusMaxPlaybackRate(int frequency_hz) {
   return 0;
 }
 
+int Channel::SetOpusDtx(bool enable_dtx) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SetOpusDtx(%d)", enable_dtx);
+  int ret = enable_dtx ? audio_coding_->EnableOpusDtx(true)
+                       : audio_coding_->DisableOpusDtx();
+  if (ret != 0) {
+    _engineStatisticsPtr->SetLastError(
+        VE_AUDIO_CODING_MODULE_ERROR, kTraceError, "SetOpusDtx() failed");
+    return -1;
+  }
+  return 0;
+}
+
 int32_t Channel::RegisterExternalTransport(Transport& transport)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
@@ -1635,7 +1625,7 @@ Channel::DeRegisterExternalTransport()
     return 0;
 }
 
-int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length,
+int32_t Channel::ReceivedRTPPacket(const int8_t* data, size_t length,
                                    const PacketTime& packet_time) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::ReceivedRTPPacket()");
@@ -1676,7 +1666,7 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length,
       } else {
         arrival_time_ms = TickTime::MillisecondTimestamp();
       }
-      int payload_length = length - header.headerLength;
+      size_t payload_length = length - header.headerLength;
       vie_network_->ReceivedBWEPacket(video_channel_, arrival_time_ms,
                                       payload_length, header);
     }
@@ -1686,15 +1676,15 @@ int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length,
 }
 
 bool Channel::ReceivePacket(const uint8_t* packet,
-                            int packet_length,
+                            size_t packet_length,
                             const RTPHeader& header,
                             bool in_order) {
-  if (rtp_payload_registry_->IsEncapsulated(header)) {
-    return HandleEncapsulation(packet, packet_length, header);
+  if (rtp_payload_registry_->IsRtx(header)) {
+    return HandleRtxPacket(packet, packet_length, header);
   }
   const uint8_t* payload = packet + header.headerLength;
-  int payload_length = packet_length - header.headerLength;
-  assert(payload_length >= 0);
+  assert(packet_length >= header.headerLength);
+  size_t payload_length = packet_length - header.headerLength;
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
@@ -1704,9 +1694,9 @@ bool Channel::ReceivePacket(const uint8_t* packet,
                                           payload_specific, in_order);
 }
 
-bool Channel::HandleEncapsulation(const uint8_t* packet,
-                                  int packet_length,
-                                  const RTPHeader& header) {
+bool Channel::HandleRtxPacket(const uint8_t* packet,
+                              size_t packet_length,
+                              const RTPHeader& header) {
   if (!rtp_payload_registry_->IsRtx(header))
     return false;
 
@@ -1752,29 +1742,27 @@ bool Channel::IsPacketRetransmitted(const RTPHeader& header,
   if (!statistician)
     return false;
   // Check if this is a retransmission.
-  uint16_t min_rtt = 0;
+  int64_t min_rtt = 0;
   _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 
-int32_t Channel::ReceivedRTCPPacket(const int8_t* data, int32_t length) {
+int32_t Channel::ReceivedRTCPPacket(const int8_t* data, size_t length) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::ReceivedRTCPPacket()");
   // Store playout timestamp for the received RTCP packet
   UpdatePlayoutTimestamp(true);
 
   // Dump the RTCP packet to a file (if RTP dump is enabled).
-  if (_rtpDumpIn.DumpPacket((const uint8_t*)data,
-                            (uint16_t)length) == -1) {
+  if (_rtpDumpIn.DumpPacket((const uint8_t*)data, length) == -1) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                  VoEId(_instanceId,_channelId),
                  "Channel::SendPacket() RTCP dump to input file failed");
   }
 
   // Deliver RTCP packet to RTP/RTCP module for parsing
-  if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data,
-                                         (uint16_t)length) == -1) {
+  if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, length) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
         "Channel::IncomingRTPPacket() RTCP packet is invalid");
@@ -1782,7 +1770,7 @@ int32_t Channel::ReceivedRTCPPacket(const int8_t* data, int32_t length) {
 
   {
     CriticalSectionScoped lock(ts_stats_lock_.get());
-    uint16_t rtt = GetRTT();
+    int64_t rtt = GetRTT();
     if (rtt == 0) {
       // Waiting for valid RTT.
       return 0;
@@ -2872,48 +2860,6 @@ Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
 
 #endif // #ifdef WEBRTC_VOICE_ENGINE_NR
 
-int
-Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::RegisterRTCPObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
-
-    if (_rtcpObserverPtr)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceError,
-            "RegisterRTCPObserver() observer already enabled");
-        return -1;
-    }
-
-    _rtcpObserverPtr = &observer;
-    _rtcpObserver = true;
-
-    return 0;
-}
-
-int
-Channel::DeRegisterRTCPObserver()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::DeRegisterRTCPObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
-
-    if (!_rtcpObserverPtr)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceWarning,
-            "DeRegisterRTCPObserver() observer already disabled");
-        return 0;
-    }
-
-    _rtcpObserver = false;
-    _rtcpObserverPtr = NULL;
-
-    return 0;
-}
-
 int
 Channel::SetLocalSSRC(unsigned int ssrc)
 {
@@ -2980,20 +2926,10 @@ int Channel::SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
   return 0;
 }
 
-int
-Channel::SetRTCPStatus(bool enable)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::SetRTCPStatus()");
-    if (_rtpRtcpModule->SetRTCPStatus(enable ?
-        kRtcpCompound : kRtcpOff) != 0)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-            "SetRTCPStatus() failed to set RTCP status");
-        return -1;
-    }
-    return 0;
+void Channel::SetRTCPStatus(bool enable) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SetRTCPStatus()");
+  _rtpRtcpModule->SetRTCPStatus(enable ? kRtcpCompound : kRtcpOff);
 }
 
 int
@@ -3112,8 +3048,8 @@ Channel::GetRemoteRTCPReceiverInfo(
                "fractionLost = %lu, cumulativeLost = %lu",
                jitter, fractionLost, cumulativeLost);
 
-  uint16_t dummy;
-  uint16_t rtt = 0;
+  int64_t dummy;
+  int64_t rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy)
       != 0)
   {
@@ -3289,16 +3225,19 @@ Channel::GetRTPStatistics(CallStatistics& stats)
 
     // --- RTT
     stats.rttMs = GetRTT();
-
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() => rttMs=%d", stats.rttMs);
+    if (stats.rttMs == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
+                   "GetRTPStatistics() failed to get RTT");
+    } else {
+      WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+                   "GetRTPStatistics() => rttMs=%" PRId64, stats.rttMs);
+    }
 
     // --- Data counters
 
-    uint32_t bytesSent(0);
+    size_t bytesSent(0);
     uint32_t packetsSent(0);
-    uint32_t bytesReceived(0);
+    size_t bytesReceived(0);
     uint32_t packetsReceived(0);
 
     if (statistician) {
@@ -3321,8 +3260,8 @@ Channel::GetRTPStatistics(CallStatistics& stats)
 
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
                  VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() => bytesSent=%d, packetsSent=%d,"
-                 " bytesReceived=%d, packetsReceived=%d)",
+                 "GetRTPStatistics() => bytesSent=%" PRIuS ", packetsSent=%d,"
+                 " bytesReceived=%" PRIuS ", packetsReceived=%d)",
                  stats.bytesSent, stats.packetsSent, stats.bytesReceived,
                  stats.packetsReceived);
 
@@ -3378,6 +3317,7 @@ Channel::GetREDStatus(bool& enabled, int& redPayloadtype)
                 "module");
             return -1;
         }
+        redPayloadtype = payloadType;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
                    VoEId(_instanceId, _channelId),
                    "GetREDStatus() => enabled=%d, redPayloadtype=%d",
@@ -3623,7 +3563,10 @@ Channel::EncodeAndSend()
 
     // The ACM resamples internally.
     _audioFrame.timestamp_ = _timeStamp;
-    if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) != 0)
+    // This call will trigger AudioPacketizationCallback::SendData if encoding
+    // is done and payload is ready for packetization and transmission.
+    // Otherwise, it will return without invoking the callback.
+    if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) < 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::EncodeAndSend() ACM encoding failed");
@@ -3631,12 +3574,7 @@ Channel::EncodeAndSend()
     }
 
     _timeStamp += _audioFrame.samples_per_channel_;
-
-    // --- Encode if complete frame is ready
-
-    // This call will trigger AudioPacketizationCallback::SendData if encoding
-    // is done and payload is ready for packetization and transmission.
-    return audio_coding_->Process();
+    return 0;
 }
 
 int Channel::RegisterExternalMediaProcessing(
@@ -3737,12 +3675,7 @@ Channel::GetNetworkStatistics(NetworkStatistics& stats)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::GetNetworkStatistics()");
-    ACMNetworkStatistics acm_stats;
-    int return_value = audio_coding_->NetworkStatistics(&acm_stats);
-    if (return_value >= 0) {
-      memcpy(&stats, &acm_stats, sizeof(NetworkStatistics));
-    }
-    return return_value;
+    return audio_coding_->GetNetworkStatistics(&stats);
 }
 
 void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
@@ -3865,47 +3798,28 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
   return 0;
 }
 
-int
-Channel::SetInitTimestamp(unsigned int timestamp)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
+int Channel::SetInitTimestamp(unsigned int timestamp) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetInitTimestamp()");
-    if (channel_state_.Get().sending)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_SENDING, kTraceError, "SetInitTimestamp() already sending");
-        return -1;
-    }
-    if (_rtpRtcpModule->SetStartTimestamp(timestamp) != 0)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-            "SetInitTimestamp() failed to set timestamp");
-        return -1;
-    }
-    return 0;
+  if (channel_state_.Get().sending) {
+    _engineStatisticsPtr->SetLastError(VE_SENDING, kTraceError,
+                                       "SetInitTimestamp() already sending");
+    return -1;
+  }
+  _rtpRtcpModule->SetStartTimestamp(timestamp);
+  return 0;
 }
 
-int
-Channel::SetInitSequenceNumber(short sequenceNumber)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::SetInitSequenceNumber()");
-    if (channel_state_.Get().sending)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_SENDING, kTraceError,
-            "SetInitSequenceNumber() already sending");
-        return -1;
-    }
-    if (_rtpRtcpModule->SetSequenceNumber(sequenceNumber) != 0)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_RTP_RTCP_MODULE_ERROR, kTraceError,
-            "SetInitSequenceNumber() failed to set sequence number");
-        return -1;
-    }
-    return 0;
+int Channel::SetInitSequenceNumber(short sequenceNumber) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SetInitSequenceNumber()");
+  if (channel_state_.Get().sending) {
+    _engineStatisticsPtr->SetLastError(
+        VE_SENDING, kTraceError, "SetInitSequenceNumber() already sending");
+    return -1;
+  }
+  _rtpRtcpModule->SetSequenceNumber(sequenceNumber);
+  return 0;
 }
 
 int
@@ -3923,7 +3837,7 @@ Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const
 int32_t
 Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
 {
-    scoped_ptr fileBuffer(new int16_t[640]);
+  rtc::scoped_ptr fileBuffer(new int16_t[640]);
     int fileSamples(0);
 
     {
@@ -3993,7 +3907,7 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
 {
     assert(mixingFrequency <= 48000);
 
-    scoped_ptr fileBuffer(new int16_t[960]);
+    rtc::scoped_ptr fileBuffer(new int16_t[960]);
     int fileSamples(0);
 
     {
@@ -4116,7 +4030,7 @@ Channel::InsertInbandDtmfTone()
 }
 
 int32_t
-Channel::SendPacketRaw(const void *data, int len, bool RTCP)
+Channel::SendPacketRaw(const void *data, size_t len, bool RTCP)
 {
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_transportPtr == NULL)
@@ -4228,47 +4142,6 @@ Channel::RegisterReceiveCodecsToRTPModule()
     }
 }
 
-int Channel::SetSecondarySendCodec(const CodecInst& codec,
-                                   int red_payload_type) {
-  // Sanity check for payload type.
-  if (red_payload_type < 0 || red_payload_type > 127) {
-    _engineStatisticsPtr->SetLastError(
-        VE_PLTYPE_ERROR, kTraceError,
-        "SetRedPayloadType() invalid RED payload type");
-    return -1;
-  }
-
-  if (SetRedPayloadType(red_payload_type) < 0) {
-    _engineStatisticsPtr->SetLastError(
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "SetSecondarySendCodec() Failed to register RED ACM");
-    return -1;
-  }
-  if (audio_coding_->RegisterSecondarySendCodec(codec) < 0) {
-    _engineStatisticsPtr->SetLastError(
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "SetSecondarySendCodec() Failed to register secondary send codec in "
-        "ACM");
-    return -1;
-  }
-
-  return 0;
-}
-
-void Channel::RemoveSecondarySendCodec() {
-  audio_coding_->UnregisterSecondarySendCodec();
-}
-
-int Channel::GetSecondarySendCodec(CodecInst* codec) {
-  if (audio_coding_->SecondarySendCodec(codec) < 0) {
-    _engineStatisticsPtr->SetLastError(
-        VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
-        "GetSecondarySendCodec() Failed to get secondary sent codec from ACM");
-    return -1;
-  }
-  return 0;
-}
-
 // Assuming this method is called with valid payload type.
 int Channel::SetRedPayloadType(int red_payload_type) {
   CodecInst codec;
@@ -4339,22 +4212,14 @@ int32_t Channel::GetPlayoutFrequency() {
   return playout_frequency;
 }
 
-int Channel::GetRTT() const {
+int64_t Channel::GetRTT() const {
   RTCPMethod method = _rtpRtcpModule->RTCP();
   if (method == kRtcpOff) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() RTCP is disabled => valid RTT "
-                 "measurements cannot be retrieved");
     return 0;
   }
   std::vector report_blocks;
   _rtpRtcpModule->RemoteRTCPStat(&report_blocks);
   if (report_blocks.empty()) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() failed to measure RTT since no "
-                 "RTCP packets have been received yet");
     return 0;
   }
 
@@ -4371,19 +4236,15 @@ int Channel::GetRTT() const {
     // the SSRC of the other end.
     remoteSSRC = report_blocks[0].remoteSSRC;
   }
-  uint16_t rtt = 0;
-  uint16_t avg_rtt = 0;
-  uint16_t max_rtt= 0;
-  uint16_t min_rtt = 0;
+  int64_t rtt = 0;
+  int64_t avg_rtt = 0;
+  int64_t max_rtt= 0;
+  int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
-    WEBRTC_TRACE(kTraceWarning, kTraceVoice,
-                 VoEId(_instanceId, _channelId),
-                 "GetRTPStatistics() failed to retrieve RTT from "
-                 "the RTP/RTCP module");
     return 0;
   }
-  return static_cast(rtt);
+  return rtt;
 }
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel.h b/media/webrtc/trunk/webrtc/voice_engine/channel.h
index e44b4420e3..5e8c50e27a 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
@@ -22,7 +23,6 @@
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/utility/interface/file_player.h"
 #include "webrtc/modules/utility/interface/file_recorder.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
@@ -59,7 +59,6 @@ class RtpRtcp;
 class TelephoneEventHandler;
 class ViENetwork;
 class VoEMediaProcess;
-class VoERTCPObserver;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
@@ -69,10 +68,11 @@ struct SenderInfo;
 
 namespace voe {
 
+class OutputMixer;
 class Statistics;
 class StatisticsProxy;
 class TransmitMixer;
-class OutputMixer;
+class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
@@ -149,14 +149,13 @@ class ChannelState {
     }
 
 private:
-    scoped_ptr lock_;
+ rtc::scoped_ptr lock_;
     State state_;
 };
 
 class Channel:
     public RtpData,
     public RtpFeedback,
-    public RtcpFeedback,
     public FileCallback, // receiving notification from file player & recorder
     public Transport,
     public RtpAudioFeedback,
@@ -165,6 +164,8 @@ class Channel:
     public MixerParticipant // supplies output mixer with audio frames
 {
 public:
+    friend class VoERtcpObserver;
+
     enum {KNumSocketThreads = 1};
     enum {KNumberOfSocketBuffers = 8};
     virtual ~Channel();
@@ -207,18 +208,14 @@ public:
     int32_t GetRecPayloadType(CodecInst& codec);
     int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency);
     int SetOpusMaxPlaybackRate(int frequency_hz);
-
-    // VoE dual-streaming.
-    int SetSecondarySendCodec(const CodecInst& codec, int red_payload_type);
-    void RemoveSecondarySendCodec();
-    int GetSecondarySendCodec(CodecInst* codec);
+    int SetOpusDtx(bool enable_dtx);
 
     // VoENetwork
     int32_t RegisterExternalTransport(Transport& transport);
     int32_t DeRegisterExternalTransport();
-    int32_t ReceivedRTPPacket(const int8_t* data, int32_t length,
+    int32_t ReceivedRTPPacket(const int8_t* data, size_t length,
                               const PacketTime& packet_time);
-    int32_t ReceivedRTCPPacket(const int8_t* data, int32_t length);
+    int32_t ReceivedRTCPPacket(const int8_t* data, size_t length);
 
     // VoEFile
     int StartPlayingFileLocally(const char* fileName, bool loop,
@@ -316,8 +313,6 @@ public:
 #endif
 
     // VoERTP_RTCP
-    int RegisterRTCPObserver(VoERTCPObserver& observer);
-    int DeRegisterRTCPObserver();
     int SetLocalSSRC(unsigned int ssrc);
     int GetLocalSSRC(unsigned int& ssrc);
     int GetRemoteSSRC(unsigned int& ssrc);
@@ -325,7 +320,7 @@ public:
     int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
-    int SetRTCPStatus(bool enable);
+    void SetRTCPStatus(bool enable);
     int GetRTCPStatus(bool& enabled);
     int SetRTCP_CNAME(const char cName[256]);
     int GetRemoteRTCP_CNAME(char cName[256]);
@@ -357,69 +352,55 @@ public:
     void SetVideoEngineBWETarget(ViENetwork* vie_network, int video_channel);
 
     // From AudioPacketizationCallback in the ACM
-    virtual int32_t SendData(
-        FrameType frameType,
-        uint8_t payloadType,
-        uint32_t timeStamp,
-        const uint8_t* payloadData,
-        uint16_t payloadSize,
-        const RTPFragmentationHeader* fragmentation) OVERRIDE;
+    int32_t SendData(FrameType frameType,
+                     uint8_t payloadType,
+                     uint32_t timeStamp,
+                     const uint8_t* payloadData,
+                     size_t payloadSize,
+                     const RTPFragmentationHeader* fragmentation) override;
 
     // From ACMVADCallback in the ACM
-    virtual int32_t InFrameType(int16_t frameType) OVERRIDE;
+    int32_t InFrameType(FrameType frame_type) override;
 
     int32_t OnRxVadDetected(int vadDecision);
 
     // From RtpData in the RTP/RTCP module
-    virtual int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        uint16_t payloadSize,
-        const WebRtcRTPHeader* rtpHeader) OVERRIDE;
-    virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                   int packet_length) OVERRIDE;
+    int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+                                  size_t payloadSize,
+                                  const WebRtcRTPHeader* rtpHeader) override;
+    bool OnRecoveredPacket(const uint8_t* packet,
+                           size_t packet_length) override;
 
     // From RtpFeedback in the RTP/RTCP module
-    virtual int32_t OnInitializeDecoder(
-        int32_t id,
-        int8_t payloadType,
-        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-        int frequency,
-        uint8_t channels,
-        uint32_t rate) OVERRIDE;
-    virtual void OnIncomingSSRCChanged(int32_t id,
-                                       uint32_t ssrc) OVERRIDE;
-    virtual void OnIncomingCSRCChanged(int32_t id,
-                                       uint32_t CSRC, bool added) OVERRIDE;
-    virtual void ResetStatistics(uint32_t ssrc) OVERRIDE;
-
-    // From RtcpFeedback in the RTP/RTCP module
-    virtual void OnApplicationDataReceived(int32_t id,
-                                           uint8_t subType,
-                                           uint32_t name,
-                                           uint16_t length,
-                                           const uint8_t* data) OVERRIDE;
+    int32_t OnInitializeDecoder(int32_t id,
+                                int8_t payloadType,
+                                const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                                int frequency,
+                                uint8_t channels,
+                                uint32_t rate) override;
+    void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
+    void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
+    void ResetStatistics(uint32_t ssrc) override;
 
     // From RtpAudioFeedback in the RTP/RTCP module
-    virtual void OnPlayTelephoneEvent(int32_t id,
-                                      uint8_t event,
-                                      uint16_t lengthMs,
-                                      uint8_t volume) OVERRIDE;
+    void OnPlayTelephoneEvent(int32_t id,
+                              uint8_t event,
+                              uint16_t lengthMs,
+                              uint8_t volume) override;
 
     // From Transport (called by the RTP/RTCP module)
-    virtual int SendPacket(int /*channel*/, const void *data, int len) OVERRIDE;
-    virtual int SendRTCPPacket(int /*channel*/,
-                               const void *data,
-                               int len) OVERRIDE;
+    int SendPacket(int /*channel*/, const void* data, size_t len) override;
+    int SendRTCPPacket(int /*channel*/, const void* data, size_t len) override;
 
     // From MixerParticipant
-    virtual int32_t GetAudioFrame(int32_t id, AudioFrame& audioFrame) OVERRIDE;
-    virtual int32_t NeededFrequency(int32_t id) OVERRIDE;
+    int32_t GetAudioFrame(int32_t id, AudioFrame& audioFrame) override;
+    int32_t NeededFrequency(int32_t id) override;
 
     // From FileCallback
-    virtual void PlayNotification(int32_t id, uint32_t durationMs) OVERRIDE;
-    virtual void RecordNotification(int32_t id, uint32_t durationMs) OVERRIDE;
-    virtual void PlayFileEnded(int32_t id) OVERRIDE;
-    virtual void RecordFileEnded(int32_t id) OVERRIDE;
+    void PlayNotification(int32_t id, uint32_t durationMs) override;
+    void RecordNotification(int32_t id, uint32_t durationMs) override;
+    void PlayFileEnded(int32_t id) override;
+    void RecordFileEnded(int32_t id) override;
 
     uint32_t InstanceId() const
     {
@@ -469,24 +450,22 @@ public:
     uint32_t PrepareEncodeAndSend(int mixingFrequency);
     uint32_t EncodeAndSend();
 
-    // From BitrateObserver (called by the RTP/RTCP module).
-    void OnNetworkChanged(const uint32_t bitrate_bps,
-                          const uint8_t fraction_lost,  // 0 - 255.
-                          const uint32_t rtt);
+protected:
+    void OnIncomingFractionLoss(int fraction_lost);
 
 private:
-    bool ReceivePacket(const uint8_t* packet, int packet_length,
+    bool ReceivePacket(const uint8_t* packet, size_t packet_length,
                        const RTPHeader& header, bool in_order);
-    bool HandleEncapsulation(const uint8_t* packet,
-                             int packet_length,
-                             const RTPHeader& header);
+    bool HandleRtxPacket(const uint8_t* packet,
+                         size_t packet_length,
+                         const RTPHeader& header);
     bool IsPacketInOrder(const RTPHeader& header) const;
     bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
     int ResendPackets(const uint16_t* sequence_numbers, int length);
     int InsertInbandDtmfTone();
     int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
     int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
-    int32_t SendPacketRaw(const void *data, int len, bool RTCP);
+    int32_t SendPacketRaw(const void *data, size_t len, bool RTCP);
     void UpdatePacketDelay(uint32_t timestamp,
                            uint16_t sequenceNumber);
     void RegisterReceiveCodecsToRTPModule();
@@ -496,7 +475,7 @@ private:
                                   unsigned char id);
 
     int32_t GetPlayoutFrequency();
-    int GetRTT() const;
+    int64_t GetRTT() const;
 
     CriticalSectionWrapper& _fileCritSect;
     CriticalSectionWrapper& _callbackCritSect;
@@ -506,20 +485,20 @@ private:
 
     ChannelState channel_state_;
 
-    scoped_ptr rtp_header_parser_;
-    scoped_ptr rtp_payload_registry_;
-    scoped_ptr rtp_receive_statistics_;
-    scoped_ptr statistics_proxy_;
-    scoped_ptr rtp_receiver_;
+    rtc::scoped_ptr rtp_header_parser_;
+    rtc::scoped_ptr rtp_payload_registry_;
+    rtc::scoped_ptr rtp_receive_statistics_;
+    rtc::scoped_ptr statistics_proxy_;
+    rtc::scoped_ptr rtp_receiver_;
     TelephoneEventHandler* telephone_event_handler_;
-    scoped_ptr _rtpRtcpModule;
-    scoped_ptr audio_coding_;
+    rtc::scoped_ptr _rtpRtcpModule;
+    rtc::scoped_ptr audio_coding_;
     RtpDump& _rtpDumpIn;
     RtpDump& _rtpDumpOut;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;
-    scoped_ptr mono_recording_audio_;
+    rtc::scoped_ptr mono_recording_audio_;
     // Downsamples to the codec rate if necessary.
     PushResampler input_resampler_;
     FilePlayer* _inputFilePlayerPtr;
@@ -548,9 +527,9 @@ private:
     uint16_t send_sequence_number_;
     uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
-    scoped_ptr ts_stats_lock_;
+    rtc::scoped_ptr ts_stats_lock_;
 
-    scoped_ptr rtp_ts_wraparound_handler_;
+    rtc::scoped_ptr rtp_ts_wraparound_handler_;
     // The rtp timestamp of the first played out audio frame.
     int64_t capture_start_rtp_time_stamp_;
     // The capture ntp time (in local timebase) of the first played out audio
@@ -567,15 +546,13 @@ private:
     CriticalSectionWrapper* _callbackCritSectPtr; // owned by base
     Transport* _transportPtr; // WebRtc socket or external transport
     RMSLevel rms_level_;
-    scoped_ptr rx_audioproc_; // far end AudioProcessing
+    rtc::scoped_ptr rx_audioproc_;  // far end AudioProcessing
     VoERxVadCallback* _rxVadObserverPtr;
     int32_t _oldVadDecision;
     int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
-    VoERTCPObserver* _rtcpObserverPtr;
     // VoEBase
     bool _externalMixing;
     bool _mixFileWithMicrophone;
-    bool _rtcpObserver;
     // VoEVolumeControl
     bool _mute;
     float _panLeft;
@@ -604,10 +581,8 @@ private:
     bool _rxNsIsEnabled;
     bool restored_packet_in_use_;
     // RtcpBandwidthObserver
-    scoped_ptr bitrate_controller_;
-    scoped_ptr rtcp_bandwidth_observer_;
-    scoped_ptr send_bitrate_observer_;
-    scoped_ptr network_predictor_;
+    rtc::scoped_ptr rtcp_observer_;
+    rtc::scoped_ptr network_predictor_;
 };
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc
index 9386257476..78c484f3dc 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.cc
@@ -46,10 +46,11 @@ ChannelOwner::ChannelRef::ChannelRef(class Channel* channel)
     : channel(channel), ref_count(1) {}
 
 ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
-    : instance_id_(instance_id),
+    : config_(config),
+      instance_id_(instance_id),
       last_channel_id_(-1),
-      lock_(CriticalSectionWrapper::CreateCriticalSection()),
-      config_(config) {}
+      lock_(CriticalSectionWrapper::CreateCriticalSection())
+      {}
 
 ChannelOwner ChannelManager::CreateChannel() {
   return CreateChannelInternal(config_);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h
index 3c0f68147e..27e2865baf 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel_manager.h
@@ -14,9 +14,9 @@
 #include 
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/atomic32.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -60,7 +60,7 @@ class ChannelOwner {
   // deleted when no references to them are held.
   struct ChannelRef {
     ChannelRef(Channel* channel);
-    const scoped_ptr channel;
+    const rtc::scoped_ptr channel;
     Atomic32 ref_count;
   };
 
@@ -109,6 +109,7 @@ class ChannelManager {
   void DestroyAllChannels();
 
   size_t NumOfChannels() const;
+  const Config& config_;
 
  private:
   // Create a channel given a configuration, |config|.
@@ -118,11 +119,9 @@ class ChannelManager {
 
   Atomic32 last_channel_id_;
 
-  scoped_ptr lock_;
+  rtc::scoped_ptr lock_;
   std::vector channels_;
 
-  const Config& config_;
-
   DISALLOW_COPY_AND_ASSIGN(ChannelManager);
 };
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h b/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h
index b7946eeec9..40f88b0076 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/mock/fake_voe_external_media.h
@@ -13,7 +13,7 @@
 
 #include 
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/fake_common.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 
@@ -56,7 +56,7 @@ class FakeVoEExternalMedia : public VoEExternalMedia {
                    int samples_per_channel, int sample_rate_hz,
                    int num_channels) {
     const int length = samples_per_channel * num_channels;
-    scoped_ptr data;
+    rtc::scoped_ptr data;
     if (!audio) {
       data.reset(new int16_t[length]);
       memset(data.get(), 0, length * sizeof(data[0]));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h
index 8b40360ae6..162848c282 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_audio_processing.h
@@ -185,9 +185,11 @@ public:
     virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;
 
     // Gets the EC internal |delay_median| and |delay_std| in ms between
-    // near-end and far-end. The values are calculated over the time period
-    // since the last GetEcDelayMetrics() call.
-    virtual int GetEcDelayMetrics(int& delay_median, int& delay_std) = 0;
+    // near-end and far-end. The metric |fraction_poor_delays| is the amount of
+    // delay values that potentially can break the EC. The values are aggregated
+    // over one second and the last updated metrics are returned.
+    virtual int GetEcDelayMetrics(int& delay_median, int& delay_std,
+                                  float& fraction_poor_delays) = 0;
 
     // Enables recording of Audio Processing (AP) debugging information.
     // The file can later be used for off-line analysis of the AP performance.
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h
index fa169ead4e..b99f36aa81 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_base.h
@@ -87,7 +87,6 @@ public:
 
 #if !defined(WEBRTC_CHROMIUM_BUILD)
     static int SetAndroidObjects(void* javaVM, void* context);
-    static int SetAndroidObjects(void* javaVM, void* env, void* context);
 #endif
 
 protected:
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h
index 234596a216..4b9f939ce1 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_codec.h
@@ -64,23 +64,6 @@ public:
     // |channel|.
     virtual int GetSendCodec(int channel, CodecInst& codec) = 0;
 
-    // Sets the |codec| as secondary codec for |channel|. Registering a
-    // secondary send codec enables dual-streaming. In dual-streaming mode,
-    // payloads of the primary and the secondary codecs are packed in RED
-    // payloads with |red_payload_type| as payload type. The Secondary codec
-    // MUST have the same sampling rate as the primary codec, otherwise the
-    // codec cannot be registered and -1 is returned. This method fails if a
-    // primary codec is not yet set.
-    virtual int SetSecondarySendCodec(int channel, const CodecInst& codec,
-                                      int red_payload_type) = 0;
-
-    // Removes the secondary codec from |channel|. This will terminate
-    // dual-streaming.
-    virtual int RemoveSecondarySendCodec(int channel) = 0;
-
-    // Gets |codec| which is used as secondary codec in |channel|.
-    virtual int GetSecondarySendCodec(int channel, CodecInst& codec) = 0;
-
     // Gets the currently received |codec| for a specific |channel|.
     virtual int GetRecCodec(int channel, CodecInst& codec) = 0;
 
@@ -134,6 +117,10 @@ public:
       return -1;
     }
 
+    // If send codec is Opus on a specified |channel|, set its DTX. Returns 0 if
+    // success, and -1 if failed.
+    virtual int SetOpusDtx(int channel, bool enable_dtx) = 0;
+
     // Don't use. To be removed.
     virtual int SetAMREncFormat(int channel, AmrMode mode) { return -1; }
     virtual int SetAMRDecFormat(int channel, AmrMode mode) { return -1; }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h
index 043efb28af..d69fa89973 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_hardware.h
@@ -69,7 +69,7 @@ public:
     // (GUID) for the playout device.
     virtual int GetPlayoutDeviceName(int index, char strNameUTF8[128],
                                      char strGuidUTF8[128]) = 0;
-
+ 
     // Checks if the sound card is available to be opened for recording.
     virtual int GetRecordingDeviceStatus(bool& isAvailable) = 0;
 
@@ -88,7 +88,7 @@ public:
 
     // Gets the currently used (active) audio device layer.
     virtual int GetAudioDeviceLayer(AudioLayers& audioLayer) = 0;
-
+ 
     // Gets the VoiceEngine's current CPU consumption in terms of the percent
     // of total CPU availability. [Windows only]
     virtual int GetCPULoad(int& loadPercent) = 0;
@@ -112,6 +112,8 @@ public:
     virtual int SetPlayoutSampleRate(unsigned int samples_per_sec) = 0;
     virtual int PlayoutSampleRate(unsigned int* samples_per_sec) const = 0;
 
+    virtual bool BuiltInAECIsAvailable() const = 0;
+
     // *Experimental - not recommended for use.*
     // Enables the Windows Core Audio built-in AEC. Fails on other platforms.
     //
@@ -131,7 +133,6 @@ public:
     virtual int EnableBuiltInAEC(bool enable) = 0;
     virtual bool BuiltInAECIsEnabled() const = 0;
 
-
 protected:
     VoEHardware() {}
     virtual ~VoEHardware() {}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h
index 67429755a7..ff8b8e1f94 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_network.h
@@ -69,10 +69,10 @@ public:
     // including the RTP-header must also be given to the VoiceEngine.
     virtual int ReceivedRTPPacket(int channel,
                                   const void* data,
-                                  unsigned int length) = 0;
+                                  size_t length) = 0;
     virtual int ReceivedRTPPacket(int channel,
                                   const void* data,
-                                  unsigned int length,
+                                  size_t length,
                                   const PacketTime& packet_time) {
       return 0;
     }
@@ -80,8 +80,9 @@ public:
     // The packets received from the network should be passed to this
     // function when external transport is enabled. Note that the data
     // including the RTCP-header must also be given to the VoiceEngine.
-    virtual int ReceivedRTCPPacket(
-        int channel, const void* data, unsigned int length) = 0;
+    virtual int ReceivedRTCPPacket(int channel,
+                                   const void* data,
+                                   size_t length) = 0;
 
 protected:
     VoENetwork() {}
diff --git a/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h b/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
index 93e6cd4dff..ee0fae0014 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/include/voe_rtp_rtcp.h
@@ -61,19 +61,6 @@ protected:
     virtual ~VoERTPObserver() {}
 };
 
-// VoERTCPObserver
-class WEBRTC_DLLEXPORT VoERTCPObserver
-{
-public:
-    virtual void OnApplicationDataReceived(
-        int channel, unsigned char subType,
-        unsigned int name, const unsigned char* data,
-        unsigned short dataLengthInBytes) = 0;
-
-protected:
-    virtual ~VoERTCPObserver() {}
-};
-
 // CallStatistics
 struct CallStatistics
 {
@@ -81,10 +68,10 @@ struct CallStatistics
     unsigned int cumulativeLost;
     unsigned int extendedMax;
     unsigned int jitterSamples;
-    int rttMs;
-    int bytesSent;
+    int64_t rttMs;
+    size_t bytesSent;
     int packetsSent;
-    int bytesReceived;
+    size_t bytesReceived;
     int packetsReceived;
     // The capture ntp time (in local timebase) of the first played out audio
     // frame.
@@ -270,9 +257,6 @@ public:
     virtual int RegisterRTPObserver(int channel,
             VoERTPObserver& observer) { return -1; };
     virtual int DeRegisterRTPObserver(int channel) { return -1; };
-    virtual int RegisterRTCPObserver(
-            int channel, VoERTCPObserver& observer) { return -1; };
-    virtual int DeRegisterRTCPObserver(int channel) { return -1; };
     virtual int GetRemoteCSRCs(int channel,
             unsigned int arrCSRC[15]) { return -1; };
     virtual int InsertExtraRTPPacket(
diff --git a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc
index 88de5c880d..ca358022a5 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.cc
@@ -28,7 +28,7 @@ MonitorModule::~MonitorModule()
     delete &_callbackCritSect;
 }
 
-int32_t 
+int32_t
 MonitorModule::RegisterObserver(MonitorObserver& observer)
 {
     CriticalSectionScoped lock(&_callbackCritSect);
@@ -40,7 +40,7 @@ MonitorModule::RegisterObserver(MonitorObserver& observer)
     return 0;
 }
 
-int32_t 
+int32_t
 MonitorModule::DeRegisterObserver()
 {
     CriticalSectionScoped lock(&_callbackCritSect);
@@ -52,22 +52,15 @@ MonitorModule::DeRegisterObserver()
     return 0;
 }
 
-int32_t 
-MonitorModule::ChangeUniqueId(int32_t id)
-{
-    return 0;
-}
-
-int32_t 
+int64_t
 MonitorModule::TimeUntilNextProcess()
 {
-    uint32_t now = TickTime::MillisecondTimestamp();
-    int32_t timeToNext =
-        kAverageProcessUpdateTimeMs - (now - _lastProcessTime);
-    return (timeToNext); 
+    int64_t now = TickTime::MillisecondTimestamp();
+    const int64_t kAverageProcessUpdateTimeMs = 1000;
+    return kAverageProcessUpdateTimeMs - (now - _lastProcessTime);
 }
 
-int32_t 
+int32_t
 MonitorModule::Process()
 {
     _lastProcessTime = TickTime::MillisecondTimestamp();
diff --git a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h
index 693b2e00a9..42ea74d7e2 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/monitor_module.h
@@ -40,16 +40,14 @@ public:
 
     virtual ~MonitorModule();
 public:	// module
-    virtual int32_t ChangeUniqueId(int32_t id) OVERRIDE;
+ int64_t TimeUntilNextProcess() override;
 
-    virtual int32_t TimeUntilNextProcess() OVERRIDE;
+ int32_t Process() override;
 
-    virtual int32_t Process() OVERRIDE;
 private:
-    enum { kAverageProcessUpdateTimeMs = 1000 };
     MonitorObserver* _observerPtr;
     CriticalSectionWrapper&	_callbackCritSect;
-    int32_t _lastProcessTime;
+    int64_t _lastProcessTime;
 };
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/network_predictor.cc b/media/webrtc/trunk/webrtc/voice_engine/network_predictor.cc
index 4093877343..f8fa2e315c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/network_predictor.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/network_predictor.cc
@@ -19,6 +19,12 @@ NetworkPredictor::NetworkPredictor(Clock* clock)
       loss_rate_filter_(new rtc::ExpFilter(0.9999f)) {
 }
 
+uint8_t NetworkPredictor::GetLossRate() {
+  float value = loss_rate_filter_->filtered();
+  return (value == rtc::ExpFilter::kValueUndefined) ? 0 :
+      static_cast(value + 0.5);
+}
+
 void NetworkPredictor::UpdatePacketLossRate(uint8_t loss_rate) {
   int64_t now_ms = clock_->TimeInMilliseconds();
   // Update the recursive average filter.
@@ -28,10 +34,5 @@ void NetworkPredictor::UpdatePacketLossRate(uint8_t loss_rate) {
   last_loss_rate_update_time_ms_ = now_ms;
 }
 
-uint8_t NetworkPredictor::GetLossRate() {
-  float value = loss_rate_filter_->filtered();
-  return (value == rtc::ExpFilter::kValueUndefined) ? 0 :
-      static_cast(value + 0.5);
-}
 }  // namespace voe
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h b/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h
index d9f0b7b144..0e5068659c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/network_predictor.h
@@ -38,7 +38,7 @@ class NetworkPredictor {
   int64_t last_loss_rate_update_time_ms_;
 
   // An exponential filter is used to predict packet loss rate.
-  scoped_ptr loss_rate_filter_;
+  rtc::scoped_ptr loss_rate_filter_;
 };
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc
index e399f6820f..55ee20da03 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/network_predictor_unittest.cc
@@ -23,7 +23,7 @@ class TestNetworkPredictor : public ::testing::Test {
       : clock_(0),
         network_predictor_(new NetworkPredictor(&clock_)) {}
   SimulatedClock clock_;
-  scoped_ptr network_predictor_;
+  rtc::scoped_ptr network_predictor_;
 };
 
 TEST_F(TestNetworkPredictor, TestPacketLossRateFilter) {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc b/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc
index 218784474e..ad00e038f6 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/shared_data.cc
@@ -29,7 +29,7 @@ SharedData::SharedData(const Config& config) :
     _channelManager(_gInstanceCounter, config),
     _engineStatistics(_gInstanceCounter),
     _audioDevicePtr(NULL),
-    _moduleProcessThreadPtr(ProcessThread::CreateProcessThread()),
+    _moduleProcessThreadPtr(ProcessThread::Create()),
     _externalRecording(false),
     _externalPlayout(false)
 {
@@ -55,7 +55,7 @@ SharedData::~SharedData()
         _audioDevicePtr->Release();
     }
     delete _apiCritPtr;
-    ProcessThread::DestroyProcessThread(_moduleProcessThreadPtr);
+    _moduleProcessThreadPtr->Stop();
     Trace::ReturnTrace();
 }
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/shared_data.h b/media/webrtc/trunk/webrtc/voice_engine/shared_data.h
index dd76e96abd..3ab1d45ac5 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/shared_data.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/shared_data.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
 #define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/interface/process_thread.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
@@ -48,7 +48,7 @@ public:
     void set_ext_recording(bool value) { _externalRecording = value; }
     bool ext_playout() const { return _externalPlayout; }
     void set_ext_playout(bool value) { _externalPlayout = value; }
-    ProcessThread* process_thread() { return _moduleProcessThreadPtr; }
+    ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
     AudioDeviceModule::AudioLayer audio_device_layer() const {
       return _audioDeviceLayer;
     }
@@ -73,8 +73,8 @@ protected:
     AudioDeviceModule* _audioDevicePtr;
     OutputMixer* _outputMixerPtr;
     TransmitMixer* _transmitMixerPtr;
-    scoped_ptr audioproc_;
-    ProcessThread* _moduleProcessThreadPtr;
+    rtc::scoped_ptr audioproc_;
+    rtc::scoped_ptr _moduleProcessThreadPtr;
 
     bool _externalRecording;
     bool _externalPlayout;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc
index 2f04cc57ab..b0a26e06a0 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/android/android_test/jni/android_test.cc
@@ -139,19 +139,20 @@ class my_transportation : public Transport
       netw(network) {
   }
 
-  virtual int SendPacket(int channel,const void *data,int len) OVERRIDE;
-  virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE;
+  int SendPacket(int channel, const void* data, size_t len) override;
+  int SendRTCPPacket(int channel, const void* data, size_t len) override;
+
  private:
   VoENetwork * netw;
 };
 
-int my_transportation::SendPacket(int channel,const void *data,int len)
+int my_transportation::SendPacket(int channel, const void *data, size_t len)
 {
   netw->ReceivedRTPPacket(channel, data, len);
   return len;
 }
 
-int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
+int my_transportation::SendRTCPPacket(int channel, const void *data, size_t len)
 {
   netw->ReceivedRTCPPacket(channel, data, len);
   return len;
@@ -176,27 +177,18 @@ private:
     static bool Run(void* ptr);
     bool Process();
 private:
-    ThreadWrapper* _thread;
+    rtc::scoped_ptr _thread;
 };
 
 ThreadTest::~ThreadTest()
 {
     if (_thread)
-    {
-        _thread->SetNotAlive();
-        if (_thread->Stop())
-        {
-            delete _thread;
-            _thread = NULL;
-        }
-    }
+        _thread->Stop();
 }
 
-ThreadTest::ThreadTest() :
-    _thread(NULL)
+ThreadTest::ThreadTest()
 {
-    _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
-                                          "ThreadTest thread");
+    _thread = ThreadWrapper::CreateThread(Run, this, "ThreadTest thread");
 }
 
 bool ThreadTest::Run(void* ptr)
@@ -270,7 +262,6 @@ bool ThreadTest::Process()
             "sending instance started from thread");
 #endif
 
-    _thread->SetNotAlive();
     _thread->Stop();
 
     //res = veData1.jvm->DetachCurrentThread();
@@ -282,8 +273,7 @@ int ThreadTest::RunTest()
 {
     if (_thread)
     {
-        unsigned int id;
-        _thread->Start(id);
+        _thread->Start();
     }
     return 0;
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/extended/ec_metrics_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/extended/ec_metrics_test.cc
index feb36ddc08..a202aea280 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/extended/ec_metrics_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/extended/ec_metrics_test.cc
@@ -41,14 +41,17 @@ TEST_F(EcMetricsTest, ManualTestEcMetrics) {
   int erl, erle, rerl, a_nlp;
   int delay_median = 0;
   int delay_std = 0;
+  float fraction_poor_delays = 0;
 
   for (int i = 0; i < 5; i++) {
     Sleep(1000);
     EXPECT_EQ(0, voe_apm_->GetEchoMetrics(erl, erle, rerl, a_nlp));
-    EXPECT_EQ(0, voe_apm_->GetEcDelayMetrics(delay_median, delay_std));
+    EXPECT_EQ(0, voe_apm_->GetEcDelayMetrics(delay_median, delay_std,
+                                             fraction_poor_delays));
     TEST_LOG("    Echo  : ERL=%5d, ERLE=%5d, RERL=%5d, A_NLP=%5d [dB], "
-        " delay median=%3d, delay std=%3d [ms]\n", erl, erle, rerl, a_nlp,
-        delay_median, delay_std);
+        " delay median=%3d, delay std=%3d [ms], "
+        "fraction_poor_delays=%3.1f [%%]\n", erl, erle, rerl, a_nlp,
+        delay_median, delay_std, fraction_poor_delays * 100);
   }
 
   EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(false));
@@ -63,8 +66,9 @@ TEST_F(EcMetricsTest, GetEcMetricsFailsIfEcNotEnabled) {
 
 TEST_F(EcMetricsTest, GetEcDelayMetricsFailsIfEcNotEnabled) {
   int dummy = 0;
+  float dummy_f = 0;
   EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(true));
-  EXPECT_EQ(-1, voe_apm_->GetEcDelayMetrics(dummy, dummy));
+  EXPECT_EQ(-1, voe_apm_->GetEcDelayMetrics(dummy, dummy, dummy_f));
   EXPECT_EQ(VE_APM_ERROR, voe_base_->LastError());
 }
 
@@ -76,8 +80,11 @@ TEST_F(EcMetricsTest, ManualVerifyEcDelayMetrics) {
 
   for (int i = 0; i < 5; i++) {
     int delay, delay_std;
-    EXPECT_EQ(0, voe_apm_->GetEcDelayMetrics(delay, delay_std));
-    TEST_LOG("Delay = %d, Delay Std = %d\n", delay, delay_std);
+    float fraction_poor_delays;
+    EXPECT_EQ(0, voe_apm_->GetEcDelayMetrics(delay, delay_std,
+                                             fraction_poor_delays));
+    TEST_LOG("Delay = %d, Delay Std = %d, Fraction poor delays = %3.1f\n",
+             delay, delay_std, fraction_poor_delays * 100);
     Sleep(1000);
   }
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc
index 164e547ab1..c825ea5861 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc
@@ -18,7 +18,6 @@
 
 FakeExternalTransport::FakeExternalTransport(webrtc::VoENetwork* ptr)
     : my_network_(ptr),
-      thread_(NULL),
       lock_(NULL),
       event_(NULL),
       length_(0),
@@ -28,26 +27,21 @@ FakeExternalTransport::FakeExternalTransport(webrtc::VoENetwork* ptr)
   const char* thread_name = "external_thread";
   lock_ = webrtc::CriticalSectionWrapper::CreateCriticalSection();
   event_ = webrtc::EventWrapper::Create();
-  thread_ = webrtc::ThreadWrapper::CreateThread(
-      Run, this, webrtc::kHighPriority, thread_name);
+  thread_ = webrtc::ThreadWrapper::CreateThread(Run, this, thread_name);
   if (thread_) {
-    unsigned int id;
-    thread_->Start(id);
+    thread_->Start();
+    thread_->SetPriority(webrtc::kHighPriority);
   }
 }
 
 FakeExternalTransport::~FakeExternalTransport() {
   if (thread_) {
-    thread_->SetNotAlive();
     event_->Set();
-    if (thread_->Stop()) {
-      delete thread_;
-      thread_ = NULL;
-      delete event_;
-      event_ = NULL;
-      delete lock_;
-      lock_ = NULL;
-    }
+    thread_->Stop();
+    delete event_;
+    event_ = NULL;
+    delete lock_;
+    lock_ = NULL;
   }
 }
 
@@ -71,7 +65,9 @@ bool FakeExternalTransport::Process() {
   return true;
 }
 
-int FakeExternalTransport::SendPacket(int channel, const void *data, int len) {
+int FakeExternalTransport::SendPacket(int channel,
+                                      const void *data,
+                                      size_t len) {
   lock_->Enter();
   if (len < 1612) {
     memcpy(packet_buffer_, (const unsigned char*) data, len);
@@ -80,17 +76,17 @@ int FakeExternalTransport::SendPacket(int channel, const void *data, int len) {
   }
   lock_->Leave();
   event_->Set();  // Triggers ReceivedRTPPacket() from worker thread.
-  return len;
+  return static_cast(len);
 }
 
 int FakeExternalTransport::SendRTCPPacket(int channel,
                                           const void *data,
-                                          int len) {
+                                          size_t len) {
   if (delay_is_enabled_) {
     webrtc::SleepMs(delay_time_in_ms_);
   }
   my_network_->ReceivedRTCPPacket(channel, data, len);
-  return len;
+  return static_cast(len);
 }
 
 void FakeExternalTransport::SetDelayStatus(bool enable,
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h
index 318afdaf68..aecc58264a 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h
@@ -23,10 +23,8 @@ class FakeExternalTransport : public webrtc::Transport {
  public:
   explicit FakeExternalTransport(webrtc::VoENetwork* ptr);
   virtual ~FakeExternalTransport();
-
-  virtual int SendPacket(int channel, const void *data, int len) OVERRIDE;
-  virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE;
-
+  int SendPacket(int channel, const void* data, size_t len) override;
+  int SendRTCPPacket(int channel, const void* data, size_t len) override;
   void SetDelayStatus(bool enabled, unsigned int delayInMs = 100);
 
   webrtc::VoENetwork* my_network_;
@@ -34,12 +32,12 @@ class FakeExternalTransport : public webrtc::Transport {
   static bool Run(void* ptr);
   bool Process();
  private:
-  webrtc::ThreadWrapper* thread_;
+  rtc::scoped_ptr thread_;
   webrtc::CriticalSectionWrapper* lock_;
   webrtc::EventWrapper* event_;
  private:
   unsigned char packet_buffer_[1612];
-  int length_;
+  size_t length_;
   int channel_;
   bool delay_is_enabled_;
   int delay_time_in_ms_;
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index 3b3878a777..cee5a58f8e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -13,10 +13,12 @@
 
 #include 
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
+#include "webrtc/system_wrappers/interface/atomic32.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
 
@@ -27,22 +29,32 @@ class LoopBackTransport : public webrtc::Transport {
   LoopBackTransport(webrtc::VoENetwork* voe_network)
       : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
         packet_event_(webrtc::EventWrapper::Create()),
-        thread_(webrtc::ThreadWrapper::CreateThread(NetworkProcess, this)),
-        voe_network_(voe_network) {
-    unsigned int id;
-    thread_->Start(id);
+        thread_(webrtc::ThreadWrapper::CreateThread(
+            NetworkProcess, this, "LoopBackTransport")),
+        voe_network_(voe_network), transmitted_packets_(0) {
+    thread_->Start();
   }
 
   ~LoopBackTransport() { thread_->Stop(); }
 
-  virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
+  int SendPacket(int channel, const void* data, size_t len) override {
     StorePacket(Packet::Rtp, channel, data, len);
-    return len;
+    return static_cast(len);
   }
 
-  virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE {
+  int SendRTCPPacket(int channel, const void* data, size_t len) override {
     StorePacket(Packet::Rtcp, channel, data, len);
-    return len;
+    return static_cast(len);
+  }
+
+  void WaitForTransmittedPackets(int32_t packet_count) {
+    enum {
+      kSleepIntervalMs = 10
+    };
+    int32_t limit = transmitted_packets_.Value() + packet_count;
+    while (transmitted_packets_.Value() < limit) {
+      webrtc::SleepMs(kSleepIntervalMs);
+    }
   }
 
  private:
@@ -50,20 +62,24 @@ class LoopBackTransport : public webrtc::Transport {
     enum Type { Rtp, Rtcp, } type;
 
     Packet() : len(0) {}
-    Packet(Type type, int channel, const void* data, int len)
+    Packet(Type type, int channel, const void* data, size_t len)
         : type(type), channel(channel), len(len) {
       assert(len <= 1500);
-      memcpy(this->data, data, static_cast(len));
+      memcpy(this->data, data, len);
     }
 
     int channel;
     uint8_t data[1500];
-    int len;
+    size_t len;
   };
 
-  void StorePacket(Packet::Type type, int channel, const void* data, int len) {
-    webrtc::CriticalSectionScoped lock(crit_.get());
-    packet_queue_.push_back(Packet(type, channel, data, len));
+  void StorePacket(Packet::Type type, int channel,
+                   const void* data,
+                   size_t len) {
+    {
+      webrtc::CriticalSectionScoped lock(crit_.get());
+      packet_queue_.push_back(Packet(type, channel, data, len));
+    }
     packet_event_->Set();
   }
 
@@ -74,7 +90,6 @@ class LoopBackTransport : public webrtc::Transport {
   bool SendPackets() {
     switch (packet_event_->Wait(10)) {
       case webrtc::kEventSignaled:
-        packet_event_->Reset();
         break;
       case webrtc::kEventTimeout:
         break;
@@ -102,15 +117,17 @@ class LoopBackTransport : public webrtc::Transport {
           voe_network_->ReceivedRTCPPacket(p.channel, p.data, p.len);
           break;
       }
+      ++transmitted_packets_;
     }
     return true;
   }
 
-  webrtc::scoped_ptr crit_;
-  webrtc::scoped_ptr packet_event_;
-  webrtc::scoped_ptr thread_;
+  const rtc::scoped_ptr crit_;
+  const rtc::scoped_ptr packet_event_;
+  const rtc::scoped_ptr thread_;
   std::deque packet_queue_ GUARDED_BY(crit_.get());
   webrtc::VoENetwork* const voe_network_;
+  webrtc::Atomic32 transmitted_packets_;
 };
 
 // This fixture initializes the voice engine in addition to the work
@@ -124,7 +141,7 @@ class AfterInitializationFixture : public BeforeInitializationFixture {
   virtual ~AfterInitializationFixture();
 
  protected:
-  webrtc::scoped_ptr error_observer_;
+  rtc::scoped_ptr error_observer_;
 };
 
 #endif  // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
index 8a7c4c3b4d..488f4489bf 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc
@@ -56,6 +56,10 @@ void BeforeStreamingFixture::ResumePlaying() {
   EXPECT_EQ(0, voe_base_->StartSend(channel_));
 }
 
+void BeforeStreamingFixture::WaitForTransmittedPackets(int32_t packet_count) {
+  transport_->WaitForTransmittedPackets(packet_count);
+}
+
 void BeforeStreamingFixture::SetUpLocalPlayback() {
   transport_ = new LoopBackTransport(voe_network_);
   EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_, *transport_));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h
index 6de583c7e7..1348e481a8 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h
@@ -41,6 +41,9 @@ class BeforeStreamingFixture : public AfterInitializationFixture {
   // Resumes all sending and playout.
   void ResumePlaying();
 
+  // Waits until packet_count packetes have been processed by recipient.
+  void WaitForTransmittedPackets(int32_t packet_count);
+
  private:
   void SetUpLocalPlayback();
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
index 66c00d459a..a733b12705 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
@@ -47,6 +47,7 @@ TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeRecognizesISAC) {
 
 TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeISACPayloadType) {
   strcpy(codec_instance_.plname, "ISAC");
+  codec_instance_.rate = 32000;
 
   codec_instance_.pltype = 123;
   EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
index 3b56bae54c..bfc2a30c5f 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
@@ -97,16 +97,17 @@ TEST_F(CodecTest, VoiceActivityDetectionTypeSettingsCanBeChanged) {
   EXPECT_EQ(vad_mode, webrtc::kVadAggressiveMid);
   EXPECT_FALSE(dtx_disabled);
 
-  // The fourth argument is the DTX disable flag.
-  EXPECT_EQ(0, voe_codec_->SetVADStatus(
-      channel_, true, webrtc::kVadAggressiveHigh, true));
+  // The fourth argument is the DTX disable flag, which is always supposed to
+  // be false.
+  EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true,
+                                        webrtc::kVadAggressiveHigh, false));
   EXPECT_EQ(0, voe_codec_->GetVADStatus(
       channel_, vad_enabled, vad_mode, dtx_disabled));
   EXPECT_EQ(vad_mode, webrtc::kVadAggressiveHigh);
-  EXPECT_TRUE(dtx_disabled);
+  EXPECT_FALSE(dtx_disabled);
 
-  EXPECT_EQ(0, voe_codec_->SetVADStatus(
-      channel_, true, webrtc::kVadConventional, true));
+  EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true,
+                                        webrtc::kVadConventional, false));
   EXPECT_EQ(0, voe_codec_->GetVADStatus(
       channel_, vad_enabled, vad_mode, dtx_disabled));
   EXPECT_EQ(vad_mode, webrtc::kVadConventional);
@@ -157,6 +158,30 @@ TEST_F(CodecTest, OpusMaxPlaybackRateCannotBeSetForNonOpus) {
   }
 }
 
+TEST_F(CodecTest, OpusDtxCanBeSetForOpus) {
+  for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
+    voe_codec_->GetCodec(i, codec_instance_);
+    if (_stricmp("opus", codec_instance_.plname)) {
+      continue;
+    }
+    voe_codec_->SetSendCodec(channel_, codec_instance_);
+    EXPECT_EQ(0, voe_codec_->SetOpusDtx(channel_, false));
+    EXPECT_EQ(0, voe_codec_->SetOpusDtx(channel_, true));
+  }
+}
+
+TEST_F(CodecTest, OpusDtxCannotBeSetForNonOpus) {
+  for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
+    voe_codec_->GetCodec(i, codec_instance_);
+    if (!_stricmp("opus", codec_instance_.plname)) {
+      continue;
+    }
+    voe_codec_->SetSendCodec(channel_, codec_instance_);
+    EXPECT_EQ(-1, voe_codec_->SetOpusDtx(channel_, false));
+    EXPECT_EQ(-1, voe_codec_->SetOpusDtx(channel_, true));
+  }
+}
+
 // TODO(xians, phoglund): Re-enable when issue 372 is resolved.
 TEST_F(CodecTest, DISABLED_ManualVerifySendCodecsForAllPacketSizes) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
index c9a913510b..f3be6355ba 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/neteq_stats_test.cc
@@ -23,31 +23,35 @@ TEST_F(NetEQStatsTest, ManualPrintStatisticsAfterRunningAWhile) {
 
   TEST_LOG("Inspect these statistics and ensure they make sense.\n");
 
-  TEST_LOG("    currentAccelerateRate     = %hu \n",
+  TEST_LOG("    currentAccelerateRate       = %hu \n",
       network_statistics.currentAccelerateRate);
-  TEST_LOG("    currentBufferSize         = %hu \n",
+  TEST_LOG("    currentBufferSize           = %hu \n",
       network_statistics.currentBufferSize);
-  TEST_LOG("    currentDiscardRate        = %hu \n",
+  TEST_LOG("    currentSecondaryDecodedRate = %hu \n",
+      network_statistics.currentSecondaryDecodedRate);
+  TEST_LOG("    currentDiscardRate          = %hu \n",
       network_statistics.currentDiscardRate);
-  TEST_LOG("    currentExpandRate         = %hu \n",
+  TEST_LOG("    currentExpandRate           = %hu \n",
       network_statistics.currentExpandRate);
-  TEST_LOG("    currentPacketLossRate     = %hu \n",
+  TEST_LOG("    currentPacketLossRate       = %hu \n",
       network_statistics.currentPacketLossRate);
-  TEST_LOG("    currentPreemptiveRate     = %hu \n",
+  TEST_LOG("    currentPreemptiveRate       = %hu \n",
       network_statistics.currentPreemptiveRate);
-  TEST_LOG("    preferredBufferSize       = %hu \n",
+  TEST_LOG("    currentSpeechExpandRate     = %hu \n",
+      network_statistics.currentSpeechExpandRate);
+  TEST_LOG("    preferredBufferSize         = %hu \n",
       network_statistics.preferredBufferSize);
-  TEST_LOG("    jitterPeaksFound          = %i \n",
+  TEST_LOG("    jitterPeaksFound            = %i \n",
       network_statistics.jitterPeaksFound);
-  TEST_LOG("    clockDriftPPM             = %i \n",
+  TEST_LOG("    clockDriftPPM               = %i \n",
       network_statistics.clockDriftPPM);
-  TEST_LOG("    meanWaitingTimeMs         = %i \n",
+  TEST_LOG("    meanWaitingTimeMs           = %i \n",
       network_statistics.meanWaitingTimeMs);
-  TEST_LOG("    medianWaitingTimeMs       = %i \n",
+  TEST_LOG("    medianWaitingTimeMs         = %i \n",
       network_statistics.medianWaitingTimeMs);
-  TEST_LOG("    minWaitingTimeMs          = %i \n",
+  TEST_LOG("    minWaitingTimeMs            = %i \n",
       network_statistics.minWaitingTimeMs);
-  TEST_LOG("    maxWaitingTimeMs          = %i \n",
+  TEST_LOG("    maxWaitingTimeMs            = %i \n",
       network_statistics.maxWaitingTimeMs);
 
   // This is only set to a non-zero value in off-mode.
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 2fc117b673..18f064eb77 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -29,11 +29,9 @@ class ExtensionVerifyTransport : public webrtc::Transport {
         audio_level_id_(-1),
         absolute_sender_time_id_(-1) {}
 
-  virtual int SendPacket(int channel, const void* data, int len) OVERRIDE {
+  int SendPacket(int channel, const void* data, size_t len) override {
     webrtc::RTPHeader header;
-    if (parser_->Parse(reinterpret_cast(data),
-                       static_cast(len),
-                       &header)) {
+    if (parser_->Parse(reinterpret_cast(data), len, &header)) {
       bool ok = true;
       if (audio_level_id_ >= 0 &&
           !header.extension.hasAudioLevel) {
@@ -51,11 +49,11 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     }
     // received_packets_ count all packets we receive.
     ++received_packets_;
-    return len;
+    return static_cast(len);
   }
 
-  virtual int SendRTCPPacket(int channel, const void* data, int len) OVERRIDE {
-    return len;
+  int SendRTCPPacket(int channel, const void* data, size_t len) override {
+    return static_cast(len);
   }
 
   void SetAudioLevelId(int id) {
@@ -74,7 +72,7 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     while (received_packets_.Value() < kPacketsExpected) {
       webrtc::SleepMs(kSleepIntervalMs);
     }
-    // Check whether any where 'bad' (didn't contain an extension when they
+    // Check whether any were 'bad' (didn't contain an extension when they
     // where supposed to).
     return bad_packets_.Value() == 0;
   }
@@ -84,7 +82,7 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     kPacketsExpected = 10,
     kSleepIntervalMs = 10
   };
-  webrtc::scoped_ptr parser_;
+  rtc::scoped_ptr parser_;
   webrtc::Atomic32 received_packets_;
   webrtc::Atomic32 bad_packets_;
   int audio_level_id_;
@@ -93,14 +91,12 @@ class ExtensionVerifyTransport : public webrtc::Transport {
 
 class SendRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture {
  protected:
-  virtual void SetUp() OVERRIDE {
+  void SetUp() override {
     EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
     EXPECT_EQ(0, voe_network_->RegisterExternalTransport(channel_,
                                                          verifying_transport_));
   }
-  virtual void TearDown() OVERRIDE {
-    PausePlaying();
-  }
+  void TearDown() override { PausePlaying(); }
 
   ExtensionVerifyTransport verifying_transport_;
 };
@@ -163,27 +159,33 @@ class MockViENetwork : public webrtc::ViENetwork {
   virtual ~MockViENetwork() {}
 
   MOCK_METHOD0(Release, int());
+  MOCK_METHOD4(SetBitrateConfig, void(int, int, int, int));
   MOCK_METHOD2(SetNetworkTransmissionState, void(const int, const bool));
   MOCK_METHOD2(RegisterSendTransport, int(const int, webrtc::Transport&));
   MOCK_METHOD1(DeregisterSendTransport, int(const int));
-  MOCK_METHOD4(ReceivedRTPPacket, int(const int, const void*, const int,
+  MOCK_METHOD4(ReceivedRTPPacket, int(const int, const void*, const size_t,
                                       const webrtc::PacketTime&));
-  MOCK_METHOD3(ReceivedRTCPPacket, int(const int, const void*, const int));
+  MOCK_METHOD3(ReceivedRTCPPacket, int(const int, const void*, const size_t));
   MOCK_METHOD2(SetMTU, int(int, unsigned int));
-  MOCK_METHOD4(ReceivedBWEPacket, int(const int, int64_t, int,
+  MOCK_METHOD4(ReceivedBWEPacket, int(const int, int64_t, size_t,
                                       const webrtc::RTPHeader&));
 };
 
 class ReceiveRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture {
  protected:
-  virtual void SetUp() OVERRIDE {
+  void SetUp() override {
     EXPECT_EQ(0,
         voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true, 11));
     EXPECT_EQ(0,
         voe_rtp_rtcp_->SetReceiveAbsoluteSenderTimeStatus(channel_, true, 11));
   }
 
+  void Wait() {
+    WaitForTransmittedPackets(kPacketsExpected);
+  }
+
   enum {
+    kPacketsExpected = 5,
     kVideoChannelId1 = 667,
     kVideoChannelId2 = 668
   };
@@ -191,8 +193,9 @@ class ReceiveRtpRtcpHeaderExtensionsTest : public BeforeStreamingFixture {
 };
 
 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTDisabled) {
+  EXPECT_CALL(mock_network_, ReceivedBWEPacket(_, _, _, _)).Times(0);
   ResumePlaying();
-  Sleep(500);
+  Wait();
 }
 
 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTFailSetTarget) {
@@ -200,6 +203,7 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTFailSetTarget) {
   EXPECT_EQ(-1, voe_rtp_rtcp_->SetVideoEngineBWETarget(-1, &mock_network_,
                                                       kVideoChannelId1));
   ResumePlaying();
+  Wait();
 }
 
 TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabled) {
@@ -211,7 +215,7 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabled) {
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId1));
   ResumePlaying();
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
 }
 
@@ -226,7 +230,7 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledBadExtensionId) {
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId1));
   ResumePlaying();
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
 }
 
@@ -241,7 +245,7 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledNotSending) {
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId1));
   ResumePlaying();
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
 }
 
@@ -256,7 +260,7 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTEnabledNotReceiving) {
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId1));
   ResumePlaying();
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
 }
 
@@ -275,10 +279,10 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTSwitchViENetwork) {
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_2,
                                                       kVideoChannelId1));
   ResumePlaying();
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId1));
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
 }
 
@@ -295,9 +299,9 @@ TEST_F(ReceiveRtpRtcpHeaderExtensionsTest, ReceiveASTSwitchVideoChannel) {
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId1));
   ResumePlaying();
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, &mock_network_,
                                                       kVideoChannelId2));
-  Sleep(500);
+  Wait();
   EXPECT_EQ(0, voe_rtp_rtcp_->SetVideoEngineBWETarget(channel_, NULL, -1));
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 9a732c0e2b..e5b2f307de 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -29,16 +29,15 @@ class TestRtpObserver : public webrtc::VoERTPObserver {
   void WaitForChangedSsrc() {
     // 10 seconds should be enough.
     EXPECT_EQ(voetest::kEventSignaled, changed_ssrc_event_->Wait(10*1000));
-    changed_ssrc_event_->Reset();
   }
   void SetIncomingSsrc(unsigned int ssrc) {
     voetest::CriticalSectionScoped lock(crit_.get());
     incoming_ssrc_ = ssrc;
   }
  public:
-  voetest::scoped_ptr crit_;
+  rtc::scoped_ptr crit_;
   unsigned int incoming_ssrc_;
-  voetest::scoped_ptr changed_ssrc_event_;
+  rtc::scoped_ptr changed_ssrc_event_;
 };
 
 void TestRtpObserver::OnIncomingSSRCChanged(int channel,
@@ -55,23 +54,6 @@ void TestRtpObserver::OnIncomingSSRCChanged(int channel,
   }
 }
 
-class RtcpAppHandler : public webrtc::VoERTCPObserver {
- public:
-  RtcpAppHandler() : length_in_bytes_(0), sub_type_(0), name_(0) {}
-  void OnApplicationDataReceived(int channel,
-                                 unsigned char sub_type,
-                                 unsigned int name,
-                                 const unsigned char* data,
-                                 unsigned short length_in_bytes);
-  void Reset();
-  ~RtcpAppHandler() {}
-  unsigned short length_in_bytes_;
-  unsigned char data_[256];
-  unsigned char sub_type_;
-  unsigned int name_;
-};
-
-
 static const char* const RTCP_CNAME = "Whatever";
 
 class RtpRtcpTest : public AfterStreamingFixture {
@@ -104,23 +86,6 @@ class RtpRtcpTest : public AfterStreamingFixture {
   LoopBackTransport* transport_;
 };
 
-void RtcpAppHandler::OnApplicationDataReceived(
-    const int /*channel*/, unsigned char sub_type,
-    unsigned int name, const unsigned char* data,
-    unsigned short length_in_bytes) {
-  length_in_bytes_ = length_in_bytes;
-  memcpy(data_, &data[0], length_in_bytes);
-  sub_type_ = sub_type;
-  name_ = name;
-}
-
-void RtcpAppHandler::Reset() {
-  length_in_bytes_ = 0;
-  memset(data_, 0, sizeof(data_));
-  sub_type_ = 0;
-  name_ = 0;
-}
-
 TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
   if (!FLAGS_include_timing_dependent_tests) {
     TEST_LOG("Skipping test - running in slow execution environment...\n");
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
index dddbb2cde9..ad6116dcdd 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_cpu_test.cc
@@ -17,7 +17,7 @@
 #include 
 #endif
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/voice_engine/test/auto_test/voe_test_defines.h"
 
@@ -64,7 +64,7 @@ int VoECpuTest::DoTest() {
   CHECK(base->Init());
   channel = base->CreateChannel();
 
-  scoped_ptr voice_socket_transport(
+  rtc::scoped_ptr voice_socket_transport(
       new VoiceChannelTransport(voe_network, channel));
 
   CHECK(voice_socket_transport->SetSendDestination("127.0.0.1", 5566));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc
index d2c19175c9..ab8fffe832 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.cc
@@ -24,9 +24,8 @@
 
 #include "webrtc/voice_engine/test/auto_test/voe_stress_test.h"
 
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
 #include "webrtc/voice_engine/test/auto_test/voe_test_defines.h"
@@ -145,7 +144,7 @@ int VoEStressTest::StartStopTest() {
   printf("Test will take approximately %d minutes. \n",
          numberOfLoops * loopSleep / 1000 / 60 + 1);
 
-  scoped_ptr voice_channel_transport(
+  rtc::scoped_ptr voice_channel_transport(
       new VoiceChannelTransport(voe_network, 0));
 
   for (i = 0; i < numberOfLoops; ++i) {
@@ -335,11 +334,9 @@ int VoEStressTest::MultipleThreadsTest() {
   int rnd(0);
 
   // Start extra thread
-  const char* threadName = "StressTest Extra API Thread";
   _ptrExtraApiThread = ThreadWrapper::CreateThread(RunExtraApi, this,
-                                                   kNormalPriority, threadName);
-  unsigned int id(0);
-  VALIDATE_STRESS(!_ptrExtraApiThread->Start(id));
+                                                   "StressTestExtraApiThread");
+  VALIDATE_STRESS(!_ptrExtraApiThread->Start());
 
   //       Some possible extensions include:
   //       Add more API calls to randomize
@@ -369,7 +366,6 @@ int VoEStressTest::MultipleThreadsTest() {
 
   // Stop extra thread
   VALIDATE_STRESS(!_ptrExtraApiThread->Stop());
-  delete _ptrExtraApiThread;
 
   ///////////// End test /////////////
 
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h
index 15653bc55e..7128b238ff 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/voe_stress_test.h
@@ -11,9 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
 #define WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
 
-namespace webrtc {
-class ThreadWrapper;
-}
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
 
 namespace voetest {
 // TODO(andrew): using directives are not permitted.
@@ -23,8 +21,7 @@ class VoETestManager;
 
 class VoEStressTest {
  public:
-  VoEStressTest(VoETestManager& mgr) :
-    _mgr(mgr), _ptrExtraApiThread(NULL) {
+  VoEStressTest(VoETestManager& mgr) : _mgr(mgr) {
   }
   ~VoEStressTest() {
   }
@@ -41,7 +38,7 @@ class VoEStressTest {
 
   VoETestManager& _mgr;
 
-  ThreadWrapper* _ptrExtraApiThread;
+  rtc::scoped_ptr _ptrExtraApiThread;
 };
 
 }  // namespace voetest
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc b/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
index 09a70a9d82..419ba55d78 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -19,9 +19,9 @@
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/channel_transport/include/channel_transport.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
@@ -140,7 +140,7 @@ int main(int argc, char** argv) {
 
   MyObserver my_observer;
 
-  scoped_ptr trace_to_stderr;
+  rtc::scoped_ptr trace_to_stderr;
   if (!FLAGS_use_log_file) {
     trace_to_stderr.reset(new test::TraceToStderr);
   } else {
@@ -232,7 +232,9 @@ void RunTest(std::string out_path) {
   bool typing_detection = false;
   bool muted = false;
   bool opus_stereo = false;
+  bool opus_dtx = false;
   bool experimental_ns_enabled = false;
+  bool debug_recording_started = false;
 
 #if defined(WEBRTC_ANDROID)
   std::string resource_path = "/sdcard/";
@@ -258,7 +260,7 @@ void RunTest(std::string out_path) {
     fflush(NULL);
   }
 
-  scoped_ptr voice_channel_transport(
+  rtc::scoped_ptr voice_channel_transport(
       new VoiceChannelTransport(netw, chan));
 
   char ip[64];
@@ -446,8 +448,10 @@ void RunTest(std::string out_path) {
       printf("%i. Toggle Opus stereo (Opus must be selected again to apply "
              "the setting) \n", option_index++);
       printf("%i. Set Opus maximum playback rate \n", option_index++);
+      printf("%i. Toggle Opus DTX \n", option_index++);
       printf("%i. Set bit rate (only take effect on codecs that allow the "
              "change) \n", option_index++);
+      printf("%i. Toggle debug recording \n", option_index++);
 
       printf("Select action or %i to stop the call: ", option_index);
       int option_selection;
@@ -457,9 +461,14 @@ void RunTest(std::string out_path) {
       if (option_selection < option_index) {
         res = codec->GetCodec(option_selection, cinst);
         VALIDATE;
-        SetStereoIfOpus(opus_stereo, &cinst);
-        printf("Set primary codec\n");
-        res = codec->SetSendCodec(chan, cinst);
+        if (strcmp(cinst.plname, "red") == 0) {
+          printf("Enabling RED\n");
+          res = rtp_rtcp->SetREDStatus(chan, true, cinst.pltype);
+        } else {
+          SetStereoIfOpus(opus_stereo, &cinst);
+          printf("Set primary codec\n");
+          res = codec->SetSendCodec(chan, cinst);
+        }
         VALIDATE;
       } else if (option_selection == option_index++) {
         enable_cng = !enable_cng;
@@ -766,6 +775,11 @@ void RunTest(std::string out_path) {
         ASSERT_EQ(1, scanf("%i", &max_playback_rate));
         res = codec->SetOpusMaxPlaybackRate(chan, max_playback_rate);
         VALIDATE;
+      } else if (option_selection == option_index++) {
+        opus_dtx = !opus_dtx;
+        res = codec->SetOpusDtx(chan, opus_dtx);
+        VALIDATE;
+        printf("Opus DTX %s.\n", opus_dtx ? "enabled" : "disabled");
       } else if (option_selection == option_index++) {
         res = codec->GetSendCodec(chan, cinst);
         VALIDATE;
@@ -773,11 +787,25 @@ void RunTest(std::string out_path) {
         ASSERT_EQ(1, scanf("%i", &cinst.rate));
         res = codec->SetSendCodec(chan, cinst);
         VALIDATE;
+      } else if (option_selection == option_index++) {
+        const char* kDebugFileName = "audio.aecdump";
+        if (debug_recording_started) {
+          apm->StopDebugRecording();
+          printf("Debug recording named %s stopped\n", kDebugFileName);
+        } else {
+          apm->StartDebugRecording(kDebugFileName);
+          printf("Debug recording named %s started\n", kDebugFileName);
+        }
+        debug_recording_started = !debug_recording_started;
       } else {
         break;
       }
     }
 
+    if (debug_recording_started) {
+      apm->StopDebugRecording();
+    }
+
     if (send) {
       printf("Stop Send \n");
       res = base1->StopSend(chan);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc
index 4929ae4af6..4436a86d86 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/test/win_test/WinTestDlg.cc
@@ -129,8 +129,9 @@ class MyTransport : public Transport
 {
 public:
     MyTransport(VoENetwork* veNetwork);
-    virtual int SendPacket(int channel, const void *data, int len) OVERRIDE;
-    virtual int SendRTCPPacket(int channel, const void *data, int len) OVERRIDE;
+    int SendPacket(int channel, const void* data, size_t len) override;
+    int SendRTCPPacket(int channel, const void* data, size_t len) override;
+
 private:
     VoENetwork* _veNetworkPtr;
 };
@@ -141,14 +142,14 @@ MyTransport::MyTransport(VoENetwork* veNetwork) :
 }
 
 int
-MyTransport::SendPacket(int channel, const void *data, int len)
+MyTransport::SendPacket(int channel, const void *data, size_t len)
 {
     _veNetworkPtr->ReceivedRTPPacket(channel, data, len);
     return len;
 }
 
 int
-MyTransport::SendRTCPPacket(int channel, const void *data, int len)
+MyTransport::SendRTCPPacket(int channel, const void *data, size_t len)
 {
     _veNetworkPtr->ReceivedRTCPPacket(channel, data, len);
     return len;
@@ -923,23 +924,27 @@ void CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt()
 {
     BOOL ret;
     int pt = GetDlgItemInt(IDC_EDIT_EVENT_RX_PT, &ret);
-    if (ret == FALSE)
+    if (ret == FALSE || pt < 0 || pt > 127)
         return;
     CodecInst codec;
     strcpy_s(codec.plname, 32, "telephone-event");
-    codec.pltype = pt; codec.channels = 1; codec.plfreq = 8000;
+    codec.pltype = pt;
+    codec.channels = 1;
+    codec.plfreq = 8000;
     TEST2(_veCodecPtr->SetRecPayloadType(_channel, codec) == 0,
-        _T("SetSendTelephoneEventPayloadType(channel=%d, codec.pltype=%u)"), _channel, codec.pltype);
+          _T("SetRecPayloadType(channel=%d, codec.pltype=%d)"), _channel,
+          codec.pltype);
 }
 
 void CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt()
 {
     BOOL ret;
     int pt = GetDlgItemInt(IDC_EDIT_EVENT_TX_PT, &ret);
-    if (ret == FALSE)
+    if (ret == FALSE || pt < 0 || pt > 127)
         return;
     TEST2(_veDTMFPtr->SetSendTelephoneEventPayloadType(_channel, pt) == 0,
-        _T("SetSendTelephoneEventPayloadType(channel=%d, type=%u)"), _channel, pt);
+          _T("SetSendTelephoneEventPayloadType(channel=%d, type=%d)"), _channel,
+          pt);
 }
 
 void CTelephonyEvent::OnBnClickedCheckDetectInband()
diff --git a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc
index fa8697e2e9..ac9247f062 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.cc
@@ -259,15 +259,8 @@ TransmitMixer::SetEngineInformation(ProcessThread& processThread,
     _engineStatisticsPtr = &engineStatistics;
     _channelManagerPtr = &channelManager;
 
-    if (_processThreadPtr->RegisterModule(&_monitorModule) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
-                     "TransmitMixer::SetEngineInformation() failed to"
-                     "register the monitor module");
-    } else
-    {
-        _monitorModule.RegisterObserver(*this);
-    }
+    _processThreadPtr->RegisterModule(&_monitorModule);
+    _monitorModule.RegisterObserver(*this);
 
     return 0;
 }
@@ -1196,7 +1189,7 @@ int32_t TransmitMixer::RecordAudioToFile(
 int32_t TransmitMixer::MixOrReplaceAudioWithFile(
     int mixingFrequency)
 {
-    scoped_ptr fileBuffer(new int16_t[640]);
+  rtc::scoped_ptr fileBuffer(new int16_t[640]);
 
     int fileSamples(0);
     {
diff --git a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h
index afeec3ac88..919de13123 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/transmit_mixer.h
@@ -11,13 +11,13 @@
 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/utility/interface/file_player.h"
 #include "webrtc/modules/utility/interface/file_recorder.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"
@@ -229,7 +229,7 @@ private:
     int32_t _remainingMuteMicTimeMs;
     bool stereo_codec_;
     bool swap_stereo_channels_;
-    scoped_ptr mono_buffer_;
+    rtc::scoped_ptr mono_buffer_;
 };
 
 }  // namespace voe
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc
index 63a4ed768a..a310628eec 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -921,9 +921,10 @@ int VoEAudioProcessingImpl::GetEchoMetrics(int& ERL,
 }
 
 int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
-                                              int& delay_std) {
+                                              int& delay_std,
+                                              float& fraction_poor_delays) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcDelayMetrics(median=?, std=?)");
+               "GetEcDelayMetrics(median=?, std=?, fraction_poor_delays=?)");
 #ifdef WEBRTC_VOICE_ENGINE_ECHO
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -937,9 +938,10 @@ int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
 
   int median = 0;
   int std = 0;
+  float poor_fraction = 0;
   // Get delay-logging values from Audio Processing Module.
   if (_shared->audio_processing()->echo_cancellation()->GetDelayMetrics(
-        &median, &std)) {
+        &median, &std, &poor_fraction)) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetEcDelayMetrics(), AudioProcessingModule delay-logging "
                  "error");
@@ -949,10 +951,12 @@ int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
   // EC delay-logging metrics
   delay_median = median;
   delay_std = std;
+  fraction_poor_delays = poor_fraction;
 
   WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetEcDelayMetrics() => delay_median=%d, delay_std=%d",
-               delay_median, delay_std);
+               "GetEcDelayMetrics() => delay_median=%d, delay_std=%d, "
+               "fraction_poor_delays=%f", delay_median, delay_std,
+               fraction_poor_delays);
   return 0;
 #else
   _shared->SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h
index 524439d5c1..26f7eec745 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_audio_processing_impl.h
@@ -76,7 +76,8 @@ class VoEAudioProcessingImpl : public VoEAudioProcessing {
 
   virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP);
 
-  virtual int GetEcDelayMetrics(int& delay_median, int& delay_std);
+  virtual int GetEcDelayMetrics(int& delay_median, int& delay_std,
+                                float& fraction_poor_delays);
 
   virtual int StartDebugRecording(const char* fileNameUTF8);
   virtual int StartDebugRecording(FILE* file_handle);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_auto_test.isolate b/media/webrtc/trunk/webrtc/voice_engine/voe_auto_test.isolate
index c13110e718..e89620dde0 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_auto_test.isolate
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_auto_test.isolate
@@ -7,14 +7,6 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
-      'variables': {
-        'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
-        ],
-      },
-    }],
     ['OS=="linux" or OS=="mac" or OS=="win"', {
       'variables': {
         'command': [
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc
index 0cf787acf1..430ee40c67 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_base_impl.cc
@@ -314,12 +314,7 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
 
     if (_shared->process_thread())
     {
-        if (_shared->process_thread()->Start() != 0)
-        {
-            _shared->SetLastError(VE_THREAD_ERROR, kTraceError,
-                "Init() failed to start module process thread");
-            return -1;
-        }
+        _shared->process_thread()->Start();
     }
 
     // Create an internal ADM if the user has not added an external
@@ -347,12 +342,9 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
 
     // Register the ADM to the process thread, which will drive the error
     // callback mechanism
-    if (_shared->process_thread() &&
-        _shared->process_thread()->RegisterModule(_shared->audio_device()) != 0)
+    if (_shared->process_thread())
     {
-        _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
-            "Init() failed to register the ADM");
-        return -1;
+        _shared->process_thread()->RegisterModule(_shared->audio_device());
     }
 
     bool available(false);
@@ -432,7 +424,7 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
     }
 
     if (!audioproc) {
-      audioproc = AudioProcessing::Create(VoEId(_shared->instance_id(), -1));
+      audioproc = AudioProcessing::Create(_shared->channel_manager().config_);
       if (!audioproc) {
         LOG(LS_ERROR) << "Failed to create AudioProcessing.";
         _shared->SetLastError(VE_NO_MEMORY);
@@ -962,18 +954,10 @@ int32_t VoEBaseImpl::TerminateInternal()
     {
         if (_shared->audio_device())
         {
-            if (_shared->process_thread()->
-                    DeRegisterModule(_shared->audio_device()) != 0)
-            {
-                _shared->SetLastError(VE_THREAD_ERROR, kTraceError,
-                    "TerminateInternal() failed to deregister ADM");
-            }
-        }
-        if (_shared->process_thread()->Stop() != 0)
-        {
-            _shared->SetLastError(VE_THREAD_ERROR, kTraceError,
-                "TerminateInternal() failed to stop module process thread");
+            _shared->process_thread()->DeRegisterModule(
+                _shared->audio_device());
         }
+        _shared->process_thread()->Stop();
     }
 
     if (_shared->audio_device())
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc
index 6057534d04..69aba712ef 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_base_unittest.cc
@@ -11,9 +11,9 @@
 #include "webrtc/voice_engine/include/voe_base.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -32,11 +32,11 @@ class VoEBaseTest : public ::testing::Test {
 
   VoiceEngine* voe_;
   VoEBase* base_;
-  scoped_ptr adm_;
+  rtc::scoped_ptr adm_;
 };
 
 TEST_F(VoEBaseTest, AcceptsAudioProcessingPtr) {
-  AudioProcessing* audioproc = AudioProcessing::Create(0);
+  AudioProcessing* audioproc = AudioProcessing::Create();
   EXPECT_EQ(0, base_->Init(adm_.get(), audioproc));
   EXPECT_EQ(audioproc, base_->audio_processing());
 }
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc
index 6f470be106..2b0141fa81 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.cc
@@ -436,6 +436,23 @@ int VoECodecImpl::SetOpusMaxPlaybackRate(int channel, int frequency_hz) {
   return channelPtr->SetOpusMaxPlaybackRate(frequency_hz);
 }
 
+int VoECodecImpl::SetOpusDtx(int channel, bool enable_dtx) {
+  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+               "SetOpusDtx(channel=%d, enable_dtx=%d)", channel, enable_dtx);
+  if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return -1;
+  }
+  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+  voe::Channel* channelPtr = ch.channel();
+  if (channelPtr == NULL) {
+    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+                          "SetOpusDtx failed to locate channel");
+    return -1;
+  }
+  return channelPtr->SetOpusDtx(enable_dtx);
+}
+
 void VoECodecImpl::ACMToExternalCodecRepresentation(CodecInst& toInst,
                                                     const CodecInst& fromInst)
 {
@@ -514,111 +531,6 @@ void VoECodecImpl::ExternalToACMCodecRepresentation(CodecInst& toInst,
     }
 }
 
-int VoECodecImpl::SetSecondarySendCodec(int channel, const CodecInst& codec,
-                                        int red_payload_type) {
-  CodecInst copy_codec;
-  ExternalToACMCodecRepresentation(copy_codec, codec);
-
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "SetSecondarySendCodec(channel=%d, codec)", channel);
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "codec: plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
-               "channels=%d, rate=%d", codec.plname, codec.pacsize,
-               codec.plfreq, codec.pltype, codec.channels, codec.rate);
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-
-  // External sanity checks performed outside the ACM
-  if ((STR_CASE_CMP(copy_codec.plname, "L16") == 0) &&
-      (copy_codec.pacsize >= 960)) {
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-                          "SetSecondarySendCodec() invalid L16 packet size");
-    return -1;
-  }
-
-  // None of the following codecs can be registered as the secondary encoder.
-  if (!STR_CASE_CMP(copy_codec.plname, "CN") ||
-      !STR_CASE_CMP(copy_codec.plname, "TELEPHONE-EVENT") ||
-      !STR_CASE_CMP(copy_codec.plname, "RED"))  {
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-                          "SetSecondarySendCodec() invalid codec name");
-    return -1;
-  }
-
-  // Only mono and stereo are supported.
-  if ((copy_codec.channels != 1) && (copy_codec.channels != 2)) {
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-                          "SetSecondarySendCodec() invalid number of channels");
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "SetSecondarySendCodec() failed to locate channel");
-    return -1;
-  }
-  if (!AudioCodingModule::IsCodecValid(copy_codec)) {
-    _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
-                          "SetSecondarySendCodec() invalid codec");
-    return -1;
-  }
-  if (channelPtr->SetSecondarySendCodec(copy_codec, red_payload_type) != 0) {
-    _shared->SetLastError(VE_CANNOT_SET_SECONDARY_SEND_CODEC, kTraceError,
-                          "SetSecondarySendCodec() failed to set secondary "
-                          "send codec");
-    return -1;
-  }
-  return 0;
-}
-
-int VoECodecImpl::GetSecondarySendCodec(int channel, CodecInst& codec) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "GetSecondarySendCodec(channel=%d, codec=?)", channel);
-  if (!_shared->statistics().Initialized()) {
-    _shared->SetLastError(VE_NOT_INITED, kTraceError);
-    return -1;
-  }
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "GetSecondarySendCodec() failed to locate channel");
-    return -1;
-  }
-  CodecInst acm_codec;
-  if (channelPtr->GetSecondarySendCodec(&acm_codec) != 0) {
-    _shared->SetLastError(VE_CANNOT_GET_SECONDARY_SEND_CODEC, kTraceError,
-                          "GetSecondarySendCodec() failed to get secondary "
-                          "send codec");
-    return -1;
-  }
-  ACMToExternalCodecRepresentation(codec, acm_codec);
-  WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
-               VoEId(_shared->instance_id(), -1),
-               "GetSecondarySendCodec() => plname=%s, pacsize=%d, plfreq=%d, "
-               "channels=%d, rate=%d", codec.plname, codec.pacsize,
-               codec.plfreq, codec.channels, codec.rate);
-  return 0;
-}
-
-int VoECodecImpl::RemoveSecondarySendCodec(int channel) {
-  WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
-               "RemoveSecondarySendCodec(channel=%d)", channel);
-  voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
-  voe::Channel* channelPtr = ch.channel();
-  if (channelPtr == NULL) {
-    _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
-                          "RemoveSecondarySendCodec() failed to locate "
-                          "channel");
-    return -1;
-  }
-  channelPtr->RemoveSecondarySendCodec();
-  return 0;
-}
-
 #endif  // WEBRTC_VOICE_ENGINE_CODEC_API
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h
index f001df8660..dad808dbbf 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_impl.h
@@ -56,13 +56,7 @@ public:
 
     virtual int SetOpusMaxPlaybackRate(int channel, int frequency_hz);
 
-    // Dual-streaming
-    virtual int SetSecondarySendCodec(int channel, const CodecInst& codec,
-                                      int red_payload_type);
-
-    virtual int RemoveSecondarySendCodec(int channel);
-
-    virtual int GetSecondarySendCodec(int channel, CodecInst& codec);
+    virtual int SetOpusDtx(int channel, bool enable_dtx);
 
 protected:
     VoECodecImpl(voe::SharedData* shared);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc
index 7e440b2ada..6eb5a5145c 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_codec_unittest.cc
@@ -11,8 +11,8 @@
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_hardware.h"
@@ -94,81 +94,13 @@ class VoECodecTest : public ::testing::Test {
   int channel_;
   CodecInst primary_;
   CodecInst valid_secondary_;
-  scoped_ptr adm_;
+  rtc::scoped_ptr adm_;
 
   // A codec which is not valid to be registered as secondary codec.
   CodecInst invalid_secondary_;
   int red_payload_type_;
 };
 
-
-TEST_F(VoECodecTest,
-       DISABLED_ON_ANDROID(DualStreamSetSecondaryBeforePrimaryFails)) {
-  // Setting secondary before a primary is registered should fail.
-  EXPECT_EQ(-1, voe_codec_->SetSecondarySendCodec(channel_, valid_secondary_,
-                                                  red_payload_type_));
-  red_payload_type_ = 1;
-}
-
-TEST_F(VoECodecTest,
-       DISABLED_ON_ANDROID(DualStreamRegisterWithWrongInputsFails)) {
-  // Register primary codec.
-  EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, primary_));
-
-  // Wrong secondary.
-  EXPECT_EQ(-1, voe_codec_->SetSecondarySendCodec(channel_, invalid_secondary_,
-                                                  red_payload_type_));
-
-  // Wrong payload.
-  EXPECT_EQ(-1, voe_codec_->SetSecondarySendCodec(channel_, valid_secondary_,
-                                                  -1));
-  // Wrong channel.
-  EXPECT_EQ(-1, voe_codec_->SetSecondarySendCodec(channel_ + 1,
-                                                  valid_secondary_,
-                                                  red_payload_type_));
-}
-
-TEST_F(VoECodecTest, DISABLED_ON_ANDROID(DualStreamGetSecodaryEncoder)) {
-  // Register primary codec.
-  EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, primary_));
-
-  // Register a valid codec.
-  EXPECT_EQ(0, voe_codec_->SetSecondarySendCodec(channel_, valid_secondary_,
-                                                 red_payload_type_));
-  CodecInst my_codec;
-
-  // Get secondary codec from wrong channel.
-  EXPECT_EQ(-1, voe_codec_->GetSecondarySendCodec(channel_ + 1, my_codec));
-
-  // Get secondary and compare.
-  memset(&my_codec, 0, sizeof(my_codec));
-  EXPECT_EQ(0, voe_codec_->GetSecondarySendCodec(channel_, my_codec));
-
-  EXPECT_EQ(valid_secondary_.plfreq, my_codec.plfreq);
-  EXPECT_EQ(valid_secondary_.channels, my_codec.channels);
-  EXPECT_EQ(valid_secondary_.pacsize, my_codec.pacsize);
-  EXPECT_EQ(valid_secondary_.rate, my_codec.rate);
-  EXPECT_EQ(valid_secondary_.pltype, my_codec.pltype);
-  EXPECT_EQ(0, STR_CASE_CMP(valid_secondary_.plname, my_codec.plname));
-}
-
-TEST_F(VoECodecTest, DISABLED_ON_ANDROID(DualStreamRemoveSecondaryCodec)) {
-  // Register primary codec.
-  EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, primary_));
-
-  // Register a valid codec.
-  EXPECT_EQ(0, voe_codec_->SetSecondarySendCodec(channel_, valid_secondary_,
-                                                 red_payload_type_));
-  // Remove from wrong channel.
-  EXPECT_EQ(-1, voe_codec_->RemoveSecondarySendCodec(channel_ + 1));
-  EXPECT_EQ(0, voe_codec_->RemoveSecondarySendCodec(channel_));
-
-  CodecInst my_codec;
-
-  // Get should fail, if secondary is removed.
-  EXPECT_EQ(-1, voe_codec_->GetSecondarySendCodec(channel_, my_codec));
-}
-
 TEST(VoECodecInst, TestCompareCodecInstances) {
   CodecInst codec1, codec2;
   memset(&codec1, 0, sizeof(CodecInst));
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc
index 921701b21f..4505f9ee86 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.cc
@@ -787,6 +787,14 @@ int VoEHardwareImpl::PlayoutSampleRate(unsigned int* samples_per_sec) const {
   return _shared->audio_device()->PlayoutSampleRate(samples_per_sec);
 }
 
+bool VoEHardwareImpl::BuiltInAECIsAvailable() const {
+if (!_shared->statistics().Initialized()) {
+    _shared->SetLastError(VE_NOT_INITED, kTraceError);
+    return false;
+  }
+  return _shared->audio_device()->BuiltInAECIsAvailable();
+}
+
 #endif  // WEBRTC_VOICE_ENGINE_HARDWARE_API
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h
index 4e06f978d9..e23e4e5df4 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_hardware_impl.h
@@ -28,15 +28,15 @@ public:
     virtual int GetRecordingDeviceName(int index,
                                        char strNameUTF8[128],
                                        char strGuidUTF8[128]);
+ 
+    virtual int GetRecordingDeviceStatus(bool& isAvailable);
+
+    virtual int GetPlayoutDeviceStatus(bool& isAvailable);
 
     virtual int GetPlayoutDeviceName(int index,
                                      char strNameUTF8[128],
                                      char strGuidUTF8[128]);
 
-    virtual int GetRecordingDeviceStatus(bool& isAvailable);
-
-    virtual int GetPlayoutDeviceStatus(bool& isAvailable);
-
     virtual int SetRecordingDevice(
         int index,
         StereoChannel recordingChannel = kStereoBoth);
@@ -44,9 +44,7 @@ public:
     virtual int SetPlayoutDevice(int index);
 
     virtual int SetAudioDeviceLayer(AudioLayers audioLayer);
-
-    virtual int GetAudioDeviceLayer(AudioLayers& audioLayer);
-
+ 
     virtual int GetCPULoad(int& loadPercent);
 
     virtual int ResetAudioDevice();
@@ -62,11 +60,15 @@ public:
     virtual int EnableBuiltInAEC(bool enable);
     virtual bool BuiltInAECIsEnabled() const;
 
+    virtual int GetAudioDeviceLayer(AudioLayers& audioLayer);
+
     virtual int SetRecordingSampleRate(unsigned int samples_per_sec);
     virtual int RecordingSampleRate(unsigned int* samples_per_sec) const;
     virtual int SetPlayoutSampleRate(unsigned int samples_per_sec);
     virtual int PlayoutSampleRate(unsigned int* samples_per_sec) const;
 
+    virtual bool BuiltInAECIsAvailable() const;
+
 protected:
     VoEHardwareImpl(voe::SharedData* shared);
     virtual ~VoEHardwareImpl();
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc
index 70c548857b..89d1b04f48 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.cc
@@ -10,6 +10,7 @@
 
 #include "webrtc/voice_engine/voe_network_impl.h"
 
+#include "webrtc/base/format_macros.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace.h"
@@ -88,17 +89,18 @@ int VoENetworkImpl::DeRegisterExternalTransport(int channel)
 
 int VoENetworkImpl::ReceivedRTPPacket(int channel,
                                       const void* data,
-                                      unsigned int length) {
+                                      size_t length) {
   return ReceivedRTPPacket(channel, data, length, webrtc::PacketTime());
 }
 
 int VoENetworkImpl::ReceivedRTPPacket(int channel,
                                       const void* data,
-                                      unsigned int length,
+                                      size_t length,
                                       const PacketTime& packet_time)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "ReceivedRTPPacket(channel=%d, length=%u)", channel, length);
+                 "ReceivedRTPPacket(channel=%d, length=%" PRIuS ")", channel,
+                 length);
     if (!_shared->statistics().Initialized())
     {
         _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -137,10 +139,11 @@ int VoENetworkImpl::ReceivedRTPPacket(int channel,
 }
 
 int VoENetworkImpl::ReceivedRTCPPacket(int channel, const void* data,
-                                       unsigned int length)
+                                       size_t length)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1),
-                 "ReceivedRTCPPacket(channel=%d, length=%u)", channel, length);
+                 "ReceivedRTCPPacket(channel=%d, length=%" PRIuS ")", channel,
+                 length);
     if (!_shared->statistics().Initialized())
     {
         _shared->SetLastError(VE_NOT_INITED, kTraceError);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h
index 0cf07ad1fd..ee9b92ec3b 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_network_impl.h
@@ -22,22 +22,17 @@ namespace webrtc
 class VoENetworkImpl: public VoENetwork
 {
 public:
-    virtual int RegisterExternalTransport(int channel,
-                                          Transport& transport) OVERRIDE;
+ int RegisterExternalTransport(int channel, Transport& transport) override;
 
-    virtual int DeRegisterExternalTransport(int channel) OVERRIDE;
+ int DeRegisterExternalTransport(int channel) override;
 
-    virtual int ReceivedRTPPacket(int channel,
-                                  const void* data,
-                                  unsigned int length) OVERRIDE;
-    virtual int ReceivedRTPPacket(int channel,
-                                  const void* data,
-                                  unsigned int length,
-                                  const PacketTime& packet_time) OVERRIDE;
+ int ReceivedRTPPacket(int channel, const void* data, size_t length) override;
+ int ReceivedRTPPacket(int channel,
+                       const void* data,
+                       size_t length,
+                       const PacketTime& packet_time) override;
 
-    virtual int ReceivedRTCPPacket(int channel,
-                                   const void* data,
-                                   unsigned int length) OVERRIDE;
+ int ReceivedRTCPPacket(int channel, const void* data, size_t length) override;
 
 protected:
     VoENetworkImpl(voe::SharedData* shared);
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
index 47a6a219de..422cd27f0e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
@@ -249,7 +249,8 @@ int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable)
             "SetRTCPStatus() failed to locate channel");
         return -1;
     }
-    return channelPtr->SetRTCPStatus(enable);
+    channelPtr->SetRTCPStatus(enable);
+    return 0;
 }
 
 int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled)
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp b/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp
index 36e9b6af74..232dff127e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine.gyp
@@ -15,6 +15,7 @@
       'target_name': 'voice_engine',
       'type': 'static_library',
       'dependencies': [
+        '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
         '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
@@ -24,15 +25,12 @@
         '<(webrtc_root)/modules/modules.gyp:media_file',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
-        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+        '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       ],
       'defines': [
         'WEBRTC_EXTERNAL_TRANSPORT',
       ],
       'sources': [
-        '../common_types.h',
-        '../engine_configurations.h',
-        '../typedefs.h',
         'include/voe_audio_processing.h',
         'include/voe_base.h',
         'include/voe_codec.h',
@@ -122,7 +120,7 @@
             '<(webrtc_root)/modules/modules.gyp:media_file',
             '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
             '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'sources': [
@@ -150,8 +148,8 @@
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
            ],
@@ -218,8 +216,8 @@
             'voice_engine',
             '<(DEPTH)/testing/gtest.gyp:gtest',
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
           ],
@@ -239,7 +237,7 @@
               'type': 'executable',
               'dependencies': [
                 'voice_engine',
-                '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+                '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
                 '<(webrtc_root)/test/test.gyp:test_support',
               ],
               'sources': [
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc
index d77ffdaec5..8fa9489d35 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.cc
@@ -154,7 +154,7 @@ bool VoiceEngine::Delete(VoiceEngine*& voiceEngine)
 }
 
 #if !defined(WEBRTC_CHROMIUM_BUILD)
-int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
+int VoiceEngine::SetAndroidObjects(void* javaVM, void* context)
 {
 #ifdef WEBRTC_ANDROID
 #ifdef WEBRTC_ANDROID_OPENSLES
@@ -165,16 +165,20 @@ int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
   typedef AudioDeviceTemplate
       AudioDeviceInstanceJni;
 #endif
-  if (javaVM && env && context) {
+  if (javaVM && context) {
 #if !defined(WEBRTC_GONK) && defined(ANDROID)
-    AudioDeviceInstanceJni::SetAndroidAudioDeviceObjects(javaVM, env, context);
+    AudioDeviceInstanceJni::SetAndroidAudioDeviceObjects(javaVM, context);
+#endif
+#ifdef WEBRTC_ANDROID_OPENSLES
+    AudioDeviceInstance::SetAndroidAudioDeviceObjects(javaVM, context);
 #endif
-    AudioDeviceInstance::SetAndroidAudioDeviceObjects(javaVM, env, context);
   } else {
 #if !defined(WEBRTC_GONK) && defined(ANDROID)
     AudioDeviceInstanceJni::ClearAndroidAudioDeviceObjects();
 #endif
+#ifdef WEBRTC_ANDROID_OPENSLES
     AudioDeviceInstance::ClearAndroidAudioDeviceObjects();
+#endif
   }
   return 0;
 #else
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h
index 1d2b9c5efd..992bc2d83e 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_impl.h
@@ -136,7 +136,7 @@ public:
 
 private:
     Atomic32 _ref_count;
-    scoped_ptr own_config_;
+    rtc::scoped_ptr own_config_;
 };
 
 }  // namespace webrtc
diff --git a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_unittests.isolate b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_unittests.isolate
index 4cd1f907e3..0d55515f99 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/voice_engine_unittests.isolate
+++ b/media/webrtc/trunk/webrtc/voice_engine/voice_engine_unittests.isolate
@@ -7,14 +7,6 @@
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
-    ['OS=="android"', {
-      'variables': {
-        'files': [
-          '<(DEPTH)/data/',
-          '<(DEPTH)/resources/',
-        ],
-      },
-    }],
     ['OS=="linux" or OS=="mac" or OS=="win"', {
       'variables': {
         'command': [
diff --git a/media/webrtc/trunk/webrtc/webrtc.gyp b/media/webrtc/trunk/webrtc/webrtc.gyp
index f9ed8838f0..1ecc0cd708 100644
--- a/media/webrtc/trunk/webrtc/webrtc.gyp
+++ b/media/webrtc/trunk/webrtc/webrtc.gyp
@@ -28,11 +28,10 @@
       'common.gyp:*',
       'common_audio/common_audio.gyp:*',
       'common_video/common_video.gyp:*',
-      'libjingle/xmllite/xmllite.gyp:*',
-      'libjingle/xmpp/xmpp.gyp:*',
       'modules/modules.gyp:*',
       'p2p/p2p.gyp:*',
-      'system_wrappers/source/system_wrappers.gyp:*',
+      'system_wrappers/system_wrappers.gyp:*',
+      'tools/tools.gyp:*',
       'video_engine/video_engine.gyp:*',
       'voice_engine/voice_engine.gyp:*',
       '<(webrtc_vp8_dir)/vp8.gyp:*',
@@ -51,11 +50,10 @@
         ['include_tests==1', {
           'dependencies': [
             'common_video/common_video_unittests.gyp:*',
-            'system_wrappers/source/system_wrappers_tests.gyp:*',
+            'system_wrappers/system_wrappers_tests.gyp:*',
             'test/metrics.gyp:*',
             'test/test.gyp:*',
             'test/webrtc_test_common.gyp:webrtc_test_common_unittests',
-            'tools/tools.gyp:*',
             'webrtc_tests',
             'rtc_unittests',
           ],
@@ -85,13 +83,13 @@
       ],
       'conditions': [
         # TODO(andresp): Chromium libpeerconnection should link directly with
-	# this and no if conditions should be needed on webrtc build files.
+        # this and no if conditions should be needed on webrtc build files.
         ['build_with_chromium==1', {
-	  'dependencies': [
-	    '<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
-	    '<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
-	  ],
-	}],
+          'dependencies': [
+            '<(webrtc_root)/modules/modules.gyp:video_capture',
+            '<(webrtc_root)/modules/modules.gyp:video_render',
+          ],
+        }],
       ],
     },
   ],
diff --git a/media/webrtc/trunk/webrtc/webrtc_examples.gyp b/media/webrtc/trunk/webrtc/webrtc_examples.gyp
index 4d63110021..5d18aa64bf 100644
--- a/media/webrtc/trunk/webrtc/webrtc_examples.gyp
+++ b/media/webrtc/trunk/webrtc/webrtc_examples.gyp
@@ -16,20 +16,22 @@
           'type': 'loadable_module',
           'dependencies': [
             '<(DEPTH)/third_party/icu/icu.gyp:icuuc',
-            '<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
-	    '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
-            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
+            '<(webrtc_root)/common.gyp:webrtc_common',
+            '<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
+            '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
-            '<(webrtc_root)/video_engine/video_engine.gyp:video_engine_core',
             '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
           ],
           'sources': [
             'examples/android/media_demo/jni/jni_helpers.cc',
             'examples/android/media_demo/jni/on_load.cc',
-            'examples/android/media_demo/jni/video_engine_jni.cc',
             'examples/android/media_demo/jni/voice_engine_jni.cc',
-            'examples/android/media_demo/jni/media_codec_video_decoder.cc',
           ],
+          'variables': {
+            # This library uses native JNI exports; tell GYP so that the
+            # required symbols will be kept.
+            'use_native_jni_exports': 1,
+          },
           'link_settings': {
             'libraries': [
               '-llog',
@@ -57,8 +59,6 @@
               },
               'inputs' : [
                 '<(PRODUCT_DIR)/lib.java/audio_device_module_java.jar',
-                '<(PRODUCT_DIR)/lib.java/video_capture_module_java.jar',
-                '<(PRODUCT_DIR)/lib.java/video_render_module_java.jar',
                 '<(PRODUCT_DIR)/libwebrtcdemo-jni.so',
                 ' {
       sendAsyncMessage("BackgroundPageThumbs:didCapture", {
         id: capture.id,
diff --git a/toolkit/devtools/server/actors/device.js b/toolkit/devtools/server/actors/device.js
index 3e50af6017..f194466124 100644
--- a/toolkit/devtools/server/actors/device.js
+++ b/toolkit/devtools/server/actors/device.js
@@ -10,6 +10,7 @@ const {Promise: promise} = Cu.import("resource://gre/modules/Promise.jsm", {});
 const {LongStringActor} = require("devtools/server/actors/string");
 const {DebuggerServer} = require("devtools/server/main");
 
+Cu.importGlobalProperties(["FileReader"]);
 Cu.import("resource://gre/modules/PermissionsTable.jsm")
 
 const APP_MAP = {
@@ -145,7 +146,6 @@ let DeviceActor = exports.DeviceActor = protocol.ActorClass({
   getWallpaper: method(function() {
     let deferred = promise.defer();
     this._getSetting("wallpaper.image").then((blob) => {
-      let FileReader = CC("@mozilla.org/files/filereader;1");
       let reader = new FileReader();
       let conn = this.conn;
       reader.addEventListener("load", function() {
diff --git a/toolkit/devtools/server/actors/webapps.js b/toolkit/devtools/server/actors/webapps.js
index c00ecb2510..44ddf976d6 100644
--- a/toolkit/devtools/server/actors/webapps.js
+++ b/toolkit/devtools/server/actors/webapps.js
@@ -11,6 +11,8 @@ Cu.import("resource://gre/modules/XPCOMUtils.jsm");
 Cu.import("resource://gre/modules/osfile.jsm");
 Cu.import("resource://gre/modules/FileUtils.jsm");
 
+Cu.importGlobalProperties(["FileReader"]);
+
 let {Promise: promise} = Cu.import("resource://gre/modules/Promise.jsm", {});
 
 let DevToolsUtils = require("devtools/toolkit/DevToolsUtils");
@@ -822,8 +824,7 @@ WebappsActor.prototype = {
       }
 
       // Convert the blog to a base64 encoded data URI
-      let reader = Cc["@mozilla.org/files/filereader;1"]
-                     .createInstance(Ci.nsIDOMFileReader);
+      let reader = new FileReader();
       reader.onload = function () {
         deferred.resolve({
           url: reader.result
diff --git a/toolkit/devtools/webconsole/test/common.js b/toolkit/devtools/webconsole/test/common.js
index 666cedf1ce..ca349a6419 100644
--- a/toolkit/devtools/webconsole/test/common.js
+++ b/toolkit/devtools/webconsole/test/common.js
@@ -3,7 +3,7 @@
 
 "use strict";
 
-const {classes: Cc, interfaces: Ci, utils: Cu} = Components;
+var {classes: Cc, interfaces: Ci, utils: Cu} = Components;
 
 const XHTML_NS = "http://www.w3.org/1999/xhtml";
 
diff --git a/toolkit/modules/CharsetMenu.jsm b/toolkit/modules/CharsetMenu.jsm
index 0a409cafba..e95de0cb22 100644
--- a/toolkit/modules/CharsetMenu.jsm
+++ b/toolkit/modules/CharsetMenu.jsm
@@ -186,12 +186,12 @@ let CharsetMenu = {
   },
 
   getCharsetInfo: function(charsets, sort=true) {
-    let list = [{
+    let list = Array.from(charsets, charset => ({
       label: this._getCharsetLabel(charset),
       accesskey: this._getCharsetAccessKey(charset),
       name: "charset",
       value: charset
-    } for (charset of charsets)];
+    }));
 
     if (sort) {
       list.sort(CharsetComparator);
diff --git a/toolkit/modules/DeferredTask.jsm b/toolkit/modules/DeferredTask.jsm
index 81b796da9f..9ca6508284 100644
--- a/toolkit/modules/DeferredTask.jsm
+++ b/toolkit/modules/DeferredTask.jsm
@@ -277,7 +277,7 @@ this.DeferredTask.prototype = {
     this._armed = false;
     this._runningPromise = runningDeferred.promise;
 
-    runningDeferred.resolve(Task.spawn(function () {
+    runningDeferred.resolve(Task.spawn(function* () {
       // Execute the provided function asynchronously.
       yield Task.spawn(this._taskFn).then(null, Cu.reportError);
 
diff --git a/toolkit/modules/FileUtils.jsm b/toolkit/modules/FileUtils.jsm
index b4409cb99a..df17d60a2b 100644
--- a/toolkit/modules/FileUtils.jsm
+++ b/toolkit/modules/FileUtils.jsm
@@ -68,7 +68,10 @@ this.FileUtils = {
     if (shouldCreate) {
       try {
         dir.create(Ci.nsIFile.DIRECTORY_TYPE, this.PERMS_DIRECTORY);
-      } catch (ex if ex.result == Cr.NS_ERROR_FILE_ALREADY_EXISTS) {
+      } catch (ex) {
+        if (ex.result != Cr.NS_ERROR_FILE_ALREADY_EXISTS) {
+          throw ex;
+        }
         // Ignore the exception due to a directory that already exists.
       }
     }
diff --git a/toolkit/modules/Log.jsm b/toolkit/modules/Log.jsm
index 80f2dac099..41059836c7 100644
--- a/toolkit/modules/Log.jsm
+++ b/toolkit/modules/Log.jsm
@@ -867,7 +867,7 @@ FileAppender.prototype = {
   __proto__: Appender.prototype,
 
   _openFile: function () {
-    return Task.spawn(function _openFile() {
+    return Task.spawn(function* _openFile() {
       try {
         this._file = yield OS.File.open(this._path,
                                         {truncate: true});
diff --git a/toolkit/modules/Promise-backend.js b/toolkit/modules/Promise-backend.js
index 3d56dad2cd..41653a92af 100644
--- a/toolkit/modules/Promise-backend.js
+++ b/toolkit/modules/Promise-backend.js
@@ -236,8 +236,7 @@ var PendingErrors = {
   flush: function() {
     // Since we are going to modify the map while walking it,
     // let's copying the keys first.
-    let keys = [key for (key of this._map.keys())];
-    for (let key of keys) {
+    for (let key of Array.from(this._map.keys())) {
       this.report(key);
     }
   },
diff --git a/toolkit/modules/PromiseUtils.jsm b/toolkit/modules/PromiseUtils.jsm
index e0a09ec6f8..8ceebced32 100644
--- a/toolkit/modules/PromiseUtils.jsm
+++ b/toolkit/modules/PromiseUtils.jsm
@@ -50,4 +50,4 @@ function Deferred() {
     this.resolve = resolve;
     this.reject = reject;
   });
-}
\ No newline at end of file
+}
diff --git a/toolkit/modules/PropertyListUtils.jsm b/toolkit/modules/PropertyListUtils.jsm
index e0fade1694..4b29988137 100644
--- a/toolkit/modules/PropertyListUtils.jsm
+++ b/toolkit/modules/PropertyListUtils.jsm
@@ -61,7 +61,7 @@ const Cc = Components.classes;
 const Ci = Components.interfaces;
 const Cu = Components.utils;
 
-Cu.importGlobalProperties(['File']);
+Cu.importGlobalProperties(['File', 'FileReader']);
 Cu.import("resource://gre/modules/XPCOMUtils.jsm");
 
 XPCOMUtils.defineLazyModuleGetter(this, "Dict",
@@ -102,8 +102,7 @@ this.PropertyListUtils = Object.freeze({
           file = new File(file);
         }
 
-        let fileReader = Cc["@mozilla.org/files/filereader;1"].
-                         createInstance(Ci.nsIDOMFileReader);
+        let fileReader = new FileReader();
         let onLoadEnd = function() {
           let root = null;
           try {
@@ -694,7 +693,7 @@ XMLPropertyListReader.prototype = {
         // Strip spaces and new lines.
         let base64str = aDOMElt.textContent.replace(/\s*/g, "");
         let decoded = atob(base64str);
-        return new Uint8Array([decoded.charCodeAt(i) for (i in decoded)]);
+        return new Uint8Array(Array.from(decoded, c => c.charCodeAt(0)));
       case "dict":
         return this._wrapDictionary(aDOMElt);
       case "array":
diff --git a/toolkit/modules/RemotePageManager.jsm b/toolkit/modules/RemotePageManager.jsm
index 354350c23e..93a1ab78cf 100644
--- a/toolkit/modules/RemotePageManager.jsm
+++ b/toolkit/modules/RemotePageManager.jsm
@@ -461,7 +461,7 @@ var RemotePageManagerInternal = {
 
   // A listener is requesting the list of currently registered urls
   initListener: function({ target: messageManager }) {
-    messageManager.sendAsyncMessage("RemotePage:Register", { urls: [u for (u of this.pages.keys())] })
+    messageManager.sendAsyncMessage("RemotePage:Register", { urls: Array.from(this.pages.keys()) })
   },
 
   // A remote page has been created and a port is ready in the content side
diff --git a/toolkit/modules/Services.jsm b/toolkit/modules/Services.jsm
index 1788b15b6e..5ed07fb2f7 100644
--- a/toolkit/modules/Services.jsm
+++ b/toolkit/modules/Services.jsm
@@ -23,9 +23,11 @@ XPCOMUtils.defineLazyGetter(Services, "appinfo", function () {
                   .getService(Ci.nsIXULRuntime);
   try {
     appinfo.QueryInterface(Ci.nsIXULAppInfo);
-  } catch (ex if ex instanceof Components.Exception &&
-                 ex.result == Cr.NS_NOINTERFACE) {
+  } catch (ex) {
     // Not all applications implement nsIXULAppInfo (e.g. xpcshell doesn't).
+    if (!(ex instanceof Components.Exception) || ex.result != Cr.NS_NOINTERFACE) {
+      throw ex;
+    }
   }
   return appinfo;
 });
diff --git a/toolkit/modules/Sqlite.jsm b/toolkit/modules/Sqlite.jsm
index 3c02acb6b4..3840c32a05 100644
--- a/toolkit/modules/Sqlite.jsm
+++ b/toolkit/modules/Sqlite.jsm
@@ -769,13 +769,15 @@ ConnectionData.prototype = Object.freeze({
 
           try {
             onRow(row);
-          } catch (e if e instanceof StopIteration) {
-            userCancelled = true;
-            pending.cancel();
-            break;
-          } catch (ex) {
+          } catch (e) {
+            if (e instanceof StopIteration) {
+              userCancelled = true;
+              pending.cancel();
+              break;
+            }
+
             self._log.warn("Exception when calling onRow callback: " +
-                           CommonUtils.exceptionStr(ex));
+                           CommonUtils.exceptionStr(e));
           }
         }
       },
@@ -811,7 +813,7 @@ ConnectionData.prototype = Object.freeze({
             break;
 
           case Ci.mozIStorageStatementCallback.REASON_ERROR:
-            let error = new Error("Error(s) encountered during statement execution: " + [error.message for (error of errors)].join(", "));
+            let error = new Error("Error(s) encountered during statement execution: " + errors.map(e => e.message).join(", "));
             error.errors = errors;
             deferred.reject(error);
             break;
diff --git a/toolkit/modules/Task.jsm b/toolkit/modules/Task.jsm
index 1bd82a949c..98e0601b45 100644
--- a/toolkit/modules/Task.jsm
+++ b/toolkit/modules/Task.jsm
@@ -237,9 +237,10 @@ function createAsyncFunction(aTask) {
       try {
         // Let's call into the function ourselves.
         result = aTask.apply(this, arguments);
-      } catch (ex if ex instanceof Task.Result) {
-        return Promise.resolve(ex.value);
       } catch (ex) {
+        if (ex instanceof Task.Result) {
+          return Promise.resolve(ex.value);
+        }
         return Promise.reject(ex);
       }
     }
@@ -330,16 +331,18 @@ TaskImpl.prototype = {
           let yielded = aSendResolved ? this._iterator.send(aSendValue)
                                       : this._iterator.throw(aSendValue);
           this._handleResultValue(yielded);
-        } catch (ex if ex instanceof Task.Result) {
-          // The generator function threw the special exception that allows it to
-          // return a specific value on resolution.
-          this.deferred.resolve(ex.value);
-        } catch (ex if ex instanceof StopIteration) {
-          // The generator function terminated with no specific result.
-          this.deferred.resolve(undefined);
         } catch (ex) {
-          // The generator function failed with an uncaught exception.
-          this._handleException(ex);
+          if (ex instanceof Task.Result) {
+            // The generator function threw the special exception that allows it to
+            // return a specific value on resolution.
+            this.deferred.resolve(ex.value);
+          } else if (ex instanceof StopIteration) {
+            // The generator function terminated with no specific result.
+            this.deferred.resolve(undefined);
+          } else {
+            // The generator function failed with an uncaught exception.
+            this._handleException(ex);
+          }
         }
       }
     } finally {
diff --git a/toolkit/modules/ZipUtils.jsm b/toolkit/modules/ZipUtils.jsm
index 4f735f89dd..decb3bc7bd 100644
--- a/toolkit/modules/ZipUtils.jsm
+++ b/toolkit/modules/ZipUtils.jsm
@@ -79,11 +79,11 @@ function saveStreamAsync(aPath, aStream, aFile) {
         input.asyncWait(readData, 0, 0, Services.tm.currentThread);
       }, readFailed);
     }
-    catch (e if e.result == Cr.NS_BASE_STREAM_CLOSED) {
-      deferred.resolve(aFile.close());
-    }
     catch (e) {
-      readFailed(e);
+      if (e.result == Cr.NS_BASE_STREAM_CLOSED)
+        deferred.resolve(aFile.close());
+      else
+        readFailed(e);
     }
   }
 
@@ -115,7 +115,7 @@ this.ZipUtils = {
       return Promise.reject(e);
     }
 
-    return Task.spawn(function() {
+    return Task.spawn(function* () {
       // Get all of the entries in the zip and sort them so we create directories
       // before files
       let entries = zipReader.findEntries(null);
diff --git a/toolkit/modules/addons/MatchPattern.jsm b/toolkit/modules/addons/MatchPattern.jsm
index 91d4bf6b30..dda7fb52df 100644
--- a/toolkit/modules/addons/MatchPattern.jsm
+++ b/toolkit/modules/addons/MatchPattern.jsm
@@ -99,7 +99,7 @@ function MatchPattern(pat)
   } else if (pat instanceof String || typeof(pat) == "string") {
     this.matchers = [new SingleMatchPattern(pat)];
   } else {
-    this.matchers = [for (p of pat) new SingleMatchPattern(p)];
+    this.matchers = pat.map(p => new SingleMatchPattern(p));
   }
 }
 
diff --git a/toolkit/modules/addons/WebRequestContent.js b/toolkit/modules/addons/WebRequestContent.js
index 2cd019a3bd..75221b2ede 100644
--- a/toolkit/modules/addons/WebRequestContent.js
+++ b/toolkit/modules/addons/WebRequestContent.js
@@ -140,7 +140,11 @@ let ContentPolicy = {
       try {
         // If e10s is disabled, this throws NS_NOINTERFACE for closed tabs.
         mm = ir.getInterface(Ci.nsIContentFrameMessageManager);
-      } catch (e if e.result == Cr.NS_NOINTERFACE) {}
+      } catch (e) {
+        if (e.result != Cr.NS_NOINTERFACE) {
+          throw e;
+        }
+      }
     }
 
     let data = {ids,
diff --git a/toolkit/modules/tests/xpcshell/test_DeferredTask.js b/toolkit/modules/tests/xpcshell/test_DeferredTask.js
index 606d9cbe19..41dbcefb0c 100644
--- a/toolkit/modules/tests/xpcshell/test_DeferredTask.js
+++ b/toolkit/modules/tests/xpcshell/test_DeferredTask.js
@@ -156,7 +156,7 @@ add_test(function test_arm_async()
   let finishedExecutionAgain = false;
 
   // Create a task that will run later.
-  let deferredTask = new DeferredTask(function () {
+  let deferredTask = new DeferredTask(function* () {
     yield promiseTimeout(4*T);
     if (!finishedExecution) {
       finishedExecution = true;
@@ -247,7 +247,7 @@ add_test(function test_disarm_async()
 {
   let finishedExecution = false;
 
-  let deferredTask = new DeferredTask(function () {
+  let deferredTask = new DeferredTask(function* () {
     deferredTask.arm();
     yield promiseTimeout(2*T);
     finishedExecution = true;
@@ -277,7 +277,7 @@ add_test(function test_disarm_immediate_async()
 {
   let executed = false;
 
-  let deferredTask = new DeferredTask(function () {
+  let deferredTask = new DeferredTask(function* () {
     do_check_false(executed);
     executed = true;
     yield promiseTimeout(2*T);
@@ -353,7 +353,7 @@ add_test(function test_finalize_executes_entirely()
   let executedAgain = false;
   let timePassed = false;
 
-  let deferredTask = new DeferredTask(function () {
+  let deferredTask = new DeferredTask(function* () {
     // The first time, we arm the timer again and set up the finalization.
     if (!executed) {
       deferredTask.arm();
diff --git a/toolkit/modules/tests/xpcshell/test_Log.js b/toolkit/modules/tests/xpcshell/test_Log.js
index 0fcfe40520..a0fb975997 100644
--- a/toolkit/modules/tests/xpcshell/test_Log.js
+++ b/toolkit/modules/tests/xpcshell/test_Log.js
@@ -233,7 +233,7 @@ function fileContents(path) {
   });
 }
 
-add_task(function test_FileAppender() {
+add_task(function* test_FileAppender() {
   // This directory does not exist yet
   let dir = OS.Path.join(do_get_profile().path, "test_Log");
   do_check_false(yield OS.File.exists(dir));
@@ -287,7 +287,7 @@ add_task(function test_FileAppender() {
               "test.FileAppender\tINFO\t5\n");
 });
 
-add_task(function test_BoundedFileAppender() {
+add_task(function* test_BoundedFileAppender() {
   let dir = OS.Path.join(do_get_profile().path, "test_Log");
 
   if (!(yield OS.File.exists(dir))) {
@@ -338,7 +338,7 @@ add_task(function test_BoundedFileAppender() {
 /*
  * Test parameter formatting.
  */
-add_task(function log_message_with_params() {
+add_task(function* log_message_with_params() {
   let formatter = new Log.BasicFormatter();
 
   function formatMessage(text, params) {
@@ -487,7 +487,7 @@ add_task(function log_message_with_params() {
  * with the object argument as parameters. This makes the log useful when the
  * caller does "catch(err) {logger.error(err)}"
  */
-add_task(function test_log_err_only() {
+add_task(function* test_log_err_only() {
   let log = Log.repository.getLogger("error.only");
   let testFormatter = { format: msg => msg };
   let appender = new MockAppender(testFormatter);
@@ -513,7 +513,7 @@ add_task(function test_log_err_only() {
 /*
  * Test logStructured() messages through basic formatter.
  */
-add_task(function test_structured_basic() {
+add_task(function* test_structured_basic() {
   let log = Log.repository.getLogger("test.logger");
   let appender = new MockAppender(new Log.BasicFormatter());
 
@@ -538,7 +538,7 @@ add_task(function test_structured_basic() {
 /*
  * Test that all the basic logger methods pass the message and params through to the appender.
  */
-add_task(function log_message_with_params() {
+add_task(function* log_message_with_params() {
   let log = Log.repository.getLogger("error.logger");
   let testFormatter = { format: msg => msg };
   let appender = new MockAppender(testFormatter);
@@ -562,7 +562,7 @@ add_task(function log_message_with_params() {
 /*
  * Check that we format JS Errors reasonably.
  */
-add_task(function format_errors() {
+add_task(function* format_errors() {
   let pFormat = new Log.ParameterFormatter();
 
   // Test that subclasses of Error are recognized as errors.
diff --git a/toolkit/modules/tests/xpcshell/test_NewTabUtils.js b/toolkit/modules/tests/xpcshell/test_NewTabUtils.js
index e9bed568ac..f6a383a415 100644
--- a/toolkit/modules/tests/xpcshell/test_NewTabUtils.js
+++ b/toolkit/modules/tests/xpcshell/test_NewTabUtils.js
@@ -11,7 +11,7 @@ function run_test() {
   run_next_test();
 }
 
-add_task(function isTopSiteGivenProvider() {
+add_task(function* isTopSiteGivenProvider() {
   let expectedLinks = makeLinks(0, 10, 2);
 
   // The lowest 2 frecencies have the same base domain.
@@ -50,7 +50,7 @@ add_task(function isTopSiteGivenProvider() {
   NewTabUtils.links.removeProvider(provider);
 });
 
-add_task(function multipleProviders() {
+add_task(function* multipleProviders() {
   // Make each provider generate NewTabUtils.links.maxNumLinks links to check
   // that no more than maxNumLinks are actually returned in the merged list.
   let evenLinks = makeLinks(0, 2 * NewTabUtils.links.maxNumLinks, 2);
@@ -76,7 +76,7 @@ add_task(function multipleProviders() {
   NewTabUtils.links.removeProvider(oddProvider);
 });
 
-add_task(function changeLinks() {
+add_task(function* changeLinks() {
   let expectedLinks = makeLinks(0, 20, 2);
   let provider = new TestProvider(done => done(expectedLinks));
 
@@ -131,7 +131,7 @@ add_task(function changeLinks() {
   NewTabUtils.links.removeProvider(provider);
 });
 
-add_task(function oneProviderAlreadyCached() {
+add_task(function* oneProviderAlreadyCached() {
   let links1 = makeLinks(0, 10, 1);
   let provider1 = new TestProvider(done => done(links1));
 
@@ -153,7 +153,7 @@ add_task(function oneProviderAlreadyCached() {
   NewTabUtils.links.removeProvider(provider2);
 });
 
-add_task(function newLowRankedLink() {
+add_task(function* newLowRankedLink() {
   // Init a provider with 10 links and make its maximum number also 10.
   let links = makeLinks(0, 10, 1);
   let provider = new TestProvider(done => done(links));
@@ -181,7 +181,7 @@ add_task(function newLowRankedLink() {
   NewTabUtils.links.removeProvider(provider);
 });
 
-add_task(function extractSite() {
+add_task(function* extractSite() {
   // All these should extract to the same site
   [ "mozilla.org",
     "m.mozilla.org",
diff --git a/toolkit/modules/tests/xpcshell/test_PromiseUtils.js b/toolkit/modules/tests/xpcshell/test_PromiseUtils.js
index 8953d2d501..ff9ba3688b 100644
--- a/toolkit/modules/tests/xpcshell/test_PromiseUtils.js
+++ b/toolkit/modules/tests/xpcshell/test_PromiseUtils.js
@@ -102,4 +102,4 @@ add_task(function* test_reject_resolved_promise() {
   let p = new Promise((resolve, reject) => reject(new Error("This on rejects")));
   def.reject(p);
   yield Assert.rejects(def.promise, Promise, "Rejection with a rejected promise uses the passed promise itself as the reason of rejection");
-});
\ No newline at end of file
+});
diff --git a/toolkit/modules/tests/xpcshell/test_ZipUtils.js b/toolkit/modules/tests/xpcshell/test_ZipUtils.js
index 35f882a643..dbd938b1dc 100644
--- a/toolkit/modules/tests/xpcshell/test_ZipUtils.js
+++ b/toolkit/modules/tests/xpcshell/test_ZipUtils.js
@@ -45,7 +45,7 @@ add_task(function test_extractFiles() {
   ensureExtracted(target);
 });
 
-add_task(function test_extractFilesAsync() {
+add_task(function* test_extractFilesAsync() {
   let target = dir.clone();
   target.append("test_extractFilesAsync");
   target.create(Components.interfaces.nsIFile.DIRECTORY_TYPE,
diff --git a/toolkit/modules/tests/xpcshell/test_sqlite.js b/toolkit/modules/tests/xpcshell/test_sqlite.js
index dff42f5cf1..9c5f3c15ca 100644
--- a/toolkit/modules/tests/xpcshell/test_sqlite.js
+++ b/toolkit/modules/tests/xpcshell/test_sqlite.js
@@ -126,7 +126,9 @@ add_task(function* test_schema_version() {
       yield db.setSchemaVersion(v);
       do_print("Schema version " + v + " should have been rejected");
       success = false;
-    } catch (ex if ex.message.startsWith("Schema version must be an integer.")) {
+    } catch (ex) {
+      if (!ex.message.startsWith("Schema version must be an integer."))
+        throw ex;
       success = true;
     }
     do_check_true(success);
@@ -918,7 +920,11 @@ add_task(function* test_cloneStorageConnection() {
   try {
     let clone = yield Sqlite.cloneStorageConnection({ connection: null });
     do_throw(new Error("Should throw on invalid connection"));
-  } catch (ex if ex.name == "TypeError") {}
+  } catch (ex) {
+    if (ex.name != "TypeError") {
+      throw ex;
+    }
+  }
 });
 
 // Test clone() method.
diff --git a/toolkit/modules/tests/xpcshell/test_sqlite_shutdown.js b/toolkit/modules/tests/xpcshell/test_sqlite_shutdown.js
index 3c6634eb2d..2c3a0cf5bb 100644
--- a/toolkit/modules/tests/xpcshell/test_sqlite_shutdown.js
+++ b/toolkit/modules/tests/xpcshell/test_sqlite_shutdown.js
@@ -25,7 +25,7 @@ function getConnection(dbName, extraOptions={}) {
   return Sqlite.openConnection(options);
 }
 
-function getDummyDatabase(name, extraOptions={}) {
+function* getDummyDatabase(name, extraOptions={}) {
   const TABLES = {
     dirs: "id INTEGER PRIMARY KEY AUTOINCREMENT, path TEXT",
     files: "id INTEGER PRIMARY KEY AUTOINCREMENT, dir_id INTEGER, path TEXT",
@@ -39,7 +39,7 @@ function getDummyDatabase(name, extraOptions={}) {
     c._initialStatementCount++;
   }
 
-  throw new Task.Result(c);
+  return c;
 }
 
 function sleep(ms) {
diff --git a/toolkit/mozapps/extensions/content/selectAddons.js b/toolkit/mozapps/extensions/content/selectAddons.js
index 33ab8a84e0..82edbc419b 100644
--- a/toolkit/mozapps/extensions/content/selectAddons.js
+++ b/toolkit/mozapps/extensions/content/selectAddons.js
@@ -123,7 +123,7 @@ var gChecking = {
     if (this._completeCount < this._addonCount)
       return;
 
-    var addons = [gAddons[id] for (id in gAddons)];
+    let addons = Object.values(gAddons);
 
     addons.sort(function sortAddons(a, b) {
       let orderA = orderForScope(a.addon.scope);
diff --git a/toolkit/mozapps/extensions/content/update.js b/toolkit/mozapps/extensions/content/update.js
index 4e44b4f5ed..72530cfdf2 100644
--- a/toolkit/mozapps/extensions/content/update.js
+++ b/toolkit/mozapps/extensions/content/update.js
@@ -131,7 +131,8 @@ var gUpdateWizard = {
     if (gMismatchPage.waiting) {
       logger.info("Dialog closed in mismatch page");
       if (gUpdateWizard.addonInstalls.size > 0) {
-        gInstallingPage.startInstalls([i for ([, i] of gUpdateWizard.addonInstalls)]);
+        gInstallingPage.startInstalls(
+          Array.from(gUpdateWizard.addonInstalls.values()));
       }
       return true;
     }
@@ -195,8 +196,10 @@ var gVersionInfoPage = {
         logger.debug("repopulateCache completed after dialog closed");
       }
     }
-    // Fetch the add-ons that are still affected by this update.
-    let idlist = [id for (id of gUpdateWizard.affectedAddonIDs)];
+    // Fetch the add-ons that are still affected by this update,
+    // excluding the hotfix add-on.
+    let idlist = Array.from(gUpdateWizard.affectedAddonIDs).filter(
+      a => a.id != AddonManager.hotfixID);
     if (idlist.length < 1) {
       gVersionInfoPage.onAllUpdatesFinished();
       return;
@@ -206,7 +209,7 @@ var gVersionInfoPage = {
     let fetchedAddons = yield new Promise((resolve, reject) =>
       AddonManager.getAddonsByIDs(idlist, resolve));
     // We shouldn't get nulls here, but let's be paranoid...
-    gUpdateWizard.addons = [a for (a of fetchedAddons) if (a)];
+    gUpdateWizard.addons = fetchedAddons.filter(a => a);
     if (gUpdateWizard.addons.length < 1) {
       gVersionInfoPage.onAllUpdatesFinished();
       return;
@@ -234,8 +237,8 @@ var gVersionInfoPage = {
     AddonManagerPrivate.recordSimpleMeasure("appUpdate_upgradeFailed", 0);
     AddonManagerPrivate.recordSimpleMeasure("appUpdate_upgradeDeclined", 0);
     // Filter out any add-ons that are now enabled.
-    logger.debug("VersionInfo updates finished: found " +
-         [addon.id + ":" + addon.appDisabled for (addon of gUpdateWizard.addons)].toSource());
+    let addonList = gUpdateWizard.addons.map(a => a.id + ":" + a.appDisabled);
+    logger.debug("VersionInfo updates finished: found " + addonList.toSource());
     let filteredAddons = [];
     for (let a of gUpdateWizard.addons) {
       if (a.appDisabled) {
@@ -252,7 +255,8 @@ var gVersionInfoPage = {
     if (gUpdateWizard.shuttingDown) {
       // jump directly to updating auto-update add-ons in the background
       if (gUpdateWizard.addonInstalls.size > 0) {
-        gInstallingPage.startInstalls([i for ([, i] of gUpdateWizard.addonInstalls)]);
+        let installs = Array.from(gUpdateWizard.addonInstalls.values());
+        gInstallingPage.startInstalls(installs);
       }
       return;
     }
@@ -478,8 +482,8 @@ var gInstallingPage = {
       return;
     }
 
-    logger.debug("Start installs for "
-                 + [i.existingAddon.id for (i of aInstallList)].toSource());
+    let installs = Array.from(aInstallList).map(a => a.existingAddon.id);
+    logger.debug("Start installs for " + installs.toSource());
     this._errors = [];
     this._installs = aInstallList;
     this._installing = true;
diff --git a/toolkit/mozapps/extensions/internal/AddonRepository.jsm b/toolkit/mozapps/extensions/internal/AddonRepository.jsm
index 14e8d04cc1..1d12b4b80d 100644
--- a/toolkit/mozapps/extensions/internal/AddonRepository.jsm
+++ b/toolkit/mozapps/extensions/internal/AddonRepository.jsm
@@ -631,7 +631,7 @@ this.AddonRepository = {
       return this._clearCache();
     }
 
-    let ids = [a.id for (a of allAddons)];
+    let ids = allAddons.map(a => a.id);
     logger.debug("Repopulate add-on cache with " + ids.toSource());
 
     let self = this;
@@ -1618,34 +1618,34 @@ var AddonDatabase = {
              schema < DB_MIN_JSON_SCHEMA) {
            throw new Error("Invalid schema value.");
          }
-       } catch (e if e instanceof OS.File.Error && e.becauseNoSuchFile) {
-         logger.debug("No " + FILE_DATABASE + " found.");
+       } catch (e) {
+         if (e instanceof OS.File.Error && e.becauseNoSuchFile) {
+           logger.debug("No " + FILE_DATABASE + " found.");
 
-         // Create a blank addons.json file
-         this._saveDBToDisk();
+           // Create a blank addons.json file
+           this._saveDBToDisk();
 
-         let dbSchema = 0;
-         try {
-           dbSchema = Services.prefs.getIntPref(PREF_GETADDONS_DB_SCHEMA);
-         } catch (e) {}
+           let dbSchema = 0;
+           try {
+             dbSchema = Services.prefs.getIntPref(PREF_GETADDONS_DB_SCHEMA);
+           } catch (e) {}
 
-         if (dbSchema < DB_MIN_JSON_SCHEMA) {
-           let results = yield new Promise((resolve, reject) => {
-             AddonRepository_SQLiteMigrator.migrate(resolve);
-           });
+           if (dbSchema < DB_MIN_JSON_SCHEMA) {
+             let results = yield new Promise((resolve, reject) => {
+               AddonRepository_SQLiteMigrator.migrate(resolve);
+             });
+
+             if (results.length) {
+               yield this._insertAddons(results);
+             }
 
-           if (results.length) {
-             yield this._insertAddons(results);
            }
 
            Services.prefs.setIntPref(PREF_GETADDONS_DB_SCHEMA, DB_SCHEMA);
+         } else {
+           logger.error("Malformed " + FILE_DATABASE + ": " + e);
+           this.databaseOk = false;
          }
-
-         return this.DB;
-       } catch (e) {
-         logger.error("Malformed " + FILE_DATABASE + ": " + e);
-         this.databaseOk = false;
-
          return this.DB;
        }
 
diff --git a/toolkit/mozapps/extensions/internal/AddonRepository_SQLiteMigrator.jsm b/toolkit/mozapps/extensions/internal/AddonRepository_SQLiteMigrator.jsm
index b98963033f..59035223a0 100644
--- a/toolkit/mozapps/extensions/internal/AddonRepository_SQLiteMigrator.jsm
+++ b/toolkit/mozapps/extensions/internal/AddonRepository_SQLiteMigrator.jsm
@@ -60,7 +60,7 @@ this.AddonRepository_SQLiteMigrator = {
 
     this._retrieveStoredData((results) => {
       this._closeConnection();
-      let resultArray = [addon for ([,addon] of Iterator(results))];
+      let resultArray = Object.values(results);
       logger.debug(resultArray.length + " addons imported.")
       aCallback(resultArray);
     });
diff --git a/toolkit/mozapps/extensions/internal/AddonUpdateChecker.jsm b/toolkit/mozapps/extensions/internal/AddonUpdateChecker.jsm
index 6bae768eec..74d9231925 100644
--- a/toolkit/mozapps/extensions/internal/AddonUpdateChecker.jsm
+++ b/toolkit/mozapps/extensions/internal/AddonUpdateChecker.jsm
@@ -651,7 +651,9 @@ UpdateParser.prototype = {
         let json = JSON.parse(request.responseText);
 
         parser = () => parseJSONManifest(this.id, this.updateKey, request, json);
-      } catch (e if e instanceof SyntaxError) {
+      } catch (e) {
+        if (!(e instanceof SyntaxError))
+          throw e;
         let domParser = Cc["@mozilla.org/xmlextras/domparser;1"].createInstance(Ci.nsIDOMParser);
         let xml = domParser.parseFromString(request.responseText, "text/xml");
 
diff --git a/toolkit/mozapps/extensions/internal/GMPProvider.jsm b/toolkit/mozapps/extensions/internal/GMPProvider.jsm
index d20ca7977f..bc17d514c9 100644
--- a/toolkit/mozapps/extensions/internal/GMPProvider.jsm
+++ b/toolkit/mozapps/extensions/internal/GMPProvider.jsm
@@ -555,7 +555,9 @@ var GMPProvider = {
         this._log.info("startup - adding gmp directory " + gmpPath);
         try {
           gmpService.addPluginDirectory(gmpPath);
-        } catch (e if e.name == 'NS_ERROR_NOT_AVAILABLE') {
+        } catch (e) {
+          if (e.name != 'NS_ERROR_NOT_AVAILABLE')
+            throw e;
           this._log.warn("startup - adding gmp directory failed with " +
                          e.name + " - sandboxing not available?", e);
         }
@@ -636,8 +638,10 @@ var GMPProvider = {
       return;
     }
 
-    let results = [p.wrapper for ([id, p] of this._plugins)
-                    if (!GMPUtils.isPluginHidden(p))];
+    let results = Array.from(this._plugins.values())
+      .filter(p => !GMPUtils.isPluginHidden(p))
+      .map(p => p.wrapper);
+
     aCallback(results);
   },
 
diff --git a/toolkit/mozapps/extensions/internal/PluginProvider.jsm b/toolkit/mozapps/extensions/internal/PluginProvider.jsm
index 7f58ceb136..86d74fda2f 100644
--- a/toolkit/mozapps/extensions/internal/PluginProvider.jsm
+++ b/toolkit/mozapps/extensions/internal/PluginProvider.jsm
@@ -39,7 +39,8 @@ function getIDHashForString(aStr) {
 
   // convert the binary hash data to a hex string.
   let binary = hasher.finish(false);
-  let hash = [toHexString(binary.charCodeAt(i)) for (i in binary)].join("").toLowerCase();
+  let hash = Array.from(binary, c => toHexString(c.charCodeAt(0)));
+  hash = hash.join("").toLowerCase();
   return "{" + hash.substr(0, 8) + "-" +
                hash.substr(8, 4) + "-" +
                hash.substr(12, 4) + "-" +
@@ -464,7 +465,9 @@ function PluginWrapper(aId, aName, aDescription, aTags) {
       dir = Services.dirsvc.get("Home", Ci.nsIFile);
       if (path.startsWith(dir.path))
         return AddonManager.SCOPE_USER;
-    } catch (e if (e.result && e.result == Components.results.NS_ERROR_FAILURE)) {
+    } catch (e) {
+      if (!e.result || e.result != Components.results.NS_ERROR_FAILURE)
+        throw e;
       // Do nothing: missing "Home".
     }
 
diff --git a/toolkit/mozapps/extensions/internal/XPIProvider.jsm b/toolkit/mozapps/extensions/internal/XPIProvider.jsm
index 90ef847c86..3d5541c083 100644
--- a/toolkit/mozapps/extensions/internal/XPIProvider.jsm
+++ b/toolkit/mozapps/extensions/internal/XPIProvider.jsm
@@ -1323,7 +1323,7 @@ function escapeAddonURI(aAddon, aUri, aUpdateType, aAppVersion)
 }
 
 function removeAsync(aFile) {
-  return Task.spawn(function () {
+  return Task.spawn(function*() {
     let info = null;
     try {
       info = yield OS.File.stat(aFile.path);
@@ -1332,7 +1332,9 @@ function removeAsync(aFile) {
       else
         yield OS.File.remove(aFile.path);
     }
-    catch (e if e instanceof OS.File.Error && e.becauseNoSuchFile) {
+    catch (e) {
+      if (!(e instanceof OS.File.Error) || ! e.becauseNoSuchFile)
+        throw e;
       // The file has already gone away
       return;
     }
@@ -2559,7 +2561,9 @@ this.XPIProvider = {
         try {
           isDir = stageDirEntry.isDirectory();
         }
-        catch (e if e.result == Cr.NS_ERROR_FILE_TARGET_DOES_NOT_EXIST) {
+        catch (e) {
+          if (e.result != Cr.NS_ERROR_FILE_TARGET_DOES_NOT_EXIST)
+            throw e;
           // If the file has already gone away then don't worry about it, this
           // can happen on OSX where the resource fork is automatically moved
           // with the data fork for the file. See bug 733436.
@@ -3608,8 +3612,8 @@ this.XPIProvider = {
 
     // XXX This will go away when we fold bootstrappedAddons into XPIStates.
     if (updateReasons.length == 0) {
-      let bootstrapDescriptors = new Set([for (b of Object.keys(this.bootstrappedAddons))
-                                          this.bootstrappedAddons[b].descriptor]);
+      let bootstrapDescriptors = new Set(Object.keys(this.bootstrappedAddons)
+        .map(b => this.bootstrappedAddons[b].descriptor));
 
       for (let location of XPIStates.db.values()) {
         for (let state of location.values()) {
@@ -3619,7 +3623,7 @@ this.XPIProvider = {
 
       if (bootstrapDescriptors.size > 0) {
         logger.warn("Bootstrap state is invalid (missing add-ons: "
-            + [for (b of bootstrapDescriptors) b] + ")");
+            + Array.from(bootstrapDescriptors).join(", ") + ")");
         updateReasons.push("missingBootstrapAddon");
       }
     }
@@ -4818,7 +4822,8 @@ function getHashStringForCrypto(aCrypto) {
 
   // convert the binary hash data to a hex string.
   let binary = aCrypto.finish(false);
-  return [toHexString(binary.charCodeAt(i)) for (i in binary)].join("").toLowerCase()
+  let hash = Array.from(binary, c => toHexString(c.charCodeAt(0)))
+  return hash.join("").toLowerCase();
 }
 
 /**
@@ -5779,7 +5784,7 @@ AddonInstall.prototype = {
     let stagingDir = this.installLocation.getStagingDir();
     let stagedAddon = stagingDir.clone();
 
-    Task.spawn((function() {
+    Task.spawn((function*() {
       let installedUnpacked = 0;
       yield this.installLocation.requestStagingDir();
 
diff --git a/toolkit/mozapps/extensions/internal/XPIProviderUtils.js b/toolkit/mozapps/extensions/internal/XPIProviderUtils.js
index d4798b726c..2e0f9275bc 100644
--- a/toolkit/mozapps/extensions/internal/XPIProviderUtils.js
+++ b/toolkit/mozapps/extensions/internal/XPIProviderUtils.js
@@ -200,7 +200,7 @@ function asyncMap(aObjects, aMethod, aCallback) {
  * @param  aStatement
  *         The statement to execute
  */
-function resultRows(aStatement) {
+function* resultRows(aStatement) {
   try {
     while (stepStatement(aStatement))
       yield aStatement.row;
@@ -390,7 +390,7 @@ function _findAddon(addonDB, aFilter) {
  * Internal interface to get a filtered list of addons from a loaded addonDB
  */
 function _filterDB(addonDB, aFilter) {
-  return [for (addon of addonDB.values()) if (aFilter(addon)) addon];
+  return Array.from(addonDB.values()).filter(aFilter);
 }
 
 this.XPIDatabase = {
@@ -918,7 +918,7 @@ this.XPIDatabase = {
 
       let reqCount = 0;
       let props = [];
-      for (let row in resultRows(stmt)) {
+      for (let row of resultRows(stmt)) {
         if (REQUIRED.indexOf(row.name) != -1) {
           reqCount++;
           props.push(row.name);
@@ -938,7 +938,7 @@ this.XPIDatabase = {
       stmt.finalize();
 
       stmt = aConnection.createStatement("SELECT " + props.join(",") + " FROM addon");
-      for (let row in resultRows(stmt)) {
+      for (let row of resultRows(stmt)) {
         if (!(row.location in migrateData))
           migrateData[row.location] = {};
         let addonData = {
@@ -965,7 +965,7 @@ this.XPIDatabase = {
         for (let id in migrateData[location]) {
           taStmt.params.internal_id = migrateData[location][id].internal_id;
           delete migrateData[location][id].internal_id;
-          for (let row in resultRows(taStmt)) {
+          for (let row of resultRows(taStmt)) {
             migrateData[location][id].targetApplications.push({
               id: row.id,
               minVersion: row.minVersion,
diff --git a/toolkit/mozapps/extensions/nsBlocklistService.js b/toolkit/mozapps/extensions/nsBlocklistService.js
index b05e1d59d8..7863bbdcbd 100644
--- a/toolkit/mozapps/extensions/nsBlocklistService.js
+++ b/toolkit/mozapps/extensions/nsBlocklistService.js
@@ -95,9 +95,11 @@ XPCOMUtils.defineLazyGetter(this, "gApp", function bls_gApp() {
                   .getService(Ci.nsIXULRuntime);
   try {
     appinfo.QueryInterface(Ci.nsIXULAppInfo);
-  } catch (ex if ex instanceof Components.Exception &&
-                 ex.result == Cr.NS_NOINTERFACE) {
+  } catch (ex) {
     // Not all applications implement nsIXULAppInfo (e.g. xpcshell doesn't).
+    if (!(ex instanceof Components.Exception) ||
+        ex.result != Cr.NS_NOINTERFACE)
+      throw ex;
   }
   return appinfo;
 });
diff --git a/toolkit/mozapps/extensions/test/browser/browser_bug580298.js b/toolkit/mozapps/extensions/test/browser/browser_bug580298.js
index 77b386c28a..2242460d86 100644
--- a/toolkit/mozapps/extensions/test/browser/browser_bug580298.js
+++ b/toolkit/mozapps/extensions/test/browser/browser_bug580298.js
@@ -9,7 +9,7 @@ var gManagerWindow;
 var gCategoryUtilities;
 var gProvider;
 
-add_task(function test() {
+add_task(function* test() {
   gProvider = new MockProvider();
 
   gProvider.createAddons([{
diff --git a/toolkit/mozapps/extensions/test/browser/browser_bug596336.js b/toolkit/mozapps/extensions/test/browser/browser_bug596336.js
index 00b7e6619b..6c9e5bb81b 100644
--- a/toolkit/mozapps/extensions/test/browser/browser_bug596336.js
+++ b/toolkit/mozapps/extensions/test/browser/browser_bug596336.js
@@ -41,7 +41,7 @@ function install_addon(aXpi) {
   });
 }
 
-let check_addon = Task.async(function*(aAddon, aVersion) {
+var check_addon = Task.async(function*(aAddon, aVersion) {
   is(get_list_item_count(), 1, "Should be one item in the list");
   is(aAddon.version, aVersion, "Add-on should have the right version");
 
@@ -51,98 +51,82 @@ let check_addon = Task.async(function*(aAddon, aVersion) {
   // Force XBL to apply
   item.clientTop;
 
-  is(get_node(item, "version").value, version, "Version should be correct");
+  let { version } = yield get_tooltip_info(item);
+  is(version, aVersion, "Version should be correct");
 
   if (aAddon.userDisabled)
     is_element_visible(get_class_node(item, "disabled-postfix"), "Disabled postfix should be hidden");
   else
     is_element_hidden(get_class_node(item, "disabled-postfix"), "Disabled postfix should be hidden");
-}
+});
 
 // Install version 1 then upgrade to version 2 with the manager open
-add_test(function() {
-  install_addon("browser_bug596336_1", function() {
-    AddonManager.getAddonByID("addon1@tests.mozilla.org", function(aAddon) {
-      check_addon(aAddon, "1.0");
-      ok(!aAddon.userDisabled, "Add-on should not be disabled");
+add_task(function*() {
+  yield install_addon("browser_bug596336_1");
+  let [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
+  yield check_addon(aAddon, "1.0");
+  ok(!aAddon.userDisabled, "Add-on should not be disabled");
 
-      install_addon("browser_bug596336_2", function() {
-        AddonManager.getAddonByID("addon1@tests.mozilla.org", function(aAddon) {
-          check_addon(aAddon, "2.0");
-          ok(!aAddon.userDisabled, "Add-on should not be disabled");
+  yield install_addon("browser_bug596336_2");
+  [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
+  yield check_addon(aAddon, "2.0");
+  ok(!aAddon.userDisabled, "Add-on should not be disabled");
 
-          aAddon.uninstall();
+  aAddon.uninstall();
 
-          is(get_list_item_count(), 0, "Should be no items in the list");
-
-          run_next_test();
-        });
-      });
-    });
-  });
+  is(get_list_item_count(), 0, "Should be no items in the list");
 });
 
 // Install version 1 mark it as disabled then upgrade to version 2 with the
 // manager open
-add_test(function() {
-  install_addon("browser_bug596336_1", function() {
-    AddonManager.getAddonByID("addon1@tests.mozilla.org", function(aAddon) {
-      aAddon.userDisabled = true;
-      check_addon(aAddon, "1.0");
-      ok(aAddon.userDisabled, "Add-on should be disabled");
+add_task(function*() {
+  yield install_addon("browser_bug596336_1");
+  let [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
+  aAddon.userDisabled = true;
+  yield check_addon(aAddon, "1.0");
+  ok(aAddon.userDisabled, "Add-on should be disabled");
 
-      install_addon("browser_bug596336_2", function() {
-        AddonManager.getAddonByID("addon1@tests.mozilla.org", function(aAddon) {
-          check_addon(aAddon, "2.0");
-          ok(aAddon.userDisabled, "Add-on should be disabled");
+  yield install_addon("browser_bug596336_2");
+  [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
+  yield check_addon(aAddon, "2.0");
+  ok(aAddon.userDisabled, "Add-on should be disabled");
 
-          aAddon.uninstall();
+  aAddon.uninstall();
 
-          is(get_list_item_count(), 0, "Should be no items in the list");
-
-          run_next_test();
-        });
-      });
-    });
-  });
+  is(get_list_item_count(), 0, "Should be no items in the list");
 });
 
 // Install version 1 click the remove button and then upgrade to version 2 with
 // the manager open
-add_test(function() {
-  install_addon("browser_bug596336_1", function() {
-    AddonManager.getAddonByID("addon1@tests.mozilla.org", function(aAddon) {
-      check_addon(aAddon, "1.0");
-      ok(!aAddon.userDisabled, "Add-on should not be disabled");
+add_task(function*() {
+  yield install_addon("browser_bug596336_1");
+  let [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
+  yield check_addon(aAddon, "1.0");
+  ok(!aAddon.userDisabled, "Add-on should not be disabled");
 
-      let item = get_addon_element(gManagerWindow, "addon1@tests.mozilla.org");
-      EventUtils.synthesizeMouseAtCenter(get_node(item, "remove-btn"), { }, gManagerWindow);
+  let item = get_addon_element(gManagerWindow, "addon1@tests.mozilla.org");
+  EventUtils.synthesizeMouseAtCenter(get_node(item, "remove-btn"), { }, gManagerWindow);
 
-      // Force XBL to apply
-      item.clientTop;
+  // Force XBL to apply
+  item.clientTop;
 
-      ok(!!(aAddon.pendingOperations & AddonManager.PENDING_UNINSTALL), "Add-on should be pending uninstall");
-      is_element_visible(get_class_node(item, "pending"), "Pending message should be visible");
+  ok(aAddon.userDisabled, "Add-on should be disabled");
+  ok(!aAddon.pendingUninstall, "Add-on should not be pending uninstall");
+  is_element_visible(get_class_node(item, "pending"), "Pending message should be visible");
 
-      install_addon("browser_bug596336_2", function() {
-        AddonManager.getAddonByID("addon1@tests.mozilla.org", function(aAddon) {
-          check_addon(aAddon, "2.0");
-          ok(!aAddon.userDisabled, "Add-on should not be disabled");
+  yield install_addon("browser_bug596336_2");
+  [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
+  yield check_addon(aAddon, "2.0");
+  ok(!aAddon.userDisabled, "Add-on should not be disabled");
 
-          aAddon.uninstall();
+  aAddon.uninstall();
 
-          is(get_list_item_count(), 0, "Should be no items in the list");
-
-          run_next_test();
-        });
-      });
-    });
-  });
+  is(get_list_item_count(), 0, "Should be no items in the list");
 });
 
 // Install version 1, disable it, click the remove button and then upgrade to
 // version 2 with the manager open
-add_task(function() {
+add_task(function*() {
   yield install_addon("browser_bug596336_1");
   let [aAddon] = yield promiseAddonsByIDs(["addon1@tests.mozilla.org"]);
   aAddon.userDisabled = true;
diff --git a/toolkit/mozapps/extensions/test/browser/browser_cancelCompatCheck.js b/toolkit/mozapps/extensions/test/browser/browser_cancelCompatCheck.js
index 1799adcdda..f4ea898945 100644
--- a/toolkit/mozapps/extensions/test/browser/browser_cancelCompatCheck.js
+++ b/toolkit/mozapps/extensions/test/browser/browser_cancelCompatCheck.js
@@ -220,7 +220,7 @@ let inactiveAddonIds = [
 
 // Make sure the addons in the list are not installed
 function* check_addons_uninstalled(aAddonList) {
-  let foundList = yield promise_addons_by_ids([addon.id for (addon of aAddonList)]);
+  let foundList = yield promise_addons_by_ids(aAddonList.map(a => a.id));
   for (let i = 0; i < aAddonList.length; i++) {
     ok(!foundList[i], "Addon " + aAddonList[i].id + " is not installed");
   }
@@ -232,7 +232,7 @@ function* check_addons_uninstalled(aAddonList) {
 // Add-ons that have updates available should not update if they were disabled before
 // For this test, addon8 became disabled during update and addon9 was previously disabled,
 // so addon8 should update and addon9 should not
-add_task(function cancel_during_repopulate() {
+add_task(function* cancel_during_repopulate() {
   let a5, a8, a9, a10;
 
   Services.prefs.setBoolPref(PREF_STRICT_COMPAT, true);
@@ -297,7 +297,7 @@ add_task(function cancel_during_repopulate() {
 // calls in gVersionInfoPage_onPageShow() to complete
 // For this test, both addon8 and addon9 were disabled by this update, but addon8
 // is set to not auto-update, so only addon9 should update in the background
-add_task(function cancel_during_findUpdates() {
+add_task(function* cancel_during_findUpdates() {
   let a5, a8, a9;
 
   Services.prefs.setBoolPref(PREF_STRICT_COMPAT, true);
@@ -357,7 +357,7 @@ add_task(function cancel_during_findUpdates() {
 // to continue updating in the background and cancels any other updates
 // Same conditions as the previous test - addon8 and addon9 have updates available,
 // addon8 is set to not auto-update so only addon9 should become compatible
-add_task(function cancel_mismatch() {
+add_task(function* cancel_mismatch() {
   let a3, a5, a7, a8, a9;
 
   Services.prefs.setBoolPref(PREF_STRICT_COMPAT, true);
@@ -416,7 +416,7 @@ add_task(function cancel_mismatch() {
 
 // Cancelling during the 'mismatch' screen with only add-ons that have
 // no updates available
-add_task(function cancel_mismatch_no_updates() {
+add_task(function* cancel_mismatch_no_updates() {
   let a3, a5, a6
 
   Services.prefs.setBoolPref(PREF_STRICT_COMPAT, true);
diff --git a/toolkit/mozapps/extensions/test/browser/browser_experiments.js b/toolkit/mozapps/extensions/test/browser/browser_experiments.js
index 3da5fd423f..b973f0b12a 100644
--- a/toolkit/mozapps/extensions/test/browser/browser_experiments.js
+++ b/toolkit/mozapps/extensions/test/browser/browser_experiments.js
@@ -331,7 +331,7 @@ add_task(function* testActivateExperiment() {
   is_element_visible(el, "Experiment info is visible on experiment tab.");
 });
 
-add_task(function testDeactivateExperiment() {
+add_task(function* testDeactivateExperiment() {
   if (!gExperiments) {
     return;
   }
@@ -379,7 +379,7 @@ add_task(function testDeactivateExperiment() {
   is_element_hidden(el, "Preferences button is not visible.");
 });
 
-add_task(function testActivateRealExperiments() {
+add_task(function* testActivateRealExperiments() {
   if (!gExperiments) {
     info("Skipping experiments test because that feature isn't available.");
     return;
@@ -532,7 +532,7 @@ add_task(function testActivateRealExperiments() {
   }
 });
 
-add_task(function testDetailView() {
+add_task(function* testDetailView() {
   if (!gExperiments) {
     info("Skipping experiments test because that feature isn't available.");
     return;
diff --git a/toolkit/mozapps/extensions/test/xpcshell/head_addons.js b/toolkit/mozapps/extensions/test/xpcshell/head_addons.js
index 0ea7b1cd5a..0329a0dcc2 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/head_addons.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/head_addons.js
@@ -194,7 +194,8 @@ function do_get_file_hash(aFile, aAlgorithm) {
   let toHexString = charCode => ("0" + charCode.toString(16)).slice(-2);
 
   let binary = crypto.finish(false);
-  return aAlgorithm + ":" + [toHexString(binary.charCodeAt(i)) for (i in binary)].join("")
+  let hash = Array.from(binary, c => toHexString(c.charCodeAt(0)));
+  return aAlgorithm + ":" + hash.join("");
 }
 
 /**
@@ -876,7 +877,9 @@ function promiseSetExtensionModifiedTime(aPath, aTime) {
     try {
       let iterator = new OS.File.DirectoryIterator(aPath);
       entries = yield iterator.nextBatch();
-    } catch (ex if ex instanceof OS.File.Error) {
+    } catch (ex) {
+      if (!(ex instanceof OS.File.Error))
+        throw ex;
       return;
     } finally {
       if (iterator) {
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_DeferredSave.js b/toolkit/mozapps/extensions/test/xpcshell/test_DeferredSave.js
index 7599c8b808..ed92afc3b3 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_DeferredSave.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_DeferredSave.js
@@ -147,7 +147,7 @@ function run_test() {
 }
 
 // Modify set data once, ask for save, make sure it saves cleanly
-add_task(function test_basic_save_succeeds() {
+add_task(function* test_basic_save_succeeds() {
   setQuickMockTimer();
   let tester = DeferredSaveTester();
   let data = "Test 1 Data";
@@ -159,7 +159,7 @@ add_task(function test_basic_save_succeeds() {
 
 // Two saves called during the same event loop, both with callbacks
 // Make sure we save only the second version of the data
-add_task(function test_two_saves() {
+add_task(function* test_two_saves() {
   setQuickMockTimer();
   let tester = DeferredSaveTester();
   let firstCallback_happened = false;
@@ -182,7 +182,7 @@ add_task(function test_two_saves() {
 
 // Two saves called with a delay in between, both with callbacks
 // Make sure we save the second version of the data
-add_task(function test_two_saves_delay() {
+add_task(function* test_two_saves_delay() {
   let timerPromise = setPromiseMockTimer();
   let tester = DeferredSaveTester();
   let firstCallback_happened = false;
@@ -219,7 +219,7 @@ add_task(function test_two_saves_delay() {
 // Test case where OS.File immediately reports an error when the write begins
 // Also check that the "error" getter correctly returns the error
 // Then do a write that succeeds, and make sure the error is cleared
-add_task(function test_error_immediate() {
+add_task(function* test_error_immediate() {
   let tester = DeferredSaveTester();
   let testError = new Error("Forced failure");
   function writeFail(aTester) {
@@ -243,7 +243,7 @@ add_task(function test_error_immediate() {
 // Save one set of changes, then while the write is in progress, modify the
 // data two more times. Test that we re-write the dirty data exactly once
 // after the first write succeeds
-add_task(function dirty_while_writing() {
+add_task(function* dirty_while_writing() {
   let tester = DeferredSaveTester();
   let firstData = "First data";
   let secondData = "Second data";
@@ -317,7 +317,7 @@ function write_then_disable(aTester) {
 
 // Flush tests. First, do an ordinary clean save and then call flush;
 // there should not be another save
-add_task(function flush_after_save() {
+add_task(function* flush_after_save() {
   setQuickMockTimer();
   let tester = DeferredSaveTester();
   let dataToSave = "Flush after save";
@@ -328,7 +328,7 @@ add_task(function flush_after_save() {
 });
 
 // Flush while a write is in progress, but the in-memory data is clean
-add_task(function flush_during_write() {
+add_task(function* flush_during_write() {
   let tester = DeferredSaveTester();
   let dataToSave = "Flush during write";
   let firstCallback_happened = false;
@@ -361,7 +361,7 @@ add_task(function flush_during_write() {
 // Flush while dirty but write not in progress
 // The data written should be the value at the time
 // flush() is called, even if it is changed later
-add_task(function flush_while_dirty() {
+add_task(function* flush_while_dirty() {
   let timerPromise = setPromiseMockTimer();
   let tester = DeferredSaveTester();
   let firstData = "Flush while dirty, valid data";
@@ -397,7 +397,7 @@ add_task(function flush_while_dirty() {
 // then flush, then modify the data again
 // Data for the second write should be taken at the time
 // flush() is called, even if it is modified later
-add_task(function flush_writing_dirty() {
+add_task(function* flush_writing_dirty() {
   let timerPromise = setPromiseMockTimer();
   let tester = DeferredSaveTester();
   let firstData = "Flush first pass data";
@@ -463,7 +463,7 @@ function badDataProvider() {
 
 // Handle cases where data provider throws
 // First, throws during a normal save
-add_task(function data_throw() {
+add_task(function* data_throw() {
   setQuickMockTimer();
   badDataError = expectedDataError;
   let tester = DeferredSaveTester(badDataProvider);
@@ -473,7 +473,7 @@ add_task(function data_throw() {
 });
 
 // Now, throws during flush
-add_task(function data_throw_during_flush() {
+add_task(function* data_throw_during_flush() {
   badDataError = expectedDataError;
   let tester = DeferredSaveTester(badDataProvider);
   let firstCallback_happened = false;
@@ -508,7 +508,7 @@ add_task(function data_throw_during_flush() {
 // actually restart timer for delayed write
 // write completes
 // delayed timer goes off, throws error because DeferredSave has been torn down
-add_task(function delay_flush_race() {
+add_task(function* delay_flush_race() {
   let timerPromise = setPromiseMockTimer();
   let tester = DeferredSaveTester();
   let firstData = "First save";
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_asyncBlocklistLoad.js b/toolkit/mozapps/extensions/test/xpcshell/test_asyncBlocklistLoad.js
index 11d9f2943e..38e563979b 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_asyncBlocklistLoad.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_asyncBlocklistLoad.js
@@ -6,7 +6,7 @@ function run_test() {
   run_next_test();
 }
 
-add_task(function () {
+add_task(function* () {
   let blocklist = AM_Cc["@mozilla.org/extensions/blocklist;1"].
                   getService().wrappedJSObject;
   let scope = Components.utils.import("resource://gre/modules/osfile.jsm");
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_locked2.js b/toolkit/mozapps/extensions/test/xpcshell/test_locked2.js
index 10b13c9f65..b5766ce4b7 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_locked2.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_locked2.js
@@ -71,7 +71,7 @@ var addon5 = {
 const profileDir = gProfD.clone();
 profileDir.append("extensions");
 
-add_task(function() {
+add_task(function*() {
 
   createAppInfo("xpcshell@tests.mozilla.org", "XPCShell", "2", "2");
 
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_mapURIToAddonID.js b/toolkit/mozapps/extensions/test/xpcshell/test_mapURIToAddonID.js
index a6f9c80523..7d8f1eed64 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_mapURIToAddonID.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_mapURIToAddonID.js
@@ -279,11 +279,12 @@ function run_test_invalidarg() {
       AddonManager.mapURIToAddonID(test);
       throw new Error("Shouldn't be able to map the URI in question");
     }
-    catch (ex if ex.result) {
-      do_check_eq(ex.result, Components.results.NS_ERROR_INVALID_ARG);
-    }
     catch (ex) {
-      do_throw(ex);
+      if (ex.result) {
+        do_check_eq(ex.result, Components.results.NS_ERROR_INVALID_ARG);
+      } else {
+        do_throw(ex);
+      }
     }
   }
 
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_no_addons.js b/toolkit/mozapps/extensions/test/xpcshell/test_no_addons.js
index ae75fbb438..ae321ccc3a 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_no_addons.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_no_addons.js
@@ -56,14 +56,14 @@ function check_empty_state() {
 // bootstrap add-ons preference is not found
 // add-on directory state preference is an empty array
 // no pending operations
-add_task(function first_run() {
+add_task(function* first_run() {
   startupManager();
   check_empty_state();
   yield true;
 });
 
 // Now do something that causes a DB load, and re-check
-function trigger_db_load() {
+function* trigger_db_load() {
   let addonDefer = Promise.defer();
   AddonManager.getAddonsByTypes(['extension'], addonDefer.resolve);
   let addonList = yield addonDefer.promise;
@@ -76,7 +76,7 @@ function trigger_db_load() {
 add_task(trigger_db_load);
 
 // Now restart the manager and check again
-add_task(function restart_and_recheck() {
+add_task(function* restart_and_recheck() {
   restartManager();
   check_empty_state();
   yield true;
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_updateCancel.js b/toolkit/mozapps/extensions/test/xpcshell/test_updateCancel.js
index d513f4adfd..8ed536ae08 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_updateCancel.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_updateCancel.js
@@ -75,7 +75,7 @@ writeInstallRDFForExtension({
   name: "Test Addon 1",
 }, profileDir);
 
-add_task(function cancel_during_check() {
+add_task(function* cancel_during_check() {
   startupManager();
 
   let a1 = yield promiseAddonByID("addon1@tests.mozilla.org");
@@ -109,7 +109,7 @@ add_task(function cancel_during_check() {
 
 // Test that update check is cancelled if the XPI provider shuts down while
 // the update check is in progress
-add_task(function shutdown_during_check() {
+add_task(function* shutdown_during_check() {
   // Reset our HTTP listener
   httpReceived = Promise.defer();
 
diff --git a/toolkit/mozapps/extensions/test/xpcshell/test_update_strictcompat.js b/toolkit/mozapps/extensions/test/xpcshell/test_update_strictcompat.js
index 3fc16a0f7e..61e0e6b00b 100644
--- a/toolkit/mozapps/extensions/test/xpcshell/test_update_strictcompat.js
+++ b/toolkit/mozapps/extensions/test/xpcshell/test_update_strictcompat.js
@@ -1023,7 +1023,7 @@ for (let test of testParams) {
     });
   });
 
-  add_task(function cleanup() {
+  add_task(function* cleanup() {
     let addons = yield new Promise(resolve => {
       AddonManager.getAddonsByTypes(["extension"], resolve);
     });
diff --git a/xpcom/reflect/xptinfo/ShimInterfaceInfo.cpp b/xpcom/reflect/xptinfo/ShimInterfaceInfo.cpp
index 8ea43d73f1..a93d5b0b2e 100644
--- a/xpcom/reflect/xptinfo/ShimInterfaceInfo.cpp
+++ b/xpcom/reflect/xptinfo/ShimInterfaceInfo.cpp
@@ -44,7 +44,6 @@
 #include "nsIDOMEvent.h"
 #include "nsIDOMEventTarget.h"
 #include "nsIDOMFileList.h"
-#include "nsIDOMFileReader.h"
 #include "nsIDOMFocusEvent.h"
 #include "nsIDOMFormData.h"
 #include "nsIDOMGeoPositionError.h"
@@ -179,7 +178,6 @@
 #include "mozilla/dom/EventBinding.h"
 #include "mozilla/dom/EventTargetBinding.h"
 #include "mozilla/dom/FileListBinding.h"
-#include "mozilla/dom/FileReaderBinding.h"
 #include "mozilla/dom/FocusEventBinding.h"
 #include "mozilla/dom/FormDataBinding.h"
 #include "mozilla/dom/HistoryBinding.h"
@@ -363,7 +361,6 @@ const ComponentsInterfaceShimEntry kComponentsInterfaceShimMap[] =
   DEFINE_SHIM(Element),
   DEFINE_SHIM(Event),
   DEFINE_SHIM(EventTarget),
-  DEFINE_SHIM(FileReader),
   DEFINE_SHIM(FileList),
   DEFINE_SHIM(FocusEvent),
   DEFINE_SHIM(FormData),