import changes from `dev' branch of rmottola/Arctic-Fox:

- bug 1179662 rename AudioNode::Stream() to GetStream() as it may return null r=padenot (d2d7e5f90)
-  bug 1197043 remove unnecessary aSampleRate parameter for AudioNodeStream creation r=padenot (3733ceb56)
- bug 1197043 rename Add/RemoveStream to Add/RemoveStreamGraphThread r=padenot (f648b8251)
- bug 1197043 introduce MediaStreamGraph::AddStream() r=padenot (ac021d4b2)
- bug 1197043 move AudioNodeStream creation to stream class r=padenot (a90a05910)
- bug 1197043 use flags to distinguish between external streams and events r=padenot (024dd96f1)
- Bug 1176300 - Move libsoundtouch to lgpllibs; r=glandium (99a546adf)
- Bug 1176300 - Add lgpllibs library to build system; r=glandium (bb4d07670)
- Bug 1176300 - Update libsoundtouch to patched r222; r=padenot (hg rev 8c32c900cb48)
- Bug 1176300 - Add soundtouch factory functions for DLL memory handling on windows; r=padenot (hg rev 84a1ffbc2db8)
- Bug 901633 - Part 1 - Implement a generic audio packetizer. r=jesup (a38c2d70b)
- Bug 901633 - Part 2 - Make AudioChannelFormat and AudioSegment more generic. r=roc (556b7349f)
- Bug 901633 - Part 3 - Fix TrackEncoder to use the new AudioChunk methods. r=jesup (56e018f83)
- Bug 901633 - Part 4 - Update AudioNodeStream to use new chunk methods. r=roc (9df19b894)
- Bug 901633 - Part 6 - Update DelayBuffer to use the new AudioChunk methods. r=karlt (6d5684334)
- Bug 901633 - Part 7 - Update AudioNodeExternalInputStream to use the new AudioChunk methods. r=karlt (caa3afa01)
- Bug 1155089: Fix hazard analysis bustage on a CLOSED TREE. r=bustage (8d23ccf39)
- Bug 1166183 - Back out the direct listener removal landed by mistake in bug 1141781. r=jesup (745d683d4)
- Bug 1166183 - Reset PipelineListener's flag after ReplaceTrack(). r=bwc (2fb38ca01)
- Bug 1170059 - Fix -Wunreachable-code clang warnings in webrtc/signaling. r=jesup (0d99b30ca)
- Bug 1139144 - Remove unused empty() definition from databuffer.h. r=mt (2fef64e3c)
- Bug 822129: don't alloc/free on every packet send in MediaPipeline r=bwc (ddfdc9455)
- Bug 1172397 - Check for Conduit/Type mismatch on every frame. r=jesup, r=bwc (a399a3336)
- Bug 1137169 - Uninitialised value uses related to mozilla::dom::WebAdioUtils::SpeexResamplerProcess. r=rjesup. (ce14ac278)
- Bug 901633 - Part 5 - Make MediaPipeline downmix and properly convert audio for webrtc.org code. r=jesup (89138b5d5)
- Bug 901633 - Part 8 - Use our new generic packetizer in the MediaPipeline so that we can packetize stereo easily. r=jesup (fb5d075b6)
- Bug 901633 - Part 9 - Make the necessary changes to VoEExternalMediaImpl::ExternalRecordingInsertData so that it the number of channels is forwarded down the webrtc.org code. r=jesup (d5d7dd4ca)
- Bug 901633 - Part 10 - Change the receiving side of the MediaPipeline so that it can detect and handle stereo. r=jesup (a73c1520f)
- Bug 901633 - Part 11 - Add an API in webrtc.org's output mixer to get the output channel count. r=jesup (4b396b85e)
- Bug 901633 - Part 12 - Add a function to deinterleave and convert an audio buffer. r=jesup (47ce3c7a5)
- Bug 901633 - Part 13 - Teach the resampler at the input of the MSG to dynamically change its channel count if needed. r=jesup (120c8d037)
- Bug 901633 - Part 14 - Add testing for our audio processing functions. r=jesup (5aa95b82e)
- Bug 901633 - Part 15 - Remove an allocation on the sending side, out of the packetizer. r=jesup (df8aed252)
- Bug 901633 - Part 16 - Remove another allocation in the sending side r=jesup (1e2fc8bca)
- Bug 1196408 - Make sure we only report a corrupt/slow video frame once. r=cpearce (5bae2f17a)
- bug 1162364 report telemetry on WMFMediaDataDecoder errors r=cpearce,f=vladan,bsmedberg (e217618ef)
- Bug 1193864 - Fixed dom/media/platforms/wmf/ compilation on mingw. r=cpearce (4e8c0ecd7)
- Bug 1141139: Enable low latency decoding on Windows. r=cpearce (9e0a36e27)
- Bug 1193547 - Fallback to software decoding explicitly if the GPU doesn't support decoding the current resolution in hardware. r=cpearce,jya (7fbab8784)
- Bug 1196417 - Make video software fallback only affect the current video instead of the entire browser. r=cpearce (3e83f0677)

dom/media/webaudio/AudioNodeExternalInputStream.cpp

@@ -13,7 +13,7 @@ using namespace mozilla::dom;
 namespace mozilla {
 
 AudioNodeExternalInputStream::AudioNodeExternalInputStream(AudioNodeEngine* aEngine, TrackRate aSampleRate, uint32_t aContextId)
-  : AudioNodeStream(aEngine, MediaStreamGraph::INTERNAL_STREAM, aSampleRate, aContextId)
+  : AudioNodeStream(aEngine, NO_STREAM_FLAGS, aSampleRate, aContextId)
 {
   MOZ_COUNT_CTOR(AudioNodeExternalInputStream);
 }
@@ -23,50 +23,52 @@ AudioNodeExternalInputStream::~AudioNodeExternalInputStream()
   MOZ_COUNT_DTOR(AudioNodeExternalInputStream);
 }
 
+/* static */ already_AddRefed<AudioNodeExternalInputStream>
+AudioNodeExternalInputStream::Create(MediaStreamGraph* aGraph,
+                                     AudioNodeEngine* aEngine)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(aGraph->GraphRate() == aEngine->NodeMainThread()->Context()->SampleRate());
+
+  nsRefPtr<AudioNodeExternalInputStream> stream =
+    new AudioNodeExternalInputStream(aEngine, aGraph->GraphRate(),
+                                     aEngine->NodeMainThread()->Context()->Id());
+  aGraph->AddStream(stream);
+  return stream.forget();
+}
+
 /**
  * Copies the data in aInput to aOffsetInBlock within aBlock.
  * aBlock must have been allocated with AllocateInputBlock and have a channel
  * count that's a superset of the channels in aInput.
  */
+template <typename T>
 static void
-CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock,
+CopyChunkToBlock(AudioChunk& aInput, AudioChunk *aBlock,
                  uint32_t aOffsetInBlock)
 {
   uint32_t blockChannels = aBlock->ChannelCount();
-  nsAutoTArray<const void*,2> channels;
+  nsAutoTArray<const T*,2> channels;
   if (aInput.IsNull()) {
     channels.SetLength(blockChannels);
     PodZero(channels.Elements(), blockChannels);
   } else {
-    channels.SetLength(aInput.ChannelCount());
-    PodCopy(channels.Elements(), aInput.mChannelData.Elements(), channels.Length());
+    const nsTArray<const T*>& inputChannels = aInput.ChannelData<T>();
+    channels.SetLength(inputChannels.Length());
+    PodCopy(channels.Elements(), inputChannels.Elements(), channels.Length());
     if (channels.Length() != blockChannels) {
       // We only need to upmix here because aBlock's channel count has been
       // chosen to be a superset of the channel count of every chunk.
-      AudioChannelsUpMix(&channels, blockChannels, nullptr);
+      AudioChannelsUpMix(&channels, blockChannels, static_cast<T*>(nullptr));
     }
   }
 
-  uint32_t duration = aInput.GetDuration();
   for (uint32_t c = 0; c < blockChannels; ++c) {
     float* outputData = aBlock->ChannelFloatsForWrite(c) + aOffsetInBlock;
     if (channels[c]) {
-      switch (aInput.mBufferFormat) {
-      case AUDIO_FORMAT_FLOAT32:
-        ConvertAudioSamplesWithScale(
-            static_cast<const float*>(channels[c]), outputData, duration,
-            aInput.mVolume);
-        break;
-      case AUDIO_FORMAT_S16:
-        ConvertAudioSamplesWithScale(
-            static_cast<const int16_t*>(channels[c]), outputData, duration,
-            aInput.mVolume);
-        break;
-      default:
-        NS_ERROR("Unhandled format");
-      }
+      ConvertAudioSamplesWithScale(channels[c], outputData, aInput.GetDuration(), aInput.mVolume);
     } else {
-      PodZero(outputData, duration);
+      PodZero(outputData, aInput.GetDuration());
     }
   }
 }
@@ -97,7 +99,18 @@ static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
 
   uint32_t duration = 0;
   for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
-    CopyChunkToBlock(*ci, aBlock, duration);
+    switch (ci->mBufferFormat) {
+      case AUDIO_FORMAT_S16: {
+        CopyChunkToBlock<int16_t>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_FLOAT32: {
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_SILENCE:
+        break;
+    }
     duration += ci->GetDuration();
   }
 }