mirror of
https://github.com/roytam1/palemoon27.git
synced 2026-05-26 05:11:03 +00:00
93f846cd1f
- Bug 1275016 - Rename Endian.h to EndianUtils.h to avoid #include confusion with Android's endian.h stdlib header. r=froydnj (b54a25f572) - add crashreporter stuff (aa7ef15337) - Bug 1261168 - Add AlignedAutoTArray type in Web Audio; r=padenot (285d2cb88b) - Bug 1273390. Part 1 - move some functions to private. r=jya. (07a3037e59) - Bug 1273390. Part 2 - add assertions. r=jya. (2cae7c596a) - Bug 1273390. Part 3 - rename some functions to be consistent with other sub-classes of MediaDataDecoder. r=jya. (c48c7060ce) - Bug 1273390. Part 4 - remove use of FlushableTaskQueue::Flush(). r=jya. (00565a65f4) - Bug 1273390. Part 5 - remove use of FlushableTaskQueue. r=jya. (30600b204e) - Bug 1273774. Part 1 - remove unused members and thread assertions. r=jya (f5177ed641) - Bug 1273774. Part 2 - do decoding jobs synchronously without dispatching. r=jya. (62d840d27c) - Bug 1273774. Part 3 - remove members no longer used. r=jya. (e957ca512a) - Bug 1244410: [ffmpeg] Ensure the last drained frame has the proper duration set. r=gerald (d5521bfdd4) - Bug 1271508. Part 1 - refactor FFmpegAudioDecoder code to be similar to FFmpegVideoDecoder::Input() so it would be easier to extract common code to the parent class. r=jya. (613e6c624c) - Bug 1271508. Part 2 - rename functions so they are the same as those of FFmpegAudioDecoder so it would be easier to extract common code to the parent class. r=jya. (cb281cba26) - Bug 1270350 - per comment 0, use SyncRunnable to repalce the boilerplate code. r=jya. (b99460e571) - Bug 1271508. Part 3 - extract code to the parent class and remove use of mTaskQueue from sub-classes. r=jya. (2a7ff4dd1e) - Bug 1274216 - remove use of FlushableTaskQueue from PlatformDecoderModule. r=jya. (eb160c5fa2) - Bug 1271517. Part 1 - remove use of FlushableTaskQueue::Flush() from FFmpegDataDecoder::Flush(). r=jya. (fdf10da4ab) - Bug 1271517. Part 2 - remove use of FlushableTaskQueue. r=jya. (a7016d8506) - Bug 1273397. Part 1 - rename some functions to be consistent with other MediaDataDecoder sub-classes. r=jya. (7eecb164be) - Bug 1273397. Part 2 - constify some members. r=jya. (e4482f9a23) - Bug 1273397. Part 3 - remove use of FlushableTaskQueue::Flush(). r=jya. (0b7ee073fe) - Bug 1273397. Part 4 - remove use of FlushableTaskQueue. r=jya. (6a612161d5) - Bug 1273397. Part 5 - add assertions. r=jya. (ff3a62a6fb) - Bug 1274199 - remove use of FlushableTaskQueue. r=cpearce. (adc4c84ede) - Bug 1273405. Part 1 - rename some functions to be consistent with other MediaDataDecoder sub-classes. r=jya. (af123d6c21) - Bug 1273405. Part 2 - remove use of FlushableTaskQueue::Flush(). r=jya. (2d144bfbcd) - Bug 1273405. Part 3 - remove use of FlushableTaskQueue. r=jya. (1e9ea3c2c7) - Bug 1273405. Part 4 - add assertions. r=jya. (b400647323) - Bug 1271491: [WMF] P1. Don't use main thread only preferences methods. r=cpearce (7177454dfb) - Bug 1262427. Don't try D3D11 harder. r=dvander (404147d6fa) - Use gfxConfig for D3D9 preferences. (bug 1270650, r=jrmuizel) (40d89c154c) - Bug 1271491: P2. Allow initialization of WMFPlatformDecoderModule from any threads. r=mattwoodrow (c8fe0bf009) - Bug 1271491: P3. Remove refcounting the number of time apple's linkers are called. r=cpearce (0324ffe876) - Bug 1271491: [ffmpeg] P4. Remove requirements to call Init on the main thread. r=cpearce (b511d7dfd5) - Bug 1271491: [GMP] P5. Allow GMPDecoderModule::Init() to be called off the main thread. r=cpearce (2131eb0b2e) - Bug 1266102 - Don't run VP9 benchmark on Android r=jya (57d7b573fe) - Bug 1271491: P6. Remove the need to call PDMFactory::Init(). r=cpearce (5726cfe49c) - Bug 1271491: P7. Remove unused members. r=alfredo (0f8a9dde73) - Bug 1268905 - Disable D3D11 with some Toshiba DLLs - r=cpearce (b5bf77442e) - Bug 1269204 - Disable D3D11 with idg10umd32 9.17.10.2857 - r=cpearce (7eb6a3d96b) - Bug 1273406 - Disable D3D11 with some iSonyVideoProcessor DLLs - r=cpearce (d9b6f0cefe) - Bug 1273406 - Ugly macros transform into beautiful constexpr goodness - r=cpearce (0671483695) - Bug 1273691 - Implement 'media.wmf.disable-d3d11-for-dlls' pref - r=cpearce (193ae53070) - Bug 1272225. Part 1 - add assertions to make thread constraints clear. r=jya. (83c620470e) - Bug 1272225. Part 2 - remove use of FlushableTaskQueue::Flush(). r=jya. (9473e092d1) - Bug 1272553. Part 1 - move code around to be able to extract common code in P2. r=jya. (d727f97ee8) - Bug 1272553. Part 2 - extract common code to the parent class. r=jya. (2fb3cd4bd9) - Bug 1272553. Part 3 - make mTaskQueue private. r=jya. (93fea98cb6) - Bug 1272232. Part 1 - move code around so we can extract common code in P2. r=jya. (8cdaab9078) - Bug 1272232. Part 2 - extract common code to the parent class. r=jya. (27156668b3) - Bug 1272232. Part 3 - constify some members and make them private when possible. r=jya. (550b963d97) - Bug 1272232. Part 4 - remove use of FlushableTaskQueue::Flush(). r=jya. (bdbfdeb6bc) - Bug 1272232. Part 5 - remove use of FlushableTaskQueue. r=jya. (640f889a9d) - Bug 1274913 - Move PDM log definition to header. r=njn (823b07f42b) - Bug 1275538: P1. Abort early if a skip request is in progress. r=gerald,kamidphish (d67b8a2236) - Bug 1272422 - Part 1: Expose control of suspending background video. r=cpearce (ec7193773f) - Bug 1272422 - Part 2: Vidoe -> Video. r=cpearce (97390aee69) - Bug 1272422 - Part 3: Don't reset audio queue. r=jya (e183db1062) - Bug 1272964: P1. Only activate skip to next keyframe logic when next keyframe time is known. r=gerald (1be74df027) - Bug 1272964: P2. Don't activate skip to next keyframe until we passed the internal seek target. r=gerald (c55b6ae003) - Bug 1258922: [MSE] P1. Initialise variable. r=gerald (56a5acb345) - Bug 1258922: [MSE] P2. Do not go over gap when attempting to find the next key frame. r=gerald (db1319f080) - Bug 1258922: [MSE] P3. Check that the data we are attempting to skip to is buffered. r=gerald (621d71d5d6) - Bug 1258922: [MSE] P4. Set draining flag to true when skip to next keyframe failed. r=gerald (6c75613faf) - Bug 1272916: [MSE] P1. Don't rely only on dts gap to establish if we have a gap in our source buffer. r=gerald (8770113b83) - Bug 1272964: [MSE] P3. Do not skip over gaps when searching for the next keyframe. r=gerald (76916c5ac6) - Bug 1272964: P4. Only flush decoder if skip to next keyframe actually succeeds. r=cpearce (5394708eef) - Bug 1270323: P1. Don't reset flag indicating that new data was received. r=cpearce (d32f06ef34) - Bug 1270323: P2. Don't process new incoming data while a skip to next keyframe is pending. r=cpearce (bca7909de9) - Bug 1270323: [ffmpeg] P3. Use the dts of the last sample input, not the dts of the last decoded sample (0d768c33ef) - Bug 1270323: P4. Don't drain decoder if we're already waiting for new data. r=cpearce (679302cb6e) - Bug 1270323: P5. Prevent potential null deref. r=cpearce (cc63270e06) - Bug 1275538: P2. Drop decoded frames that we know are already too late. r=kamidphish (4e7af9398c) - Bug 1273018: P1. Rename some members. r=gerald (3a92fbd994) - Bug 1273018: P2. Don't reject audio waiting promise when performing a video only seek. r=gerald (34e4988db1) - Bug 1273018: P3. Adjust range of audio assertions. r=gerald (feb2afd0ae) - Bug 1249706 - Backout a085ea2d24bb for blowing telemetry server's mind. r=backout (d61fb51f52) - Bug 1249706 - Fix 8fe22dd4fc8a (backout of a085ea2d24bb). r=bustage (ba65251db7) - Bug 1272964: [MSE] P5. Default to skipping to the next keyframe if no keyframe was found past currentTime. (29086fcf56) - Bug 1272964: P6. Exclude frames dropped due to internal seeking from calculations. r=cpearce (bf6faa7612) - Bug 1068151 - keep decoding a corrupted video. r=jya (3b5462e5b6) - Bug 1273947 - Update ResetDecode() to ResetDecode(TargetQueue) r=jya (6c28d46974) - Bug 1277508: P1. Don't attempt to demux new samples while we're currently draining. r=kamidphish (64f200b921) - Bug 1274933: Reject data promise when EOS is encountered following waiting for data. r=gerald (5bba4a7853) - Bug 1277508: P2. Add HasPendingDrain convenience method. r=kamidphish (3d89a90a97)
696 lines
22 KiB
C++
696 lines
22 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AppleUtils.h"
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#include "MP4Decoder.h"
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#include "mp4_demuxer/Adts.h"
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#include "MediaInfo.h"
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#include "AppleATDecoder.h"
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#include "mozilla/Logging.h"
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#include "mozilla/SyncRunnable.h"
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#include "mozilla/UniquePtr.h"
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#define LOG(...) MOZ_LOG(sPDMLog, mozilla::LogLevel::Debug, (__VA_ARGS__))
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#define FourCC2Str(n) ((char[5]){(char)(n >> 24), (char)(n >> 16), (char)(n >> 8), (char)(n), 0})
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namespace mozilla {
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AppleATDecoder::AppleATDecoder(const AudioInfo& aConfig,
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FlushableTaskQueue* aAudioTaskQueue,
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MediaDataDecoderCallback* aCallback)
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: mConfig(aConfig)
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, mFileStreamError(false)
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, mTaskQueue(aAudioTaskQueue)
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, mCallback(aCallback)
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, mConverter(nullptr)
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, mStream(nullptr)
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, mIsFlushing(false)
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{
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MOZ_COUNT_CTOR(AppleATDecoder);
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LOG("Creating Apple AudioToolbox decoder");
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LOG("Audio Decoder configuration: %s %d Hz %d channels %d bits per channel",
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mConfig.mMimeType.get(),
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mConfig.mRate,
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mConfig.mChannels,
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mConfig.mBitDepth);
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if (mConfig.mMimeType.EqualsLiteral("audio/mpeg")) {
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mFormatID = kAudioFormatMPEGLayer3;
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} else if (mConfig.mMimeType.EqualsLiteral("audio/mp4a-latm")) {
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mFormatID = kAudioFormatMPEG4AAC;
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} else {
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mFormatID = 0;
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}
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}
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AppleATDecoder::~AppleATDecoder()
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{
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MOZ_COUNT_DTOR(AppleATDecoder);
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MOZ_ASSERT(!mConverter);
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}
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RefPtr<MediaDataDecoder::InitPromise>
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AppleATDecoder::Init()
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{
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if (!mFormatID) {
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NS_ERROR("Non recognised format");
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return InitPromise::CreateAndReject(DecoderFailureReason::INIT_ERROR, __func__);
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}
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return InitPromise::CreateAndResolve(TrackType::kAudioTrack, __func__);
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}
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nsresult
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AppleATDecoder::Input(MediaRawData* aSample)
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{
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MOZ_ASSERT(mCallback->OnReaderTaskQueue());
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LOG("mp4 input sample %p %lld us %lld pts%s %llu bytes audio",
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aSample,
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aSample->mDuration,
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aSample->mTime,
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aSample->mKeyframe ? " keyframe" : "",
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(unsigned long long)aSample->Size());
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// Queue a task to perform the actual decoding on a separate thread.
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nsCOMPtr<nsIRunnable> runnable =
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NewRunnableMethod<RefPtr<MediaRawData>>(
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this,
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&AppleATDecoder::SubmitSample,
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RefPtr<MediaRawData>(aSample));
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mTaskQueue->Dispatch(runnable.forget());
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return NS_OK;
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}
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void
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AppleATDecoder::ProcessFlush()
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{
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MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
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mQueuedSamples.Clear();
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OSStatus rv = AudioConverterReset(mConverter);
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if (rv) {
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LOG("Error %d resetting AudioConverter", rv);
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}
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}
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nsresult
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AppleATDecoder::Flush()
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{
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MOZ_ASSERT(mCallback->OnReaderTaskQueue());
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LOG("Flushing AudioToolbox AAC decoder");
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mIsFlushing = true;
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nsCOMPtr<nsIRunnable> runnable =
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NewRunnableMethod(this, &AppleATDecoder::ProcessFlush);
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SyncRunnable::DispatchToThread(mTaskQueue, runnable);
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mIsFlushing = false;
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return NS_OK;
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}
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nsresult
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AppleATDecoder::Drain()
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{
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MOZ_ASSERT(mCallback->OnReaderTaskQueue());
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LOG("Draining AudioToolbox AAC decoder");
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mTaskQueue->AwaitIdle();
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mCallback->DrainComplete();
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return Flush();
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}
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nsresult
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AppleATDecoder::Shutdown()
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{
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MOZ_ASSERT(mCallback->OnReaderTaskQueue());
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LOG("Shutdown: Apple AudioToolbox AAC decoder");
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mQueuedSamples.Clear();
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OSStatus rv = AudioConverterDispose(mConverter);
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if (rv) {
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LOG("error %d disposing of AudioConverter", rv);
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return NS_ERROR_FAILURE;
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}
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mConverter = nullptr;
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if (mStream) {
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rv = AudioFileStreamClose(mStream);
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if (rv) {
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LOG("error %d disposing of AudioFileStream", rv);
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return NS_ERROR_FAILURE;
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}
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mStream = nullptr;
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}
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return NS_OK;
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}
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struct PassthroughUserData {
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UInt32 mChannels;
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UInt32 mDataSize;
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const void* mData;
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AudioStreamPacketDescription mPacket;
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};
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// Error value we pass through the decoder to signal that nothing
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// has gone wrong during decoding and we're done processing the packet.
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const uint32_t kNoMoreDataErr = 'MOAR';
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static OSStatus
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_PassthroughInputDataCallback(AudioConverterRef aAudioConverter,
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UInt32* aNumDataPackets /* in/out */,
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AudioBufferList* aData /* in/out */,
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AudioStreamPacketDescription** aPacketDesc,
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void* aUserData)
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{
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PassthroughUserData* userData = (PassthroughUserData*)aUserData;
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if (!userData->mDataSize) {
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*aNumDataPackets = 0;
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return kNoMoreDataErr;
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}
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if (aPacketDesc) {
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userData->mPacket.mStartOffset = 0;
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userData->mPacket.mVariableFramesInPacket = 0;
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userData->mPacket.mDataByteSize = userData->mDataSize;
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*aPacketDesc = &userData->mPacket;
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}
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aData->mBuffers[0].mNumberChannels = userData->mChannels;
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aData->mBuffers[0].mDataByteSize = userData->mDataSize;
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aData->mBuffers[0].mData = const_cast<void*>(userData->mData);
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// No more data to provide following this run.
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userData->mDataSize = 0;
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return noErr;
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}
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void
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AppleATDecoder::SubmitSample(MediaRawData* aSample)
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{
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MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
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if (mIsFlushing) {
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return;
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}
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nsresult rv = NS_OK;
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if (!mConverter) {
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rv = SetupDecoder(aSample);
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if (rv != NS_OK && rv != NS_ERROR_NOT_INITIALIZED) {
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mCallback->Error(MediaDataDecoderError::FATAL_ERROR);
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return;
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}
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}
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mQueuedSamples.AppendElement(aSample);
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if (rv == NS_OK) {
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for (size_t i = 0; i < mQueuedSamples.Length(); i++) {
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if (NS_FAILED(DecodeSample(mQueuedSamples[i]))) {
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mQueuedSamples.Clear();
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mCallback->Error(MediaDataDecoderError::DECODE_ERROR);
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return;
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}
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}
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mQueuedSamples.Clear();
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}
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if (mTaskQueue->IsEmpty()) {
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mCallback->InputExhausted();
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}
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}
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nsresult
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AppleATDecoder::DecodeSample(MediaRawData* aSample)
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{
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MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
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// Array containing the queued decoded audio frames, about to be output.
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nsTArray<AudioDataValue> outputData;
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UInt32 channels = mOutputFormat.mChannelsPerFrame;
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// Pick a multiple of the frame size close to a power of two
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// for efficient allocation.
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const uint32_t MAX_AUDIO_FRAMES = 128;
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const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
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// Descriptions for _decompressed_ audio packets. ignored.
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auto packets = MakeUnique<AudioStreamPacketDescription[]>(MAX_AUDIO_FRAMES);
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// This API insists on having packets spoon-fed to it from a callback.
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// This structure exists only to pass our state.
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PassthroughUserData userData =
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{ channels, (UInt32)aSample->Size(), aSample->Data() };
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// Decompressed audio buffer
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AlignedAudioBuffer decoded(maxDecodedSamples);
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if (!decoded) {
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return NS_ERROR_OUT_OF_MEMORY;
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}
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do {
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AudioBufferList decBuffer;
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decBuffer.mNumberBuffers = 1;
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decBuffer.mBuffers[0].mNumberChannels = channels;
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decBuffer.mBuffers[0].mDataByteSize =
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maxDecodedSamples * sizeof(AudioDataValue);
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decBuffer.mBuffers[0].mData = decoded.get();
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// in: the max number of packets we can handle from the decoder.
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// out: the number of packets the decoder is actually returning.
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UInt32 numFrames = MAX_AUDIO_FRAMES;
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OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
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_PassthroughInputDataCallback,
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&userData,
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&numFrames /* in/out */,
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&decBuffer,
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packets.get());
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if (rv && rv != kNoMoreDataErr) {
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LOG("Error decoding audio stream: %d\n", rv);
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return NS_ERROR_FAILURE;
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}
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if (numFrames) {
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outputData.AppendElements(decoded.get(), numFrames * channels);
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}
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if (rv == kNoMoreDataErr) {
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break;
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}
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} while (true);
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if (outputData.IsEmpty()) {
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return NS_OK;
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}
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size_t numFrames = outputData.Length() / channels;
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int rate = mOutputFormat.mSampleRate;
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media::TimeUnit duration = FramesToTimeUnit(numFrames, rate);
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if (!duration.IsValid()) {
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NS_WARNING("Invalid count of accumulated audio samples");
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return NS_ERROR_FAILURE;
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}
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#ifdef LOG_SAMPLE_DECODE
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LOG("pushed audio at time %lfs; duration %lfs\n",
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(double)aSample->mTime / USECS_PER_S,
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duration.ToSeconds());
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#endif
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AudioSampleBuffer data(outputData.Elements(), outputData.Length());
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if (!data.Data()) {
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return NS_ERROR_OUT_OF_MEMORY;
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}
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if (mChannelLayout && !mAudioConverter) {
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AudioConfig in(*mChannelLayout.get(), rate);
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AudioConfig out(channels, rate);
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if (!in.IsValid() || !out.IsValid()) {
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return NS_ERROR_FAILURE;
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}
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mAudioConverter = MakeUnique<AudioConverter>(in, out);
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}
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if (mAudioConverter) {
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MOZ_ASSERT(mAudioConverter->CanWorkInPlace());
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data = mAudioConverter->Process(Move(data));
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}
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RefPtr<AudioData> audio = new AudioData(aSample->mOffset,
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aSample->mTime,
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duration.ToMicroseconds(),
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numFrames,
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data.Forget(),
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channels,
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rate);
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mCallback->Output(audio);
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return NS_OK;
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}
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nsresult
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AppleATDecoder::GetInputAudioDescription(AudioStreamBasicDescription& aDesc,
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const nsTArray<uint8_t>& aExtraData)
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{
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MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
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// Request the properties from CoreAudio using the codec magic cookie
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AudioFormatInfo formatInfo;
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PodZero(&formatInfo.mASBD);
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formatInfo.mASBD.mFormatID = mFormatID;
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|
if (mFormatID == kAudioFormatMPEG4AAC) {
|
|
formatInfo.mASBD.mFormatFlags = mConfig.mExtendedProfile;
|
|
}
|
|
formatInfo.mMagicCookieSize = aExtraData.Length();
|
|
formatInfo.mMagicCookie = aExtraData.Elements();
|
|
|
|
UInt32 formatListSize;
|
|
// Attempt to retrieve the default format using
|
|
// kAudioFormatProperty_FormatInfo method.
|
|
// This method only retrieves the FramesPerPacket information required
|
|
// by the decoder, which depends on the codec type and profile.
|
|
aDesc.mFormatID = mFormatID;
|
|
aDesc.mChannelsPerFrame = mConfig.mChannels;
|
|
aDesc.mSampleRate = mConfig.mRate;
|
|
UInt32 inputFormatSize = sizeof(aDesc);
|
|
OSStatus rv = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
|
|
0,
|
|
NULL,
|
|
&inputFormatSize,
|
|
&aDesc);
|
|
if (NS_WARN_IF(rv)) {
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
// If any of the methods below fail, we will return the default format as
|
|
// created using kAudioFormatProperty_FormatInfo above.
|
|
rv = AudioFormatGetPropertyInfo(kAudioFormatProperty_FormatList,
|
|
sizeof(formatInfo),
|
|
&formatInfo,
|
|
&formatListSize);
|
|
if (rv || (formatListSize % sizeof(AudioFormatListItem))) {
|
|
return NS_OK;
|
|
}
|
|
size_t listCount = formatListSize / sizeof(AudioFormatListItem);
|
|
auto formatList = MakeUnique<AudioFormatListItem[]>(listCount);
|
|
|
|
rv = AudioFormatGetProperty(kAudioFormatProperty_FormatList,
|
|
sizeof(formatInfo),
|
|
&formatInfo,
|
|
&formatListSize,
|
|
formatList.get());
|
|
if (rv) {
|
|
return NS_OK;
|
|
}
|
|
LOG("found %u available audio stream(s)",
|
|
formatListSize / sizeof(AudioFormatListItem));
|
|
// Get the index number of the first playable format.
|
|
// This index number will be for the highest quality layer the platform
|
|
// is capable of playing.
|
|
UInt32 itemIndex;
|
|
UInt32 indexSize = sizeof(itemIndex);
|
|
rv = AudioFormatGetProperty(kAudioFormatProperty_FirstPlayableFormatFromList,
|
|
formatListSize,
|
|
formatList.get(),
|
|
&indexSize,
|
|
&itemIndex);
|
|
if (rv) {
|
|
return NS_OK;
|
|
}
|
|
|
|
aDesc = formatList[itemIndex].mASBD;
|
|
|
|
return NS_OK;
|
|
}
|
|
|
|
AudioConfig::Channel
|
|
ConvertChannelLabel(AudioChannelLabel id)
|
|
{
|
|
switch (id) {
|
|
case kAudioChannelLabel_Mono:
|
|
return AudioConfig::CHANNEL_MONO;
|
|
case kAudioChannelLabel_Left:
|
|
return AudioConfig::CHANNEL_LEFT;
|
|
case kAudioChannelLabel_Right:
|
|
return AudioConfig::CHANNEL_RIGHT;
|
|
case kAudioChannelLabel_Center:
|
|
return AudioConfig::CHANNEL_CENTER;
|
|
case kAudioChannelLabel_LFEScreen:
|
|
return AudioConfig::CHANNEL_LFE;
|
|
case kAudioChannelLabel_LeftSurround:
|
|
return AudioConfig::CHANNEL_LS;
|
|
case kAudioChannelLabel_RightSurround:
|
|
return AudioConfig::CHANNEL_RS;
|
|
case kAudioChannelLabel_CenterSurround:
|
|
return AudioConfig::CHANNEL_RCENTER;
|
|
case kAudioChannelLabel_RearSurroundLeft:
|
|
return AudioConfig::CHANNEL_RLS;
|
|
case kAudioChannelLabel_RearSurroundRight:
|
|
return AudioConfig::CHANNEL_RRS;
|
|
default:
|
|
return AudioConfig::CHANNEL_INVALID;
|
|
}
|
|
}
|
|
|
|
// Will set mChannelLayout if a channel layout could properly be identified
|
|
// and is supported.
|
|
nsresult
|
|
AppleATDecoder::SetupChannelLayout()
|
|
{
|
|
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
|
|
|
|
// Determine the channel layout.
|
|
UInt32 propertySize;
|
|
UInt32 size;
|
|
OSStatus status =
|
|
AudioConverterGetPropertyInfo(mConverter,
|
|
kAudioConverterOutputChannelLayout,
|
|
&propertySize, NULL);
|
|
if (status || !propertySize) {
|
|
LOG("Couldn't get channel layout property (%s)", FourCC2Str(status));
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
auto data = MakeUnique<uint8_t[]>(propertySize);
|
|
size = propertySize;
|
|
status =
|
|
AudioConverterGetProperty(mConverter, kAudioConverterInputChannelLayout,
|
|
&size, data.get());
|
|
if (status || size != propertySize) {
|
|
LOG("Couldn't get channel layout property (%s)",
|
|
FourCC2Str(status));
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
AudioChannelLayout* layout =
|
|
reinterpret_cast<AudioChannelLayout*>(data.get());
|
|
AudioChannelLayoutTag tag = layout->mChannelLayoutTag;
|
|
|
|
// if tag is kAudioChannelLayoutTag_UseChannelDescriptions then the structure
|
|
// directly contains the the channel layout mapping.
|
|
// If tag is kAudioChannelLayoutTag_UseChannelBitmap then the layout will
|
|
// be defined via the bitmap and can be retrieved using
|
|
// kAudioFormatProperty_ChannelLayoutForBitmap property.
|
|
// Otherwise the tag itself describes the layout.
|
|
if (tag != kAudioChannelLayoutTag_UseChannelDescriptions) {
|
|
AudioFormatPropertyID property =
|
|
tag == kAudioChannelLayoutTag_UseChannelBitmap
|
|
? kAudioFormatProperty_ChannelLayoutForBitmap
|
|
: kAudioFormatProperty_ChannelLayoutForTag;
|
|
|
|
if (property == kAudioFormatProperty_ChannelLayoutForBitmap) {
|
|
status =
|
|
AudioFormatGetPropertyInfo(property,
|
|
sizeof(UInt32), &layout->mChannelBitmap,
|
|
&propertySize);
|
|
} else {
|
|
status =
|
|
AudioFormatGetPropertyInfo(property,
|
|
sizeof(AudioChannelLayoutTag), &tag,
|
|
&propertySize);
|
|
}
|
|
if (status || !propertySize) {
|
|
LOG("Couldn't get channel layout property info (%s:%s)",
|
|
FourCC2Str(property), FourCC2Str(status));
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
data = MakeUnique<uint8_t[]>(propertySize);
|
|
layout = reinterpret_cast<AudioChannelLayout*>(data.get());
|
|
size = propertySize;
|
|
|
|
if (property == kAudioFormatProperty_ChannelLayoutForBitmap) {
|
|
status = AudioFormatGetProperty(property,
|
|
sizeof(UInt32), &layout->mChannelBitmap,
|
|
&size, layout);
|
|
} else {
|
|
status = AudioFormatGetProperty(property,
|
|
sizeof(AudioChannelLayoutTag), &tag,
|
|
&size, layout);
|
|
}
|
|
if (status || size != propertySize) {
|
|
LOG("Couldn't get channel layout property (%s:%s)",
|
|
FourCC2Str(property), FourCC2Str(status));
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
// We have retrieved the channel layout from the tag or bitmap.
|
|
// We can now directly use the channel descriptions.
|
|
layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
|
|
}
|
|
|
|
if (layout->mNumberChannelDescriptions > MAX_AUDIO_CHANNELS ||
|
|
layout->mNumberChannelDescriptions != mOutputFormat.mChannelsPerFrame) {
|
|
LOG("Nonsensical channel layout or not matching the original channel number");
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
AudioConfig::Channel channels[MAX_AUDIO_CHANNELS];
|
|
for (uint32_t i = 0; i < layout->mNumberChannelDescriptions; i++) {
|
|
AudioChannelLabel id = layout->mChannelDescriptions[i].mChannelLabel;
|
|
AudioConfig::Channel channel = ConvertChannelLabel(id);
|
|
channels[i] = channel;
|
|
}
|
|
mChannelLayout =
|
|
MakeUnique<AudioConfig::ChannelLayout>(mOutputFormat.mChannelsPerFrame,
|
|
channels);
|
|
return NS_OK;
|
|
}
|
|
|
|
nsresult
|
|
AppleATDecoder::SetupDecoder(MediaRawData* aSample)
|
|
{
|
|
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
|
|
|
|
if (mFormatID == kAudioFormatMPEG4AAC &&
|
|
mConfig.mExtendedProfile == 2) {
|
|
// Check for implicit SBR signalling if stream is AAC-LC
|
|
// This will provide us with an updated magic cookie for use with
|
|
// GetInputAudioDescription.
|
|
if (NS_SUCCEEDED(GetImplicitAACMagicCookie(aSample)) &&
|
|
!mMagicCookie.Length()) {
|
|
// nothing found yet, will try again later
|
|
return NS_ERROR_NOT_INITIALIZED;
|
|
}
|
|
// An error occurred, fallback to using default stream description
|
|
}
|
|
|
|
LOG("Initializing Apple AudioToolbox decoder");
|
|
|
|
AudioStreamBasicDescription inputFormat;
|
|
PodZero(&inputFormat);
|
|
nsresult rv =
|
|
GetInputAudioDescription(inputFormat,
|
|
mMagicCookie.Length() ?
|
|
mMagicCookie : *mConfig.mExtraData);
|
|
if (NS_FAILED(rv)) {
|
|
return rv;
|
|
}
|
|
// Fill in the output format manually.
|
|
PodZero(&mOutputFormat);
|
|
mOutputFormat.mFormatID = kAudioFormatLinearPCM;
|
|
mOutputFormat.mSampleRate = inputFormat.mSampleRate;
|
|
mOutputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
|
|
#if defined(MOZ_SAMPLE_TYPE_FLOAT32)
|
|
mOutputFormat.mBitsPerChannel = 32;
|
|
mOutputFormat.mFormatFlags =
|
|
kLinearPCMFormatFlagIsFloat |
|
|
0;
|
|
#elif defined(MOZ_SAMPLE_TYPE_S16)
|
|
mOutputFormat.mBitsPerChannel = 16;
|
|
mOutputFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | 0;
|
|
#else
|
|
# error Unknown audio sample type
|
|
#endif
|
|
// Set up the decoder so it gives us one sample per frame
|
|
mOutputFormat.mFramesPerPacket = 1;
|
|
mOutputFormat.mBytesPerPacket = mOutputFormat.mBytesPerFrame
|
|
= mOutputFormat.mChannelsPerFrame * mOutputFormat.mBitsPerChannel / 8;
|
|
|
|
OSStatus status = AudioConverterNew(&inputFormat, &mOutputFormat, &mConverter);
|
|
if (status) {
|
|
LOG("Error %d constructing AudioConverter", status);
|
|
mConverter = nullptr;
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
|
|
if (NS_FAILED(SetupChannelLayout())) {
|
|
NS_WARNING("Couldn't retrieve channel layout, will use default layout");
|
|
}
|
|
|
|
return NS_OK;
|
|
}
|
|
|
|
static void
|
|
_MetadataCallback(void* aAppleATDecoder,
|
|
AudioFileStreamID aStream,
|
|
AudioFileStreamPropertyID aProperty,
|
|
UInt32* aFlags)
|
|
{
|
|
AppleATDecoder* decoder = static_cast<AppleATDecoder*>(aAppleATDecoder);
|
|
LOG("MetadataCallback receiving: '%s'", FourCC2Str(aProperty));
|
|
if (aProperty == kAudioFileStreamProperty_MagicCookieData) {
|
|
UInt32 size;
|
|
Boolean writeable;
|
|
OSStatus rv = AudioFileStreamGetPropertyInfo(aStream,
|
|
aProperty,
|
|
&size,
|
|
&writeable);
|
|
if (rv) {
|
|
LOG("Couldn't get property info for '%s' (%s)",
|
|
FourCC2Str(aProperty), FourCC2Str(rv));
|
|
decoder->mFileStreamError = true;
|
|
return;
|
|
}
|
|
auto data = MakeUnique<uint8_t[]>(size);
|
|
rv = AudioFileStreamGetProperty(aStream, aProperty,
|
|
&size, data.get());
|
|
if (rv) {
|
|
LOG("Couldn't get property '%s' (%s)",
|
|
FourCC2Str(aProperty), FourCC2Str(rv));
|
|
decoder->mFileStreamError = true;
|
|
return;
|
|
}
|
|
decoder->mMagicCookie.AppendElements(data.get(), size);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_SampleCallback(void* aSBR,
|
|
UInt32 aNumBytes,
|
|
UInt32 aNumPackets,
|
|
const void* aData,
|
|
AudioStreamPacketDescription* aPackets)
|
|
{
|
|
}
|
|
|
|
nsresult
|
|
AppleATDecoder::GetImplicitAACMagicCookie(const MediaRawData* aSample)
|
|
{
|
|
MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
|
|
|
|
// Prepend ADTS header to AAC audio.
|
|
RefPtr<MediaRawData> adtssample(aSample->Clone());
|
|
if (!adtssample) {
|
|
return NS_ERROR_OUT_OF_MEMORY;
|
|
}
|
|
int8_t frequency_index =
|
|
mp4_demuxer::Adts::GetFrequencyIndex(mConfig.mRate);
|
|
|
|
bool rv = mp4_demuxer::Adts::ConvertSample(mConfig.mChannels,
|
|
frequency_index,
|
|
mConfig.mProfile,
|
|
adtssample);
|
|
if (!rv) {
|
|
NS_WARNING("Failed to apply ADTS header");
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
if (!mStream) {
|
|
OSStatus rv = AudioFileStreamOpen(this,
|
|
_MetadataCallback,
|
|
_SampleCallback,
|
|
kAudioFileAAC_ADTSType,
|
|
&mStream);
|
|
if (rv) {
|
|
NS_WARNING("Couldn't open AudioFileStream");
|
|
return NS_ERROR_FAILURE;
|
|
}
|
|
}
|
|
|
|
OSStatus status = AudioFileStreamParseBytes(mStream,
|
|
adtssample->Size(),
|
|
adtssample->Data(),
|
|
0 /* discontinuity */);
|
|
if (status) {
|
|
NS_WARNING("Couldn't parse sample");
|
|
}
|
|
|
|
if (status || mFileStreamError || mMagicCookie.Length()) {
|
|
// We have decoded a magic cookie or an error occurred as such
|
|
// we won't need the stream any longer.
|
|
AudioFileStreamClose(mStream);
|
|
mStream = nullptr;
|
|
}
|
|
|
|
return (mFileStreamError || status) ? NS_ERROR_FAILURE : NS_OK;
|
|
}
|
|
|
|
} // namespace mozilla
|