/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */ /* vim:set ts=2 sw=2 sts=2 et cindent: */ /* 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 "AudioSink.h" #include "MediaDecoderStateMachine.h" #include "AudioStream.h" #include "prenv.h" namespace mozilla { extern PRLogModuleInfo* gMediaDecoderLog; #define SINK_LOG(msg, ...) \ MOZ_LOG(gMediaDecoderLog, PR_LOG_DEBUG, ("AudioSink=%p " msg, this, ##__VA_ARGS__)) #define SINK_LOG_V(msg, ...) \ MOZ_LOG(gMediaDecoderLog, PR_LOG_DEBUG+1, ("AudioSink=%p " msg, this, ##__VA_ARGS__)) AudioSink::OnAudioEndTimeUpdateTask::OnAudioEndTimeUpdateTask( MediaDecoderStateMachine* aStateMachine) : mMutex("OnAudioEndTimeUpdateTask") , mEndTime(0) , mStateMachine(aStateMachine) { } NS_IMETHODIMP AudioSink::OnAudioEndTimeUpdateTask::Run() { MutexAutoLock lock(mMutex); if (mStateMachine) { mStateMachine->OnAudioEndTimeUpdate(mEndTime); } return NS_OK; } void AudioSink::OnAudioEndTimeUpdateTask::Dispatch(int64_t aEndTime) { MutexAutoLock lock(mMutex); if (mStateMachine) { mEndTime = aEndTime; nsRefPtr runnable(this); mStateMachine->TaskQueue()->Dispatch(runnable.forget()); } } void AudioSink::OnAudioEndTimeUpdateTask::Cancel() { MutexAutoLock lock(mMutex); mStateMachine = nullptr; } // The amount of audio frames that is used to fuzz rounding errors. static const int64_t AUDIO_FUZZ_FRAMES = 1; AudioSink::AudioSink(MediaDecoderStateMachine* aStateMachine, int64_t aStartTime, AudioInfo aInfo, dom::AudioChannel aChannel) : mStateMachine(aStateMachine) , mStartTime(aStartTime) , mWritten(0) , mLastGoodPosition(0) , mInfo(aInfo) , mChannel(aChannel) , mVolume(1.0) , mPlaybackRate(1.0) , mPreservesPitch(false) , mStopAudioThread(false) , mSetVolume(false) , mSetPlaybackRate(false) , mSetPreservesPitch(false) , mPlaying(true) { NS_ASSERTION(mStartTime != -1, "Should have audio start time by now"); mOnAudioEndTimeUpdateTask = new OnAudioEndTimeUpdateTask(aStateMachine); } nsresult AudioSink::Init() { nsresult rv = NS_NewNamedThread("Media Audio", getter_AddRefs(mThread), nullptr, MEDIA_THREAD_STACK_SIZE); if (NS_FAILED(rv)) { mStateMachine->OnAudioSinkError(); return rv; } nsCOMPtr event = NS_NewRunnableMethod(this, &AudioSink::AudioLoop); rv = mThread->Dispatch(event, NS_DISPATCH_NORMAL); if (NS_FAILED(rv)) { mStateMachine->OnAudioSinkError(); return rv; } return NS_OK; } int64_t AudioSink::GetPosition() { AssertCurrentThreadInMonitor(); int64_t pos; if (mAudioStream && (pos = mAudioStream->GetPosition()) >= 0) { // Update the last good position when we got a good one. mLastGoodPosition = pos; } return mLastGoodPosition; } bool AudioSink::HasUnplayedFrames() { AssertCurrentThreadInMonitor(); // Experimentation suggests that GetPositionInFrames() is zero-indexed, // so we need to add 1 here before comparing it to mWritten. return mAudioStream && mAudioStream->GetPositionInFrames() + 1 < mWritten; } void AudioSink::PrepareToShutdown() { AssertCurrentThreadInMonitor(); mStopAudioThread = true; if (mAudioStream) { mAudioStream->Cancel(); } GetReentrantMonitor().NotifyAll(); } void AudioSink::Shutdown() { mOnAudioEndTimeUpdateTask->Cancel(); mThread->Shutdown(); mThread = nullptr; MOZ_ASSERT(!mAudioStream); } void AudioSink::SetVolume(double aVolume) { AssertCurrentThreadInMonitor(); mVolume = aVolume; mSetVolume = true; } void AudioSink::SetPlaybackRate(double aPlaybackRate) { AssertCurrentThreadInMonitor(); NS_ASSERTION(mPlaybackRate != 0, "Don't set the playbackRate to 0 on AudioStream"); mPlaybackRate = aPlaybackRate; mSetPlaybackRate = true; } void AudioSink::SetPreservesPitch(bool aPreservesPitch) { AssertCurrentThreadInMonitor(); mPreservesPitch = aPreservesPitch; mSetPreservesPitch = true; } void AudioSink::StartPlayback() { AssertCurrentThreadInMonitor(); mPlaying = true; GetReentrantMonitor().NotifyAll(); } void AudioSink::StopPlayback() { AssertCurrentThreadInMonitor(); mPlaying = false; GetReentrantMonitor().NotifyAll(); } void AudioSink::AudioLoop() { AssertOnAudioThread(); SINK_LOG("AudioLoop started"); if (NS_FAILED(InitializeAudioStream())) { NS_WARNING("Initializing AudioStream failed."); mStateMachine->DispatchOnAudioSinkError(); return; } while (1) { { ReentrantMonitorAutoEnter mon(GetReentrantMonitor()); WaitForAudioToPlay(); if (!IsPlaybackContinuing()) { break; } } // See if there's a gap in the audio. If there is, push silence into the // audio hardware, so we can play across the gap. // Calculate the timestamp of the next chunk of audio in numbers of // samples. NS_ASSERTION(AudioQueue().GetSize() > 0, "Should have data to play"); CheckedInt64 sampleTime = UsecsToFrames(AudioQueue().PeekFront()->mTime, mInfo.mRate); // Calculate the number of frames that have been pushed onto the audio hardware. CheckedInt64 playedFrames = UsecsToFrames(mStartTime, mInfo.mRate) + mWritten; CheckedInt64 missingFrames = sampleTime - playedFrames; if (!missingFrames.isValid() || !sampleTime.isValid()) { NS_WARNING("Int overflow adding in AudioLoop"); break; } if (missingFrames.value() > AUDIO_FUZZ_FRAMES) { // The next audio chunk begins some time after the end of the last chunk // we pushed to the audio hardware. We must push silence into the audio // hardware so that the next audio chunk begins playback at the correct // time. missingFrames = std::min(UINT32_MAX, missingFrames.value()); mWritten += PlaySilence(static_cast(missingFrames.value())); } else { mWritten += PlayFromAudioQueue(); } int64_t endTime = GetEndTime(); if (endTime != -1) { mOnAudioEndTimeUpdateTask->Dispatch(endTime); } } ReentrantMonitorAutoEnter mon(GetReentrantMonitor()); MOZ_ASSERT(mStopAudioThread || AudioQueue().AtEndOfStream()); if (!mStopAudioThread && mPlaying) { Drain(); } SINK_LOG("AudioLoop complete"); Cleanup(); SINK_LOG("AudioLoop exit"); } nsresult AudioSink::InitializeAudioStream() { // AudioStream initialization can block for extended periods in unusual // circumstances, so we take care to drop the decoder monitor while // initializing. RefPtr audioStream(new AudioStream()); nsresult rv = audioStream->Init(mInfo.mChannels, mInfo.mRate, mChannel, AudioStream::HighLatency); if (NS_FAILED(rv)) { audioStream->Shutdown(); return rv; } ReentrantMonitorAutoEnter mon(GetReentrantMonitor()); mAudioStream = audioStream; UpdateStreamSettings(); return NS_OK; } void AudioSink::Drain() { MOZ_ASSERT(mPlaying && !mAudioStream->IsPaused()); AssertCurrentThreadInMonitor(); // If the media was too short to trigger the start of the audio stream, // start it now. mAudioStream->Start(); { ReentrantMonitorAutoExit exit(GetReentrantMonitor()); mAudioStream->Drain(); } } void AudioSink::Cleanup() { AssertCurrentThreadInMonitor(); nsRefPtr audioStream; audioStream.swap(mAudioStream); // Suppress the callback when the stop is requested by MediaDecoderStateMachine. // See Bug 115334. if (!mStopAudioThread) { mStateMachine->DispatchOnAudioSinkComplete(); } ReentrantMonitorAutoExit exit(GetReentrantMonitor()); audioStream->Shutdown(); } bool AudioSink::ExpectMoreAudioData() { return AudioQueue().GetSize() == 0 && !AudioQueue().IsFinished(); } void AudioSink::WaitForAudioToPlay() { // Wait while we're not playing, and we're not shutting down, or we're // playing and we've got no audio to play. AssertCurrentThreadInMonitor(); while (!mStopAudioThread && (!mPlaying || ExpectMoreAudioData())) { if (!mPlaying && !mAudioStream->IsPaused()) { mAudioStream->Pause(); } GetReentrantMonitor().Wait(); } } bool AudioSink::IsPlaybackContinuing() { AssertCurrentThreadInMonitor(); if (mPlaying && mAudioStream->IsPaused()) { mAudioStream->Resume(); } // If we're shutting down, captured, or at EOS, break out and exit the audio // thread. if (mStopAudioThread || AudioQueue().AtEndOfStream()) { return false; } UpdateStreamSettings(); return true; } uint32_t AudioSink::PlaySilence(uint32_t aFrames) { // Maximum number of bytes we'll allocate and write at once to the audio // hardware when the audio stream contains missing frames and we're // writing silence in order to fill the gap. We limit our silence-writes // to 32KB in order to avoid allocating an impossibly large chunk of // memory if we encounter a large chunk of silence. const uint32_t SILENCE_BYTES_CHUNK = 32 * 1024; AssertOnAudioThread(); NS_ASSERTION(!mAudioStream->IsPaused(), "Don't play when paused"); uint32_t maxFrames = SILENCE_BYTES_CHUNK / mInfo.mChannels / sizeof(AudioDataValue); uint32_t frames = std::min(aFrames, maxFrames); SINK_LOG_V("playing %u frames of silence", aFrames); WriteSilence(frames); return frames; } uint32_t AudioSink::PlayFromAudioQueue() { AssertOnAudioThread(); NS_ASSERTION(!mAudioStream->IsPaused(), "Don't play when paused"); nsRefPtr audio(AudioQueue().PopFront()); SINK_LOG_V("playing %u frames of audio at time %lld", audio->mFrames, audio->mTime); mAudioStream->Write(audio->mAudioData, audio->mFrames); StartAudioStreamPlaybackIfNeeded(); if (audio->mOffset != -1) { mStateMachine->DispatchOnPlaybackOffsetUpdate(audio->mOffset); } return audio->mFrames; } void AudioSink::UpdateStreamSettings() { AssertCurrentThreadInMonitor(); bool setVolume = mSetVolume; bool setPlaybackRate = mSetPlaybackRate; bool setPreservesPitch = mSetPreservesPitch; double volume = mVolume; double playbackRate = mPlaybackRate; bool preservesPitch = mPreservesPitch; mSetVolume = false; mSetPlaybackRate = false; mSetPreservesPitch = false; { ReentrantMonitorAutoExit exit(GetReentrantMonitor()); if (setVolume) { mAudioStream->SetVolume(volume); } if (setPlaybackRate && NS_FAILED(mAudioStream->SetPlaybackRate(playbackRate))) { NS_WARNING("Setting the playback rate failed in AudioSink."); } if (setPreservesPitch && NS_FAILED(mAudioStream->SetPreservesPitch(preservesPitch))) { NS_WARNING("Setting the pitch preservation failed in AudioSink."); } } } void AudioSink::StartAudioStreamPlaybackIfNeeded() { // This value has been chosen empirically. const uint32_t MIN_WRITE_BEFORE_START_USECS = 200000; // We want to have enough data in the buffer to start the stream. if (static_cast(mAudioStream->GetWritten()) / mAudioStream->GetRate() >= static_cast(MIN_WRITE_BEFORE_START_USECS) / USECS_PER_S) { mAudioStream->Start(); } } void AudioSink::WriteSilence(uint32_t aFrames) { uint32_t numSamples = aFrames * mInfo.mChannels; nsAutoTArray buf; buf.SetLength(numSamples); memset(buf.Elements(), 0, numSamples * sizeof(AudioDataValue)); mAudioStream->Write(buf.Elements(), aFrames); StartAudioStreamPlaybackIfNeeded(); } int64_t AudioSink::GetEndTime() { CheckedInt64 playedUsecs = FramesToUsecs(mWritten, mInfo.mRate) + mStartTime; if (!playedUsecs.isValid()) { NS_WARNING("Int overflow calculating audio end time"); return -1; } return playedUsecs.value(); } MediaQueue& AudioSink::AudioQueue() { return mStateMachine->AudioQueue(); } ReentrantMonitor& AudioSink::GetReentrantMonitor() { return mStateMachine->mDecoder->GetReentrantMonitor(); } void AudioSink::AssertCurrentThreadInMonitor() { return mStateMachine->AssertCurrentThreadInMonitor(); } void AudioSink::AssertOnAudioThread() { MOZ_ASSERT(IsCurrentThread(mThread)); } } // namespace mozilla