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https://github.com/mozilla/gecko-dev.git
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437 lines
11 KiB
C++
437 lines
11 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 "AudioSink.h"
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#include "MediaDecoderStateMachine.h"
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#include "AudioStream.h"
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#include "prenv.h"
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namespace mozilla {
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#ifdef PR_LOGGING
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extern PRLogModuleInfo* gMediaDecoderLog;
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#define SINK_LOG(msg, ...) \
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PR_LOG(gMediaDecoderLog, PR_LOG_DEBUG, ("AudioSink=%p " msg, this, ##__VA_ARGS__))
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#define SINK_LOG_V(msg, ...) \
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PR_BEGIN_MACRO \
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if (!PR_GetEnv("MOZ_QUIET")) { \
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SINK_LOG(msg, ##__VA_ARGS__); \
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} \
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PR_END_MACRO
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#else
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#define SINK_LOG(msg, ...)
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#define SINK_LOG_V(msg, ...)
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#endif
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// The amount of audio frames that is used to fuzz rounding errors.
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static const int64_t AUDIO_FUZZ_FRAMES = 1;
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AudioSink::AudioSink(MediaDecoderStateMachine* aStateMachine,
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int64_t aStartTime, AudioInfo aInfo, dom::AudioChannel aChannel)
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: mStateMachine(aStateMachine)
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, mStartTime(aStartTime)
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, mWritten(0)
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, mLastGoodPosition(0)
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, mInfo(aInfo)
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, mChannel(aChannel)
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, mVolume(1.0)
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, mPlaybackRate(1.0)
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, mPreservesPitch(false)
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, mStopAudioThread(false)
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, mSetVolume(false)
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, mSetPlaybackRate(false)
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, mSetPreservesPitch(false)
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, mPlaying(true)
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{
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NS_ASSERTION(mStartTime != -1, "Should have audio start time by now");
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}
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nsresult
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AudioSink::Init()
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{
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nsresult rv = NS_NewNamedThread("Media Audio",
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getter_AddRefs(mThread),
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nullptr,
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MEDIA_THREAD_STACK_SIZE);
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if (NS_FAILED(rv)) {
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mStateMachine->OnAudioSinkError();
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return rv;
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}
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nsCOMPtr<nsIRunnable> event = NS_NewRunnableMethod(this, &AudioSink::AudioLoop);
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rv = mThread->Dispatch(event, NS_DISPATCH_NORMAL);
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if (NS_FAILED(rv)) {
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mStateMachine->OnAudioSinkError();
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return rv;
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}
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return NS_OK;
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}
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int64_t
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AudioSink::GetPosition()
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{
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AssertCurrentThreadInMonitor();
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int64_t pos;
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if (mAudioStream &&
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(pos = mAudioStream->GetPosition()) >= 0) {
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// Update the last good position when we got a good one.
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mLastGoodPosition = pos;
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}
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return mLastGoodPosition;
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}
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bool
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AudioSink::HasUnplayedFrames()
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{
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AssertCurrentThreadInMonitor();
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// Experimentation suggests that GetPositionInFrames() is zero-indexed,
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// so we need to add 1 here before comparing it to mWritten.
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return mAudioStream && mAudioStream->GetPositionInFrames() + 1 < mWritten;
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}
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void
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AudioSink::PrepareToShutdown()
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{
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AssertCurrentThreadInMonitor();
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mStopAudioThread = true;
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if (mAudioStream) {
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mAudioStream->Cancel();
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}
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GetReentrantMonitor().NotifyAll();
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}
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void
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AudioSink::Shutdown()
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{
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mThread->Shutdown();
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mThread = nullptr;
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MOZ_ASSERT(!mAudioStream);
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}
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void
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AudioSink::SetVolume(double aVolume)
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{
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AssertCurrentThreadInMonitor();
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mVolume = aVolume;
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mSetVolume = true;
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}
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void
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AudioSink::SetPlaybackRate(double aPlaybackRate)
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{
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AssertCurrentThreadInMonitor();
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NS_ASSERTION(mPlaybackRate != 0, "Don't set the playbackRate to 0 on AudioStream");
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mPlaybackRate = aPlaybackRate;
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mSetPlaybackRate = true;
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}
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void
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AudioSink::SetPreservesPitch(bool aPreservesPitch)
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{
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AssertCurrentThreadInMonitor();
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mPreservesPitch = aPreservesPitch;
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mSetPreservesPitch = true;
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}
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void
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AudioSink::StartPlayback()
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{
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AssertCurrentThreadInMonitor();
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mPlaying = true;
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GetReentrantMonitor().NotifyAll();
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}
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void
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AudioSink::StopPlayback()
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{
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AssertCurrentThreadInMonitor();
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mPlaying = false;
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GetReentrantMonitor().NotifyAll();
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}
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void
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AudioSink::AudioLoop()
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{
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AssertOnAudioThread();
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SINK_LOG("AudioLoop started");
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if (NS_FAILED(InitializeAudioStream())) {
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NS_WARNING("Initializing AudioStream failed.");
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mStateMachine->DispatchOnAudioSinkError();
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return;
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}
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while (1) {
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{
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ReentrantMonitorAutoEnter mon(GetReentrantMonitor());
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WaitForAudioToPlay();
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if (!IsPlaybackContinuing()) {
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break;
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}
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}
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// See if there's a gap in the audio. If there is, push silence into the
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// audio hardware, so we can play across the gap.
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// Calculate the timestamp of the next chunk of audio in numbers of
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// samples.
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NS_ASSERTION(AudioQueue().GetSize() > 0, "Should have data to play");
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CheckedInt64 sampleTime = UsecsToFrames(AudioQueue().PeekFront()->mTime, mInfo.mRate);
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// Calculate the number of frames that have been pushed onto the audio hardware.
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CheckedInt64 playedFrames = UsecsToFrames(mStartTime, mInfo.mRate) + mWritten;
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CheckedInt64 missingFrames = sampleTime - playedFrames;
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if (!missingFrames.isValid() || !sampleTime.isValid()) {
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NS_WARNING("Int overflow adding in AudioLoop");
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break;
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}
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if (missingFrames.value() > AUDIO_FUZZ_FRAMES) {
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// The next audio chunk begins some time after the end of the last chunk
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// we pushed to the audio hardware. We must push silence into the audio
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// hardware so that the next audio chunk begins playback at the correct
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// time.
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missingFrames = std::min<int64_t>(UINT32_MAX, missingFrames.value());
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mWritten += PlaySilence(static_cast<uint32_t>(missingFrames.value()));
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} else {
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mWritten += PlayFromAudioQueue();
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}
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int64_t endTime = GetEndTime();
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if (endTime != -1) {
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mStateMachine->OnAudioEndTimeUpdate(endTime);
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}
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}
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ReentrantMonitorAutoEnter mon(GetReentrantMonitor());
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MOZ_ASSERT(mStopAudioThread || AudioQueue().AtEndOfStream());
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if (!mStopAudioThread && mPlaying) {
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Drain();
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}
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SINK_LOG("AudioLoop complete");
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Cleanup();
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SINK_LOG("AudioLoop exit");
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}
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nsresult
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AudioSink::InitializeAudioStream()
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{
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// AudioStream initialization can block for extended periods in unusual
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// circumstances, so we take care to drop the decoder monitor while
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// initializing.
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RefPtr<AudioStream> audioStream(new AudioStream());
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nsresult rv = audioStream->Init(mInfo.mChannels, mInfo.mRate,
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mChannel, AudioStream::HighLatency);
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if (NS_FAILED(rv)) {
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audioStream->Shutdown();
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return rv;
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}
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ReentrantMonitorAutoEnter mon(GetReentrantMonitor());
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mAudioStream = audioStream;
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UpdateStreamSettings();
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return NS_OK;
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}
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void
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AudioSink::Drain()
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{
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MOZ_ASSERT(mPlaying && !mAudioStream->IsPaused());
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AssertCurrentThreadInMonitor();
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// If the media was too short to trigger the start of the audio stream,
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// start it now.
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mAudioStream->Start();
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{
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ReentrantMonitorAutoExit exit(GetReentrantMonitor());
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mAudioStream->Drain();
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}
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}
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void
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AudioSink::Cleanup()
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{
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AssertCurrentThreadInMonitor();
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nsRefPtr<AudioStream> audioStream;
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audioStream.swap(mAudioStream);
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mStateMachine->OnAudioSinkComplete();
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ReentrantMonitorAutoExit exit(GetReentrantMonitor());
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audioStream->Shutdown();
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}
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bool
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AudioSink::ExpectMoreAudioData()
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{
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return AudioQueue().GetSize() == 0 && !AudioQueue().IsFinished();
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}
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void
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AudioSink::WaitForAudioToPlay()
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{
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// Wait while we're not playing, and we're not shutting down, or we're
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// playing and we've got no audio to play.
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AssertCurrentThreadInMonitor();
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while (!mStopAudioThread && (!mPlaying || ExpectMoreAudioData())) {
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if (!mPlaying && !mAudioStream->IsPaused()) {
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mAudioStream->Pause();
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}
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GetReentrantMonitor().Wait();
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}
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}
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bool
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AudioSink::IsPlaybackContinuing()
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{
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AssertCurrentThreadInMonitor();
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if (mPlaying && mAudioStream->IsPaused()) {
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mAudioStream->Resume();
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}
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// If we're shutting down, captured, or at EOS, break out and exit the audio
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// thread.
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if (mStopAudioThread || AudioQueue().AtEndOfStream()) {
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return false;
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}
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UpdateStreamSettings();
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return true;
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}
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uint32_t
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AudioSink::PlaySilence(uint32_t aFrames)
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{
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// Maximum number of bytes we'll allocate and write at once to the audio
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// hardware when the audio stream contains missing frames and we're
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// writing silence in order to fill the gap. We limit our silence-writes
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// to 32KB in order to avoid allocating an impossibly large chunk of
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// memory if we encounter a large chunk of silence.
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const uint32_t SILENCE_BYTES_CHUNK = 32 * 1024;
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AssertOnAudioThread();
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NS_ASSERTION(!mAudioStream->IsPaused(), "Don't play when paused");
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uint32_t maxFrames = SILENCE_BYTES_CHUNK / mInfo.mChannels / sizeof(AudioDataValue);
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uint32_t frames = std::min(aFrames, maxFrames);
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SINK_LOG_V("playing %u frames of silence", aFrames);
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WriteSilence(frames);
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return frames;
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}
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uint32_t
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AudioSink::PlayFromAudioQueue()
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{
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AssertOnAudioThread();
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NS_ASSERTION(!mAudioStream->IsPaused(), "Don't play when paused");
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nsRefPtr<AudioData> audio(AudioQueue().PopFront());
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SINK_LOG_V("playing %u frames of audio at time %lld",
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audio->mFrames, audio->mTime);
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mAudioStream->Write(audio->mAudioData, audio->mFrames);
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StartAudioStreamPlaybackIfNeeded();
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if (audio->mOffset != -1) {
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mStateMachine->OnPlaybackOffsetUpdate(audio->mOffset);
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}
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return audio->mFrames;
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}
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void
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AudioSink::UpdateStreamSettings()
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{
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AssertCurrentThreadInMonitor();
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bool setVolume = mSetVolume;
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bool setPlaybackRate = mSetPlaybackRate;
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bool setPreservesPitch = mSetPreservesPitch;
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double volume = mVolume;
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double playbackRate = mPlaybackRate;
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bool preservesPitch = mPreservesPitch;
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mSetVolume = false;
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mSetPlaybackRate = false;
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mSetPreservesPitch = false;
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{
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ReentrantMonitorAutoExit exit(GetReentrantMonitor());
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if (setVolume) {
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mAudioStream->SetVolume(volume);
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}
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if (setPlaybackRate &&
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NS_FAILED(mAudioStream->SetPlaybackRate(playbackRate))) {
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NS_WARNING("Setting the playback rate failed in AudioSink.");
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}
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if (setPreservesPitch &&
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NS_FAILED(mAudioStream->SetPreservesPitch(preservesPitch))) {
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NS_WARNING("Setting the pitch preservation failed in AudioSink.");
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}
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}
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}
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void
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AudioSink::StartAudioStreamPlaybackIfNeeded()
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{
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// This value has been chosen empirically.
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const uint32_t MIN_WRITE_BEFORE_START_USECS = 200000;
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// We want to have enough data in the buffer to start the stream.
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if (static_cast<double>(mAudioStream->GetWritten()) / mAudioStream->GetRate() >=
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static_cast<double>(MIN_WRITE_BEFORE_START_USECS) / USECS_PER_S) {
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mAudioStream->Start();
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}
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}
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void
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AudioSink::WriteSilence(uint32_t aFrames)
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{
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uint32_t numSamples = aFrames * mInfo.mChannels;
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nsAutoTArray<AudioDataValue, 1000> buf;
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buf.SetLength(numSamples);
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memset(buf.Elements(), 0, numSamples * sizeof(AudioDataValue));
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mAudioStream->Write(buf.Elements(), aFrames);
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StartAudioStreamPlaybackIfNeeded();
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}
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int64_t
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AudioSink::GetEndTime()
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{
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CheckedInt64 playedUsecs = FramesToUsecs(mWritten, mInfo.mRate) + mStartTime;
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if (!playedUsecs.isValid()) {
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NS_WARNING("Int overflow calculating audio end time");
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return -1;
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}
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return playedUsecs.value();
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}
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MediaQueue<AudioData>&
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AudioSink::AudioQueue()
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{
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return mStateMachine->AudioQueue();
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}
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ReentrantMonitor&
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AudioSink::GetReentrantMonitor()
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{
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return mStateMachine->mDecoder->GetReentrantMonitor();
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}
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void
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AudioSink::AssertCurrentThreadInMonitor()
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{
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return mStateMachine->AssertCurrentThreadInMonitor();
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}
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void
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AudioSink::AssertOnAudioThread()
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{
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MOZ_ASSERT(IsCurrentThread(mThread));
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}
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} // namespace mozilla
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