gecko-dev/dom/media/AudioSink.cpp

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11 KiB
C++
Исходник Обычный вид История

/* -*- 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 {
#ifdef PR_LOGGING
extern PRLogModuleInfo* gMediaDecoderLog;
#define SINK_LOG(msg, ...) \
PR_LOG(gMediaDecoderLog, PR_LOG_DEBUG, ("AudioSink=%p " msg, this, ##__VA_ARGS__))
#define SINK_LOG_V(msg, ...) \
PR_BEGIN_MACRO \
if (!PR_GetEnv("MOZ_QUIET")) { \
SINK_LOG(msg, ##__VA_ARGS__); \
} \
PR_END_MACRO
#else
#define SINK_LOG(msg, ...)
#define SINK_LOG_V(msg, ...)
#endif
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// 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");
}
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<nsIRunnable> 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()
{
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;
}
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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<int64_t>(UINT32_MAX, missingFrames.value());
mWritten += PlaySilence(static_cast<uint32_t>(missingFrames.value()));
} else {
mWritten += PlayFromAudioQueue();
}
int64_t endTime = GetEndTime();
if (endTime != -1) {
mStateMachine->DispatchOnAudioEndTimeUpdate(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> 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;
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<AudioData> 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<double>(mAudioStream->GetWritten()) / mAudioStream->GetRate() >=
static_cast<double>(MIN_WRITE_BEFORE_START_USECS) / USECS_PER_S) {
mAudioStream->Start();
}
}
void
AudioSink::WriteSilence(uint32_t aFrames)
{
uint32_t numSamples = aFrames * mInfo.mChannels;
nsAutoTArray<AudioDataValue, 1000> 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<AudioData>&
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