mirror of
https://github.com/mozilla/gecko-dev.git
synced 2024-10-30 05:35:31 +00:00
428 lines
11 KiB
C++
428 lines
11 KiB
C++
/* -*- 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
|
|
|
|
// 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;
|
|
}
|
|
|
|
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.");
|
|
ReentrantMonitorAutoEnter mon(GetReentrantMonitor());
|
|
mStateMachine->OnAudioSinkError();
|
|
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<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->OnAudioEndTimeUpdate(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()
|
|
{
|
|
// Must hold lock while shutting down and anulling the audio stream to prevent
|
|
// state machine thread trying to use it while we're destroying it.
|
|
AssertCurrentThreadInMonitor();
|
|
mAudioStream->Shutdown();
|
|
mAudioStream = nullptr;
|
|
mStateMachine->OnAudioSinkComplete();
|
|
}
|
|
|
|
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->OnPlaybackOffsetUpdate(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
|