gecko-dev/dom/media/mediasink/DecodedAudioDataSink.cpp

280 lines
8.4 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 "MediaQueue.h"
#include "DecodedAudioDataSink.h"
#include "VideoUtils.h"
#include "mozilla/CheckedInt.h"
#include "mozilla/DebugOnly.h"
namespace mozilla {
extern LazyLogModule gMediaDecoderLog;
#define SINK_LOG(msg, ...) \
MOZ_LOG(gMediaDecoderLog, LogLevel::Debug, \
("DecodedAudioDataSink=%p " msg, this, ##__VA_ARGS__))
#define SINK_LOG_V(msg, ...) \
MOZ_LOG(gMediaDecoderLog, LogLevel::Verbose, \
("DecodedAudioDataSink=%p " msg, this, ##__VA_ARGS__))
namespace media {
// The amount of audio frames that is used to fuzz rounding errors.
static const int64_t AUDIO_FUZZ_FRAMES = 1;
DecodedAudioDataSink::DecodedAudioDataSink(MediaQueue<MediaData>& aAudioQueue,
int64_t aStartTime,
const AudioInfo& aInfo,
dom::AudioChannel aChannel)
: AudioSink(aAudioQueue)
, mStartTime(aStartTime)
, mWritten(0)
, mLastGoodPosition(0)
, mInfo(aInfo)
, mChannel(aChannel)
, mPlaying(true)
{
}
DecodedAudioDataSink::~DecodedAudioDataSink()
{
}
RefPtr<GenericPromise>
DecodedAudioDataSink::Init(const PlaybackParams& aParams)
{
RefPtr<GenericPromise> p = mEndPromise.Ensure(__func__);
nsresult rv = InitializeAudioStream(aParams);
if (NS_FAILED(rv)) {
mEndPromise.Reject(rv, __func__);
}
return p;
}
int64_t
DecodedAudioDataSink::GetPosition()
{
int64_t pos;
if (mAudioStream &&
(pos = mAudioStream->GetPosition()) >= 0) {
NS_ASSERTION(pos >= mLastGoodPosition,
"AudioStream position shouldn't go backward");
// Update the last good position when we got a good one.
if (pos >= mLastGoodPosition) {
mLastGoodPosition = pos;
}
}
return mStartTime + mLastGoodPosition;
}
bool
DecodedAudioDataSink::HasUnplayedFrames()
{
// 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
DecodedAudioDataSink::Shutdown()
{
if (mAudioStream) {
mAudioStream->Shutdown();
mAudioStream = nullptr;
}
mEndPromise.ResolveIfExists(true, __func__);
}
void
DecodedAudioDataSink::SetVolume(double aVolume)
{
if (mAudioStream) {
mAudioStream->SetVolume(aVolume);
}
}
void
DecodedAudioDataSink::SetPlaybackRate(double aPlaybackRate)
{
MOZ_ASSERT(aPlaybackRate != 0, "Don't set the playbackRate to 0 on AudioStream");
if (mAudioStream) {
mAudioStream->SetPlaybackRate(aPlaybackRate);
}
}
void
DecodedAudioDataSink::SetPreservesPitch(bool aPreservesPitch)
{
if (mAudioStream) {
mAudioStream->SetPreservesPitch(aPreservesPitch);
}
}
void
DecodedAudioDataSink::SetPlaying(bool aPlaying)
{
if (!mAudioStream || mPlaying == aPlaying) {
return;
}
// pause/resume AudioStream as necessary.
if (!aPlaying && !mAudioStream->IsPaused()) {
mAudioStream->Pause();
} else if (aPlaying && mAudioStream->IsPaused()) {
mAudioStream->Resume();
}
mPlaying = aPlaying;
}
nsresult
DecodedAudioDataSink::InitializeAudioStream(const PlaybackParams& aParams)
{
mAudioStream = new AudioStream(*this);
nsresult rv = mAudioStream->Init(mInfo.mChannels, mInfo.mRate, mChannel);
if (NS_FAILED(rv)) {
mAudioStream->Shutdown();
mAudioStream = nullptr;
return rv;
}
// Set playback params before calling Start() so they can take effect
// as soon as the 1st DataCallback of the AudioStream fires.
mAudioStream->SetVolume(aParams.mVolume);
mAudioStream->SetPlaybackRate(aParams.mPlaybackRate);
mAudioStream->SetPreservesPitch(aParams.mPreservesPitch);
mAudioStream->Start();
return NS_OK;
}
int64_t
DecodedAudioDataSink::GetEndTime() const
{
CheckedInt64 playedUsecs = FramesToUsecs(mWritten, mInfo.mRate) + mStartTime;
if (!playedUsecs.isValid()) {
NS_WARNING("Int overflow calculating audio end time");
return -1;
}
return playedUsecs.value();
}
UniquePtr<AudioStream::Chunk>
DecodedAudioDataSink::PopFrames(uint32_t aFrames)
{
class Chunk : public AudioStream::Chunk {
public:
Chunk(AudioData* aBuffer, uint32_t aFrames, AudioDataValue* aData)
: mBuffer(aBuffer), mFrames(aFrames), mData(aData) {}
Chunk() : mFrames(0), mData(nullptr) {}
const AudioDataValue* Data() const { return mData; }
uint32_t Frames() const { return mFrames; }
AudioDataValue* GetWritable() const { return mData; }
private:
const RefPtr<AudioData> mBuffer;
const uint32_t mFrames;
AudioDataValue* const mData;
};
class SilentChunk : public AudioStream::Chunk {
public:
SilentChunk(uint32_t aFrames, uint32_t aChannels)
: mFrames(aFrames)
, mData(MakeUnique<AudioDataValue[]>(aChannels * aFrames)) {
memset(mData.get(), 0, aChannels * aFrames * sizeof(AudioDataValue));
}
const AudioDataValue* Data() const { return mData.get(); }
uint32_t Frames() const { return mFrames; }
AudioDataValue* GetWritable() const { return mData.get(); }
private:
const uint32_t mFrames;
UniquePtr<AudioDataValue[]> mData;
};
if (!mCurrentData) {
// No data in the queue. Return an empty chunk.
if (AudioQueue().GetSize() == 0) {
return MakeUnique<Chunk>();
}
// 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.
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) +
static_cast<int64_t>(mWritten);
CheckedInt64 missingFrames = sampleTime - playedFrames;
if (!missingFrames.isValid() || !sampleTime.isValid()) {
NS_WARNING("Int overflow in DecodedAudioDataSink");
mErrored = true;
return MakeUnique<Chunk>();
}
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());
auto framesToPop = std::min<uint32_t>(missingFrames.value(), aFrames);
mWritten += framesToPop;
return MakeUnique<SilentChunk>(framesToPop, mInfo.mChannels);
}
mCurrentData = dont_AddRef(AudioQueue().PopFront().take()->As<AudioData>());
mCursor = MakeUnique<AudioBufferCursor>(mCurrentData->mAudioData.get(),
mCurrentData->mChannels,
mCurrentData->mFrames);
}
auto framesToPop = std::min(aFrames, mCursor->Available());
SINK_LOG_V("playing audio at time=%lld offset=%u length=%u",
mCurrentData->mTime, mCurrentData->mFrames - mCursor->Available(), framesToPop);
UniquePtr<AudioStream::Chunk> chunk;
if (mCurrentData->mRate == mInfo.mRate &&
mCurrentData->mChannels == mInfo.mChannels) {
chunk = MakeUnique<Chunk>(mCurrentData, framesToPop, mCursor->Ptr());
} else {
SINK_LOG_V("mismatched sample format mInfo=[%uHz/%u channels] audio=[%uHz/%u channels]",
mInfo.mRate, mInfo.mChannels, mCurrentData->mRate, mCurrentData->mChannels);
chunk = MakeUnique<SilentChunk>(framesToPop, mInfo.mChannels);
}
mWritten += framesToPop;
mCursor->Advance(framesToPop);
// All frames are popped. Reset mCurrentData so we can pop new elements from
// the audio queue in next calls to PopFrames().
if (mCursor->Available() == 0) {
mCurrentData = nullptr;
}
return chunk;
}
bool
DecodedAudioDataSink::Ended() const
{
// Return true when error encountered so AudioStream can start draining.
return AudioQueue().IsFinished() || mErrored;
}
void
DecodedAudioDataSink::Drained()
{
SINK_LOG("Drained");
mEndPromise.Resolve(true, __func__);
}
} // namespace media
} // namespace mozilla