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/* 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
#ifndef AudioDecoderInputTrack_h
#define AudioDecoderInputTrack_h
#include "AudioSegment.h"
#include "MediaEventSource.h"
#include "MediaTimer.h"
#include "MediaTrackGraph.h"
#include "MediaSegment.h"
#include "TimeUnits.h"
#include "mozilla/SPSCQueue.h"
#include "mozilla/StateMirroring.h"
#include "nsISerialEventTarget.h"
namespace mozilla {
class AudioData;
class AudioInfo;
class RLBoxSoundTouch;
/**
* AudioDecoderInputTrack is used as a source for the audio decoder data, which
* supports adjusting playback rate and preserve pitch.
* The owner of this track would be responsible to push audio data via
* `AppendData()` into a SPSC queue, which is a thread-safe queue between the
* decoder thread (producer) and the graph thread (consumer). MediaTrackGraph
* requires data via `ProcessInput()`, then AudioDecoderInputTrack would convert
* (based on sample rate and playback rate) and append the amount of needed
* audio frames onto the output segment that would be used by MediaTrackGraph.
*/
class AudioDecoderInputTrack final : public ProcessedMediaTrack {
public:
static AudioDecoderInputTrack* Create(MediaTrackGraph* aGraph,
nsISerialEventTarget* aDecoderThread,
const AudioInfo& aInfo,
float aPlaybackRate, float aVolume,
bool aPreservesPitch);
// SPSCData suppports filling different supported type variants, and is used
// to achieve a thread-safe information exchange between the decoder thread
// and the graph thread.
struct SPSCData final {
struct Empty {};
struct ClearFutureData {};
struct DecodedData {
DecodedData()
: mStartTime(media::TimeUnit::Invalid()),
mEndTime(media::TimeUnit::Invalid()) {}
DecodedData(DecodedData&& aDecodedData)
: mSegment(std::move(aDecodedData.mSegment)) {
mStartTime = aDecodedData.mStartTime;
mEndTime = aDecodedData.mEndTime;
aDecodedData.Clear();
}
DecodedData(media::TimeUnit aStartTime, media::TimeUnit aEndTime)
: mStartTime(aStartTime), mEndTime(aEndTime) {}
DecodedData(const DecodedData&) = delete;
DecodedData& operator=(const DecodedData&) = delete;
void Clear() {
mSegment.Clear();
mStartTime = media::TimeUnit::Invalid();
mEndTime = media::TimeUnit::Invalid();
}
AudioSegment mSegment;
media::TimeUnit mStartTime;
media::TimeUnit mEndTime;
};
struct EOS {};
SPSCData() : mData(Empty()){};
explicit SPSCData(ClearFutureData&& aArg) : mData(std::move(aArg)){};
explicit SPSCData(DecodedData&& aArg) : mData(std::move(aArg)){};
explicit SPSCData(EOS&& aArg) : mData(std::move(aArg)){};
bool HasData() const { return !mData.is<Empty>(); }
bool IsClearFutureData() const { return mData.is<ClearFutureData>(); }
bool IsDecodedData() const { return mData.is<DecodedData>(); }
bool IsEOS() const { return mData.is<EOS>(); }
DecodedData* AsDecodedData() {
return IsDecodedData() ? &mData.as<DecodedData>() : nullptr;
}
Variant<Empty, ClearFutureData, DecodedData, EOS> mData;
};
// Decoder thread API
void AppendData(AudioData* aAudio, const PrincipalHandle& aPrincipalHandle);
void AppendData(nsTArray<RefPtr<AudioData>>& aAudioArray,
const PrincipalHandle& aPrincipalHandle);
void NotifyEndOfStream();
void ClearFutureData();
void SetVolume(float aVolume);
void SetPlaybackRate(float aPlaybackRate);
void SetPreservesPitch(bool aPreservesPitch);
// After calling this, the track are not expected to receive any new data.
void Close();
bool HasBatchedData() const;
MediaEventSource<int64_t>& OnOutput() { return mOnOutput; }
MediaEventSource<void>& OnEnd() { return mOnEnd; }
// Graph Thread API
void DestroyImpl() override;
void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
uint32_t NumberOfChannels() const override;
// The functions below are only used for testing.
TrackTime WrittenFrames() const {
AssertOnGraphThread();
return mWrittenFrames;
}
float Volume() const {
AssertOnGraphThread();
return mVolume;
}
float PlaybackRate() const {
AssertOnGraphThread();
return mPlaybackRate;
}
protected:
~AudioDecoderInputTrack();
private:
AudioDecoderInputTrack(nsISerialEventTarget* aDecoderThread,
TrackRate aGraphRate, const AudioInfo& aInfo,
float aPlaybackRate, float aVolume,
bool aPreservesPitch);
// Return false if the converted segment contains zero duration.
bool ConvertAudioDataToSegment(AudioData* aAudio, AudioSegment& aSegment,
const PrincipalHandle& aPrincipalHandle);
void HandleSPSCData(SPSCData& aData);
// These methods would return the total frames that we consumed from
// `mBufferedData`.
TrackTime AppendBufferedDataToOutput(TrackTime aExpectedDuration);
TrackTime FillDataToTimeStretcher(TrackTime aExpectedDuration);
TrackTime AppendTimeStretchedDataToSegment(TrackTime aExpectedDuration,
AudioSegment& aOutput);
TrackTime AppendUnstretchedDataToSegment(TrackTime aExpectedDuration,
AudioSegment& aOutput);
// Return the total frames that we retrieve from the time stretcher.
TrackTime DrainStretchedDataIfNeeded(TrackTime aExpectedDuration,
AudioSegment& aOutput);
TrackTime GetDataFromTimeStretcher(TrackTime aExpectedDuration,
AudioSegment& aOutput);
void NotifyInTheEndOfProcessInput(TrackTime aFillDuration);
bool HasSentAllData() const;
bool ShouldBatchData() const;
void BatchData(AudioData* aAudio, const PrincipalHandle& aPrincipalHandle);
void DispatchPushBatchedDataIfNeeded();
void PushBatchedDataIfNeeded();
void PushDataToSPSCQueue(SPSCData& data);
void SetVolumeImpl(float aVolume);
void SetPlaybackRateImpl(float aPlaybackRate);
void SetPreservesPitchImpl(bool aPreservesPitch);
void EnsureTimeStretcher();
void SetTempoAndRateForTimeStretcher();
uint32_t GetChannelCountForTimeStretcher() const;
inline void AssertOnDecoderThread() const {
MOZ_ASSERT(mDecoderThread->IsOnCurrentThread());
}
const RefPtr<nsISerialEventTarget> mDecoderThread;
// Notify the amount of audio frames which have been sent to the track.
MediaEventProducer<int64_t> mOnOutput;
// Notify when the track is ended.
MediaEventProducer<void> mOnEnd;
// These variables are ONLY used in the decoder thread.
nsAutoRef<SpeexResamplerState> mResampler;
uint32_t mResamplerChannelCount;
const uint32_t mInitialInputChannels;
TrackRate mInputSampleRate;
DelayedScheduler mDelayedScheduler;
bool mShutdownSPSCQueue = false;
// These attributes are ONLY used in the graph thread.
bool mReceivedEOS = false;
TrackTime mWrittenFrames = 0;
float mPlaybackRate;
float mVolume;
bool mPreservesPitch;
// A thread-safe queue shared by the decoder thread and the graph thread.
// The decoder thread is the producer side, and the graph thread is the
// consumer side. This queue should NEVER get full. In order to achieve that,
// we would batch input samples when SPSC queue doesn't have many available
// capacity.
// In addition, as the media track isn't guaranteed to be destroyed on the
// graph thread (it could be destroyed on the main thread as well) so we might
// not clear all data in SPSC queue when the track's `DestroyImpl()` gets
// called. We leave to destroy the queue later when the track gets destroyed.
SPSCQueue<SPSCData> mSPSCQueue{40};
// When the graph requires the less amount of audio frames than the amount of
// frames an audio data has, then the remaining part of frames would be stored
// and used in next iteration.
// This is ONLY used in the graph thread.
AudioSegment mBufferedData;
// In order to prevent SPSC queue from being full, we want to batch multiple
// data into one to control the density of SPSC queue, the length of batched
// data would be dynamically adjusted by queue's available capacity.
// This is ONLY used in the decoder thread.
SPSCData::DecodedData mBatchedData;
// True if we've sent all data to the graph, then the track will be marked as
// ended in the next iteration.
bool mSentAllData = false;
// This is used to adjust the playback rate and pitch.
RLBoxSoundTouch* mTimeStretcher = nullptr;
// Buffers that would be used for the time stretching.
AutoTArray<AudioDataValue, 2> mInterleavedBuffer;
};
} // namespace mozilla
#endif // AudioDecoderInputTrack_h