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/* -*- 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
#if !defined(AudioStream_h_)
# define AudioStream_h_
# include "AudioSampleFormat.h"
# include "CubebUtils.h"
# include "MediaInfo.h"
# include "MediaSink.h"
# include "mozilla/Atomics.h"
# include "mozilla/Monitor.h"
# include "mozilla/MozPromise.h"
# include "mozilla/ProfilerUtils.h"
# include "mozilla/RefPtr.h"
# include "mozilla/Result.h"
# include "mozilla/TimeStamp.h"
# include "mozilla/UniquePtr.h"
# include "mozilla/SPSCQueue.h"
# include "nsCOMPtr.h"
# include "nsThreadUtils.h"
# include "WavDumper.h"
namespace mozilla {
struct CubebDestroyPolicy {
void operator()(cubeb_stream* aStream) const {
cubeb_stream_destroy(aStream);
}
};
class AudioStream;
class FrameHistory;
class AudioConfig;
class RLBoxSoundTouch;
// A struct that contains the number of frames serviced or underrun by a
// callback, alongside the sample-rate for this callback (in case of playback
// rate change, it can be variable).
struct CallbackInfo {
CallbackInfo() = default;
CallbackInfo(uint32_t aServiced, uint32_t aUnderrun, uint32_t aOutputRate)
: mServiced(aServiced), mUnderrun(aUnderrun), mOutputRate(aOutputRate) {}
uint32_t mServiced = 0;
uint32_t mUnderrun = 0;
uint32_t mOutputRate = 0;
};
class AudioClock {
public:
explicit AudioClock(uint32_t aInRate);
// Update the number of samples that has been written in the audio backend.
// Called on the audio thread only.
void UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun,
bool aAudioThreadChanged);
/**
* @param aFrames The playback position in frames of the audio engine.
* @return The playback position in frames of the stream,
* adjusted by playback rate changes and underrun frames.
*/
int64_t GetPositionInFrames(int64_t aFrames);
/**
* @param frames The playback position in frames of the audio engine.
* @return The playback position in microseconds of the stream,
* adjusted by playback rate changes and underrun frames.
*/
int64_t GetPosition(int64_t frames);
// Set the playback rate.
// Called on the audio thread only.
void SetPlaybackRate(double aPlaybackRate);
// Get the current playback rate.
// Called on the audio thread only.
double GetPlaybackRate() const;
// Set if we are preserving the pitch.
// Called on the audio thread only.
void SetPreservesPitch(bool aPreservesPitch);
// Get the current pitch preservation state.
// Called on the audio thread only.
bool GetPreservesPitch() const;
// Called on either thread.
uint32_t GetInputRate() const { return mInRate; }
uint32_t GetOutputRate() const { return mOutRate; }
private:
// Output rate in Hz (characteristic of the playback rate). Written on the
// audio thread, read on either thread.
Atomic<uint32_t> mOutRate;
// Input rate in Hz (characteristic of the media being played).
const uint32_t mInRate;
// True if the we are timestretching, false if we are resampling. Accessed on
// the audio thread only.
bool mPreservesPitch;
// The history of frames sent to the audio engine in each DataCallback.
// Only accessed from non-audio threads on macOS, accessed on both threads and
// protected by the AudioStream monitor on other platforms.
const UniquePtr<FrameHistory> mFrameHistory
# ifndef XP_MACOSX
MOZ_GUARDED_BY(mMutex)
# endif
;
# ifdef XP_MACOSX
// Enqueued on the audio thread, dequeued from the other thread. The maximum
// size of this queue has been chosen empirically.
SPSCQueue<CallbackInfo> mCallbackInfoQueue{100};
// If it isn't possible to send the callback info to the non-audio thread,
// store them here until it's possible to send them. This is an unlikely
// fallback path. The size of this array has been chosen empirically. Only
// ever accessed on the audio thread.
AutoTArray<CallbackInfo, 5> mAudioThreadCallbackInfo;
# else
Mutex mMutex{"AudioClock"};
# endif
};
/*
* A bookkeeping class to track the read/write position of an audio buffer.
*/
class AudioBufferCursor {
public:
AudioBufferCursor(Span<AudioDataValue> aSpan, uint32_t aChannels,
uint32_t aFrames)
: mChannels(aChannels), mSpan(aSpan), mFrames(aFrames) {}
// Advance the cursor to account for frames that are consumed.
uint32_t Advance(uint32_t aFrames) {
MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
MOZ_ASSERT(mFrames >= aFrames);
mFrames -= aFrames;
mOffset += mChannels * aFrames;
return aFrames;
}
// The number of frames available for read/write in this buffer.
uint32_t Available() const { return mFrames; }
// Return a pointer where read/write should begin.
AudioDataValue* Ptr() const {
MOZ_DIAGNOSTIC_ASSERT(mOffset <= mSpan.Length());
return mSpan.Elements() + mOffset;
}
protected:
bool Contains(uint32_t aFrames) const {
return mSpan.Length() >= mOffset + mChannels * aFrames;
}
const uint32_t mChannels;
private:
const Span<AudioDataValue> mSpan;
size_t mOffset = 0;
uint32_t mFrames;
};
/*
* A helper class to encapsulate pointer arithmetic and provide means to modify
* the underlying audio buffer.
*/
class AudioBufferWriter : public AudioBufferCursor {
public:
AudioBufferWriter(Span<AudioDataValue> aSpan, uint32_t aChannels,
uint32_t aFrames)
: AudioBufferCursor(aSpan, aChannels, aFrames) {}
uint32_t WriteZeros(uint32_t aFrames) {
MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
memset(Ptr(), 0, sizeof(AudioDataValue) * mChannels * aFrames);
return Advance(aFrames);
}
uint32_t Write(const AudioDataValue* aPtr, uint32_t aFrames) {
MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
memcpy(Ptr(), aPtr, sizeof(AudioDataValue) * mChannels * aFrames);
return Advance(aFrames);
}
// Provide a write fuction to update the audio buffer with the following
// signature: uint32_t(const AudioDataValue* aPtr, uint32_t aFrames)
// aPtr: Pointer to the audio buffer.
// aFrames: The number of frames available in the buffer.
// return: The number of frames actually written by the function.
template <typename Function>
uint32_t Write(const Function& aFunction, uint32_t aFrames) {
MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
return Advance(aFunction(Ptr(), aFrames));
}
using AudioBufferCursor::Available;
};
// Access to a single instance of this class must be synchronized by
// callers, or made from a single thread. One exception is that access to
// GetPosition, GetPositionInFrames, SetVolume, and Get{Rate,Channels},
// SetMicrophoneActive is thread-safe without external synchronization.
class AudioStream final {
virtual ~AudioStream();
public:
NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream)
class Chunk {
public:
// Return a pointer to the audio data.
virtual const AudioDataValue* Data() const = 0;
// Return the number of frames in this chunk.
virtual uint32_t Frames() const = 0;
// Return the number of audio channels.
virtual uint32_t Channels() const = 0;
// Return the sample rate of this chunk.
virtual uint32_t Rate() const = 0;
// Return a writable pointer for downmixing.
virtual AudioDataValue* GetWritable() const = 0;
virtual ~Chunk() = default;
};
class DataSource {
public:
// Attempt to acquire aFrames frames of audio, and returns the number of
// frames successfuly acquired.
virtual uint32_t PopFrames(AudioDataValue* aAudio, uint32_t aFrames,
bool aAudioThreadChanged) = 0;
// Return true if no more data will be added to the source.
virtual bool Ended() const = 0;
protected:
virtual ~DataSource() = default;
};
// aOutputChannels is the number of audio channels (1 for mono, 2 for stereo,
// etc), aChannelMap is the indicator for channel layout(mono, stereo, 5.1 or
// 7.1 ). Initialize the audio stream.and aRate is the sample rate
// (22050Hz, 44100Hz, etc).
AudioStream(DataSource& aSource, uint32_t aInRate, uint32_t aOutputChannels,
AudioConfig::ChannelLayout::ChannelMap aChannelMap);
nsresult Init(AudioDeviceInfo* aSinkInfo);
// Closes the stream. All future use of the stream is an error.
void ShutDown();
// Set the current volume of the audio playback. This is a value from
// 0 (meaning muted) to 1 (meaning full volume). Thread-safe.
void SetVolume(double aVolume);
void SetStreamName(const nsAString& aStreamName);
// Start the stream.
RefPtr<MediaSink::EndedPromise> Start();
// Pause audio playback.
void Pause();
// Resume audio playback.
void Resume();
// Return the position in microseconds of the audio frame being played by
// the audio hardware, compensated for playback rate change. Thread-safe.
int64_t GetPosition();
// Return the position, measured in audio frames played since the stream
// was opened, of the audio hardware. Thread-safe.
int64_t GetPositionInFrames();
uint32_t GetOutChannels() const { return mOutChannels; }
// Set playback rate as a multiple of the intrinsic playback rate. This is
// to be called only with aPlaybackRate > 0.0.
nsresult SetPlaybackRate(double aPlaybackRate);
// Switch between resampling (if false) and time stretching (if true,
// default).
nsresult SetPreservesPitch(bool aPreservesPitch);
size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const;
bool IsPlaybackCompleted() const;
// Returns true if at least one DataCallback has been called.
bool CallbackStarted() const { return mCallbacksStarted; }
protected:
friend class AudioClock;
// Return the position, measured in audio frames played since the stream was
// opened, of the audio hardware, not adjusted for the changes of playback
// rate or underrun frames.
// Caller must own the monitor.
int64_t GetPositionInFramesUnlocked();
private:
nsresult OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
TimeStamp aStartTime, bool aIsFirst);
static long DataCallback_S(cubeb_stream*, void* aThis,
const void* /* aInputBuffer */,
void* aOutputBuffer, long aFrames) {
return static_cast<AudioStream*>(aThis)->DataCallback(aOutputBuffer,
aFrames);
}
static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState) {
static_cast<AudioStream*>(aThis)->StateCallback(aState);
}
long DataCallback(void* aBuffer, long aFrames);
void StateCallback(cubeb_state aState);
// Audio thread only
nsresult EnsureTimeStretcherInitialized();
void GetUnprocessed(AudioBufferWriter& aWriter);
void GetTimeStretched(AudioBufferWriter& aWriter);
void UpdatePlaybackRateIfNeeded();
// Return true if audio frames are valid (correct sampling rate and valid
// channel count) otherwise false.
bool IsValidAudioFormat(Chunk* aChunk) MOZ_REQUIRES(mMonitor);
template <typename Function, typename... Args>
int InvokeCubeb(Function aFunction, Args&&... aArgs) MOZ_REQUIRES(mMonitor);
bool CheckThreadIdChanged();
void AssertIsOnAudioThread() const;
RLBoxSoundTouch* mTimeStretcher;
AudioClock mAudioClock;
WavDumper mDumpFile;
const AudioConfig::ChannelLayout::ChannelMap mChannelMap;
// The monitor is held to protect all access to member variables below.
Monitor mMonitor MOZ_UNANNOTATED;
const uint32_t mOutChannels;
// mCubebStream holds a bare pointer to cubeb, so we hold a ref on its behalf
RefPtr<CubebUtils::CubebHandle> mCubeb;
// Owning reference to a cubeb_stream. Set in Init(), cleared in ShutDown, so
// no lock is needed to access.
UniquePtr<cubeb_stream, CubebDestroyPolicy> mCubebStream;
enum StreamState {
INITIALIZED, // Initialized, playback has not begun.
STARTED, // cubeb started.
STOPPED, // Stopped by a call to Pause().
DRAINED, // StateCallback has indicated that the drain is complete.
ERRORED, // Stream disabled due to an internal error.
SHUTDOWN // ShutDown has been called
};
std::atomic<StreamState> mState;
// DataSource::PopFrames can never be called concurrently.
// DataSource::IsEnded uses only atomics.
DataSource& mDataSource;
// The device info of the current sink. If null
// the default device is used. It is set
// during the Init() in decoder thread.
RefPtr<AudioDeviceInfo> mSinkInfo;
// Contains the id of the audio thread, from profiler_get_thread_id.
std::atomic<ProfilerThreadId> mAudioThreadId;
const bool mSandboxed = false;
MozPromiseHolder<MediaSink::EndedPromise> mEndedPromise
MOZ_GUARDED_BY(mMonitor);
std::atomic<bool> mPlaybackComplete;
// Both written on the MDSM thread, read on the audio thread.
std::atomic<float> mPlaybackRate;
std::atomic<bool> mPreservesPitch;
// Audio thread only
bool mAudioThreadChanged = false;
Atomic<bool> mCallbacksStarted;
};
} // namespace mozilla
#endif