gecko-dev/dom/media/AudioStream.h

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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
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
#if !defined(AudioStream_h_)
#define AudioStream_h_
#include "AudioSampleFormat.h"
#include "nsAutoPtr.h"
#include "nsCOMPtr.h"
#include "nsThreadUtils.h"
#include "mozilla/dom/AudioChannelBinding.h"
#include "mozilla/Monitor.h"
#include "mozilla/RefPtr.h"
#include "mozilla/TimeStamp.h"
#include "mozilla/UniquePtr.h"
#include "CubebUtils.h"
#include "soundtouch/SoundTouchFactory.h"
namespace mozilla {
struct CubebDestroyPolicy
{
void operator()(cubeb_stream* aStream) const {
cubeb_stream_destroy(aStream);
}
};
class AudioStream;
class FrameHistory;
class AudioConfig;
class AudioConverter;
class AudioClock
{
public:
explicit AudioClock(AudioStream* aStream);
// Initialize the clock with the current AudioStream. Need to be called
// before querying the clock. Called on the audio thread.
void Init();
// Update the number of samples that has been written in the audio backend.
// Called on the state machine thread.
void UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun);
// Get the read position of the stream, in microseconds.
// Called on the state machine thead.
// Assumes the AudioStream lock is held and thus calls Unlocked versions
// of AudioStream funcs.
int64_t GetPositionUnlocked() const;
// Get the read position of the stream, in frames.
// Called on the state machine thead.
int64_t GetPositionInFrames() const;
// Set the playback rate.
// Called on the audio thread.
// Assumes the AudioStream lock is held and thus calls Unlocked versions
// of AudioStream funcs.
void SetPlaybackRateUnlocked(double aPlaybackRate);
// Get the current playback rate.
// Called on the audio thread.
double GetPlaybackRate() const;
// Set if we are preserving the pitch.
// Called on the audio thread.
void SetPreservesPitch(bool aPreservesPitch);
// Get the current pitch preservation state.
// Called on the audio thread.
bool GetPreservesPitch() const;
private:
// This AudioStream holds a strong reference to this AudioClock. This
// pointer is garanteed to always be valid.
AudioStream* const mAudioStream;
// Output rate in Hz (characteristic of the playback rate)
uint32_t mOutRate;
// Input rate in Hz (characteristic of the media being played)
uint32_t mInRate;
// True if the we are timestretching, false if we are resampling.
bool mPreservesPitch;
// The history of frames sent to the audio engine in each DataCallback.
const nsAutoPtr<FrameHistory> mFrameHistory;
};
class CircularByteBuffer
{
public:
CircularByteBuffer()
: mBuffer(nullptr), mCapacity(0), mStart(0), mCount(0)
{}
// Set the capacity of the buffer in bytes. Must be called before any
// call to append or pop elements.
void SetCapacity(uint32_t aCapacity) {
MOZ_ASSERT(!mBuffer, "Buffer allocated.");
mCapacity = aCapacity;
mBuffer = MakeUnique<uint8_t[]>(mCapacity);
}
uint32_t Length() {
return mCount;
}
uint32_t Capacity() {
return mCapacity;
}
uint32_t Available() {
return Capacity() - Length();
}
// Append aLength bytes from aSrc to the buffer. Caller must check that
// sufficient space is available.
void AppendElements(const uint8_t* aSrc, uint32_t aLength) {
MOZ_ASSERT(mBuffer && mCapacity, "Buffer not initialized.");
MOZ_ASSERT(aLength <= Available(), "Buffer full.");
uint32_t end = (mStart + mCount) % mCapacity;
uint32_t toCopy = std::min(mCapacity - end, aLength);
memcpy(&mBuffer[end], aSrc, toCopy);
memcpy(&mBuffer[0], aSrc + toCopy, aLength - toCopy);
mCount += aLength;
}
// Remove aSize bytes from the buffer. Caller must check returned size in
// aSize{1,2} before using the pointer returned in aData{1,2}. Caller
// must not specify an aSize larger than Length().
void PopElements(uint32_t aSize, void** aData1, uint32_t* aSize1,
void** aData2, uint32_t* aSize2) {
MOZ_ASSERT(mBuffer && mCapacity, "Buffer not initialized.");
MOZ_ASSERT(aSize <= Length(), "Request too large.");
*aData1 = &mBuffer[mStart];
*aSize1 = std::min(mCapacity - mStart, aSize);
*aData2 = &mBuffer[0];
*aSize2 = aSize - *aSize1;
mCount -= *aSize1 + *aSize2;
mStart += *aSize1 + *aSize2;
mStart %= mCapacity;
}
size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = 0;
amount += aMallocSizeOf(mBuffer.get());
return amount;
}
private:
UniquePtr<uint8_t[]> mBuffer;
uint32_t mCapacity;
uint32_t mStart;
uint32_t mCount;
};
/*
* A bookkeeping class to track the read/write position of an audio buffer.
*/
class AudioBufferCursor {
public:
AudioBufferCursor(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames)
: mPtr(aPtr), mChannels(aChannels), mFrames(aFrames) {}
// Advance the cursor to account for frames that are consumed.
uint32_t Advance(uint32_t aFrames) {
MOZ_ASSERT(mFrames >= aFrames);
mFrames -= aFrames;
mPtr += 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 { return mPtr; }
protected:
AudioDataValue* mPtr;
const uint32_t mChannels;
uint32_t mFrames;
};
/*
* A helper class to encapsulate pointer arithmetic and provide means to modify
* the underlying audio buffer.
*/
class AudioBufferWriter : private AudioBufferCursor {
public:
AudioBufferWriter(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames)
: AudioBufferCursor(aPtr, aChannels, aFrames) {}
uint32_t WriteZeros(uint32_t aFrames) {
memset(mPtr, 0, sizeof(AudioDataValue) * mChannels * aFrames);
return Advance(aFrames);
}
uint32_t Write(const AudioDataValue* aPtr, uint32_t aFrames) {
memcpy(mPtr, 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) {
return Advance(aFunction(mPtr, 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() {}
};
class DataSource {
public:
// Return a chunk which contains at most aFrames frames or zero if no
// frames in the source at all.
virtual UniquePtr<Chunk> PopFrames(uint32_t aFrames) = 0;
// Return true if no more data will be added to the source.
virtual bool Ended() const = 0;
// Notify that all data is drained by the AudioStream.
virtual void Drained() = 0;
protected:
virtual ~DataSource() {}
};
explicit AudioStream(DataSource& aSource);
// Initialize the audio stream. aNumChannels is the number of audio
// channels (1 for mono, 2 for stereo, etc) and aRate is the sample rate
// (22050Hz, 44100Hz, etc).
nsresult Init(uint32_t aNumChannels, uint32_t aRate,
const dom::AudioChannel aAudioStreamChannel);
// Closes the stream. All future use of the stream is an error.
void Shutdown();
void Reset();
// 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);
// Start the stream.
void 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();
// Returns true when the audio stream is paused.
bool IsPaused();
static uint32_t GetPreferredRate()
{
CubebUtils::InitPreferredSampleRate();
return CubebUtils::PreferredSampleRate();
}
uint32_t GetRate() { return mOutRate; }
uint32_t GetChannels() { return mChannels; }
uint32_t GetOutChannels() { 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;
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_stream_params &aParams);
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);
nsresult EnsureTimeStretcherInitializedUnlocked();
// Return true if audio frames are valid (correct sampling rate and valid
// channel count) otherwise false.
bool IsValidAudioFormat(Chunk* aChunk);
void GetUnprocessed(AudioBufferWriter& aWriter);
void GetTimeStretched(AudioBufferWriter& aWriter);
void StartUnlocked();
// The monitor is held to protect all access to member variables.
Monitor mMonitor;
// Input rate in Hz (characteristic of the media being played)
uint32_t mInRate;
// Output rate in Hz (characteristic of the playback rate)
uint32_t mOutRate;
uint32_t mChannels;
uint32_t mOutChannels;
AudioClock mAudioClock;
soundtouch::SoundTouch* mTimeStretcher;
// Stream start time for stream open delay telemetry.
TimeStamp mStartTime;
// Output file for dumping audio
FILE* mDumpFile;
// Owning reference to a cubeb_stream.
UniquePtr<cubeb_stream, CubebDestroyPolicy> mCubebStream;
enum StreamState {
INITIALIZED, // Initialized, playback has not begun.
STARTED, // cubeb started, but callbacks haven't started
RUNNING, // DataCallbacks have started after STARTED, or after Resume().
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
};
StreamState mState;
bool mIsFirst;
DataSource& mDataSource;
};
} // namespace mozilla
#endif