mirror of
https://github.com/mozilla/gecko-dev.git
synced 2024-10-31 14:15:30 +00:00
33f592b6a5
Performing all audio processing operations in the same place, allows to simplify the code. Additionally, if accessibility.monoaudio.enable is not set, we always upmix mono to stereo so that if the first audio stream seen was mono, we aren't stuck playing all future streams in mono. MozReview-Commit-ID: 5yANN6PLFhX --HG-- extra : rebase_source : 8b2138368d97f4ec2857c021ed9605c633282ed2
385 lines
12 KiB
C++
385 lines
12 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/. */
|
|
#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
|