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 "Latency.h"
#include "mozilla/dom/AudioChannelBinding.h"
#include "mozilla/RefPtr.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 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)
int mOutRate;
// Input rate in Hz (characteristic of the media being played)
int 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 = new 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;
}
// Throw away all but aSize bytes from the buffer. Returns new size, which
// may be less than aSize
uint32_t ContractTo(uint32_t aSize) {
MOZ_ASSERT(mBuffer && mCapacity, "Buffer not initialized.");
if (aSize >= mCount) {
return mCount;
}
mStart += (mCount - aSize);
mCount = aSize;
mStart %= mCapacity;
return mCount;
}
size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = 0;
amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
return amount;
}
void Reset()
{
mBuffer = nullptr;
mCapacity = 0;
mStart = 0;
mCount = 0;
}
private:
nsAutoArrayPtr<uint8_t> mBuffer;
uint32_t mCapacity;
uint32_t mStart;
uint32_t mCount;
};
class AudioInitTask;
// 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)
AudioStream();
enum LatencyRequest {
HighLatency,
LowLatency
};
// 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(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannel aAudioStreamChannel,
LatencyRequest aLatencyRequest);
// Closes the stream. All future use of the stream is an error.
void Shutdown();
void Reset();
// Write audio data to the audio hardware. aBuf is an array of AudioDataValues
// AudioDataValue of length aFrames*mChannels. If aFrames is larger
// than the result of Available(), the write will block until sufficient
// buffer space is available. aTime is the time in ms associated with the first sample
// for latency calculations
nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp* aTime = nullptr);
// Return the number of audio frames that can be written without blocking.
uint32_t Available();
// 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);
// Informs the AudioStream that a microphone is being used by someone in the
// application.
void SetMicrophoneActive(bool aActive);
void PanOutputIfNeeded(bool aMicrophoneActive);
void ResetStreamIfNeeded();
// Block until buffered audio data has been consumed.
void Drain();
// Break any blocking operation and set the stream to shutdown.
void Cancel();
// Start the stream.
void Start();
// Return the number of frames written so far in the stream. This allow the
// caller to check if it is safe to start the stream, if needed.
int64_t GetWritten();
// 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();
int GetRate() { return mOutRate; }
int GetChannels() { return mChannels; }
int 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:
friend class AudioInitTask;
// So we can call it asynchronously from AudioInitTask
nsresult OpenCubeb(cubeb_stream_params &aParams,
LatencyRequest aLatencyRequest);
void AudioInitTaskFinished();
void CheckForStart();
static long DataCallback_S(cubeb_stream*, void* aThis, void* aBuffer, long aFrames)
{
return static_cast<AudioStream*>(aThis)->DataCallback(aBuffer, aFrames);
}
static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState)
{
static_cast<AudioStream*>(aThis)->StateCallback(aState);
}
static void DeviceChangedCallback_s(void * aThis) {
static_cast<AudioStream*>(aThis)->DeviceChangedCallback();
}
long DataCallback(void* aBuffer, long aFrames);
void StateCallback(cubeb_state aState);
void DeviceChangedCallback();
nsresult EnsureTimeStretcherInitializedUnlocked();
// aTime is the time in ms the samples were inserted into MediaStreamGraph
long GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTime);
long GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTime);
long GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t &aTime);
int64_t GetLatencyInFrames();
void GetBufferInsertTime(int64_t &aTimeMs);
void StartUnlocked();
// The monitor is held to protect all access to member variables. Write()
// waits while mBuffer is full; DataCallback() notifies as it consumes
// data from mBuffer. Drain() waits while mState is DRAINING;
// StateCallback() notifies when mState is DRAINED.
Monitor mMonitor;
// Input rate in Hz (characteristic of the media being played)
int mInRate;
// Output rate in Hz (characteristic of the playback rate)
int mOutRate;
int mChannels;
int mOutChannels;
#if defined(__ANDROID__)
dom::AudioChannel mAudioChannel;
#endif
// Number of frames written to the buffers.
int64_t mWritten;
AudioClock mAudioClock;
soundtouch::SoundTouch* mTimeStretcher;
nsRefPtr<AsyncLatencyLogger> mLatencyLog;
// copy of Latency logger's starting time for offset calculations
TimeStamp mStartTime;
// Whether we are playing a low latency stream, or a normal stream.
LatencyRequest mLatencyRequest;
// Where in the current mInserts[0] block cubeb has read to
int64_t mReadPoint;
// Keep track of each inserted block of samples and the time it was inserted
// so we can estimate the clock time for a specific sample's insertion (for when
// we send data to cubeb). Blocks are aged out as needed.
struct Inserts {
int64_t mTimeMs;
int64_t mFrames;
};
nsAutoTArray<Inserts, 8> mInserts;
// Output file for dumping audio
FILE* mDumpFile;
// Temporary audio buffer. Filled by Write() and consumed by
// DataCallback(). Once mBuffer is full, Write() blocks until sufficient
// space becomes available in mBuffer. mBuffer is sized in bytes, not
// frames.
CircularByteBuffer mBuffer;
// Owning reference to a cubeb_stream.
UniquePtr<cubeb_stream, CubebDestroyPolicy> mCubebStream;
uint32_t mBytesPerFrame;
uint32_t BytesToFrames(uint32_t aBytes) {
NS_ASSERTION(aBytes % mBytesPerFrame == 0,
"Byte count not aligned on frames size.");
return aBytes / mBytesPerFrame;
}
uint32_t FramesToBytes(uint32_t aFrames) {
return aFrames * mBytesPerFrame;
}
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().
DRAINING, // Drain requested. DataCallback will indicate end of stream
// once the remaining contents of mBuffer are requested by
// cubeb, after which StateCallback will indicate drain
// completion.
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 mNeedsStart; // needed in case Start() is called before cubeb is open
bool mIsFirst;
// True if a microphone is active.
bool mMicrophoneActive;
// When we are in the process of changing the output device, and the callback
// is not going to be called for a little while, simply drop incoming frames.
// This is only on OSX for now, because other systems handle this gracefully.
bool mShouldDropFrames;
// True if there is a pending AudioInitTask. Shutdown() will wait until the
// pending AudioInitTask is finished.
bool mPendingAudioInitTask;
// The last good position returned by cubeb_stream_get_position(). Used to
// check if the cubeb position is going backward.
uint64_t mLastGoodPosition;
};
class AudioInitTask : public nsRunnable
{
public:
AudioInitTask(AudioStream *aStream,
AudioStream::LatencyRequest aLatencyRequest,
const cubeb_stream_params &aParams)
: mAudioStream(aStream)
, mLatencyRequest(aLatencyRequest)
, mParams(aParams)
{}
nsresult Dispatch()
{
// Can't add 'this' as the event to run, since mThread may not be set yet
nsresult rv = NS_NewNamedThread("CubebInit", getter_AddRefs(mThread));
if (NS_SUCCEEDED(rv)) {
// Note: event must not null out mThread!
rv = mThread->Dispatch(this, NS_DISPATCH_NORMAL);
}
return rv;
}
protected:
virtual ~AudioInitTask() {};
private:
NS_IMETHOD Run() override final;
RefPtr<AudioStream> mAudioStream;
AudioStream::LatencyRequest mLatencyRequest;
cubeb_stream_params mParams;
nsCOMPtr<nsIThread> mThread;
};
} // namespace mozilla
#endif