gecko-dev/content/media/AudioStream.h

183 lines
7.0 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 "nscore.h"
#include "AudioSampleFormat.h"
#include "AudioChannelCommon.h"
#include "soundtouch/SoundTouch.h"
#include "nsAutoPtr.h"
namespace mozilla {
class AudioStream;
class AudioClock
{
public:
AudioClock(mozilla::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 UpdateWritePosition(uint32_t aCount);
// Get the read position of the stream, in microseconds.
// Called on the state machine thead.
uint64_t GetPosition();
// Get the read position of the stream, in frames.
// Called on the state machine thead.
uint64_t GetPositionInFrames();
// Set the playback rate.
// Called on the audio thread.
void SetPlaybackRate(double aPlaybackRate);
// Get the current playback rate.
// Called on the audio thread.
double GetPlaybackRate();
// 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();
private:
// This AudioStream holds a strong reference to this AudioClock. This
// pointer is garanteed to always be valid.
AudioStream* mAudioStream;
// The old output rate, to compensate audio latency for the period inbetween
// the moment resampled buffers are pushed to the hardware and the moment the
// clock should take the new rate into account for A/V sync.
int mOldOutRate;
// Position at which the last playback rate change occured
int64_t mBasePosition;
// Offset, in frames, at which the last playback rate change occured
int64_t mBaseOffset;
// Old base offset (number of samples), used when changing rate to compute the
// position in the stream.
int64_t mOldBaseOffset;
// Old base position (number of microseconds), when changing rate. This is the
// time in the media, not wall clock position.
int64_t mOldBasePosition;
// Write position at which the playbackRate change occured.
int64_t mPlaybackRateChangeOffset;
// The previous position reached in the media, used when compensating
// latency, to have the position at which the playbackRate change occured.
int64_t mPreviousPosition;
// Number of samples effectivelly written in backend, i.e. write position.
int64_t mWritten;
// 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 current playback rate.
double mPlaybackRate;
// True if we are playing at the old playbackRate after it has been changed.
bool mCompensatingLatency;
};
// 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}
// is thread-safe without external synchronization.
class AudioStream
{
public:
AudioStream();
virtual ~AudioStream();
// Initialize Audio Library. Some Audio backends require initializing the
// library before using it.
static void InitLibrary();
// Shutdown Audio Library. Some Audio backends require shutting down the
// library after using it.
static void ShutdownLibrary();
// AllocateStream will return either a local stream or a remoted stream
// depending on where you call it from. If you call this from a child process,
// you may receive an implementation which forwards to a compositing process.
static AudioStream* AllocateStream();
// 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).
virtual nsresult Init(int32_t aNumChannels, int32_t aRate,
const dom::AudioChannelType aAudioStreamType) = 0;
// Closes the stream. All future use of the stream is an error.
virtual void Shutdown() = 0;
// 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.
virtual nsresult Write(const mozilla::AudioDataValue* aBuf, uint32_t aFrames) = 0;
// Return the number of audio frames that can be written without blocking.
virtual uint32_t Available() = 0;
// Set the current volume of the audio playback. This is a value from
// 0 (meaning muted) to 1 (meaning full volume). Thread-safe.
virtual void SetVolume(double aVolume) = 0;
// Block until buffered audio data has been consumed.
virtual void Drain() = 0;
// Pause audio playback
virtual void Pause() = 0;
// Resume audio playback
virtual void Resume() = 0;
// Return the position in microseconds of the audio frame being played by
// the audio hardware, compensated for playback rate change. Thread-safe.
virtual int64_t GetPosition() = 0;
// Return the position, measured in audio frames played since the stream
// was opened, of the audio hardware. Thread-safe.
virtual int64_t GetPositionInFrames() = 0;
// Return the position, measured in audio framed played since the stream was
// opened, of the audio hardware, not adjusted for the changes of playback
// rate.
virtual int64_t GetPositionInFramesInternal() = 0;
// Returns true when the audio stream is paused.
virtual bool IsPaused() = 0;
// Returns the minimum number of audio frames which must be written before
// you can be sure that something will be played.
virtual int32_t GetMinWriteSize() = 0;
int GetRate() { return mOutRate; }
int GetChannels() { return mChannels; }
// This should be called before attempting to use the time stretcher.
void EnsureTimeStretcherInitialized();
// Set playback rate as a multiple of the intrinsic playback rate. This is to
// be called only with aPlaybackRate > 0.0.
virtual nsresult SetPlaybackRate(double aPlaybackRate);
// Switch between resampling (if false) and time stretching (if true, default).
virtual nsresult SetPreservesPitch(bool aPreservesPitch);
protected:
// 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;
AudioClock mAudioClock;
nsAutoPtr<soundtouch::SoundTouch> mTimeStretcher;
};
} // namespace mozilla
#endif