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* Copyright (C) 2010, Google Inc. All rights reserved.
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#ifndef AudioBufferSourceNode_h
#define AudioBufferSourceNode_h
#include "platform/audio/AudioBus.h"
#include "modules/webaudio/AudioBuffer.h"
#include "modules/webaudio/AudioParam.h"
#include "modules/webaudio/AudioScheduledSourceNode.h"
#include "modules/webaudio/PannerNode.h"
#include "wtf/OwnPtr.h"
#include "wtf/PassRefPtr.h"
#include "wtf/RefPtr.h"
#include "wtf/Threading.h"
namespace WebCore {
class AudioContext;
// AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer.
// It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways).
class AudioBufferSourceNode FINAL : public AudioScheduledSourceNode {
public:
static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
virtual ~AudioBufferSourceNode();
// AudioNode
virtual void process(size_t framesToProcess) OVERRIDE;
// setBuffer() is called on the main thread. This is the buffer we use for playback.
void setBuffer(AudioBuffer*, ExceptionState&);
AudioBuffer* buffer() { return m_buffer.get(); }
// numberOfChannels() returns the number of output channels. This value equals the number of channels from the buffer.
// If a new buffer is set with a different number of channels, then this value will dynamically change.
unsigned numberOfChannels();
// Play-state
void start(ExceptionState& exceptionState) { start(0, exceptionState); }
void start(double when, ExceptionState&);
void start(double when, double grainOffset, ExceptionState&);
void start(double when, double grainOffset, double grainDuration, ExceptionState&);
void noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionState&);
// Note: the attribute was originally exposed as .looping, but to be more consistent in naming with <audio>
// and with how it's described in the specification, the proper attribute name is .loop
// The old attribute is kept for backwards compatibility.
bool loop() const { return m_isLooping; }
void setLoop(bool looping) { m_isLooping = looping; }
// Loop times in seconds.
double loopStart() const { return m_loopStart; }
double loopEnd() const { return m_loopEnd; }
void setLoopStart(double loopStart) { m_loopStart = loopStart; }
void setLoopEnd(double loopEnd) { m_loopEnd = loopEnd; }
AudioParam* gain() { return m_gain.get(); }
AudioParam* playbackRate() { return m_playbackRate.get(); }
// If a panner node is set, then we can incorporate doppler shift into the playback pitch rate.
void setPannerNode(PannerNode*);
void clearPannerNode();
// If we are no longer playing, propogate silence ahead to downstream nodes.
virtual bool propagatesSilence() const OVERRIDE;
// AudioScheduledSourceNode
virtual void finish() OVERRIDE;
private:
AudioBufferSourceNode(AudioContext*, float sampleRate);
// Returns true on success.
bool renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames);
// Render silence starting from "index" frame in AudioBus.
inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess);
// m_buffer holds the sample data which this node outputs.
RefPtr<AudioBuffer> m_buffer;
// Pointers for the buffer and destination.
OwnPtr<const float*[]> m_sourceChannels;
OwnPtr<float*[]> m_destinationChannels;
// Used for the "gain" and "playbackRate" attributes.
RefPtr<AudioParam> m_gain;
RefPtr<AudioParam> m_playbackRate;
// If m_isLooping is false, then this node will be done playing and become inactive after it reaches the end of the sample data in the buffer.
// If true, it will wrap around to the start of the buffer each time it reaches the end.
bool m_isLooping;
double m_loopStart;
double m_loopEnd;
// m_virtualReadIndex is a sample-frame index into our buffer representing the current playback position.
// Since it's floating-point, it has sub-sample accuracy.
double m_virtualReadIndex;
// Granular playback
bool m_isGrain;
double m_grainOffset; // in seconds
double m_grainDuration; // in seconds
// totalPitchRate() returns the instantaneous pitch rate (non-time preserving).
// It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node.
double totalPitchRate();
// m_lastGain provides continuity when we dynamically adjust the gain.
float m_lastGain;
// We optionally keep track of a panner node which has a doppler shift that is incorporated into
// the pitch rate. We manually manage ref-counting because we want to use RefTypeConnection.
PannerNode* m_pannerNode;
// This synchronizes process() with setBuffer() which can cause dynamic channel count changes.
mutable Mutex m_processLock;
};
} // namespace WebCore
#endif // AudioBufferSourceNode_h