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/*
* Copyright (C) 2010, Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef AudioBufferSourceNode_h
#define AudioBufferSourceNode_h
#include "AudioBuffer.h"
#include "AudioBus.h"
#include "AudioParam.h"
#include "AudioScheduledSourceNode.h"
#include "ExceptionCode.h"
#include "PannerNode.h"
#include <memory>
#include <wtf/Lock.h>
#include <wtf/PassRefPtr.h>
#include <wtf/RefPtr.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 : public AudioScheduledSourceNode {
public:
static Ref<AudioBufferSourceNode> create(AudioContext&, float sampleRate);
virtual ~AudioBufferSourceNode();
// AudioNode
virtual void process(size_t framesToProcess) override;
virtual void reset() override;
// setBuffer() is called on the main thread. This is the buffer we use for playback.
// returns true on success.
bool setBuffer(AudioBuffer*);
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(ExceptionCode&);
void start(double when, ExceptionCode&);
void start(double when, double grainOffset, ExceptionCode&);
void start(double when, double grainOffset, double grainDuration, ExceptionCode&);
#if ENABLE(LEGACY_WEB_AUDIO)
void noteGrainOn(double when, double grainOffset, double grainDuration, ExceptionCode&);
#endif
// 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; }
// Deprecated.
bool looping();
void setLooping(bool);
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);
virtual double tailTime() const override { return 0; }
virtual double latencyTime() const override { return 0; }
enum BufferPlaybackMode {
Entire,
Partial
};
void startPlaying(BufferPlaybackMode, double when, double grainOffset, double grainDuration, ExceptionCode&);
// 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.
std::unique_ptr<const float*[]> m_sourceChannels;
std::unique_ptr<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 Lock m_processMutex;
};
} // namespace WebCore
#endif // AudioBufferSourceNode_h
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