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25 #ifndef AudioBufferSourceNode_h
26 #define AudioBufferSourceNode_h
28 #include "platform/audio/AudioBus.h"
29 #include "modules/webaudio/AudioBuffer.h"
30 #include "modules/webaudio/AudioParam.h"
31 #include "modules/webaudio/AudioScheduledSourceNode.h"
32 #include "modules/webaudio/PannerNode.h"
33 #include "wtf/OwnPtr.h"
34 #include "wtf/PassRefPtr.h"
35 #include "wtf/RefPtr.h"
36 #include "wtf/Threading.h"
42 // AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer.
43 // It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways).
45 class AudioBufferSourceNode FINAL : public AudioScheduledSourceNode {
47 static PassRefPtrWillBeRawPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
49 virtual ~AudioBufferSourceNode();
52 virtual void process(size_t framesToProcess) OVERRIDE;
54 // setBuffer() is called on the main thread. This is the buffer we use for playback.
55 void setBuffer(AudioBuffer*, ExceptionState&);
56 AudioBuffer* buffer() { return m_buffer.get(); }
58 // numberOfChannels() returns the number of output channels. This value equals the number of channels from the buffer.
59 // If a new buffer is set with a different number of channels, then this value will dynamically change.
60 unsigned numberOfChannels();
63 void start(ExceptionState& exceptionState) { start(0, exceptionState); }
64 void start(double when, ExceptionState&);
65 void start(double when, double grainOffset, ExceptionState&);
66 void start(double when, double grainOffset, double grainDuration, ExceptionState&);
68 // Note: the attribute was originally exposed as .looping, but to be more consistent in naming with <audio>
69 // and with how it's described in the specification, the proper attribute name is .loop
70 // The old attribute is kept for backwards compatibility.
71 bool loop() const { return m_isLooping; }
72 void setLoop(bool looping) { m_isLooping = looping; }
74 // Loop times in seconds.
75 double loopStart() const { return m_loopStart; }
76 double loopEnd() const { return m_loopEnd; }
77 void setLoopStart(double loopStart) { m_loopStart = loopStart; }
78 void setLoopEnd(double loopEnd) { m_loopEnd = loopEnd; }
80 AudioParam* playbackRate() { return m_playbackRate.get(); }
82 // If a panner node is set, then we can incorporate doppler shift into the playback pitch rate.
83 void setPannerNode(PannerNode*);
84 void clearPannerNode();
86 // If we are no longer playing, propogate silence ahead to downstream nodes.
87 virtual bool propagatesSilence() const OVERRIDE;
89 // AudioScheduledSourceNode
90 virtual void finish() OVERRIDE;
92 virtual void trace(Visitor*) OVERRIDE;
95 AudioBufferSourceNode(AudioContext*, float sampleRate);
97 // Returns true on success.
98 bool renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames);
100 // Render silence starting from "index" frame in AudioBus.
101 inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess);
103 // m_buffer holds the sample data which this node outputs.
104 RefPtrWillBeMember<AudioBuffer> m_buffer;
106 // Pointers for the buffer and destination.
107 OwnPtr<const float*[]> m_sourceChannels;
108 OwnPtr<float*[]> m_destinationChannels;
110 // Used for the "playbackRate" attributes.
111 RefPtrWillBeMember<AudioParam> m_playbackRate;
113 // 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.
114 // If true, it will wrap around to the start of the buffer each time it reaches the end.
120 // m_virtualReadIndex is a sample-frame index into our buffer representing the current playback position.
121 // Since it's floating-point, it has sub-sample accuracy.
122 double m_virtualReadIndex;
126 double m_grainOffset; // in seconds
127 double m_grainDuration; // in seconds
129 // totalPitchRate() returns the instantaneous pitch rate (non-time preserving).
130 // It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node.
131 double totalPitchRate();
133 // We optionally keep track of a panner node which has a doppler shift that is incorporated into
134 // the pitch rate. We manually manage ref-counting because we want to use RefTypeConnection.
135 PannerNode* m_pannerNode;
137 // This synchronizes process() with setBuffer() which can cause dynamic channel count changes.
138 mutable Mutex m_processLock;
141 } // namespace WebCore
143 #endif // AudioBufferSourceNode_h