1 | /* |
2 | * Copyright (C) 2010, Google Inc. All rights reserved. |
3 | * |
4 | * Redistribution and use in source and binary forms, with or without |
5 | * modification, are permitted provided that the following conditions |
6 | * are met: |
7 | * 1. Redistributions of source code must retain the above copyright |
8 | * notice, this list of conditions and the following disclaimer. |
9 | * 2. Redistributions in binary form must reproduce the above copyright |
10 | * notice, this list of conditions and the following disclaimer in the |
11 | * documentation and/or other materials provided with the distribution. |
12 | * |
13 | * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY |
14 | * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED |
15 | * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE |
16 | * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY |
17 | * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES |
18 | * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; |
19 | * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON |
20 | * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT |
21 | * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS |
22 | * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
23 | */ |
24 | |
25 | #pragma once |
26 | |
27 | #include "AudioScheduledSourceNode.h" |
28 | #include <wtf/Lock.h> |
29 | #include <wtf/UniqueArray.h> |
30 | |
31 | namespace WebCore { |
32 | |
33 | class AudioBuffer; |
34 | class PannerNode; |
35 | |
36 | // AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer. |
37 | // It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways). |
38 | |
39 | class AudioBufferSourceNode final : public AudioScheduledSourceNode { |
40 | WTF_MAKE_ISO_ALLOCATED(AudioBufferSourceNode); |
41 | public: |
42 | static Ref<AudioBufferSourceNode> create(AudioContext&, float sampleRate); |
43 | |
44 | virtual ~AudioBufferSourceNode(); |
45 | |
46 | // AudioNode |
47 | void process(size_t framesToProcess) final; |
48 | void reset() final; |
49 | |
50 | // setBuffer() is called on the main thread. This is the buffer we use for playback. |
51 | // returns true on success. |
52 | void setBuffer(RefPtr<AudioBuffer>&&); |
53 | AudioBuffer* buffer() { return m_buffer.get(); } |
54 | |
55 | // numberOfChannels() returns the number of output channels. This value equals the number of channels from the buffer. |
56 | // If a new buffer is set with a different number of channels, then this value will dynamically change. |
57 | unsigned numberOfChannels(); |
58 | |
59 | // Play-state |
60 | ExceptionOr<void> start(double when, double grainOffset, Optional<double> grainDuration); |
61 | |
62 | // Note: the attribute was originally exposed as .looping, but to be more consistent in naming with <audio> |
63 | // and with how it's described in the specification, the proper attribute name is .loop |
64 | // The old attribute is kept for backwards compatibility. |
65 | bool loop() const { return m_isLooping; } |
66 | void setLoop(bool looping) { m_isLooping = looping; } |
67 | |
68 | // Loop times in seconds. |
69 | double loopStart() const { return m_loopStart; } |
70 | double loopEnd() const { return m_loopEnd; } |
71 | void setLoopStart(double loopStart) { m_loopStart = loopStart; } |
72 | void setLoopEnd(double loopEnd) { m_loopEnd = loopEnd; } |
73 | |
74 | // Deprecated. |
75 | bool looping(); |
76 | void setLooping(bool); |
77 | |
78 | AudioParam* gain() { return m_gain.get(); } |
79 | AudioParam* playbackRate() { return m_playbackRate.get(); } |
80 | |
81 | // If a panner node is set, then we can incorporate doppler shift into the playback pitch rate. |
82 | void setPannerNode(PannerNode*); |
83 | void clearPannerNode(); |
84 | |
85 | // If we are no longer playing, propogate silence ahead to downstream nodes. |
86 | bool propagatesSilence() const final; |
87 | |
88 | // AudioScheduledSourceNode |
89 | void finish() final; |
90 | |
91 | private: |
92 | AudioBufferSourceNode(AudioContext&, float sampleRate); |
93 | |
94 | double tailTime() const final { return 0; } |
95 | double latencyTime() const final { return 0; } |
96 | |
97 | enum BufferPlaybackMode { |
98 | Entire, |
99 | Partial |
100 | }; |
101 | |
102 | ExceptionOr<void> startPlaying(BufferPlaybackMode, double when, double grainOffset, double grainDuration); |
103 | |
104 | // Returns true on success. |
105 | bool renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames); |
106 | |
107 | // Render silence starting from "index" frame in AudioBus. |
108 | inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess); |
109 | |
110 | // m_buffer holds the sample data which this node outputs. |
111 | RefPtr<AudioBuffer> m_buffer; |
112 | |
113 | // Pointers for the buffer and destination. |
114 | UniqueArray<const float*> m_sourceChannels; |
115 | UniqueArray<float*> m_destinationChannels; |
116 | |
117 | // Used for the "gain" and "playbackRate" attributes. |
118 | RefPtr<AudioParam> m_gain; |
119 | RefPtr<AudioParam> m_playbackRate; |
120 | |
121 | // 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. |
122 | // If true, it will wrap around to the start of the buffer each time it reaches the end. |
123 | bool m_isLooping; |
124 | |
125 | double m_loopStart; |
126 | double m_loopEnd; |
127 | |
128 | // m_virtualReadIndex is a sample-frame index into our buffer representing the current playback position. |
129 | // Since it's floating-point, it has sub-sample accuracy. |
130 | double m_virtualReadIndex; |
131 | |
132 | // Granular playback |
133 | bool m_isGrain; |
134 | double m_grainOffset; // in seconds |
135 | double m_grainDuration; // in seconds |
136 | |
137 | // totalPitchRate() returns the instantaneous pitch rate (non-time preserving). |
138 | // It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node. |
139 | double totalPitchRate(); |
140 | |
141 | // m_lastGain provides continuity when we dynamically adjust the gain. |
142 | float m_lastGain; |
143 | |
144 | // We optionally keep track of a panner node which has a doppler shift that is incorporated into |
145 | // the pitch rate. We manually manage ref-counting because we want to use RefTypeConnection. |
146 | PannerNode* m_pannerNode; |
147 | |
148 | // This synchronizes process() with setBuffer() which can cause dynamic channel count changes. |
149 | mutable Lock m_processMutex; |
150 | }; |
151 | |
152 | } // namespace WebCore |
153 | |