Move parseFontFaceDescriptor to CSSPropertyParser.cpp
[chromium-blink-merge.git] / third_party / WebKit / Source / platform / audio / AudioBus.h
blob5657f4f704597a487dfbc43edd65c7288fc59f09
1 /*
2 * Copyright (C) 2010 Google Inc. All rights reserved.
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5 * modification, are permitted provided that the following conditions
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29 #ifndef AudioBus_h
30 #define AudioBus_h
32 #include "platform/audio/AudioChannel.h"
33 #include "wtf/Noncopyable.h"
34 #include "wtf/PassOwnPtr.h"
35 #include "wtf/ThreadSafeRefCounted.h"
36 #include "wtf/Vector.h"
38 namespace blink {
40 // An AudioBus represents a collection of one or more AudioChannels.
41 // The data layout is "planar" as opposed to "interleaved".
42 // An AudioBus with one channel is mono, an AudioBus with two channels is stereo, etc.
43 class PLATFORM_EXPORT AudioBus : public ThreadSafeRefCounted<AudioBus> {
44 WTF_MAKE_NONCOPYABLE(AudioBus);
45 public:
46 enum {
47 ChannelLeft = 0,
48 ChannelRight = 1,
49 ChannelCenter = 2, // center and mono are the same
50 ChannelMono = 2,
51 ChannelLFE = 3,
52 ChannelSurroundLeft = 4,
53 ChannelSurroundRight = 5,
56 enum {
57 LayoutCanonical = 0
58 // Can define non-standard layouts here
61 enum ChannelInterpretation {
62 Speakers,
63 Discrete,
66 // allocate indicates whether or not to initially have the AudioChannels created with managed storage.
67 // Normal usage is to pass true here, in which case the AudioChannels will memory-manage their own storage.
68 // If allocate is false then setChannelMemory() has to be called later on for each channel before the AudioBus is useable...
69 static PassRefPtr<AudioBus> create(unsigned numberOfChannels, size_t length, bool allocate = true);
71 // Tells the given channel to use an externally allocated buffer.
72 void setChannelMemory(unsigned channelIndex, float* storage, size_t length);
74 // Channels
75 unsigned numberOfChannels() const { return m_channels.size(); }
77 AudioChannel* channel(unsigned channel) { return m_channels[channel].get(); }
78 const AudioChannel* channel(unsigned channel) const { return const_cast<AudioBus*>(this)->m_channels[channel].get(); }
79 AudioChannel* channelByType(unsigned type);
80 const AudioChannel* channelByType(unsigned type) const;
82 // Number of sample-frames
83 size_t length() const { return m_length; }
85 // resizeSmaller() can only be called with a new length <= the current length.
86 // The data stored in the bus will remain undisturbed.
87 void resizeSmaller(size_t newLength);
89 // Sample-rate : 0.0 if unknown or "don't care"
90 float sampleRate() const { return m_sampleRate; }
91 void setSampleRate(float sampleRate) { m_sampleRate = sampleRate; }
93 // Zeroes all channels.
94 void zero();
96 // Clears the silent flag on all channels.
97 void clearSilentFlag();
99 // Returns true if the silent bit is set on all channels.
100 bool isSilent() const;
102 // Returns true if the channel count and frame-size match.
103 bool topologyMatches(const AudioBus &sourceBus) const;
105 // Creates a new buffer from a range in the source buffer.
106 // 0 may be returned if the range does not fit in the sourceBuffer
107 static PassRefPtr<AudioBus> createBufferFromRange(const AudioBus* sourceBuffer, unsigned startFrame, unsigned endFrame);
110 // Creates a new AudioBus by sample-rate converting sourceBus to the newSampleRate.
111 // setSampleRate() must have been previously called on sourceBus.
112 // Note: sample-rate conversion is already handled in the file-reading code for the mac port, so we don't need this.
113 static PassRefPtr<AudioBus> createBySampleRateConverting(const AudioBus* sourceBus, bool mixToMono, double newSampleRate);
115 // Creates a new AudioBus by mixing all the channels down to mono.
116 // If sourceBus is already mono, then the returned AudioBus will simply be a copy.
117 static PassRefPtr<AudioBus> createByMixingToMono(const AudioBus* sourceBus);
119 // Scales all samples by the same amount.
120 void scale(float scale);
122 void reset() { m_isFirstTime = true; } // for de-zippering
124 // Copies the samples from the source bus to this one.
125 // This is just a simple per-channel copy if the number of channels match, otherwise an up-mix or down-mix is done.
126 void copyFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);
128 // Sums the samples from the source bus to this one.
129 // This is just a simple per-channel summing if the number of channels match, otherwise an up-mix or down-mix is done.
130 void sumFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);
132 // Copy each channel from sourceBus into our corresponding channel.
133 // We scale by targetGain (and our own internal gain m_busGain), performing "de-zippering" to smoothly change from *lastMixGain to (targetGain*m_busGain).
134 // The caller is responsible for setting up lastMixGain to point to storage which is unique for every "stream" which will be applied to this bus.
135 // This represents the dezippering memory.
136 void copyWithGainFrom(const AudioBus &sourceBus, float* lastMixGain, float targetGain);
138 // Copies the sourceBus by scaling with sample-accurate gain values.
139 void copyWithSampleAccurateGainValuesFrom(const AudioBus &sourceBus, float* gainValues, unsigned numberOfGainValues);
141 // Returns maximum absolute value across all channels (useful for normalization).
142 float maxAbsValue() const;
144 // Makes maximum absolute value == 1.0 (if possible).
145 void normalize();
147 static PassRefPtr<AudioBus> loadPlatformResource(const char* name, float sampleRate);
149 protected:
150 AudioBus() { }
152 AudioBus(unsigned numberOfChannels, size_t length, bool allocate);
154 void speakersCopyFrom(const AudioBus&);
155 void discreteCopyFrom(const AudioBus&);
156 void speakersSumFrom(const AudioBus&);
157 void discreteSumFrom(const AudioBus&);
158 void speakersSumFrom5_1_ToMono(const AudioBus&);
160 size_t m_length;
161 Vector<OwnPtr<AudioChannel>> m_channels;
162 int m_layout;
163 float m_busGain;
164 OwnPtr<AudioFloatArray> m_dezipperGainValues;
165 bool m_isFirstTime;
166 float m_sampleRate; // 0.0 if unknown or N/A
169 } // namespace blink
171 #endif // AudioBus_h