| /external/webkit/Source/WebCore/webaudio/ |
| DefaultAudioDestinationNode.h | 49 double sampleRate() const { return m_destination->sampleRate(); }
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| OfflineAudioDestinationNode.h | 52 double sampleRate() const { return m_renderTarget->sampleRate(); }
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| AudioBuffer.h | 44 static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
47 static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, double sampleRate);
51 double duration() const { return length() / sampleRate(); }
52 double sampleRate() const { return m_sampleRate; }
69 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
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| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| bitenc.h | 35 Word32 sampleRate;
47 Word16 samplerate
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| /external/aac/libMpegTPEnc/src/ |
| tpenc_adif.cpp | 109 INT sampleRate = adif->samplingRate;
147 transportEnc_writePCE(hBs, adif->cm, sampleRate, adif->instanceTag, adif->profile, 0, 0, alignAnchor);
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| /external/webkit/Source/WebCore/platform/audio/ |
| AudioDSPKernel.h | 44 , m_sampleRate(kernelProcessor->sampleRate())
48 AudioDSPKernel(double sampleRate)
50 , m_sampleRate(sampleRate)
60 double sampleRate() const { return m_sampleRate; }
61 double nyquist() const { return 0.5 * sampleRate(); }
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| AudioProcessor.h | 44 AudioProcessor(double sampleRate)
46 , m_sampleRate(sampleRate)
66 double sampleRate() const { return m_sampleRate; }
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| HRTFDatabase.h | 49 static PassOwnPtr<HRTFDatabase> create(double sampleRate);
60 double sampleRate() const { return m_sampleRate; }
63 explicit HRTFDatabase(double sampleRate);
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| HRTFElevation.h | 53 static PassOwnPtr<HRTFElevation> createForSubject(const String& subjectName, int elevation, double sampleRate);
56 static PassOwnPtr<HRTFElevation> createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, double x, double sampleRate);
64 double sampleRate() const { return m_sampleRate; }
86 static bool calculateKernelsForAzimuthElevation(int azimuth, int elevation, double sampleRate, const String& subjectName,
92 static bool calculateSymmetricKernelsForAzimuthElevation(int azimuth, int elevation, double sampleRate, const String& subjectName,
95 HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, double sampleRate)
99 , m_sampleRate(sampleRate)
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| HRTFKernel.h | 54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost)
56 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate, bassBoost));
59 static PassRefPtr<HRTFKernel> create(PassOwnPtr<FFTFrame> fftFrame, double frameDelay, double sampleRate)
61 return adoptRef(new HRTFKernel(fftFrame, frameDelay, sampleRate));
72 double sampleRate() const { return m_sampleRate; }
73 double nyquist() const { return 0.5 * sampleRate(); }
80 HRTFKernel(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost);
82 HRTFKernel(PassOwnPtr<FFTFrame> fftFrame, double frameDelay, double sampleRate)
85 , m_sampleRate(sampleRate)
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| HRTFPanner.h | 36 explicit HRTFPanner(double sampleRate);
44 static size_t fftSizeForSampleRate(double sampleRate);
46 double sampleRate() const { return m_sampleRate; }
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| /external/webkit/Source/WebCore/platform/audio/mac/ |
| AudioDestinationMac.h | 42 AudioDestinationMac(AudioSourceProvider&, double sampleRate);
49 double sampleRate() const { return m_sampleRate; }
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| /external/jmonkeyengine/engine/src/terrain/com/jme3/terrain/noise/modulator/ |
| CatRom2.java | 39 private int sampleRate = 100;
45 public CatRom2(final int sampleRate) {
46 this.sampleRate = sampleRate;
47 this.table = new float[4 * sampleRate + 1];
48 for (int i = 0; i < 4 * sampleRate + 1; i++) {
49 float x = i / (float) sampleRate;
59 public static CatRom2 getInstance(final int sampleRate) {
60 if (!CatRom2.instances.containsKey(sampleRate)) {
61 CatRom2.instances.put(sampleRate, new CatRom2(sampleRate));
[all...] |
| /external/webkit/Source/WebCore/bindings/js/ |
| JSAudioContextCustom.cpp | 64 // new AudioContext(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
70 float sampleRate = exec->argument(2).toFloat(exec);
72 audioContext = AudioContext::createOfflineContext(document, numberOfChannels, numberOfFrames, sampleRate);
107 // AudioBuffer createBuffer(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
113 float sampleRate = exec->argument(2).toFloat(exec);
115 RefPtr<AudioBuffer> audioBuffer = audioContext->createBuffer(numberOfChannels, numberOfFrames, sampleRate);
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| /external/webkit/Source/WebCore/bindings/v8/custom/ |
| V8AudioContextCustom.cpp | 61 // new AudioContext(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
75 float sampleRate = toFloat(args[2]);
77 audioContext = AudioContext::createOfflineContext(document, numberOfChannels, numberOfFrames, sampleRate);
119 // AudioBuffer createBuffer(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
133 float sampleRate = toFloat(args[2]);
135 RefPtr<AudioBuffer> audioBuffer = audioContext->createBuffer(numberOfChannels, numberOfFrames, sampleRate);
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| /external/webkit/Source/WebKit/chromium/src/ |
| AudioDestinationChromium.h | 45 AudioDestinationChromium(AudioSourceProvider&, double sampleRate);
52 double sampleRate() const { return m_sampleRate; }
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| WebAudioBus.cpp | 47 void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double sampleRate)
51 audioBus->setSampleRate(sampleRate);
95 double WebAudioBus::sampleRate() const
100 return m_private->sampleRate();
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| /frameworks/av/media/libnbaio/ |
| AudioStreamInSource.cpp | 47 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
50 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
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| AudioStreamOutSink.cpp | 44 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
47 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
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| /frameworks/av/media/libstagefright/codecs/common/include/ |
| voAAC.h | 45 int sampleRate; /*! audio file sample rate */
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| /external/aac/libSYS/include/ |
| wav_file.h | 127 UINT sampleRate;
179 * \param sampleRate Desired samplerate of the resulting WAV file.
185 INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample);
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| /external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
| AudioData.java | 47 protected int sampleRate;
92 return sampleRate;
99 * @param sampleRate Sample rate, 44100, 22050, etc.
101 public void setupFormat(int channels, int bitsPerSample, int sampleRate){
107 this.sampleRate = sampleRate;
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| /external/sonivox/arm-fm-22k/lib_src/ |
| eas_pcm.h | 45 EAS_U32 sampleRate;
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| /external/sonivox/arm-hybrid-22k/lib_src/ |
| eas_pcm.h | 45 EAS_U32 sampleRate;
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| /external/sonivox/arm-wt-22k/lib_src/ |
| eas_pcm.h | 45 EAS_U32 sampleRate;
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