/external/webkit/Source/WebCore/webaudio/ |
DefaultAudioDestinationNode.h | 49 double sampleRate() const { return m_destination->sampleRate(); }
|
OfflineAudioDestinationNode.h | 52 double sampleRate() const { return m_renderTarget->sampleRate(); }
|
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);
|
/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
bitenc.h | 35 Word32 sampleRate; 47 Word16 samplerate
|
/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);
|
/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(); }
|
AudioProcessor.h | 44 AudioProcessor(double sampleRate) 46 , m_sampleRate(sampleRate) 66 double sampleRate() const { return m_sampleRate; }
|
HRTFDatabase.h | 49 static PassOwnPtr<HRTFDatabase> create(double sampleRate); 60 double sampleRate() const { return m_sampleRate; } 63 explicit HRTFDatabase(double sampleRate);
|
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)
|
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)
|
HRTFPanner.h | 36 explicit HRTFPanner(double sampleRate); 44 static size_t fftSizeForSampleRate(double sampleRate); 46 double sampleRate() const { return m_sampleRate; }
|
/external/webkit/Source/WebCore/platform/audio/mac/ |
AudioDestinationMac.h | 42 AudioDestinationMac(AudioSourceProvider&, double sampleRate); 49 double sampleRate() const { return m_sampleRate; }
|
/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);
|
/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);
|
/external/webkit/Source/WebKit/chromium/src/ |
AudioDestinationChromium.h | 45 AudioDestinationChromium(AudioSourceProvider&, double sampleRate); 52 double sampleRate() const { return m_sampleRate; }
|
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();
|
/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));
|
AudioStreamOutSink.cpp | 44 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); 47 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
|
/frameworks/av/media/libstagefright/codecs/common/include/ |
voAAC.h | 45 int sampleRate; /*! audio file sample rate */
|
/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);
|
/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;
|
/external/sonivox/arm-fm-22k/lib_src/ |
eas_pcm.h | 45 EAS_U32 sampleRate;
|
/external/sonivox/arm-hybrid-22k/lib_src/ |
eas_pcm.h | 45 EAS_U32 sampleRate;
|
/external/sonivox/arm-wt-22k/lib_src/ |
eas_pcm.h | 45 EAS_U32 sampleRate;
|