| /external/chromium_org/third_party/WebKit/Source/testing/runner/ |
| MockWebAudioDevice.cpp | 37 MockWebAudioDevice::MockWebAudioDevice(double sampleRate)
38 : m_sampleRate(sampleRate)
54 double MockWebAudioDevice::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/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
| AudioDSPKernel.h | 44 , m_sampleRate(kernelProcessor->sampleRate())
48 AudioDSPKernel(float sampleRate)
50 , m_sampleRate(sampleRate)
60 float sampleRate() const { return m_sampleRate; }
61 double nyquist() const { return 0.5 * sampleRate(); }
|
| AudioProcessor.h | 44 AudioProcessor(float sampleRate, unsigned numberOfChannels)
47 , m_sampleRate(sampleRate)
68 float sampleRate() const { return m_sampleRate; }
|
| HRTFDatabase.h | 46 static PassOwnPtr<HRTFDatabase> create(float sampleRate);
57 float sampleRate() const { return m_sampleRate; }
63 explicit HRTFDatabase(float sampleRate);
|
| HRTFElevation.h | 53 static PassOwnPtr<HRTFElevation> createForSubject(const String& subjectName, int elevation, float sampleRate);
56 static PassOwnPtr<HRTFElevation> createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate);
64 float sampleRate() const { return m_sampleRate; }
86 static bool calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
92 static bool calculateSymmetricKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
96 HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, float sampleRate)
100 , m_sampleRate(sampleRate)
|
| HRTFKernel.h | 54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, float sampleRate)
56 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate));
59 static PassRefPtr<HRTFKernel> create(PassOwnPtr<FFTFrame> fftFrame, float frameDelay, float sampleRate)
61 return adoptRef(new HRTFKernel(fftFrame, frameDelay, sampleRate));
72 float sampleRate() const { return m_sampleRate; }
73 double nyquist() const { return 0.5 * sampleRate(); }
80 HRTFKernel(AudioChannel*, size_t fftSize, float sampleRate);
82 HRTFKernel(PassOwnPtr<FFTFrame> fftFrame, float frameDelay, float sampleRate)
85 , m_sampleRate(sampleRate)
|
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| OfflineAudioDestinationNode.h | 56 virtual float sampleRate() const { return m_renderTarget->sampleRate(); }
|
| AudioBuffer.h | 46 static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
49 static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate);
53 double duration() const { return length() / sampleRate(); }
54 float sampleRate() const { return m_sampleRate; }
72 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
|
| PeriodicWave.h | 45 static PassRefPtr<PeriodicWave> createSine(float sampleRate);
46 static PassRefPtr<PeriodicWave> createSquare(float sampleRate);
47 static PassRefPtr<PeriodicWave> createSawtooth(float sampleRate);
48 static PassRefPtr<PeriodicWave> createTriangle(float sampleRate);
51 static PassRefPtr<PeriodicWave> create(float sampleRate, Float32Array* real, Float32Array* imag);
65 float sampleRate() const { return m_sampleRate; }
68 explicit PeriodicWave(float sampleRate);
|
| AsyncAudioDecoder.h | 50 void decodeAsync(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback);
56 static PassOwnPtr<DecodingTask> create(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback);
61 DecodingTask(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback);
64 float sampleRate() const { return m_sampleRate; }
|
| /frameworks/base/tests/Camera2Tests/SmartCamera/SimpleCamera/src/androidx/media/filterfw/decoder/ |
| AudioSample.java | 21 public final int sampleRate;
25 public AudioSample(int sampleRate, int channelCount, byte[] bytes) {
26 this.sampleRate = sampleRate;
|
| /external/chromium_org/content/renderer/media/ |
| renderer_webaudiodevice_impl.cc | 70 double RendererWebAudioDeviceImpl::sampleRate() {
|
| /external/chromium_org/third_party/WebKit/Source/core/platform/audio/chromium/ |
| AudioDestinationChromium.h | 48 AudioDestinationChromium(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate);
55 float sampleRate() const { return m_sampleRate; }
|
| /external/chromium_org/third_party/WebKit/Source/core/platform/chromium/support/ |
| WebAudioBus.cpp | 49 void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double sampleRate)
53 audioBus->setSampleRate(sampleRate);
114 double WebAudioBus::sampleRate() const
119 return m_private->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...] |
| /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/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;
|
| /frameworks/av/media/libstagefright/rtsp/ |
| ARawAudioAssembler.cpp | 134 int32_t sampleRate, numChannels;
136 desc, &sampleRate, &numChannels);
138 format->setInt32(kKeySampleRate, sampleRate);
|
