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  /external/chromium_org/content/shell/renderer/test_runner/
mock_web_audio_device.cc 18 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/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 47 AudioProcessor(float sampleRate, unsigned numberOfChannels)
50 , m_sampleRate(sampleRate)
72 float sampleRate() const { return m_sampleRate; }
AudioDestination.h 50 AudioDestination(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate);
55 static PassOwnPtr<AudioDestination> create(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate);
61 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,
90 HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, float sampleRate)
94 , 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)
AudioBus.h 90 float sampleRate() const { return m_sampleRate; }
91 void setSampleRate(float sampleRate) { m_sampleRate = sampleRate; }
147 static PassRefPtr<AudioBus> loadPlatformResource(const char* name, float 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 74 double RendererWebAudioDeviceImpl::sampleRate() {
  /external/chromium_org/third_party/WebKit/Source/modules/webaudio/
AudioBuffer.h 46 static AudioBuffer* create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
47 static AudioBuffer* create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
50 static AudioBuffer* createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate);
56 double duration() const { return length() / static_cast<double>(sampleRate()); }
57 float sampleRate() const { return m_sampleRate; }
70 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
AudioScheduledSourceNode.cpp 44 AudioScheduledSourceNode::AudioScheduledSourceNode(AudioContext* context, float sampleRate)
45 : AudioSourceNode(context, sampleRate)
66 double sampleRate = this->sampleRate();
74 size_t startFrame = AudioUtilities::timeToSampleFrame(m_startTime, sampleRate);
75 size_t endFrame = m_endTime == UnknownTime ? 0 : AudioUtilities::timeToSampleFrame(m_endTime, sampleRate);
AudioParam.cpp 158 double sampleRate = context()->sampleRate();
160 double endTime = startTime + numberOfValues / sampleRate;
164 m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues, sampleRate, sampleRate);
ScriptProcessorNode.cpp 59 ScriptProcessorNode* ScriptProcessorNode::create(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
87 return adoptRefCountedGarbageCollectedWillBeNoop(new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels));
90 ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
91 : AudioNode(context, sampleRate)
130 float sampleRate = context()->sampleRate();
135 AudioBuffer* inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0;
136 AudioBuffer* outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0;
259 double playbackTime = (context()->currentSampleFrame() + m_bufferSize) / static_cast<double>(context()->sampleRate());
  /external/chromium_org/third_party/WebKit/Source/platform/exported/
WebAudioBus.cpp 47 void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double sampleRate)
51 audioBus->setSampleRate(sampleRate);
112 double WebAudioBus::sampleRate() const
117 return m_private->sampleRate();
  /frameworks/av/media/libnbaio/
AudioStreamInSource.cpp 46 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
49 mFormat = Format_from_SR_C(sampleRate,
AudioStreamOutSink.cpp 43 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
46 mFormat = Format_from_SR_C(sampleRate,
  /frameworks/av/media/libstagefright/codecs/common/include/
voAAC.h 45 int sampleRate; /*! audio file sample rate */
  /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 136 int32_t sampleRate, numChannels;
138 desc, &sampleRate, &numChannels);
140 format->setInt32(kKeySampleRate, sampleRate);
  /hardware/libhardware_legacy/audio/
AudioHardwareStub.h 33 virtual uint32_t sampleRate() const { return 44100; }
50 virtual uint32_t sampleRate() const { return 8000; }
83 uint32_t *sampleRate=0,
91 uint32_t *sampleRate,

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