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  /frameworks/av/media/libeffects/testlibs/
AudioBiquadFilter.cpp 28 AudioBiquadFilter::AudioBiquadFilter(int nChannels, int sampleRate) {
29 configure(nChannels, sampleRate);
33 void AudioBiquadFilter::configure(int nChannels, int sampleRate) {
35 assert(sampleRate > 0);
39 / sampleRate;
  /external/chromium_org/third_party/WebKit/Source/modules/webaudio/
AudioParamTimeline.h 61 // It should equal sampleRate for sample-accurate parameter changes, and otherwise will usually match
63 float valuesForTimeRange(double startTime, double endTime, float defaultValue, float* values, unsigned numberOfValues, double sampleRate, double controlRate);
106 float valuesForTimeRangeImpl(double startTime, double endTime, float defaultValue, float* values, unsigned numberOfValues, double sampleRate, double controlRate);
BiquadFilterNode.cpp 33 BiquadFilterNode::BiquadFilterNode(AudioContext* context, float sampleRate)
34 : AudioBasicProcessorNode(context, sampleRate)
38 m_processor = adoptPtr(new BiquadProcessor(context, sampleRate, 1, false));
BiquadProcessor.cpp 35 BiquadProcessor::BiquadProcessor(AudioContext* context, float sampleRate, size_t numberOfChannels, bool autoInitialize)
36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
45 double nyquist = 0.5 * this->sampleRate();
DynamicsCompressorNode.cpp 41 DynamicsCompressorNode::DynamicsCompressorNode(AudioContext* context, float sampleRate)
42 : AudioNode(context, sampleRate)
99 m_dynamicsCompressor = adoptPtr(new DynamicsCompressor(sampleRate(), defaultNumberOfOutputChannels));
MediaStreamAudioSourceNode.cpp 44 : AudioSourceNode(context, context->sampleRate())
66 if (numberOfChannels != m_sourceNumberOfChannels || sourceSampleRate != sampleRate()) {
68 if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels() || sourceSampleRate != sampleRate()) {
MediaStreamAudioDestinationNode.cpp 46 : AudioBasicInspectorNode(context, context->sampleRate(), numberOfChannels)
59 m_source->setAudioFormat(numberOfChannels, context->sampleRate());
OfflineAudioDestinationNode.h 56 virtual float sampleRate() const { return m_renderTarget->sampleRate(); }
ScriptProcessorNode.cpp 59 PassRefPtr<ScriptProcessorNode> ScriptProcessorNode::create(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
87 return adoptRef(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)
126 float sampleRate = context()->sampleRate();
131 RefPtr<AudioBuffer> inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0;
132 RefPtr<AudioBuffer> outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0;
WaveShaperNode.cpp 39 : AudioBasicProcessorNode(context, context->sampleRate())
42 m_processor = adoptPtr(new WaveShaperProcessor(context->sampleRate(), 1));
  /external/chromium_org/third_party/WebKit/Source/platform/audio/
DynamicsCompressor.h 70 DynamicsCompressor(float sampleRate, unsigned numberOfChannels);
79 float sampleRate() const { return m_sampleRate; }
83 double latencyTime() const { return m_compressor.latencyFrames() / static_cast<double>(sampleRate()); }
HRTFDatabase.h 46 static PassOwnPtr<HRTFDatabase> create(float sampleRate);
57 float sampleRate() const { return m_sampleRate; }
63 explicit HRTFDatabase(float sampleRate);
HRTFPanner.h 37 HRTFPanner(float sampleRate, HRTFDatabaseLoader*);
45 static size_t fftSizeForSampleRate(float sampleRate);
47 float sampleRate() const { return m_sampleRate; }
  /frameworks/av/libvideoeditor/vss/src/
VideoEditorResampler.cpp 79 M4OSA_Int32 sampleRate, M4OSA_Int32 quality) {
83 bitDepth, inChannelCount, sampleRate);
90 context->outSamplingRate = sampleRate;
  /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/
LVCS_Equaliser.c 72 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
78 Offset = (LVM_UINT16)(pParams->SampleRate + (pParams->SpeakerType * (1+LVM_FS_48000)));
  /frameworks/av/media/libstagefright/rtsp/
ARawAudioAssembler.cpp 134 int32_t sampleRate, numChannels;
136 desc, &sampleRate, &numChannels);
138 format->setInt32(kKeySampleRate, sampleRate);
  /hardware/libhardware_legacy/audio/
AudioHardwareInterface.cpp 113 size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
115 if (sampleRate != 8000) {
116 ALOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
  /hardware/qcom/audio/legacy/alsa_sound/
AudioUsbALSA.h 90 struct pcm * configureDevice(unsigned flags, char* hw, int sampleRate, int channelCount, int periodSize, bool playback);
109 status_t setHardwareParams(pcm *local_handle, uint32_t sampleRate, uint32_t channels, int periodSize);
115 status_t getCap(char * type, int &channels, int &sampleRate);
  /frameworks/av/media/libstagefright/
AudioSource.cpp 51 audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount)
53 mSampleRate(sampleRate),
57 ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
62 sampleRate,
77 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
  /external/chromium_org/third_party/WebKit/Source/platform/audio/chromium/
AudioDestinationChromium.h 50 AudioDestinationChromium(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate);
57 float sampleRate() const { return m_sampleRate; }
  /external/chromium_org/third_party/WebKit/Source/testing/runner/
WebTestInterfaces.cpp 113 WebAudioDevice* WebTestInterfaces::createAudioDevice(double sampleRate)
115 return new MockWebAudioDevice(sampleRate);
  /external/chromium_org/third_party/WebKit/public/platform/
WebAudioBus.h 50 void initialize(unsigned numberOfChannels, size_t length, double sampleRate);
61 double sampleRate() const;
  /frameworks/av/media/libeffects/lvm/lib/Common/src/
LVC_Mixer_SetTimeConstant.c 29 /* Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) */
74 pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
LVC_Mixer_VarSlope_SetTimeConstant.c 30 /* Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) */
94 pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
  /frameworks/av/media/libnbaio/
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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