| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| HRTFDatabaseLoader.h | 50 static PassRefPtr<HRTFDatabaseLoader> createAndLoadAsynchronouslyIfNecessary(float sampleRate);
70 explicit HRTFDatabaseLoader(float sampleRate);
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| 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);
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| AudioDestinationConsumer.h | 45 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
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| EqualPowerPanner.h | 36 EqualPowerPanner(float sampleRate);
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| /external/chromium_org/third_party/WebKit/Source/platform/mediastream/ |
| MediaStreamSource.cpp | 91 void MediaStreamSource::setAudioFormat(size_t numberOfChannels, float sampleRate)
96 (*it)->setFormat(numberOfChannels, sampleRate);
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| /external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
| AudioBuffer.java | 71 int bytesPerSec = (bitsPerSample / 8) * channels * sampleRate;
82 ", rate="+sampleRate+", duration="+getDuration()+"]";
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| /frameworks/av/services/audioflinger/ |
| AudioResamplerCubic.h | 31 AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) :
32 AudioResampler(bitDepth, inChannelCount, sampleRate, MED_QUALITY) {
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| TrackBase.h | 42 uint32_t sampleRate,
79 virtual uint32_t sampleRate() const { return mSampleRate; }
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| FastMixer.cpp | 75 unsigned sampleRate = 0;
233 sampleRate = 0;
236 sampleRate = Format_sampleRate(format);
239 dumpState->mSampleRate = sampleRate;
248 if (frameCount > 0 && sampleRate > 0) {
252 mixer = new AudioMixer(frameCount, sampleRate, FastMixerState::kMaxFastTracks);
254 periodNs = (frameCount * 1000000000LL) / sampleRate; // 1.00
255 underrunNs = (frameCount * 1750000000LL) / sampleRate; // 1.75
256 overrunNs = (frameCount * 500000000LL) / sampleRate; // 0.50
257 forceNs = (frameCount * 950000000LL) / sampleRate; // 0.9
[all...] |
| /external/chromium_org/content/renderer/media/ |
| renderer_webaudiodevice_impl.h | 33 virtual double sampleRate();
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| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| AudioBuffer.idl | 34 readonly attribute float sampleRate; // in sample-frames per second
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| DelayDSPKernel.h | 42 virtual double delayTime(float sampleRate) OVERRIDE;
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| DelayProcessor.h | 39 DelayProcessor(AudioContext*, float sampleRate, unsigned numberOfChannels, double maxDelayTime);
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| /external/chromium_org/third_party/WebKit/public/platform/ |
| WebAudioDestinationConsumer.h | 36 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
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| WebAudioDevice.h | 53 virtual double sampleRate() = 0;
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| /frameworks/av/include/media/ |
| IAudioFlinger.h | 64 uint32_t sampleRate,
82 uint32_t sampleRate,
94 virtual uint32_t sampleRate(audio_io_handle_t output) const = 0;
142 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
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| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_BypassMix.c | 113 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
121 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
172 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
174 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
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| LVCS_ReverbGenerator.c | 79 if(pInstance->Params.SampleRate != pParams->SampleRate ) /* Sample rate change test */
85 Delay = (LVM_UINT16)LVCS_StereoDelayCS[(LVM_UINT16)pParams->SampleRate];
97 Offset = (LVM_UINT16)pParams->SampleRate;
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| /frameworks/av/media/libeffects/testlibs/ |
| AudioBiquadFilter.h | 44 // sampleRate Sample rate, in Hz.
45 AudioBiquadFilter(int nChannels, int sampleRate);
49 // sampleRate Sample rate, in Hz.
50 void configure(int nChannels, int sampleRate);
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| /frameworks/av/media/libnbaio/ |
| LibsndfileSource.cpp | 27 NBAIO_Source(Format_from_SR_C(sfinfo.samplerate, sfinfo.channels)),
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| /frameworks/av/media/libstagefright/codecs/aacenc/ |
| AACEncoder.cpp | 84 params.sampleRate = mSampleRate;
96 static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) {
103 if (sampleRate == kSampleRateTable[i]) {
109 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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| /frameworks/av/media/libstagefright/codecs/common/include/ |
| voAAC.h | 45 int sampleRate; /*! audio file sample rate */
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| /frameworks/wilhelm/src/desktop/ |
| SndFile.c | 116 switch (sfinfo->samplerate) {
220 audioPlayer->mSndFile.mSfInfo.samplerate) / 1000LL), SEEK_SET);
262 1000LL) / thiz->mSndFile.mSfInfo.samplerate);
264 thiz->mSampleRateMilliHz = thiz->mSndFile.mSfInfo.samplerate * 1000;
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| /packages/apps/VoiceDialer/src/com/android/voicedialer/ |
| RecognizerLogger.java | 211 * @param sampleRate
214 public InputStream logInputStream(final InputStream inputStream, final int sampleRate) {
215 final ByteArrayOutputStream baos = new ByteArrayOutputStream(sampleRate * 2 * 20);
251 (short)1, sampleRate, (short)16, pcm.length);
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| /system/media/audio_utils/include/audio_utils/ |
| sndfile.h | 36 int samplerate; member in struct:__anon61058
|
