| /external/webkit/Source/WebCore/webaudio/ |
| AudioChannelMerger.cpp | 44 AudioChannelMerger::AudioChannelMerger(AudioContext* context, double sampleRate)
45 : AudioNode(context, sampleRate)
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| AudioGainNode.cpp | 37 AudioGainNode::AudioGainNode(AudioContext* context, double sampleRate)
38 : AudioNode(context, sampleRate)
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| JavaScriptAudioNode.h | 55 static PassRefPtr<JavaScriptAudioNode> create(AudioContext*, double sampleRate, size_t bufferSize, unsigned numberOfInputs = 1, unsigned numberOfOutputs = 1);
80 JavaScriptAudioNode(AudioContext*, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs);
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| AudioBuffer.idl | 35 readonly attribute float sampleRate; // in sample-frames per second
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| AudioPannerNode.cpp | 49 AudioPannerNode::AudioPannerNode(AudioContext* context, double sampleRate)
50 : AudioNode(context, sampleRate)
134 m_panner = Panner::create(m_panningModel, sampleRate());
156 OwnPtr<Panner> newPanner = Panner::create(model, sampleRate());
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| DelayDSPKernel.h | 39 DelayDSPKernel(double maxDelayTime, double sampleRate);
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| DelayProcessor.h | 40 DelayProcessor(double sampleRate, unsigned numberOfChannels);
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| AudioPannerNode.h | 56 static PassRefPtr<AudioPannerNode> create(AudioContext* context, double sampleRate)
58 return adoptRef(new AudioPannerNode(context, sampleRate));
120 AudioPannerNode(AudioContext*, double sampleRate);
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| /external/webkit/Source/WebKit/android/plugins/ |
| ANPSoundInterface.cpp | 90 static ANPAudioTrack* ANPCreateTrack(uint32_t sampleRate,
101 sampleRate,
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| /frameworks/av/services/audioflinger/ |
| AudioResampler.h | 45 int32_t sampleRate, int quality=DEFAULT);
73 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
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| AudioResamplerCubic.h | 31 AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) :
32 AudioResampler(bitDepth, inChannelCount, sampleRate) {
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| LibsndfileSource.cpp | 27 NBAIO_Source(Format_from_SR_C(sfinfo.samplerate, sfinfo.channels)),
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| /external/webkit/Source/WebCore/platform/audio/ |
| EqualPowerPanner.h | 36 EqualPowerPanner(double sampleRate);
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| Panner.h | 50 static PassOwnPtr<Panner> create(PanningModel model, double sampleRate);
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| /external/webkit/Source/WebKit/chromium/public/ |
| WebAudioDevice.h | 51 virtual double sampleRate() = 0;
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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/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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| /external/srec/audio/AudioIn/UNIX/src/ |
| audioinwrapper.cpp | 60 static int sampleRate = 8000;
70 sampleRate = sample_rate;
100 sampleRate,
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| /external/srec/srec/include/ |
| sample.h | 181 int samplerate; member in struct:__anon12652
215 void add_riff_header(PFile* waveFile, int samplerate, int bitspersample);
216 void fix_riff_header(PFile* waveFile, int samplerate, int bitspersample);
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| utteranc.h | 221 unsigned long sampleRate;
229 int update_utb_header(file_utterance_info *utt, int frames, int samplerate,
231 void init_utt_v5_header(UttHeader *uhead, int dim, int samplerate, int framerate);
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