/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
H A D | EqualPowerPanner.cpp | 41 EqualPowerPanner::EqualPowerPanner(float sampleRate) argument 47 m_smoothingConstant = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
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H A D | HRTFDatabase.cpp | 44 PassOwnPtr<HRTFDatabase> HRTFDatabase::create(float sampleRate) argument 46 OwnPtr<HRTFDatabase> hrtfDatabase = adoptPtr(new HRTFDatabase(sampleRate)); 50 HRTFDatabase::HRTFDatabase(float sampleRate) argument 52 , m_sampleRate(sampleRate) 56 OwnPtr<HRTFElevation> hrtfElevation = HRTFElevation::createForSubject("Composite", elevation, sampleRate); 75 m_elevations[i + jj] = HRTFElevation::createByInterpolatingSlices(m_elevations[i].get(), m_elevations[j].get(), x, sampleRate);
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H A D | AudioDelayDSPKernel.cpp | 47 AudioDelayDSPKernel::AudioDelayDSPKernel(double maxDelayTime, float sampleRate) argument 48 : AudioDSPKernel(sampleRate) 57 size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate); 65 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate); 68 size_t AudioDelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const 72 return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate); 85 double AudioDelayDSPKernel::delayTime(float sampleRate) argument 87 return m_desiredDelayFrames / sampleRate; 103 float sampleRate = this->sampleRate(); local [all...] |
H A D | AudioDestination.cpp | 48 PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate) argument 50 return adoptPtr(new AudioDestination(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate)); 53 AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate) argument 58 , m_sampleRate(sampleRate) 85 m_audioDevice = adoptPtr(Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
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H A D | DynamicsCompressor.h | 70 DynamicsCompressor(float sampleRate, unsigned numberOfChannels); 79 float sampleRate() const { return m_sampleRate; } function in class:blink::DynamicsCompressor 83 double latencyTime() const { return m_compressor.latencyFrames() / static_cast<double>(sampleRate()); }
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H A D | DynamicsCompressorKernel.h | 43 DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels); 72 float sampleRate() const { return m_sampleRate; } function in class:blink::DynamicsCompressorKernel
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H A D | HRTFPanner.h | 37 HRTFPanner(float sampleRate, HRTFDatabaseLoader*); 45 static size_t fftSizeForSampleRate(float sampleRate); 47 float sampleRate() const { return m_sampleRate; } function in class:blink::FINAL
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/external/chromium_org/third_party/WebKit/Source/platform/mediastream/ |
H A D | MediaStreamSource.cpp | 90 void MediaStreamSource::setAudioFormat(size_t numberOfChannels, float sampleRate) argument 95 (*it)->setFormat(numberOfChannels, sampleRate);
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/external/chromium_org/third_party/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
H A D | WebRtcAudioRecord.java | 50 private int InitRecording(int audioSource, int sampleRate) { argument 54 sampleRate, 64 _bufferedRecSamples = sampleRate / 200; 76 sampleRate, 88 // DoLog("rec not initialized " + sampleRate); 92 // DoLog("rec sample rate set to " + sampleRate);
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H A D | WebRtcAudioTrack.java | 54 private int InitPlayback(int sampleRate) { argument 57 sampleRate, 79 sampleRate, 90 // DoLog("play not initialized " + sampleRate); 94 // DoLog("play sample rate set to " + sampleRate);
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/external/srec/audio/AudioIn/UNIX/src/ |
H A D | audioinwrapper.cpp | 60 static int sampleRate = 8000; variable 70 sampleRate = sample_rate; 100 sampleRate,
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/external/aac/libAACenc/src/ |
H A D | bandwidth.cpp | 198 const INT sampleRate, 213 switch (sampleRate) { 289 INT sampleRate, 334 *bandWidth = FDKmin(proposedBandWidth, FDKmin(20000, sampleRate>>1)); 362 sampleRate, 376 *bandWidth = FDKmin(*bandWidth, sampleRate/2); 196 GetBandwidthEntry( const INT frameLength, const INT sampleRate, const INT chanBitRate, const INT entryNo) argument 285 FDKaacEnc_DetermineBandWidth(INT* bandWidth, INT proposedBandWidth, INT bitrate, AACENC_BITRATE_MODE bitrateMode, INT sampleRate, INT frameLength, CHANNEL_MAPPING* cm, CHANNEL_MODE encoderMode) argument
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H A D | pnsparam.cpp | 188 int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int isLC) { argument 213 switch (sampleRate) { 242 INT sampleRate, 267 hUsePns = FDKaacEnc_lookUpPnsUse (bitRate, sampleRate, numChan, isLC); 279 sampleRate, 240 FDKaacEnc_GetPnsParam(NOISEPARAMS *np, INT bitRate, INT sampleRate, INT sfbCnt, const INT *sfbOffset, INT *usePns, INT numChan, const int isLC) argument
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H A D | aacenc_pns.cpp | 134 INT sampleRate, 146 sampleRate, 132 FDKaacEnc_InitPnsConfiguration(PNS_CONFIG *pnsConf, INT bitRate, INT sampleRate, INT usePns, INT sfbCnt, const INT *sfbOffset, const INT numChan, const INT isLC) argument
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/external/aac/libSYS/include/ |
H A D | wav_file.h | 147 UINT sampleRate; member in struct:WAV_HEADER 200 * \param sampleRate Desired samplerate of the resulting WAV file. 206 INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample);
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/external/chromium_org/media/base/android/java/src/org/chromium/media/ |
H A D | AudioRecordInput.java | 98 int sampleRate, int channels, int bitsPerSample, int bytesPerBuffer, 100 return new AudioRecordInput(nativeAudioRecordInputStream, sampleRate, channels, 104 private AudioRecordInput(long nativeAudioRecordInputStream, int sampleRate, int channels, argument 107 mSampleRate = sampleRate; 110 mHardwareDelayBytes = HARDWARE_DELAY_MS * sampleRate / 1000 * bitsPerSample / 8; 97 createAudioRecordInput(long nativeAudioRecordInputStream, int sampleRate, int channels, int bitsPerSample, int bytesPerBuffer, boolean usePlatformAEC) argument
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
H A D | PannerNode.h | 56 static PannerNode* create(AudioContext* context, float sampleRate) argument 58 return adoptRefCountedGarbageCollectedWillBeNoop(new PannerNode(context, sampleRate)); 113 PannerNode(AudioContext*, float sampleRate);
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H A D | AudioContext.h | 97 float sampleRate() const { return m_destinationNode->sampleRate(); } function in class:blink::AudioContext 99 AudioBuffer* createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&); 240 AudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
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H A D | AudioNode.cpp | 48 AudioNode::AudioNode(AudioContext* context, float sampleRate) argument 52 , m_sampleRate(sampleRate)
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H A D | AudioNode.h | 59 AudioNode(AudioContext*, float sampleRate); 136 virtual float sampleRate() const { return m_sampleRate; } function in class:blink::AudioNode
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H A D | AudioParamTimeline.cpp | 214 double sampleRate = context->sampleRate(); local 216 double endTime = startTime + 1.1 / sampleRate; // time just beyond one sample-frame 217 double controlRate = sampleRate / AudioNode::ProcessingSizeInFrames; // one parameter change per render quantum 218 value = valuesForTimeRange(startTime, endTime, defaultValue, &value, 1, sampleRate, controlRate); 230 double sampleRate, 243 float value = valuesForTimeRangeImpl(startTime, endTime, defaultValue, values, numberOfValues, sampleRate, controlRate); 254 double sampleRate, 277 unsigned fillToFrame = AudioUtilities::timeToSampleFrame(fillToTime - startTime, sampleRate); 307 double sampleFrameTimeIncr = 1 / sampleRate; 224 valuesForTimeRange( double startTime, double endTime, float defaultValue, float* values, unsigned numberOfValues, double sampleRate, double controlRate) argument 248 valuesForTimeRangeImpl( double startTime, double endTime, float defaultValue, float* values, unsigned numberOfValues, double sampleRate, double controlRate) argument [all...] |
/external/srec/srec_jni/ |
H A D | android_speech_srec_MicrophoneInputStream.cpp | 69 (JNIEnv *env, jclass clazz, jint sampleRate, jint fifoFrames) { 72 AUDIO_SOURCE_VOICE_RECOGNITION, sampleRate, 68 Java_android_speech_srec_Recognizer_AudioRecordNew(JNIEnv *env, jclass clazz, jint sampleRate, jint fifoFrames) argument
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/external/aac/libMpegTPEnc/src/ |
H A D | tpenc_asc.cpp | 221 INT sampleRate, 233 sampleRateIndex = getSamplingRateIndex(sampleRate); 357 static void writeSampleRate(HANDLE_FDK_BITSTREAM hBitstreamBuffer, int sampleRate) argument 359 int sampleRateIndex = getSamplingRateIndex(sampleRate); 363 FDKwriteBits( hBitstreamBuffer, sampleRate, 24 ); 219 transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs, CHANNEL_MODE channelMode, INT sampleRate, int instanceTagPCE, int profile, int matrixMixdownA, int pseudoSurroundEnable, UINT alignAnchor) argument
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/external/aac/libPCMutils/src/ |
H A D | limiter.cpp | 100 unsigned int sampleRate, maxSampleRate; member in struct:TDLimiter 164 limiter->sampleRate = maxSampleRate; 406 TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate) argument 414 if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER; 417 attack = (unsigned int)(limiter->attackMs * sampleRate / 1000); 418 release = (unsigned int)(limiter->releaseMs * sampleRate / 1000); 433 limiter->sampleRate = sampleRate; 453 attack = (unsigned int)(attackMs * limiter->sampleRate / 1000); 477 release = (unsigned int)(releaseMs * limiter->sampleRate / 100 [all...] |
/external/aac/libSYS/src/ |
H A D | wav_file.cpp | 163 FDKfread_EL(&(wav->header.sampleRate), 4, 1, wav->fp); 378 * \param sampleRate desired samplerate of the resulting WAV file 385 INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample) argument 419 wav->header.sampleRate = LittleEndian32(sampleRate); 420 wav->header.bytesPerSecond = LittleEndian32(sampleRate * wav->header.blockAlign);
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