Searched defs:sampleRate (Results 51 - 75 of 129) sorted by relevance

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/external/chromium_org/third_party/WebKit/Source/platform/audio/
H A DEqualPowerPanner.cpp41 EqualPowerPanner::EqualPowerPanner(float sampleRate) argument
47 m_smoothingConstant = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
H A DHRTFDatabase.cpp44 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);
H A DAudioDelayDSPKernel.cpp47 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
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H A DAudioDestination.cpp48 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));
H A DDynamicsCompressor.h70 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()); }
H A DDynamicsCompressorKernel.h43 DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels);
72 float sampleRate() const { return m_sampleRate; } function in class:blink::DynamicsCompressorKernel
H A DHRTFPanner.h37 HRTFPanner(float sampleRate, HRTFDatabaseLoader*);
45 static size_t fftSizeForSampleRate(float sampleRate);
47 float sampleRate() const { return m_sampleRate; } function in class:blink::FINAL
/external/chromium_org/third_party/WebKit/Source/platform/mediastream/
H A DMediaStreamSource.cpp90 void MediaStreamSource::setAudioFormat(size_t numberOfChannels, float sampleRate) argument
95 (*it)->setFormat(numberOfChannels, sampleRate);
/external/chromium_org/third_party/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/
H A DWebRtcAudioRecord.java50 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);
H A DWebRtcAudioTrack.java54 private int InitPlayback(int sampleRate) { argument
57 sampleRate,
79 sampleRate,
90 // DoLog("play not initialized " + sampleRate);
94 // DoLog("play sample rate set to " + sampleRate);
/external/srec/audio/AudioIn/UNIX/src/
H A Daudioinwrapper.cpp60 static int sampleRate = 8000; variable
70 sampleRate = sample_rate;
100 sampleRate,
/external/aac/libAACenc/src/
H A Dbandwidth.cpp198 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
H A Dpnsparam.cpp188 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
H A Daacenc_pns.cpp134 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
/external/aac/libSYS/include/
H A Dwav_file.h147 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);
/external/chromium_org/media/base/android/java/src/org/chromium/media/
H A DAudioRecordInput.java98 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
/external/chromium_org/third_party/WebKit/Source/modules/webaudio/
H A DPannerNode.h56 static PannerNode* create(AudioContext* context, float sampleRate) argument
58 return adoptRefCountedGarbageCollectedWillBeNoop(new PannerNode(context, sampleRate));
113 PannerNode(AudioContext*, float sampleRate);
H A DAudioContext.h97 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);
H A DAudioNode.cpp48 AudioNode::AudioNode(AudioContext* context, float sampleRate) argument
52 , m_sampleRate(sampleRate)
H A DAudioNode.h59 AudioNode(AudioContext*, float sampleRate);
136 virtual float sampleRate() const { return m_sampleRate; } function in class:blink::AudioNode
H A DAudioParamTimeline.cpp214 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
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/external/srec/srec_jni/
H A Dandroid_speech_srec_MicrophoneInputStream.cpp69 (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
/external/aac/libMpegTPEnc/src/
H A Dtpenc_asc.cpp221 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
/external/aac/libPCMutils/src/
H A Dlimiter.cpp100 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
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/external/aac/libSYS/src/
H A Dwav_file.cpp163 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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