Searched defs:sampleRate (Results 76 - 100 of 129) sorted by relevance

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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/
H A DOscillatorNode.cpp44 OscillatorNode* OscillatorNode::create(AudioContext* context, float sampleRate) argument
46 return adoptRefCountedGarbageCollectedWillBeNoop(new OscillatorNode(context, sampleRate));
49 OscillatorNode::OscillatorNode(AudioContext* context, float sampleRate) argument
50 : AudioScheduledSourceNode(context, sampleRate)
118 float sampleRate = this->sampleRate(); local
122 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveSine, (PeriodicWave::createSine(sampleRate)));
127 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveSquare, (PeriodicWave::createSquare(sampleRate)));
132 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveSawtooth, (PeriodicWave::createSawtooth(sampleRate)));
137 DEFINE_STATIC_LOCAL(Persistent<PeriodicWave>, periodicWaveTriangle, (PeriodicWave::createTriangle(sampleRate)));
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H A DPeriodicWave.cpp48 PeriodicWave* PeriodicWave::create(float sampleRate, Float32Array* real, Float32Array* imag) argument
53 PeriodicWave* periodicWave = new PeriodicWave(sampleRate);
61 PeriodicWave* PeriodicWave::createSine(float sampleRate) argument
63 PeriodicWave* periodicWave = new PeriodicWave(sampleRate);
68 PeriodicWave* PeriodicWave::createSquare(float sampleRate) argument
70 PeriodicWave* periodicWave = new PeriodicWave(sampleRate);
75 PeriodicWave* PeriodicWave::createSawtooth(float sampleRate) argument
77 PeriodicWave* periodicWave = new PeriodicWave(sampleRate);
82 PeriodicWave* PeriodicWave::createTriangle(float sampleRate) argument
84 PeriodicWave* periodicWave = new PeriodicWave(sampleRate);
89 PeriodicWave(float sampleRate) argument
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H A DAudioBufferSourceNode.cpp50 AudioBufferSourceNode* AudioBufferSourceNode::create(AudioContext* context, float sampleRate) argument
52 return adoptRefCountedGarbageCollectedWillBeNoop(new AudioBufferSourceNode(context, sampleRate));
55 AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context, float sampleRate) argument
56 : AudioScheduledSourceNode(context, sampleRate)
203 double bufferSampleRate = buffer()->sampleRate();
228 double loopStartFrame = m_loopStart * buffer()->sampleRate();
229 double loopEndFrame = m_loopEnd * buffer()->sampleRate();
435 m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset, buffer()->sampleRate());
450 sampleRateFactor = buffer()->sampleRate() / sampleRate();
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/external/chromium_org/third_party/WebKit/Source/platform/audio/
H A DHRTFElevation.cpp104 bool HRTFElevation::calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName, argument
163 RefPtr<AudioBus> response(AudioBus::createBySampleRateConverting(preSampleRateConvertedResponse.get(), false, sampleRate));
169 RefPtr<AudioBus> impulseResponse(AudioBus::loadPlatformResource(resourceName.utf8().data(), sampleRate));
176 size_t expectedLength = static_cast<size_t>(256 * (sampleRate / 44100.0));
189 const size_t fftSize = HRTFPanner::fftSizeForSampleRate(sampleRate);
190 kernelL = HRTFKernel::create(leftEarImpulseResponse, fftSize, sampleRate);
191 kernelR = HRTFKernel::create(rightEarImpulseResponse, fftSize, sampleRate);
228 PassOwnPtr<HRTFElevation> HRTFElevation::createForSubject(const String& subjectName, int elevation, float sampleRate)
245 bool success = calculateKernelsForAzimuthElevation(rawIndex * AzimuthSpacing, actualElevation, sampleRate, subjectName, kernelListL->at(interpolatedIndex), kernelListR->at(interpolatedIndex));
265 OwnPtr<HRTFElevation> hrtfElevation = adoptPtr(new HRTFElevation(kernelListL.release(), kernelListR.release(), elevation, sampleRate));
269 createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate) argument
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H A DHRTFPanner.cpp47 HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader) argument
50 , m_sampleRate(sampleRate)
58 , m_convolverL1(fftSizeForSampleRate(sampleRate))
59 , m_convolverR1(fftSizeForSampleRate(sampleRate))
60 , m_convolverL2(fftSizeForSampleRate(sampleRate))
61 , m_convolverR2(fftSizeForSampleRate(sampleRate))
62 , m_delayLineL(MaxDelayTimeSeconds, sampleRate)
63 , m_delayLineR(MaxDelayTimeSeconds, sampleRate)
76 size_t HRTFPanner::fftSizeForSampleRate(float sampleRate) argument
86 ASSERT(AudioUtilities::isValidAudioBufferSampleRate(sampleRate));
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/external/chromium_org/third_party/WebKit/Source/platform/exported/
H A DWebMediaStreamSource.cpp176 virtual void setFormat(size_t numberOfChannels, float sampleRate) OVERRIDE;
188 void ConsumerWrapper::setFormat(size_t numberOfChannels, float sampleRate) argument
190 m_consumer->setFormat(numberOfChannels, sampleRate);
/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/apple/
H A DAppleLosslessSpecificBox.java32 private long sampleRate; // 32bit field in class:AppleLosslessSpecificBox
115 return sampleRate;
118 public void setSampleRate(int sampleRate) { argument
119 this.sampleRate = sampleRate;
136 sampleRate = IsoTypeReader.readUInt32(content);
152 IsoTypeWriter.writeUInt32(byteBuffer, sampleRate);
/external/sonivox/arm-fm-22k/host_src/
H A Deas.h55 EAS_I32 sampleRate; member in struct:__anon30830
/external/sonivox/arm-hybrid-22k/host_src/
H A Deas.h55 EAS_I32 sampleRate; member in struct:__anon30880
/external/sonivox/arm-wt-22k/host_src/
H A Deas.h55 EAS_I32 sampleRate; member in struct:__anon30931
/external/srec/srec/include/
H A Dutteranc.h221 unsigned long sampleRate; member in struct:_UttHeader
/external/aac/libAACenc/src/
H A Daacenc.cpp114 INT sampleRate);
401 switch (config->sampleRate)
429 config->sampleRate,
478 config->sampleRate);
484 config->ancDataBitRate += ( (hAacEnc->ancillaryBitsPerFrame * config->sampleRate) / config->framelength );
492 FIXP_DBL tmp = fDivNorm(config->framelength, config->sampleRate, &q_res);
516 config->sampleRate,
540 config->sampleRate,
581 qcInit.sampleRate = config->sampleRate;
999 FDKaacEnc_InitCheckAncillary(INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, INT sampleRate) argument
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H A Daacenc.h183 INT sampleRate; /* encoder sample rate */ member in struct:AACENC_CONFIG
H A Dmetadata_main.cpp407 const UINT sampleRate,
470 sampleRate,
401 FDK_MetadataEnc_Init( HANDLE_FDK_METADATA_ENCODER hMetaData, const INT resetStates, const INT metadataMode, const INT audioDelay, const UINT frameLength, const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, const CHANNEL_ORDER channelOrder ) argument
/external/aac/libMpegTPDec/src/
H A Dtpdec_asc.cpp984 INT sampleRate; local
992 sampleRate = FDKreadBits(bs,24);
994 sampleRate = SamplingRateTable[idx];
999 return sampleRate;
/external/chromium_org/third_party/WebKit/Source/web/
H A DWebMediaPlayerClientImpl.cpp301 void WebMediaPlayerClientImpl::AudioClientImpl::setFormat(size_t numberOfChannels, float sampleRate) argument
304 m_client->setFormat(numberOfChannels, sampleRate);
/external/chromium_org/third_party/webrtc/modules/audio_device/test/
H A Daudio_device_test_api.cc91 const uint32_t sampleRate,
117 const uint32_t sampleRate,
1722 uint32_t sampleRate(0);
1725 EXPECT_EQ(0, audio_device_->RecordingSampleRate(&sampleRate));
1727 EXPECT_EQ(48000, sampleRate);
1729 TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
1730 EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000));
1732 TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
1733 EXPECT_TRUE((sampleRate
86 RecordedDataIsAvailable( const void* audioSamples, const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t sampleRate, const uint32_t totalDelay, const int32_t clockSkew, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument
113 NeedMorePlayData( const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t sampleRate, void* audioSamples, uint32_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
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/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/sampleentry/
H A DAudioSampleEntry.java55 private long sampleRate; field in class:AudioSampleEntry
82 return sampleRate;
133 public void setSampleRate(long sampleRate) { argument
134 this.sampleRate = sampleRate;
198 //sampleRate = in.readFixedPoint1616();
199 sampleRate = IsoTypeReader.readUInt32(content);
201 sampleRate = sampleRate >>> 16;
242 ", sampleRate
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/external/sonivox/arm-fm-22k/lib_src/
H A Deas_pcm.c102 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate);
371 pState->sampleRate = (EAS_U16) pParams->sampleRate;
374 pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15;
879 * sampleRate - sample rate in samples/sec
888 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate) argument
895 if (srcConvRate[i][0] == sampleRate)
900 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Sample rate %u not in table, calculating by division\n", sampleRate); */ }
902 return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15;
1374 temp = (msecs * pState->sampleRate);
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H A Deas_pcmdata.h113 EAS_U16 sampleRate; /* input sample rate */ member in struct:s_pcm_state_tag
/external/sonivox/arm-hybrid-22k/lib_src/
H A Deas_pcm.c102 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate);
371 pState->sampleRate = (EAS_U16) pParams->sampleRate;
374 pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15;
879 * sampleRate - sample rate in samples/sec
888 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate) argument
895 if (srcConvRate[i][0] == sampleRate)
900 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Sample rate %u not in table, calculating by division\n", sampleRate); */ }
902 return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15;
1374 temp = (msecs * pState->sampleRate);
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H A Deas_pcmdata.h113 EAS_U16 sampleRate; /* input sample rate */ member in struct:s_pcm_state_tag
/external/sonivox/arm-wt-22k/lib_src/
H A Deas_pcm.c102 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate);
371 pState->sampleRate = (EAS_U16) pParams->sampleRate;
374 pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15;
879 * sampleRate - sample rate in samples/sec
888 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate) argument
895 if (srcConvRate[i][0] == sampleRate)
900 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Sample rate %u not in table, calculating by division\n", sampleRate); */ }
902 return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15;
1374 temp = (msecs * pState->sampleRate);
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H A Deas_pcmdata.h113 EAS_U16 sampleRate; /* input sample rate */ member in struct:s_pcm_state_tag
/external/aac/libAACdec/include/
H A Daacdecoder_lib.h220 While the members sampleRate, frameSize and numChannels might be quite self explaining,
535 INT sampleRate; /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing). */ member in struct:__anon3

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