| /external/adhd/scripts/mic_testing/frontend/ |
| D | source.js | 19 this.setSampleRate = function(sampleRate) { argument
20 this.sampleRate = sampleRate;
58 this.sampleRate * this.duration, this.sampleRate);
71 phi = f * 2 * Math.PI * i / this.sampleRate;
90 this.getAppendTone = function(sampleRate) { argument
95 this.setSampleRate(sampleRate);
107 var sampleRate;
117 sampleRate = this.audioContext.sampleRate;
164 playCallback(tmpLeft, tmpRight, sampleRate);
211 var appendTone = tonegen.getAppendTone(buffer.sampleRate);
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| D | analysis.js | 56 var AudioCurve = function(buffers, filename, sampleRate) { argument
60 this.sampleRate = sampleRate;
244 this.drawCurve = function(buffer, curveColor, sampleRate) { argument
251 var f = i * sampleRate / 2 / nyquist / buffer.length;
276 curveBuffer[i].sampleRate);
288 this.drawInstantCurve = function(leftData, rightData, sampleRate) { argument
294 this.drawCurve(fftLeft.spectrum, "#FF0000", sampleRate);
295 this.drawCurve(fftRight.spectrum, "#00FF00", sampleRate);
299 function calcIndex(freq, length, sampleRate) { argument
300 var idx = parseInt(freq * length * 2 / sampleRate);
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| D | audio.js | 78 drawContext = new DrawCanvas(canvas, audioContext.sampleRate / 2);
158 if (freqEnd > audioContext.sampleRate / 2) {
171 tonegen.setSampleRate(audioContext.sampleRate);
204 function getInstantBuffer(leftData, rightData, sampleRate) { argument
205 drawContext.drawInstantCurve(leftData, rightData, sampleRate);
268 drawContext.add(new AudioCurve(newBuffer, filename, buffer.sampleRate));
282 function getInstantBuffer(leftData, rightData, sampleRate, stop) { argument
283 drawContext.drawInstantCurve(leftData, rightData, sampleRate);
320 drawContext.add(new AudioCurve(buffer, filename, audioContext.sampleRate));
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| D | recorder.js | 12 var sampleRate = context.sampleRate;
42 recordCallback(tmpLeft, tmpRight, sampleRate, stop);
87 var freq = sampleRate / (i - prevPeak);
213 freqString = sampleRate + '\n';
345 view.setUint32(24, sampleRate, true);
347 view.setUint32(28, sampleRate * 4, true);
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| /external/sonic/ |
| D | wave.c | 21 int sampleRate; member
169 int sampleRate) in writeHeader() argument
181 writeInt(file, sampleRate); /* 24 - samples per second (numbers per second) */ in writeHeader()
182 writeInt(file, sampleRate * 2); /* 28 - bytes per second */ in writeHeader()
210 file->sampleRate = readInt(file); /* 24 - samples per second (numbers per second) */ in readHeader()
242 int *sampleRate, in openInputWaveFile() argument
259 *sampleRate = file->sampleRate; in openInputWaveFile()
267 int sampleRate, in openOutputWaveFile() argument
279 file->sampleRate = sampleRate; in openOutputWaveFile()
281 writeHeader(file, sampleRate); in openOutputWaveFile()
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| D | Sonic.java | 41 private int sampleRate; field in Sonic
178 int sampleRate, in allocateStreamBuffers() argument
181 minPeriod = sampleRate/SONIC_MAX_PITCH; in allocateStreamBuffers()
182 maxPeriod = sampleRate/SONIC_MIN_PITCH; in allocateStreamBuffers()
191 this.sampleRate = sampleRate; in allocateStreamBuffers()
200 int sampleRate, in Sonic() argument
203 allocateStreamBuffers(sampleRate, numChannels); in Sonic()
217 return sampleRate; in getSampleRate()
222 int sampleRate) in setSampleRate() argument
224 allocateStreamBuffers(sampleRate, numChannels); in setSampleRate()
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| D | main.c | 26 int sampleRate, in runSonic() argument
29 sonicStream stream = sonicCreateStream(sampleRate, numChannels); in runSonic()
83 int sampleRate, numChannels; in main() local
125 inFile = openInputWaveFile(inFileName, &sampleRate, &numChannels); in main()
129 outFile = openOutputWaveFile(outFileName, sampleRate, numChannels); in main()
135 sampleRate, numChannels); in main()
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| D | sonic.c | 45 int sampleRate; member
209 int sampleRate, in allocateStreamBuffers() argument
212 int minPeriod = sampleRate/SONIC_MAX_PITCH; in allocateStreamBuffers()
213 int maxPeriod = sampleRate/SONIC_MIN_PITCH; in allocateStreamBuffers()
239 stream->sampleRate = sampleRate; in allocateStreamBuffers()
253 int sampleRate, in sonicCreateStream() argument
261 if(!allocateStreamBuffers(stream, sampleRate, numChannels)) { in sonicCreateStream()
279 return stream->sampleRate; in sonicGetSampleRate()
286 int sampleRate) in sonicSetSampleRate() argument
289 allocateStreamBuffers(stream, sampleRate, stream->numChannels); in sonicSetSampleRate()
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| D | sonic.h | 74 sonicStream sonicCreateStream(int sampleRate, int numChannels);
129 void sonicSetSampleRate(sonicStream stream, int sampleRate);
138 float rate, float volume, int useChordPitch, int sampleRate, int numChannels);
143 float rate, float volume, int useChordPitch, int sampleRate, int numChannels);
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| D | Main.java | 29 int sampleRate, in runSonic() argument
32 Sonic sonic = new Sonic(sampleRate, numChannels); in runSonic()
72 int sampleRate = (int)format.getSampleRate(); in main() local
80 sampleRate, numChannels); in main()
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| D | wave.h | 12 waveFile openInputWaveFile(char *fileName, int *sampleRate, int *numChannels);
13 waveFile openOutputWaveFile(char *fileName, int sampleRate, int numChannels);
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| /external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/sampleentry/ |
| D | AudioSampleEntry.java | 55 private long sampleRate; field in AudioSampleEntry
82 return sampleRate; in getSampleRate()
133 public void setSampleRate(long sampleRate) { in setSampleRate() argument
134 this.sampleRate = sampleRate; in setSampleRate()
199 sampleRate = IsoTypeReader.readUInt32(content); in _parseDetails()
201 sampleRate = sampleRate >>> 16; in _parseDetails()
242 ", sampleRate=" + sampleRate + in toString()
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| /external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/apple/ |
| D | AppleLosslessSpecificBox.java | 32 private long sampleRate; // 32bit field in AppleLosslessSpecificBox
115 return sampleRate; in getSampleRate()
118 public void setSampleRate(int sampleRate) { in setSampleRate() argument
119 this.sampleRate = sampleRate; in setSampleRate()
136 sampleRate = IsoTypeReader.readUInt32(content); in _parseDetails()
152 IsoTypeWriter.writeUInt32(byteBuffer, sampleRate); in getContent()
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| /external/aac/libAACenc/src/ |
| D | bandwidth.cpp | 164 static INT GetBandwidthEntry(const INT frameLength, const INT sampleRate, in GetBandwidthEntry() argument
182 switch (sampleRate) { in GetBandwidthEntry()
262 const AACENC_BITRATE_MODE bitrateMode, const INT sampleRate, in FDKaacEnc_DetermineBandWidth() argument
307 *bandWidth = fMin(proposedBandWidth, fMin(20000, sampleRate >> 1)); in FDKaacEnc_DetermineBandWidth()
336 GetBandwidthEntry(frameLength, sampleRate, chanBitRate, entryNo); in FDKaacEnc_DetermineBandWidth()
357 *bandWidth = fMin(*bandWidth, sampleRate / 2); in FDKaacEnc_DetermineBandWidth()
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| D | pnsparam.cpp | 437 int FDKaacEnc_lookUpPnsUse(int bitRate, int sampleRate, int numChan, in FDKaacEnc_lookUpPnsUse() argument
461 switch (sampleRate) { in FDKaacEnc_lookUpPnsUse()
502 INT sampleRate, INT sfbCnt, in FDKaacEnc_GetPnsParam() argument
516 hUsePns = FDKaacEnc_lookUpPnsUse(bitRate, sampleRate, numChan, isLC); in FDKaacEnc_GetPnsParam()
534 hUsePns = FDKaacEnc_lookUpPnsUse(bitRate, sampleRate, numChan, isLC); in FDKaacEnc_GetPnsParam()
546 pnsInfo->startFreq, sampleRate, sfbCnt, sfbOffset); in FDKaacEnc_GetPnsParam()
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| D | aacenc.cpp | 146 INT sampleRate);
411 switch (config->sampleRate) { in FDKaacEnc_Initialize()
437 hTpEnc, config->audioObjectType, config->sampleRate, in FDKaacEnc_Initialize()
478 &hAacEnc->ancillaryBitsPerFrame, config->sampleRate); in FDKaacEnc_Initialize()
484 config->framelength, config->sampleRate); in FDKaacEnc_Initialize()
491 config->framelength, config->sampleRate) >> in FDKaacEnc_Initialize()
510 hAacEnc->bitrateMode, config->sampleRate, config->framelength, cm, in FDKaacEnc_Initialize()
529 hAacEnc->psyKernel, config->audioObjectType, cm, config->sampleRate, in FDKaacEnc_Initialize()
599 qcInit.sampleRate = config->sampleRate; in FDKaacEnc_Initialize()
602 qcInit.padding.paddingRest = config->sampleRate; in FDKaacEnc_Initialize()
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| D | pnsparam.h | 139 int FDKaacEnc_lookUpPnsUse(int bitRate, int sampleRate, int numChan,
145 INT sampleRate, INT sfbCnt,
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| /external/webrtc/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
| D | WebRtcAudioManager.java | 84 private int sampleRate; field in WebRtcAudioManager
100 sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS, in WebRtcAudioManager()
140 sampleRate = getNativeOutputSampleRate(); in storeAudioParameters()
147 getMinOutputFrameSize(sampleRate, channels); in storeAudioParameters()
149 inputBufferSize = getMinInputFrameSize(sampleRate, channels); in storeAudioParameters()
289 int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC, in nativeCacheAudioParameters() argument
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| D | WebRtcAudioTrack.java | 155 private void initPlayout(int sampleRate, int channels) { in initPlayout() argument
156 Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels=" in initPlayout()
160 bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND)); in initPlayout()
172 sampleRate, in initPlayout()
187 sampleRate, in initPlayout()
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| D | WebRtcAudioRecord.java | 153 private int initRecording(int sampleRate, int channels) { in initRecording() argument
154 Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + in initRecording()
166 final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND; in initRecording()
178 sampleRate, in initRecording()
196 sampleRate, in initRecording()
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| /external/aac/libPCMutils/src/ |
| D | limiter.cpp | 158 limiter->sampleRate = maxSampleRate; in pcmLimiter_Create()
455 UINT sampleRate) { in pcmLimiter_SetSampleRate() argument
462 if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER; in pcmLimiter_SetSampleRate()
465 attack = (unsigned int)(limiter->attackMs * sampleRate / 1000); in pcmLimiter_SetSampleRate()
466 release = (unsigned int)(limiter->releaseMs * sampleRate / 1000); in pcmLimiter_SetSampleRate()
481 limiter->sampleRate = sampleRate; in pcmLimiter_SetSampleRate()
501 attack = (unsigned int)(attackMs * limiter->sampleRate / 1000); in pcmLimiter_SetAttack()
525 release = (unsigned int)(releaseMs * limiter->sampleRate / 1000); in pcmLimiter_SetRelease()
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| /external/speex/libspeexdsp/ |
| D | testecho.c | 23 int sampleRate = 8000; in main() local
35 den = speex_preprocess_state_init(NN, sampleRate); in main()
36 speex_echo_ctl(st, SPEEX_ECHO_SET_SAMPLING_RATE, &sampleRate); in main()
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| /external/webrtc/webrtc/modules/audio_device/test/ |
| D | audio_device_test_api.cc | 89 const uint32_t sampleRate, in RecordedDataIsAvailable() argument
114 const uint32_t sampleRate, in NeedMorePlayData() argument
1702 uint32_t sampleRate(0); in TEST_F() local
1705 EXPECT_EQ(0, audio_device_->RecordingSampleRate(&sampleRate)); in TEST_F()
1707 EXPECT_EQ(48000, sampleRate); in TEST_F()
1709 TEST_LOG("Recording sample rate is %u\n\n", sampleRate); in TEST_F()
1710 EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000)); in TEST_F()
1712 TEST_LOG("Recording sample rate is %u\n\n", sampleRate); in TEST_F()
1713 EXPECT_TRUE((sampleRate == 44000) || (sampleRate == 16000) || in TEST_F()
1714 (sampleRate == 8000)); in TEST_F()
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| /external/sonivox/arm-wt-22k/lib_src/ |
| D | eas_pcm.c | 102 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate);
371 pState->sampleRate = (EAS_U16) pParams->sampleRate; in EAS_PEOpenStream()
374 pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15; in EAS_PEOpenStream()
888 static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate) in CalcBaseFreq() argument
895 if (srcConvRate[i][0] == sampleRate) in CalcBaseFreq()
902 return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15; in CalcBaseFreq()
1374 temp = (msecs * pState->sampleRate); in LinearPCMLocate()
1378 temp += secs * pState->sampleRate; in LinearPCMLocate()
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| /external/aac/libSACdec/src/ |
| D | sac_dec_lib.cpp | 242 const INT coreCodec, const INT sampleRate,
251 const SPATIAL_SPECIFIC_CONFIG *pSsc, UINT sampleRate) { in mpegSurroundDecoder_GetNrOfQmfBands() argument
252 UINT samplingFrequency = sampleRate; in mpegSurroundDecoder_GetNrOfQmfBands()
1009 int frameLength, int sampleRate) { in sscCheckInBand() argument
1021 if (pSsc->samplingFreq != sampleRate) { in sscCheckInBand()
1100 const INT sampleRate, const INT frameSize) { in sscCheckOutOfBand() argument
1121 if (pSsc->samplingFreq != sampleRate) { in sscCheckOutOfBand()
1288 AUDIO_OBJECT_TYPE coreCodec, int sampleRate, in mpegSurroundDecoder_Parse() argument
1400 frameSize, sampleRate); in mpegSurroundDecoder_Parse()
1512 int *nChannels, int *frameSize, int sampleRate, in mpegSurroundDecoder_Apply() argument
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