| /external/chromium_org/media/base/android/java/src/org/chromium/media/ |
| 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,
107 mSampleRate = sampleRate;
110 mHardwareDelayBytes = HARDWARE_DELAY_MS * sampleRate / 1000 * bitsPerSample / 8;
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| /frameworks/av/media/libstagefright/rtsp/ |
| AMPEG4ElementaryAssembler.cpp | 89 static bool GetSampleRateIndex(int32_t sampleRate, size_t *tableIndex) {
99 if (sampleRate == kSampleRateTable[index]) {
189 int32_t sampleRate, numChannels;
191 desc.c_str(), &sampleRate, &numChannels);
194 CHECK(GetSampleRateIndex(sampleRate, &mSampleRateIndex));
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| /frameworks/av/services/audioflinger/ |
| PlaybackTracks.h | 28 uint32_t sampleRate,
50 virtual uint32_t sampleRate() const;
164 uint32_t sampleRate,
208 uint32_t sampleRate,
258 uint32_t sampleRate,
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| /frameworks/opt/net/voip/src/jni/rtp/ |
| AmrCodec.cpp | 53 int set(int sampleRate, const char *fmtp);
67 int AmrCodec::set(int sampleRate, const char *fmtp)
97 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1;
211 int set(int sampleRate, const char *fmtp) {
212 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1;
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| /external/aac/libAACenc/src/ |
| psy_configuration.cpp | 100 LONG sampleRate;
211 static AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(LONG sampleRate, INT blockType, INT granuleLength, INT *sfbOffset, INT *sfbCnt)
240 if(sfbInfo[i].sampleRate == sampleRate){
459 const LONG samplerate,
484 barcFactor = fDivNorm(fixMin(FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfbActive], samplerate), MAX_BARC),
489 pePerWindow = fDivNorm(bitrate, samplerate, &qperwin);
520 barcWidth = FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb+1], samplerate) -
521 FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb], samplerate);
569 INT samplerate,
[all...] |
| 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);
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| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| DynamicsCompressorKernel.cpp | 51 DynamicsCompressorKernel::DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels)
52 : m_sampleRate(sampleRate)
71 m_meteringReleaseK = static_cast<float>(discreteTimeConstantForSampleRate(meteringReleaseTimeConstant, sampleRate));
87 unsigned preDelayFrames = preDelayTime * sampleRate();
226 float sampleRate = this->sampleRate();
244 float attackFrames = attackTime * sampleRate;
247 float releaseFrames = sampleRate * releaseTime;
251 float satReleaseFrames = satReleaseTime * sampleRate;
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| AudioDestination.h | 49 static PassOwnPtr<AudioDestination> create(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate);
58 virtual float sampleRate() const = 0;
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| DynamicsCompressorKernel.h | 43 DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels);
72 float sampleRate() const { return m_sampleRate; }
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| qc_main.c | 61 Word32 sampleRate,
69 quot = result / sampleRate;
73 result -= quot * sampleRate;
90 Word32 sampleRate,
99 sampleRate,
106 *paddingRest = *paddingRest + sampleRate;
546 Word32 sampleRate) /* output sampling rate */
555 sampleRate,
560 sampleRate,
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| /hardware/qcom/audio/legacy/alsa_sound/ |
| AudioHardwareALSA.h | 174 uint32_t sampleRate;
274 uint32_t sampleRate() const;
298 virtual uint32_t sampleRate() const
300 return ALSAStreamOps::sampleRate();
354 virtual uint32_t sampleRate() const
356 return ALSAStreamOps::sampleRate();
487 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channels);
507 uint32_t *sampleRate=0,
516 uint32_t *sampleRate,
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| /system/media/audio_utils/ |
| tinysndfile.c | 159 unsigned samplerate = little4u(&fmt[4]); local
160 if (samplerate == 0) {
161 fprintf(stderr, "samplerate %u == 0\n", samplerate);
173 handle->info.samplerate = samplerate;
246 (info->samplerate > 0) &&
268 write4u(&wav[24], info->samplerate);
285 unsigned byteRate = info->samplerate * blockAlignment;
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| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| AudioContext.h | 84 static PassRefPtr<AudioContext> create(Document&, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
106 float sampleRate() const { return m_destinationNode->sampleRate(); }
112 PassRefPtr<AudioBuffer> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
249 AudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
251 static bool isSampleRateRangeGood(float sampleRate);
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| AnalyserNode.cpp | 39 AnalyserNode::AnalyserNode(AudioContext* context, float sampleRate)
40 : AudioBasicInspectorNode(context, sampleRate, 2)
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| AudioBasicInspectorNode.cpp | 37 AudioBasicInspectorNode::AudioBasicInspectorNode(AudioContext* context, float sampleRate, unsigned outputChannelCount)
38 : AudioNode(context, sampleRate)
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| AudioBasicProcessorNode.cpp | 39 AudioBasicProcessorNode::AudioBasicProcessorNode(AudioContext* context, float sampleRate)
40 : AudioNode(context, sampleRate)
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| AudioDestinationNode.cpp | 39 AudioDestinationNode::AudioDestinationNode(AudioContext* context, float sampleRate)
40 : AudioNode(context, sampleRate)
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| AudioDestinationNode.h | 41 AudioDestinationNode(AudioContext*, float sampleRate);
53 double currentTime() const { return currentSampleFrame() / static_cast<double>(sampleRate()); }
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| GainNode.cpp | 37 GainNode::GainNode(AudioContext* context, float sampleRate)
38 : AudioNode(context, sampleRate)
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| OscillatorNode.h | 55 static PassRefPtr<OscillatorNode> create(AudioContext*, float sampleRate);
74 OscillatorNode(AudioContext*, float sampleRate);
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| ScriptProcessorNode.h | 53 static PassRefPtr<ScriptProcessorNode> create(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
72 ScriptProcessorNode(AudioContext*, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
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| WaveShaperProcessor.cpp | 35 WaveShaperProcessor::WaveShaperProcessor(float sampleRate, size_t numberOfChannels)
36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
|
| /frameworks/av/libvideoeditor/lvpp/ |
| VideoEditorPlayer.cpp | 392 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
407 ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
422 frameCount = (sampleRate*afFrameCount*bufferCount)/afSampleRate;
441 sampleRate,
451 sampleRate,
465 mMsecsPerFrame = 1.e3 / (float) sampleRate;
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| /frameworks/av/media/libeffects/testlibs/ |
| AudioEqualizer.h | 70 // sampleRate The input/output sample rate, in Hz.
81 int sampleRate,
88 // sampleRate The input/output sample rate, in Hz.
89 void configure(int nChannels, int sampleRate);
232 // sampleRate The input/output sample rate, in Hz.
240 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate,
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| /frameworks/av/media/libmedia/ |
| IMediaPlayerService.cpp | 227 uint32_t sampleRate;
231 status_t status = decode(url, &sampleRate, &numChannels, &format, heap, &size);
234 reply->writeInt32(sampleRate);
247 uint32_t sampleRate;
251 status_t status = decode(fd, offset, length, &sampleRate, &numChannels, &format,
255 reply->writeInt32(sampleRate);
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