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  /external/chromium_org/third_party/WebKit/Source/modules/webaudio/
PannerNode.h 64 static PassRefPtr<PannerNode> create(AudioContext* context, float sampleRate)
66 return adoptRef(new PannerNode(context, sampleRate));
133 PannerNode(AudioContext*, float sampleRate);
AudioBasicInspectorNode.h 39 AudioBasicInspectorNode(AudioContext*, float sampleRate, unsigned outputChannelCount);
  /external/chromium_org/third_party/WebKit/Source/platform/exported/
WebMediaStreamSource.cpp 179 virtual void setFormat(size_t numberOfChannels, float sampleRate) OVERRIDE;
191 void ConsumerWrapper::setFormat(size_t numberOfChannels, float sampleRate)
193 m_consumer->setFormat(numberOfChannels, sampleRate);
  /external/srec/audio/AudioIn/UNIX/src/
audioinwrapper.cpp 60 static int sampleRate = 8000;
70 sampleRate = sample_rate;
100 sampleRate,
  /external/srec/srec/include/
sample.h 181 int samplerate; member in struct:__anon26505
215 void add_riff_header(PFile* waveFile, int samplerate, int bitspersample);
216 void fix_riff_header(PFile* waveFile, int samplerate, int bitspersample);
utteranc.h 221 unsigned long sampleRate;
229 int update_utb_header(file_utterance_info *utt, int frames, int samplerate,
231 void init_utt_v5_header(UttHeader *uhead, int dim, int samplerate, int framerate);
  /frameworks/av/include/media/
SoundPool.h 60 int sampleRate() { return mSampleRate; }
70 void init(int numChannels, int sampleRate, audio_format_t format, size_t size,
72 mNumChannels = numChannels; mSampleRate = sampleRate; mFormat = format; mSize = size;
  /frameworks/av/include/media/stagefright/
ACodec.h 253 int32_t numChannels, int32_t sampleRate, int32_t bitRate,
263 bool encoder, int32_t numChannels, int32_t sampleRate, int32_t compressionLevel);
266 OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
  /frameworks/av/media/libstagefright/
AMRWriter.cpp 91 int32_t sampleRate;
94 CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
95 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
Utils.cpp 106 int32_t numChannels, sampleRate;
108 CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
111 msg->setInt32("sample-rate", sampleRate);
409 int32_t sampleRate;
410 if (msg->findInt32("sample-rate", &sampleRate)) {
411 meta->setInt32(kKeySampleRate, sampleRate);
482 int32_t sampleRate = 0;
490 if (meta->findInt32(kKeySampleRate, &sampleRate)) {
491 param.addInt(String8(AUDIO_OFFLOAD_CODEC_SAMPLE_RATE), sampleRate);
506 ALOGV("sendMetaDataToHal: bitRate %d, sampleRate %d, chanMask %d,
    [all...]
  /frameworks/av/services/audioflinger/
AudioResampler.cpp 42 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
43 AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
136 int32_t sampleRate, src_quality quality) {
189 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
193 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
197 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
201 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
211 int32_t sampleRate, src_quality quality) :
213 mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0)
    [all...]
  /hardware/libhardware_legacy/include/hardware_legacy/
AudioHardwareInterface.h 53 virtual uint32_t sampleRate() const = 0;
131 virtual uint32_t sampleRate() const = 0;
244 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
251 uint32_t *sampleRate=0,
260 uint32_t *sampleRate,
  /external/srec/srec/cfront/
frontobj.c 302 waveobj->samplerate = parameters->samplerate;
358 ASSERT(parameters->samplerate);
361 freqobj->frame_period = parameters->samplerate / freqobj->framerate;
362 freqobj->samplerate = parameters->samplerate;
378 high_cut = parameters->samplerate / 2;
380 bandwidth = parameters->samplerate / 2;
394 * ((double)parameters->samplerate / (double)11025)
398 * ((double)parameters->samplerate / (double)11025
    [all...]
spec_anl.c 328 ASSERT(freqobj->samplerate > 0);
330 freq_step = (freqobj->samplerate << 12) / (2 * freqobj->fft.size);
337 lo = (((freq[ii] - spread[ii]) * 2 * freqobj->fft.size) + freqobj->samplerate / 2) / freqobj->samplerate;
338 hi = (((freq[ii] + spread[ii]) * 2 * freqobj->fft.size) + freqobj->samplerate / 2) / freqobj->samplerate;
  /frameworks/av/media/libmedia/
AudioRecord.cpp 36 uint32_t sampleRate,
48 status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
55 ALOGE("Unsupported configuration: sampleRate %u, format %d, channelMask %#x",
56 sampleRate, format, channelMask);
81 uint32_t sampleRate,
96 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
124 uint32_t sampleRate,
166 ALOGV("set(): sampleRate %u, channelMask %#x, frameCount %u", sampleRate, channelMask,
181 if (sampleRate == 0)
    [all...]
IAudioFlinger.cpp 89 uint32_t sampleRate,
106 data.writeInt32(sampleRate);
150 uint32_t sampleRate,
163 data.writeInt32(sampleRate);
207 virtual uint32_t sampleRate(audio_io_handle_t output) const
372 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
377 data.writeInt32(sampleRate);
754 uint32_t sampleRate = data.readInt32();
777 (audio_stream_type_t) streamType, sampleRate, format,
791 uint32_t sampleRate = data.readInt32()
    [all...]
AudioTrack.cpp 41 uint32_t sampleRate)
75 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
76 afFrameCount * minBufCount * sampleRate / afSampleRate;
95 uint32_t sampleRate,
113 mStatus = set(streamType, sampleRate, format, channelMask,
121 uint32_t sampleRate,
139 mStatus = set(streamType, sampleRate, format, channelMask,
166 uint32_t sampleRate,
243 if (sampleRate == 0) {
248 sampleRate = afSampleRate
    [all...]
  /external/chromium_org/third_party/WebKit/Source/platform/audio/
DynamicsCompressor.cpp 43 DynamicsCompressor::DynamicsCompressor(float sampleRate, unsigned numberOfChannels)
45 , m_sampleRate(sampleRate)
46 , m_compressor(sampleRate, numberOfChannels)
  /frameworks/av/media/libeffects/lvm/lib/Bass/src/
LVDBE_Init.c 208 pInstance->Params.SampleRate = LVDBE_FS_8000;
273 LVDBE_BYPASS_MIXER_TC,pInstance->Params.SampleRate,2);
283 LVDBE_BYPASS_MIXER_TC,pInstance->Params.SampleRate,2);
  /external/aac/libAACenc/src/
aacenc.cpp 114 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;
    [all...]
  /external/aac/libMpegTPDec/include/
tp_data.h 251 UINT m_samplingFrequency; /**< Samplerate. */
256 UINT m_extensionSamplingFrequency; /**< Samplerate */
267 UCHAR m_samplingFrequencyIndex; /**< Samplerate index */
268 UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
  /external/aac/libMpegTPEnc/include/
tp_data.h 251 UINT m_samplingFrequency; /**< Samplerate. */
256 UINT m_extensionSamplingFrequency; /**< Samplerate */
267 UCHAR m_samplingFrequencyIndex; /**< Samplerate index */
268 UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
  /external/aac/libSYS/include/
wav_file.h 147 UINT sampleRate;
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/chrome/browser/resources/app_list/
audio_manager.js 72 return this.audioContext_.sampleRate;
  /external/chromium_org/chrome/common/extensions/api/
webrtc_audio_private.idl 29 long sampleRate;

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