/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);
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AudioBasicInspectorNode.h | 39 AudioBasicInspectorNode(AudioContext*, float sampleRate, unsigned outputChannelCount);
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/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);
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/external/srec/audio/AudioIn/UNIX/src/ |
audioinwrapper.cpp | 60 static int sampleRate = 8000; 70 sampleRate = sample_rate; 100 sampleRate,
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/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);
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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);
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/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;
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/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);
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/frameworks/av/media/libstagefright/ |
AMRWriter.cpp | 91 int32_t sampleRate; 94 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 95 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
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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,
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/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;
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/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)
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/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);
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/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 */
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/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 */
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/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);
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/external/chromium_org/chrome/browser/resources/app_list/ |
audio_manager.js | 72 return this.audioContext_.sampleRate;
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/external/chromium_org/chrome/common/extensions/api/ |
webrtc_audio_private.idl | 29 long sampleRate;
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