| /external/aac/libAACenc/src/ |
| aacenc_tns.cpp | 273 const INT sampleRate,
304 if (sampleRate >= pMaxBandsTab[i].samplingRate) {
364 input: bitrate, samplerate, number of channels,
373 INT sampleRate,
397 tC->lpcStopBand = getTnsMaxBands(sampleRate, granuleLength, (blockType == SHORT_WINDOW) ? 1 : 0);
410 tC->lpcStartBand[LOFILT] = (blockType == SHORT_WINDOW) ? 0 : ((sampleRate < 18783) ? 4 : 8);
455 tC->lpcStartBand[HIFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, pC->sfbOffset);
457 tC->lpcStartBand[LOFILT] = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, pC->sfbOffset);
476 FDKaacEnc_CalcGaussWindow(tC->acfWindow[HIFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE);
477 FDKaacEnc_CalcGaussWindow(tC->acfWindow[LOFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE)
[all...] |
| aacenc_pns.cpp | 126 bitrate, samplerate, usePns,
134 INT sampleRate,
146 sampleRate,
|
| metadata_compressor.h | 159 * \param sampleRate Sampling rate in Hz.
173 const UINT sampleRate,
|
| metadata_main.h | 141 * \param sampleRate Sampling rat in Hz of audio input signal.
156 const UINT sampleRate,
|
| /frameworks/av/services/audioflinger/ |
| AudioMixer.cpp | 99 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks)
101 mSampleRate(sampleRate)
213 t->sampleRate = mSampleRate;
296 pDbp->mDownmixConfig.inputCfg.samplingRate = pTrack->sampleRate;
297 pDbp->mDownmixConfig.outputCfg.samplingRate = pTrack->sampleRate;
486 track.sampleRate = mSampleRate;
550 if (sampleRate != value) {
551 sampleRate = value;
776 t->resampler->setSampleRate(t->sampleRate);
[all...] |
| AudioFlinger.cpp | 430 uint32_t sampleRate,
500 track = thread->createTrack_l(client, streamType, sampleRate, format,
542 uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
547 ALOGW("sampleRate() unknown thread %d", output);
550 return thread->sampleRate();
974 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
985 sample_rate: sampleRate,
[all...] |
| /frameworks/av/media/libstagefright/ |
| ACodec.cpp | 962 int32_t numChannels, sampleRate;
964 || !msg->findInt32("sample-rate", &sampleRate)) {
971 sampleRate,
975 int32_t numChannels, sampleRate;
977 || !msg->findInt32("sample-rate", &sampleRate)) {
989 encoder, numChannels, sampleRate, bitRate, aacProfile,
1008 int32_t numChannels, sampleRate, compressionLevel = -1;
1011 || !msg->findInt32("sample-rate", &sampleRate))) {
1032 encoder, numChannels, sampleRate, compressionLevel);
1035 int32_t numChannels, sampleRate;
[all...] |
| /external/srec/srec/include/ |
| frontapi.h | 72 int samplerate; member in struct:__anon15430
543 int samplerate);
551 * samplerate File's sample rate (Hz)
588 int samplerate,
595 * samplerate Device sample rate (Hz)
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/Bundle/src/ |
| LVM_Init.c | 235 DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000;
268 EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000;
559 pInstance->Params.SampleRate = LVM_FS_8000;
712 DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000;
[all...] |
| /frameworks/av/media/libmedia/ |
| AudioSystem.cpp | 237 *samplingRate = af->sampleRate(output);
329 status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
335 if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
342 inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
345 gPrevInSamplingRate = sampleRate;
|
| /external/webkit/Source/WebKit/chromium/public/ |
| WebKitClient.h | 227 // A sample-rate conversion to sampleRate will occur if the file data is at a different sample-rate.
229 virtual bool loadAudioResource(WebAudioBus* destinationBus, const char* audioFileData, size_t dataSize, double sampleRate) { return false; }
290 virtual WebAudioDevice* createAudioDevice(size_t bufferSize, unsigned numberOfChannels, double sampleRate, WebAudioDevice::RenderCallback*) { return 0; }
|
| /frameworks/av/cmds/stagefright/ |
| SimplePlayer.cpp | 583 int32_t sampleRate;
585 CHECK(format->findInt32("sample-rate", &sampleRate));
589 sampleRate,
|
| sf2.cpp | 284 int32_t numChannels, sampleRate;
286 CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
289 msg->setInt32("sample-rate", sampleRate);
|
| /frameworks/av/media/libmediaplayerservice/ |
| MediaPlayerService.h | 94 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
195 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
209 uint32_t sampleRate() const { return mSampleRate; }
|
| /frameworks/av/media/libstagefright/codecs/aacenc/ |
| SoftAACEncoder.cpp | 324 params.sampleRate = mSampleRate;
337 static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) {
346 if (sampleRate == kSampleRateTable[i]) {
|
| /frameworks/wilhelm/tests/sandbox/ |
| playbq.c | 252 switch (sfinfo.samplerate) {
264 fprintf(stderr, "unsupported sample rate %d\n", sfinfo.samplerate);
329 format_pcm.samplesPerSec = sfinfo.samplerate * 1000;
|
| playbq.cpp | 252 switch (sfinfo.samplerate) {
264 fprintf(stderr, "unsupported sample rate %d\n", sfinfo.samplerate);
329 format_pcm.samplesPerSec = sfinfo.samplerate * 1000;
|
| /external/webkit/Source/WebCore/webaudio/ |
| AudioNode.cpp | 38 AudioNode::AudioNode(AudioContext* context, double sampleRate)
42 , m_sampleRate(sampleRate)
|
| /frameworks/base/core/java/android/speech/tts/ |
| BlockingAudioTrack.java | 76 BlockingAudioTrack(int streamType, int sampleRate,
80 mSampleRateInHz = sampleRate;
|
| /frameworks/av/media/libeffects/lvm/wrapper/Reverb/ |
| EffectReverb.cpp | 149 LVM_Fs_en SampleRate;
604 LVM_Fs_en SampleRate;
625 SampleRate = LVM_FS_8000;
628 SampleRate = LVM_FS_16000;
631 SampleRate = LVM_FS_22050;
634 SampleRate = LVM_FS_32000;
637 SampleRate = LVM_FS_44100;
640 SampleRate = LVM_FS_48000;
647 if (pContext->SampleRate != SampleRate) {
[all...] |
| /device/asus/flo/ |
| snd_soc_msm_2x | 867 'Internal BTSCO SampleRate':0:8000
888 'Internal BTSCO SampleRate':0:8000
916 'Internal BTSCO SampleRate':0:8000
937 'Internal BTSCO SampleRate':0:16000
1003 'Internal BTSCO SampleRate':0:8000
1018 'Internal BTSCO SampleRate':0:8000
1039 'Internal BTSCO SampleRate':0:8000
1054 'Internal BTSCO SampleRate':0:16000
[all...] |
| snd_soc_msm_2x_Fusion3 | 867 'Internal BTSCO SampleRate':0:8000
888 'Internal BTSCO SampleRate':0:8000
916 'Internal BTSCO SampleRate':0:8000
937 'Internal BTSCO SampleRate':0:16000
1003 'Internal BTSCO SampleRate':0:8000
1018 'Internal BTSCO SampleRate':0:8000
1039 'Internal BTSCO SampleRate':0:8000
1054 'Internal BTSCO SampleRate':0:16000
[all...] |
| /device/lge/mako/ |
| snd_soc_msm_2x_Fusion3 | 867 'Internal BTSCO SampleRate':0:8000
888 'Internal BTSCO SampleRate':0:8000
916 'Internal BTSCO SampleRate':0:8000
937 'Internal BTSCO SampleRate':0:16000
1003 'Internal BTSCO SampleRate':0:8000
1018 'Internal BTSCO SampleRate':0:8000
1039 'Internal BTSCO SampleRate':0:8000
1054 'Internal BTSCO SampleRate':0:16000
[all...] |
| /docs/source.android.com/src/devices/ |
| audio_preprocessing.jd | 64 sampleRate=44100 frameCount=256 measuredWarmup=X ms, warmupCycles=X
|
| audio_warmup.jd | 64 sampleRate=44100 frameCount=256 measuredWarmup=X ms, warmupCycles=X
|
