| /external/aac/libAACenc/src/ |
| pnsparam.cpp | 188 int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int isLC) {
213 switch (sampleRate) {
242 INT sampleRate,
267 hUsePns = FDKaacEnc_lookUpPnsUse (bitRate, sampleRate, numChan, isLC);
279 sampleRate,
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| metadata_compressor.cpp | 169 UINT sampleRate; /*!< Sample rate. */
386 * \param sampleRate Sampling rate in Hz.
393 const INT sampleRate,
403 /* f = sampleRate/blockLength */
404 sampleRateFract = (FIXP_DBL)(sampleRate<<(DFRACT_BITS-1-METADATA_LINT_BITS));
479 const UINT sampleRate,
492 drcComp->sampleRate = sampleRate;
496 if (FDK_DRC_Generator_setDrcProfile(drcComp, profileLine, profileRF)!=0) { /* expects initialized blockLength and sampleRate */
631 drcComp->fastAttack[i] = tc2Coeff(tabFastAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength)
[all...] |
| /external/aac/libSBRdec/include/ |
| sbrdecoder.h | 171 * \param sampleRateIn Input samplerate of the SBR decoder instance.
172 * \param sampleRateOut Output samplerate of the SBR decoder instance.
299 * \param sampleRate Output samplerate.
310 int *sampleRate,
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| /external/aac/libSBRenc/include/ |
| sbr_encoder.h | 144 UINT sampleRate; /*!< */
314 * \param sampleRate Input: Encoder samplerate. output core encoder samplerate.
333 INT *sampleRate,
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| /external/chromium_org/media/base/android/java/src/org/chromium/media/ |
| MediaCodecBridge.java | 453 private static MediaFormat createAudioFormat(String mime, int sampleRate, int channelCount) {
454 return MediaFormat.createAudioFormat(mime, sampleRate, channelCount);
497 int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
503 int minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig,
505 mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig,
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| /frameworks/av/media/libstagefright/ |
| AudioPlayer.cpp | 727 uint32_t sampleRate;
730 sampleRate = mAudioSink->getSampleRate();
733 sampleRate = mAudioTrack->getSampleRate();
735 if (sampleRate != 0) {
736 mSampleRate = sampleRate;
|
| OMXCodec.cpp | 502 int32_t numChannels, sampleRate, aacProfile;
504 CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
515 status_t err = setAACFormat(numChannels, sampleRate, bitRate, aacProfile, isADTS);
521 int32_t numChannels, sampleRate;
523 && meta->findInt32(kKeySampleRate, &sampleRate)) {
528 sampleRate,
543 int32_t numChannels, sampleRate;
545 CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
547 setRawAudioFormat(kPortIndexInput, sampleRate, numChannels);
[all...] |
| /frameworks/wilhelm/tests/examples/ |
| slesTestFeedback.cpp | 38 static SLuint32 sampleRate = 44100; // -s#
214 sampleRate = atoi(&arg[2]);
215 switch (sampleRate) {
228 (unsigned) sampleRate);
303 pcm.samplesPerSec = sampleRate * 1000;
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| /frameworks/av/media/libmediaplayerservice/ |
| MediaPlayerService.cpp | [all...] |
| StagefrightRecorder.h | 163 status_t setParamAudioSamplingRate(int32_t sampleRate);
|
| /external/aac/libSYS/src/ |
| wav_file.cpp | 163 FDKfread_EL(&(wav->header.sampleRate), 4, 1, wav->fp);
378 * \param sampleRate desired samplerate of the resulting WAV file
385 INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate, INT numChannels, INT bitsPerSample)
419 wav->header.sampleRate = LittleEndian32(sampleRate);
420 wav->header.bytesPerSecond = LittleEndian32(sampleRate * wav->header.blockAlign);
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| psy_main.c | 187 Word32 sampleRate,
197 sampleRate,
203 err = InitTnsConfigurationLong(bitRate, sampleRate, channels,
209 sampleRate,
213 err = InitTnsConfigurationShort(bitRate, sampleRate, channels,
252 Word32 sampleRate)
270 sampleRate,
|
| block_switch.c | 111 Word32 sampleRate,
137 /* if the samplerate is less than 16000, it should be all the short block, avoid pre&post echo */
138 if(sampleRate >= 16000) {
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| /frameworks/av/media/libeffects/lvm/lib/Eq/lib/ |
| LVEQNB.h | 170 /* Capabilities.SampleRate = LVEQNB_CAP_32000 + LVEQNB_CAP_44100; */
256 LVEQNB_Fs_en SampleRate;
270 LVM_UINT16 SampleRate;
|
| /frameworks/av/services/audioflinger/ |
| AudioMixer.h | 40 AudioMixer(size_t frameCount, uint32_t sampleRate,
194 uint32_t sampleRate;
208 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
|
| Tracks.cpp | 65 uint32_t sampleRate,
79 mSampleRate(sampleRate),
323 uint32_t sampleRate,
331 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
511 uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
[all...] |
| AudioFlinger.h | 102 uint32_t sampleRate,
117 uint32_t sampleRate,
126 virtual uint32_t sampleRate(audio_io_handle_t output) const;
156 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
|
| AudioResamplerSinc.h | 37 AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate,
|
| /external/chromium_org/content/renderer/media/ |
| renderer_webaudiodevice_impl.cc | 70 double RendererWebAudioDeviceImpl::sampleRate() {
|
| /external/chromium_org/ppapi/examples/audio/ |
| audio.cc | 33 const char kSampleRateAttributeName[] = "samplerate";
|
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| OfflineAudioDestinationNode.cpp | 46 : AudioDestinationNode(context, renderTarget->sampleRate())
|
| /external/chromium_org/third_party/WebKit/Source/platform/mediastream/ |
| MediaStreamSource.h | 89 void setAudioFormat(size_t numberOfChannels, float sampleRate);
|
| /external/chromium_org/third_party/WebKit/Source/testing/runner/ |
| MockWebMediaStreamCenter.cpp | 100 virtual void setFormat(size_t numberOfChannels, float sampleRate) OVERRIDE { }
|
| /frameworks/av/libvideoeditor/lvpp/ |
| VideoEditorPlayer.h | 54 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
|
| /frameworks/av/media/libeffects/lvm/lib/Common/src/ |
| LVM_GetOmega.c | 58 /* LVM_Fs_en Fs The SampleRate */
|
