/external/aac/libAACdec/include/ |
aacdecoder_lib.h | 221 While the members sampleRate, frameSize and numChannels might be quite self explaining, 489 INT sampleRate; /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing). */ [all...] |
/external/aac/libAACenc/src/ |
qc_data.h | 146 INT sampleRate; /* output sample rate */ 147 INT advancedBitsToPe; /* if set, calc bits2PE factor depending on 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/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 | 230 private static MediaFormat createAudioFormat(String mime, int SampleRate, int ChannelCount) { 231 return MediaFormat.createAudioFormat(mime, SampleRate, ChannelCount); 263 int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE); 269 int minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfig, 271 mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfig,
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/external/sonivox/arm-wt-22k/lib_src/ |
eas_mdls.h | 271 EAS_INT sampleRate;
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/external/srec/srec/Recognizer/include/ |
SR_RecognizerImpl.h | 247 size_t sampleRate;
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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/libvideoeditor/vss/stagefrightshells/src/ |
VideoEditorAudioDecoder.cpp | 740 int32_t sampleRate, channelCount; 742 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 744 ALOGV("VideoEditorAudioDecoder_step: samplingFreq = %d", sampleRate); 747 (uint32_t)sampleRate; [all...] |
/frameworks/av/media/libstagefright/httplive/ |
PlaylistFetcher.cpp | 925 int32_t sampleRate; 926 CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate)); 943 int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate;
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/frameworks/av/media/libstagefright/mpeg2ts/ |
ESQueue.cpp | 444 int32_t sampleRate; 446 CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate)); 450 sampleRate, numChannels);
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/frameworks/av/media/libstagefright/wifi-display/source/ |
TSPacketizer.cpp | 318 int32_t sampleRate; 319 CHECK(mFormat->findInt32("sample-rate", &sampleRate)); 320 CHECK(sampleRate == 44100 || sampleRate == 48000); 327 unsigned sampling_frequency = (sampleRate == 44100) ? 1 : 2;
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/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);
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Threads.h | 124 uint32_t sampleRate() const { return mSampleRate; } 419 uint32_t sampleRate, [all...] |
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...] |
/hardware/qcom/audio/legacy/alsa_sound/ |
AudioUsbALSA.cpp | 91 status_t AudioUsbALSA::getCap(char * type, int &channels, int &sampleRate) 104 sampleRate = 0; 216 sampleRate = ratesSupported[i]; 220 ALOGD("sampleRate: %d", sampleRate); 278 status_t AudioUsbALSA::setHardwareParams(pcm *txHandle, uint32_t sampleRate, uint32_t channels, int periodBytes) 284 unsigned int requestedRate = sampleRate; 299 ALOGV("Setting period size:%d samplerate:%d, channels: %d",periodBytes,sampleRate, channels); 306 param_set_int(params, SNDRV_PCM_HW_PARAM_RATE, sampleRate); [all...] |
/frameworks/av/cmds/stagefright/ |
stagefright.cpp | 985 long sampleRate = strtol(filename + 5, &end, 10); 988 sampleRate = 44100; 990 mediaSource = new SineSource(sampleRate, 1);
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/frameworks/av/media/libmedia/ |
AudioTrack.cpp | 41 uint32_t sampleRate) 75 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 76 afFrameCount * minBufCount * sampleRate / afSampleRate; 94 uint32_t sampleRate, 111 mStatus = set(streamType, sampleRate, format, channelMask, 119 uint32_t sampleRate, 136 mStatus = set(streamType, sampleRate, format, channelMask, 163 uint32_t sampleRate, 240 if (sampleRate == 0) { 245 sampleRate = afSampleRate [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...] |
/frameworks/av/media/libmediaplayerservice/ |
MediaPlayerService.h | 91 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 200 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 218 uint32_t sampleRate() const { return mSampleRate; }
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/frameworks/av/media/libmediaplayerservice/nuplayer/ |
NuPlayer.cpp | 600 int32_t sampleRate; 601 CHECK(codecRequest->findInt32("sample-rate", &sampleRate)); 604 sampleRate, numChannels); 630 sampleRate, [all...] |
/frameworks/av/media/libstagefright/ |
AVIExtractor.cpp | 339 int sampleRate; 342 header, &frameSize, &sampleRate, NULL, NULL, &numSamples)) { 353 int64_t timeUs = mBaseTimeUs + (mNumSamplesRead * 1000000ll) / sampleRate; 713 uint32_t sampleRate = U32LE_AT(&data[4]); 716 track->mMeta->setInt32(kKeySampleRate, sampleRate); [all...] |
/frameworks/ex/variablespeed/jni/ |
variablespeed.cc | 245 SLuint32 sampleRate = *(reinterpret_cast<SLuint32*>(value->data)); 246 LOGD("sample Rate: %d", sampleRate); 247 *sampleRateOut = sampleRate;
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/frameworks/opt/net/voip/src/jni/rtp/ |
AudioGroup.cpp | 100 AudioCodec *codec, int sampleRate, int sampleCount, 104 bool mix(int32_t *output, int head, int tail, int sampleRate); 166 AudioCodec *codec, int sampleRate, int sampleCount, 178 mSampleRate = sampleRate / 1000; 234 bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate) 253 if (sampleRate == mSampleRate) { 476 bool set(int sampleRate, int sampleCount); 572 bool AudioGroup::set(int sampleRate, int sampleCount) 580 mSampleRate = sampleRate; 594 sampleRate, sampleCount, -1, -1)) [all...] |
/frameworks/wilhelm/src/android/ |
AudioPlayer_to_android.cpp | [all...] |