| /frameworks/wilhelm/tools/permute/ |
| permute.c | 173 switch (sfinfo_in.samplerate) {
183 fprintf(stderr, "%s: unsupported sample rate %d\n", path_in, sfinfo_in.samplerate);
212 double durationSeconds = (double) sfinfo_in.frames / (double) sfinfo_in.samplerate;
214 s.mMinSegmentLengthFrames = minSegmentLengthSeconds * sfinfo_in.samplerate;
259 sfinfo_out.samplerate = sfinfo_in.samplerate;
294 sfinfo_in.samplerate), (unsigned) ((s.mSegmentArray[i].mFrameLength * 1000.0) /
295 sfinfo_in.samplerate));
|
| /prebuilts/python/darwin-x86/2.7.5/lib/python2.7/plat-mac/Carbon/ |
| MediaDescr.py | 72 'compressionID', 'packetSize', ('sampleRate', _tofixed, _fromfixed)),
81 'compressionID', 'packetSize', ('sampleRate', _tofixed, _fromfixed), 'samplesPerPacket',
|
| /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...] |
| AudioCodec.h | 29 virtual int set(int sampleRate, const char *fmtp) = 0;
|
| G711Codec.cpp | 37 int set(int sampleRate, const char *fmtp) {
38 mSampleCount = sampleRate / 50;
88 int set(int sampleRate, const char *fmtp) {
89 mSampleCount = sampleRate / 50;
|
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| ChannelMergerNode.cpp | 43 PassRefPtr<ChannelMergerNode> ChannelMergerNode::create(AudioContext* context, float sampleRate, unsigned numberOfInputs)
48 return adoptRef(new ChannelMergerNode(context, sampleRate, numberOfInputs));
51 ChannelMergerNode::ChannelMergerNode(AudioContext* context, float sampleRate, unsigned numberOfInputs)
52 : AudioNode(context, sampleRate)
|
| OfflineAudioContext.idl | 27 Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
|
| AudioBufferSourceNode.cpp | 53 PassRefPtr<AudioBufferSourceNode> AudioBufferSourceNode::create(AudioContext* context, float sampleRate)
55 return adoptRef(new AudioBufferSourceNode(context, sampleRate));
58 AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context, float sampleRate)
59 : AudioScheduledSourceNode(context, sampleRate)
206 double bufferSampleRate = buffer()->sampleRate();
231 double loopStartFrame = m_loopStart * buffer()->sampleRate();
232 double loopEndFrame = m_loopEnd * buffer()->sampleRate();
441 m_virtualReadIndex = AudioUtilities::timeToSampleFrame(m_grainOffset, buffer()->sampleRate());
464 sampleRateFactor = buffer()->sampleRate() / sampleRate();
[all...] |
| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| AudioSourceProviderClient.h | 32 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
|
| /external/chromium_org/third_party/WebKit/public/platform/ |
| WebAudioSourceProviderClient.h | 32 virtual void setFormat(size_t numberOfChannels, float sampleRate) = 0;
|
| /external/jmonkeyengine/engine/src/core-plugins/com/jme3/audio/plugins/ |
| WAVLoader.java | 80 int sampleRate = in.readInt();
87 int expectedBytesPerSec = (bitsPerSample * channels * sampleRate) / 8;
99 if (bytesPerSample * sampleRate != bytesPerSec)
102 audioData.setupFormat(channels, bitsPerSample, sampleRate);
|
| /frameworks/av/libvideoeditor/vss/common/inc/ |
| VideoEditorResampler.h | 26 M4OSA_Int32 sampleRate, M4OSA_Int32 quality);
|
| /frameworks/av/media/libeffects/lvm/lib/Common/lib/ |
| Filter.h | 54 LVM_Fs_en SampleRate);
|
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_StereoEnhancer.c | 69 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
76 Offset = (LVM_UINT16)pParams->SampleRate;
99 Offset = (LVM_UINT16)(pParams->SampleRate);
|
| /frameworks/av/media/libeffects/testlibs/ |
| AudioPeakingFilter.h | 43 // sampleRate The input/output sample rate, in Hz.
44 AudioPeakingFilter(int nChannels, int sampleRate);
49 // sampleRate The input/output sample rate, in Hz.
50 void configure(int nChannels, int sampleRate);
|
| AudioShelvingFilter.h | 50 // sampleRate The input/output sample rate, in Hz.
51 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate);
56 // sampleRate The input/output sample rate, in Hz.
57 void configure(int nChannels, int sampleRate);
|
| /frameworks/av/media/libmedia/ |
| SoundPool.cpp | 491 uint32_t sampleRate;
499 status = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format, mHeap, &mSize);
501 status = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
511 ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
512 mHeap->getBase(), mSize, sampleRate, numChannels);
514 if (sampleRate > kMaxSampleRate) {
515 ALOGE("Sample rate (%u) out of range", sampleRate);
527 mSampleRate = sampleRate;
579 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5)
[all...] |
| /frameworks/av/media/libnbaio/ |
| LibsndfileSink.cpp | 27 NBAIO_Sink(Format_from_SR_C(sfinfo.samplerate, sfinfo.channels)),
|
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| block_switch.h | 63 Word32 sampleRate,
|
| qc_main.h | 62 Word32 sampleRate);
|
| /frameworks/av/services/audioflinger/ |
| RecordTracks.h | 27 uint32_t sampleRate,
|
| /hardware/libhardware_legacy/audio/ |
| AudioHardwareGeneric.h | 50 virtual uint32_t sampleRate() const { return 44100; }
84 virtual uint32_t sampleRate() const { return 8000; }
124 uint32_t *sampleRate=0,
132 uint32_t *sampleRate,
|
| /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...] |
| /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
| AC3TrackImpl.java | 21 int samplerate; field in class:AC3TrackImpl
62 audioSampleEntry.setSampleRate(samplerate);
81 trackMetaData.setTimescale(samplerate); // Audio tracks always use samplerate as timescale
149 samplerate = 48000;
153 samplerate = 44100;
157 samplerate = 32000;
161 samplerate = 0;
165 if (samplerate == 0) {
183 samplerate /= 2
[all...] |
| /frameworks/av/media/libstagefright/codecs/aacdec/ |
| SoftAAC2.cpp | 188 aacParams->nSampleRate = mStreamInfo->sampleRate;
221 pcmParams->nSamplingRate = mStreamInfo->sampleRate;
340 if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
343 mStreamInfo->sampleRate,
470 int prevSampleRate = mStreamInfo->sampleRate;
535 if (mStreamInfo->sampleRate != prevSampleRate ||
539 prevSampleRate, mStreamInfo->sampleRate,
546 } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
562 + (mNumSamplesOutput * 1000000ll) / mStreamInfo->sampleRate;
606 mStreamInfo->sampleRate = 0
[all...] |
