| /external/webrtc/webrtc/modules/audio_device/ |
| mock_audio_device_buffer.h | 23 MOCK_METHOD1(RequestPlayoutData, int32_t(size_t nSamples));
26 int32_t(const void* audioBuffer, size_t nSamples));
|
| audio_device_buffer.cc | 379 // 16-bit,48kHz mono, 10ms => nSamples=480 => _recSize=2*480=960 bytes
380 // 16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes
384 size_t nSamples)
394 _recSamples = nSamples;
395 _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
486 int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples)
509 _playSamples = nSamples;
510 _playSize = playBytesPerSample * nSamples; // {2,4}*nSamples
[all...] |
| audio_device_buffer.h | 53 size_t nSamples);
61 virtual int32_t RequestPlayoutData(size_t nSamples);
|
| /external/libopus/silk/ |
| resampler.c | 181 opus_int nSamples;
188 nSamples = S->Fs_in_kHz - S->inputDelay;
191 silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) );
196 silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
200 silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
204 silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz );
208 silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16 ) );
|
| /frameworks/av/media/libstagefright/ |
| FLACExtractor.cpp | 125 void (*mCopy)(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels);
386 unsigned nSamples,
388 for (unsigned i = 0; i < nSamples; ++i) {
396 unsigned nSamples,
398 for (unsigned i = 0; i < nSamples; ++i) {
404 static void copyMultiCh8(short *dst, const int *const *src, unsigned nSamples, unsigned nChannels)
406 for (unsigned i = 0; i < nSamples; ++i) {
416 unsigned nSamples,
418 for (unsigned i = 0; i < nSamples; ++i) {
426 unsigned nSamples,
[all...] |
| AudioSource.cpp | 399 void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
400 for (int i = nSamples; i > 0; --i) {
|
| /external/webrtc/webrtc/modules/audio_device/test/ |
| func_test_manager.h | 50 size_t nSamples;
89 const size_t nSamples,
99 int32_t NeedMorePlayData(const size_t nSamples,
|
| func_test_manager.cc | 195 const size_t nSamples,
208 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
209 packet->nSamples = nSamples;
340 const size_t nSamples,
354 memset(audioSamples, 0, nBytesPerSample * nSamples);
367 const size_t nSamplesIn = packet->nSamples;
392 2 * nSamples, lenOut);
397 2 * nSamplesIn, tmpBuf_96kHz, 2 * nSamples,
404 for (size_t i = 0; i < nSamples; i++
[all...] |
| /external/aac/libPCMutils/include/ |
| limiter.h | 167 * gain_delay <= nSamples *
170 * nSamples: number of samples per channel *
179 const UINT nSamples);
|
| /external/webrtc/talk/app/webrtc/test/ |
| fakeaudiocapturemodule_unittest.cc | 59 const size_t nSamples,
69 rec_buffer_bytes_ = nSamples * nBytesPerSample;
83 int32_t NeedMorePlayData(const size_t nSamples,
93 const size_t audio_buffer_size = nSamples * nBytesPerSample;
|
| /frameworks/av/media/libstagefright/codecs/aacenc/ |
| AACEncoder.cpp | 240 const int32_t nSamples = mChannels * kNumSamplesPerFrame;
241 while (mNumInputSamples < nSamples) {
250 sizeof(int16_t) * (nSamples - mNumInputSamples));
269 size_t copy = (nSamples - mNumInputSamples) * sizeof(int16_t);
289 if (mNumInputSamples >= nSamples) {
290 mNumInputSamples %= nSamples;
298 inputData.Length = nSamples * sizeof(int16_t);
|
| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 50 const size_t nSamples,
60 virtual int32_t NeedMorePlayData(const size_t nSamples,
|
| /external/webrtc/webrtc/voice_engine/ |
| voe_base_impl.h | 58 const size_t nSamples,
67 int32_t NeedMorePlayData(const size_t nSamples,
|
| transmit_mixer.h | 54 size_t nSamples,
176 size_t nSamples,
|
| voe_base_impl.cc | 83 const size_t nSamples,
93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples,
98 int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
106 GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true,
|
| /external/pdfium/third_party/lcms2-2.6/src/ |
| cmscgats.c | 123 int nSamples, nPatches; // Cols, Rows
[all...] |
| cmslut.c | 514 Data ->Params ->nSamples,
754 cmsUInt32Number* nSamples;
764 nSamples = clut->Params ->nSamples;
773 nTotalPoints = CubeSize(nSamples, nInputs);
782 cmsUInt32Number Colorant = rest % nSamples[t];
784 rest /= nSamples[t];
786 In[t] = _cmsQuantizeVal(Colorant, nSamples[t]);
816 cmsUInt32Number* nSamples;
820 nSamples = clut->Params ->nSamples
[all...] |
| cmsintrp.c | 104 const cmsUInt32Number nSamples[],
132 p -> nSamples[i] = nSamples[i];
133 p -> Domain[i] = nSamples[i] - 1;
139 p ->opta[i] = p ->opta[i-1] * nSamples[InputChan-i];
154 cmsInterpParams* _cmsComputeInterpParams(cmsContext ContextID, int nSamples, int InputChan, int OutputChan, const void* Table, cmsUInt32Number dwFlags)
161 Samples[i] = nSamples;
[all...] |
| /frameworks/av/include/media/stagefright/ |
| AudioSource.h | 97 void trackMaxAmplitude(int16_t *data, int nSamples);
|
| /external/libopus/silk/float/ |
| noise_shape_analysis_FLP.c | 136 opus_int k, nSamples;
182 nSamples = 2 * psEnc->sCmn.fs_kHz;
187 nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples );
193 pitch_res_ptr += nSamples;
|
| /external/libopus/silk/fixed/ |
| noise_shape_analysis_FIX.c | 153 opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0;
210 nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 );
215 silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples );
216 nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/
223 pitch_res_ptr += nSamples;
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_device_unittest.cc | 384 const size_t nSamples,
394 int32_t(const size_t nSamples,
424 const size_t nSamples,
438 audio_stream_->Write(audioSamples, nSamples);
446 int32_t RealNeedMorePlayData(const size_t nSamples,
456 nSamplesOut = nSamples;
460 audio_stream_->Read(audioSamples, nSamples);
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ios/ |
| audio_device_unittest_ios.cc | 374 const size_t nSamples,
384 int32_t(const size_t nSamples,
414 const size_t nSamples,
428 audio_stream_->Write(audioSamples, nSamples);
438 int32_t RealNeedMorePlayData(const size_t nSamples,
448 nSamplesOut = nSamples;
452 audio_stream_->Read(audioSamples, nSamples);
[all...] |
| /external/aac/libPCMutils/src/ |
| limiter.cpp | 222 const UINT nSamples)
230 FDK_ASSERT(gain_delay <= nSamples);
252 for (i = 0; i < nSamples; i++) {
|
| /external/aac/libAACdec/src/ |
| block.cpp | 684 int fr, fl, tl, nSamples, nSpec;
720 nSamples = imdct_block(
740 FDK_ASSERT(nSamples == frameLen);
|
