Searched defs:nSamples (Results 1 - 25 of 27) sorted by relevance

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/external/chromium_org/third_party/opus/src/silk/
H A Dresampler.c181 opus_int nSamples; local
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 ) );
/external/libopus/silk/
H A Dresampler.c181 opus_int nSamples; local
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 ) );
/external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/
H A Dfake_audio_device_buffer.cc58 uint32_t nSamples) {
61 assert(nSamples == kDefaultBufSizeInSamples);
64 memcpy(buffer, audioBuffer, nSamples * sizeof(int16_t));
69 int32_t FakeAudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples) { argument
70 assert(nSamples == kDefaultBufSizeInSamples);
57 SetRecordedBuffer(const void* audioBuffer, uint32_t nSamples) argument
/external/srec/audio/test/AudioInRecord/src/
H A DAudioInRecord.c55 unsigned int nSamples; local
89 nSamples = 0;
90 while (nSamples <= N_SAMPLES_TO_RECORD - N_SAMPLES_PER_BUFFER)
95 lhsErr = lhs_audioinGetSamples(hAudioIn, &u32NbrOfSamples, &(recordedSamples[nSamples]), &AudioInInfo);
97 nSamples += u32NbrOfSamples;
139 fwrite(recordedSamples, sizeof(typeSample), nSamples, fpOutput);
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/
H A Dfakeaudiocapturemodule_unittest.cc59 const uint32_t nSamples,
68 rec_buffer_bytes_ = nSamples * nBytesPerSample;
82 virtual int32_t NeedMorePlayData(const uint32_t nSamples, argument
95 const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
58 RecordedDataIsAvailable(const void* audioSamples, const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument
/external/aac/libPCMutils/src/
H A Dlimiter.cpp222 const UINT nSamples)
230 FDK_ASSERT(gain_delay <= nSamples);
252 for (i = 0; i < nSamples; i++) {
216 applyLimiter(TDLimiterPtr limiter, INT_PCM* samples, FIXP_DBL* pGain, const INT* gain_scale, const UINT gain_size, const UINT gain_delay, const UINT nSamples) argument
/external/chromium_org/third_party/opus/src/silk/fixed/
H A Dnoise_shape_analysis_FIX.c153 opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; local
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/chromium_org/third_party/opus/src/silk/float/
H A Dnoise_shape_analysis_FLP.c136 opus_int k, nSamples; local
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/chromium_org/third_party/webrtc/modules/audio_device/
H A Daudio_device_buffer.cc387 // 16-bit,48kHz mono, 10ms => nSamples=480 => _recSize=2*480=960 bytes
388 // 16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes
392 uint32_t nSamples)
402 _recSamples = nSamples;
403 _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
410 if (nSamples != _recSamples)
412 WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples);
500 int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples) argument
523 _playSamples = nSamples;
391 SetRecordedBuffer(const void* audioBuffer, uint32_t nSamples) argument
[all...]
/external/libopus/silk/fixed/
H A Dnoise_shape_analysis_FIX.c153 opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; local
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/libopus/silk/float/
H A Dnoise_shape_analysis_FLP.c136 opus_int k, nSamples; local
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/chromium_org/third_party/webrtc/modules/audio_device/test/
H A Daudio_device_test_api.cc88 const uint32_t nSamples,
114 const uint32_t nSamples,
86 RecordedDataIsAvailable( const void* audioSamples, const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t sampleRate, const uint32_t totalDelay, const int32_t clockSkew, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument
113 NeedMorePlayData( const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t sampleRate, void* audioSamples, uint32_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
H A Dfunc_test_manager.h66 uint16_t nSamples; member in struct:AudioPacket
106 const uint32_t nSamples,
116 virtual int32_t NeedMorePlayData(const uint32_t nSamples,
H A Dfunc_test_manager.cc145 const uint32_t nSamples,
158 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
159 packet->nSamples = (uint16_t) nSamples;
290 const uint32_t nSamples,
304 memset(audioSamples, 0, nBytesPerSample * nSamples);
317 const uint16_t nSamplesIn = packet->nSamples;
345 * nSamples, lenOut);
351 * nSamples, lenOut);
357 for (unsigned int i = 0; i < nSamples;
143 RecordedDataIsAvailable( const void* audioSamples, const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument
289 NeedMorePlayData( const uint32_t nSamples, const uint8_t nBytesPerSample, const uint8_t nChannels, const uint32_t samplesPerSec, void* audioSamples, uint32_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
[all...]
/external/chromium_org/third_party/webrtc/voice_engine/
H A Dvoe_base_impl.cc122 uint32_t nSamples,
133 "VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, "
136 nSamples, nBytesPerSample, nChannels, samplesPerSec,
139 NULL, 0, audioSamples, samplesPerSec, nChannels, nSamples,
146 uint32_t nSamples,
156 "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
158 nSamples, nBytesPerSample, nChannels, samplesPerSec);
162 static_cast<int>(nSamples), true, audioSamples,
120 RecordedDataIsAvailable( const void* audioSamples, uint32_t nSamples, uint8_t nBytesPerSample, uint8_t nChannels, uint32_t samplesPerSec, uint32_t totalDelayMS, int32_t clockDrift, uint32_t micLevel, bool keyPressed, uint32_t& newMicLevel) argument
145 NeedMorePlayData( uint32_t nSamples, uint8_t nBytesPerSample, uint8_t nChannels, uint32_t samplesPerSec, void* audioSamples, uint32_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
H A Dtransmit_mixer.cc321 uint32_t nSamples,
330 "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
332 "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
337 nSamples, local
320 PrepareDemux(const void* audioSamples, uint32_t nSamples, uint8_t nChannels, uint32_t samplesPerSec, uint16_t totalDelayMS, int32_t clockDrift, uint16_t currentMicLevel, bool keyPressed) argument
/external/aac/libAACdec/src/
H A Dblock.cpp684 int fr, fl, tl, nSamples, nSpec; local
720 nSamples = imdct_block(
740 FDK_ASSERT(nSamples == frameLen);
/external/aac/libAACenc/src/
H A Dpsy_main.cpp418 INT nSamples,
423 for (k=0; k<nSamples; k++) {
416 FDKaacEnc_deinterleaveInputBuffer(INT_PCM *pOutputSamples, INT_PCM *pInputSamples, INT nSamples, INT nChannels) argument
/external/chromium_org/third_party/webrtc/modules/audio_device/android/
H A Daudio_track_jni.cc1324 uint32_t nSamples = local
1335 nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
1336 if (nSamples != samplesToPlay)
1339 " invalid number of output samples(%d)", nSamples);
1345 memcpy(_javaDirectPlayBuffer, playBuffer, nSamples * 2);
1353 2 * nSamples);
/external/pdfium/core/src/fxcodec/lcms2/lcms2-2.6/include/
H A Dlcms2_plugin.h293 cmsUInt32Number nSamples[MAX_INPUT_DIMENSIONS]; // Valid on all kinds of tables member in struct:_cms_interp_struc
294 cmsUInt32Number Domain[MAX_INPUT_DIMENSIONS]; // Domain = nSamples - 1
/external/pdfium/core/src/fxcodec/lcms2/lcms2-2.6/src/
H A Dcmsintrp.c104 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) argument
161 Samples[i] = nSamples;
103 _cmsComputeInterpParamsEx(cmsContext ContextID, const cmsUInt32Number nSamples[], int InputChan, int OutputChan, const void *Table, cmsUInt32Number dwFlags) argument
H A Dcmslut.c514 Data ->Params ->nSamples,
754 cmsUInt32Number* nSamples; local
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; local
820 nSamples
[all...]
/external/chromium_org/third_party/webrtc/modules/audio_device/linux/
H A Daudio_device_pulse_linux.cc2809 uint32_t nSamples = local
2820 nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
2821 if (nSamples != numPlaySamples)
2825 nSamples);
/external/chromium_org/third_party/webrtc/modules/audio_device/mac/
H A Daudio_device_mac.cc3113 uint32_t nSamples = local
3116 nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
3117 if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
3120 " invalid number of output samples(%d)", nSamples);
3123 uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
/external/chromium_org/third_party/webrtc/modules/audio_device/win/
H A Daudio_device_wave_win.cc3509 uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES); local
3518 nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
3519 if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES)
3521 WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples);
3576 int32_t AudioDeviceWindowsWave::Write(int8_t* data, uint16_t nSamples) argument
3591 const int16_t nBytes = (2*_playChannels)*nSamples;
3623 _writtenSamples += nSamples; // each sample is 2 or 4 bytes

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