/external/chromium_org/third_party/opus/src/silk/ |
H A D | resampler.c | 181 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 ) );
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/external/libopus/silk/ |
H A D | resampler.c | 181 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 ) );
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/external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/ |
H A D | fake_audio_device_buffer.cc | 58 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
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/external/srec/audio/test/AudioInRecord/src/ |
H A D | AudioInRecord.c | 55 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);
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | fakeaudiocapturemodule_unittest.cc | 59 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
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/external/aac/libPCMutils/src/ |
H A D | limiter.cpp | 222 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
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/external/chromium_org/third_party/opus/src/silk/fixed/ |
H A D | noise_shape_analysis_FIX.c | 153 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;
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/external/chromium_org/third_party/opus/src/silk/float/ |
H A D | noise_shape_analysis_FLP.c | 136 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;
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/external/chromium_org/third_party/webrtc/modules/audio_device/ |
H A D | audio_device_buffer.cc | 387 // 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 D | noise_shape_analysis_FIX.c | 153 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;
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/external/libopus/silk/float/ |
H A D | noise_shape_analysis_FLP.c | 136 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;
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/external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
H A D | audio_device_test_api.cc | 88 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
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H A D | func_test_manager.h | 66 uint16_t nSamples; member in struct:AudioPacket 106 const uint32_t nSamples, 116 virtual int32_t NeedMorePlayData(const uint32_t nSamples,
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H A D | func_test_manager.cc | 145 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 D | voe_base_impl.cc | 122 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
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H A D | transmit_mixer.cc | 321 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
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/external/aac/libAACdec/src/ |
H A D | block.cpp | 684 int fr, fl, tl, nSamples, nSpec; local 720 nSamples = imdct_block( 740 FDK_ASSERT(nSamples == frameLen);
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/external/aac/libAACenc/src/ |
H A D | psy_main.cpp | 418 INT nSamples, 423 for (k=0; k<nSamples; k++) { 416 FDKaacEnc_deinterleaveInputBuffer(INT_PCM *pOutputSamples, INT_PCM *pInputSamples, INT nSamples, INT nChannels) argument
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/external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
H A D | audio_track_jni.cc | 1324 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);
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/external/pdfium/core/src/fxcodec/lcms2/lcms2-2.6/include/ |
H A D | lcms2_plugin.h | 293 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
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/external/pdfium/core/src/fxcodec/lcms2/lcms2-2.6/src/ |
H A D | 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) argument 161 Samples[i] = nSamples; 103 _cmsComputeInterpParamsEx(cmsContext ContextID, const cmsUInt32Number nSamples[], int InputChan, int OutputChan, const void *Table, cmsUInt32Number dwFlags) argument
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H A D | cmslut.c | 514 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 D | audio_device_pulse_linux.cc | 2809 uint32_t nSamples = local 2820 nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer); 2821 if (nSamples != numPlaySamples) 2825 nSamples);
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/external/chromium_org/third_party/webrtc/modules/audio_device/mac/ |
H A D | audio_device_mac.cc | 3113 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;
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/external/chromium_org/third_party/webrtc/modules/audio_device/win/ |
H A D | audio_device_wave_win.cc | 3509 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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