/external/webrtc/sdk/objc/unittests/ |
D | RTCAudioDeviceModule_xctest.mm | 19 typedef int32_t(^NeedMorePlayDataBlock)(const size_t nSamples, 29 const size_t nSamples, 54 int32_t NeedMorePlayData(const size_t nSamples, 62 return needMorePlayDataBlock(nSamples, 73 const size_t nSamples, 83 nSamples, 249 mock.expectNeedMorePlayData(^int32_t(const size_t nSamples, 257 nSamplesOut = nSamples; 258 XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer()); 286 mock2.expectNeedMorePlayData(^int32_t(const size_t nSamples, [all …]
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/external/pdfium/third_party/lcms/ |
D | 0032-cgats-allocation.patch | 6 t-> nSamples = atoi(cmsIT8GetProperty(it8, "NUMBER_OF_FIELDS")); 9 - t-> Data = (char**)AllocChunk (it8, ((cmsUInt32Number) t->nSamples + 1) * ((cmsUInt32Number) t… 11 + if (t -> nSamples < 0 || t->nSamples > 0x7ffe || t->nPatches < 0 || t->nPatches > 0x7ffe) 16 + t->Data = (char**)AllocChunk(it8, ((cmsUInt32Number)t->nSamples + 1) * ((cmsUInt32Number)t…
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/external/libopus/silk/ |
D | resampler.c | 181 opus_int nSamples; in silk_resampler() local 188 nSamples = S->Fs_in_kHz - S->inputDelay; in silk_resampler() 191 silk_memcpy( &S->delayBuf[ S->inputDelay ], in, nSamples * sizeof( opus_int16 ) ); in silk_resampler() 196 …silk_resampler_private_up2_HQ_wrapper( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in… in silk_resampler() 200 … silk_resampler_private_IIR_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); in silk_resampler() 204 …silk_resampler_private_down_FIR( S, &out[ S->Fs_out_kHz ], &in[ nSamples ], inLen - S->Fs_in_kHz ); in silk_resampler() 208 …silk_memcpy( &out[ S->Fs_out_kHz ], &in[ nSamples ], ( inLen - S->Fs_in_kHz ) * sizeof( opus_int16… in silk_resampler()
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/external/aac/libSACenc/src/ |
D | sacenc_vectorfunctions.h | 471 const INT nSamples) { in FDKmemcpy_flex() argument 474 for (i = 0; i < nSamples; i++) { in FDKmemcpy_flex() 480 inline void FDKmemset_flex(T *const x, const T c, const INT nSamples) { in FDKmemset_flex() argument 483 for (i = 0; i < nSamples; i++) { in FDKmemset_flex()
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/external/pdfium/third_party/lcms/src/ |
D | cmscgats.c | 123 int nSamples, nPatches; // Cols, Rows member 1448 t -> nSamples = (int) cmsIT8GetPropertyDbl(it8, "NUMBER_OF_FIELDS"); in AllocateDataFormat() 1450 if (t -> nSamples <= 0) { in AllocateDataFormat() 1453 t -> nSamples = 10; in AllocateDataFormat() 1456 … t -> DataFormat = (char**) AllocChunk (it8, ((cmsUInt32Number) t->nSamples + 1) * sizeof(char *)); in AllocateDataFormat() 1483 if (n > t -> nSamples) { in SetDataFormat() 1509 t-> nSamples = atoi(cmsIT8GetProperty(it8, "NUMBER_OF_FIELDS")); in AllocateDataSet() 1512 if (t -> nSamples < 0 || t->nSamples > 0x7ffe || t->nPatches < 0 || t->nPatches > 0x7ffe) in AllocateDataSet() 1517 …t->Data = (char**)AllocChunk(it8, ((cmsUInt32Number)t->nSamples + 1) * ((cmsUInt32Number)t->nPatch… in AllocateDataSet() 1530 int nSamples = t -> nSamples; in GetData() local [all …]
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D | cmslut.c | 514 Data ->Params ->nSamples, in CLUTElemDup() 755 cmsUInt32Number* nSamples; in cmsStageSampleCLut16bit() local 765 nSamples = clut->Params ->nSamples; in cmsStageSampleCLut16bit() 777 nTotalPoints = CubeSize(nSamples, nInputs); in cmsStageSampleCLut16bit() 786 cmsUInt32Number Colorant = rest % nSamples[t]; in cmsStageSampleCLut16bit() 788 rest /= nSamples[t]; in cmsStageSampleCLut16bit() 790 In[t] = _cmsQuantizeVal(Colorant, nSamples[t]); in cmsStageSampleCLut16bit() 821 cmsUInt32Number* nSamples; in cmsStageSampleCLutFloat() local 825 nSamples = clut->Params ->nSamples; in cmsStageSampleCLutFloat() 834 nTotalPoints = CubeSize(nSamples, nInputs); in cmsStageSampleCLutFloat() [all …]
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/external/webrtc/pc/test/ |
D | fake_audio_capture_module_unittest.cc | 39 const size_t nSamples, in RecordedDataIsAvailable() argument 49 rec_buffer_bytes_ = nSamples * nBytesPerSample; in RecordedDataIsAvailable() 72 int32_t NeedMorePlayData(const size_t nSamples, in NeedMorePlayData() argument 82 const size_t audio_buffer_size = nSamples * nBytesPerSample; in NeedMorePlayData()
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/external/webrtc/modules/audio_device/ |
D | mock_audio_device_buffer.h | 23 MOCK_METHOD(int32_t, RequestPlayoutData, (size_t nSamples), (override)); 27 (const void* audioBuffer, size_t nSamples),
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D | audio_device_data_observer.cc | 50 const size_t nSamples, in RecordedDataIsAvailable() argument 62 observer_->OnCaptureData(audioSamples, nSamples, nBytesPerSample, in RecordedDataIsAvailable() 69 audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec, in RecordedDataIsAvailable() 76 int32_t NeedMorePlayData(const size_t nSamples, in NeedMorePlayData() argument 93 nSamples, nBytesPerSample, nChannels, samples_per_sec, audioSamples, in NeedMorePlayData() 99 observer_->OnRenderData(audioSamples, nSamples, nBytesPerSample, in NeedMorePlayData()
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/external/webrtc/audio/ |
D | audio_transport_impl.cc | 173 int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples, in NeedMorePlayData() argument 189 RTC_DCHECK_EQ(nSamples * 100, samplesPerSec); in NeedMorePlayData() 190 RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels, in NeedMorePlayData() 205 RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples); in NeedMorePlayData()
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D | audio_transport_impl.h | 36 const size_t nSamples, 46 int32_t NeedMorePlayData(const size_t nSamples,
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/external/webrtc/modules/audio_device/include/ |
D | mock_audio_transport.h | 28 const size_t nSamples, 41 (const size_t nSamples,
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D | audio_device_defines.h | 38 const size_t nSamples, 49 virtual int32_t NeedMorePlayData(const size_t nSamples,
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/external/libopus/silk/float/ |
D | noise_shape_analysis_FLP.c | 155 opus_int k, nSamples, nSegs; in silk_noise_shape_analysis_FLP() local 201 nSamples = 2 * psEnc->sCmn.fs_kHz; in silk_noise_shape_analysis_FLP() 207 nrg = ( silk_float )nSamples + ( silk_float )silk_energy_FLP( pitch_res_ptr, nSamples ); in silk_noise_shape_analysis_FLP() 213 pitch_res_ptr += nSamples; in silk_noise_shape_analysis_FLP()
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/external/libopus/silk/fixed/mips/ |
D | noise_shape_analysis_FIX_mipsr1.h | 43 opus_int k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0; in silk_noise_shape_analysis_FIX() local 100 nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 ); in silk_noise_shape_analysis_FIX() 105 silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples ); in silk_noise_shape_analysis_FIX() 106 nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/ in silk_noise_shape_analysis_FIX() 113 pitch_res_ptr += nSamples; in silk_noise_shape_analysis_FIX()
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/external/exoplayer/tree/extensions/flac/src/main/jni/ |
D | flac_parser.cc | 227 unsigned bytesPerSample, unsigned nSamples, in copyToByteArrayBigEndian() argument 229 for (unsigned i = 0; i < nSamples; ++i) { in copyToByteArrayBigEndian() 244 unsigned nSamples, unsigned nChannels) { in copyToByteArrayLittleEndian() argument 245 for (unsigned i = 0; i < nSamples; ++i) { in copyToByteArrayLittleEndian()
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/external/oss-fuzz/projects/flac/ |
D | fuzzer_exo.cpp | 214 unsigned bytesPerSample, unsigned nSamples, in copyToByteArrayBigEndian() argument 216 for (unsigned i = 0; i < nSamples; ++i) { in copyToByteArrayBigEndian() 231 unsigned nSamples, unsigned nChannels) { in copyToByteArrayLittleEndian() argument 232 for (unsigned i = 0; i < nSamples; ++i) { in copyToByteArrayLittleEndian()
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/external/libopus/silk/fixed/ |
D | noise_shape_analysis_FIX.c | 149 opus_int k, i, nSamples, nSegs, Qnrg, b_Q14, warping_Q16, scale = 0; in silk_noise_shape_analysis_FIX() local 204 nSamples = silk_LSHIFT( psEnc->sCmn.fs_kHz, 1 ); in silk_noise_shape_analysis_FIX() 210 silk_sum_sqr_shift( &nrg, &scale, pitch_res_ptr, nSamples ); in silk_noise_shape_analysis_FIX() 211 nrg += silk_RSHIFT( nSamples, scale ); /* Q(-scale)*/ in silk_noise_shape_analysis_FIX() 218 pitch_res_ptr += nSamples; in silk_noise_shape_analysis_FIX()
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/external/aac/libAACdec/src/ |
D | aacdec_drc.cpp | 1408 const UINT nSamples, const UINT channels, in applyDrcLevelNormalization() argument 1414 FDK_ASSERT(gain_delay <= nSamples); in applyDrcLevelNormalization() 1433 scaleValuesSaturate(samplesIn, channels * nSamples, in applyDrcLevelNormalization() 1440 for (i = 0; i < nSamples; i++) { in applyDrcLevelNormalization() 1444 for (i = 0; i < channels * nSamples; i++) { in applyDrcLevelNormalization() 1454 for (i = 0; i < nSamples; i++) { in applyDrcLevelNormalization()
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D | aacdec_drc.h | 238 const UINT nSamples, const UINT channels,
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D | block.cpp | 1024 LONG nSamples; in CBlock_FrequencyToTime() local 1132 nSamples = in CBlock_FrequencyToTime() 1151 nSamples = in CBlock_FrequencyToTime() 1167 FDK_ASSERT(nSamples == frameLen); in CBlock_FrequencyToTime() 1225 nSamples = in CBlock_FrequencyToTime() 1245 FDK_ASSERT(nSamples == frameLen); in CBlock_FrequencyToTime()
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/external/webrtc/modules/audio_device/android/ |
D | audio_device_unittest.cc | 390 const size_t nSamples, in RealRecordedDataIsAvailable() argument 404 audio_stream_->Write(audioSamples, nSamples); in RealRecordedDataIsAvailable() 412 int32_t RealNeedMorePlayData(const size_t nSamples, in RealNeedMorePlayData() argument 422 nSamplesOut = nSamples; in RealNeedMorePlayData() 426 audio_stream_->Read(audioSamples, nSamples); in RealNeedMorePlayData()
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/external/aac/libPCMutils/include/ |
D | limiter.h | 263 const INT scaling, const UINT nSamples);
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/external/aac/libSACdec/src/ |
D | sac_dec_interface.h | 313 UINT nSamples, UINT *pControlFlags, int numInputChannels,
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/external/webrtc/sdk/android/native_unittests/audio_device/ |
D | audio_device_unittest.cc | 387 const size_t nSamples, in RealRecordedDataIsAvailable() argument 401 audio_stream_->Write(audioSamples, nSamples); in RealRecordedDataIsAvailable() 409 int32_t RealNeedMorePlayData(const size_t nSamples, in RealNeedMorePlayData() argument 419 nSamplesOut = nSamples; in RealNeedMorePlayData() 423 audio_stream_->Read(audioSamples, nSamples); in RealNeedMorePlayData()
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