/external/webrtc/src/common_audio/signal_processing/ |
D | levinson_durbin.c | 46 temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm); in WebRtcSpl_LevinsonDurbin() 50 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1); in WebRtcSpl_LevinsonDurbin() 55 temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16) in WebRtcSpl_LevinsonDurbin() 56 + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31 in WebRtcSpl_LevinsonDurbin() 68 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1); in WebRtcSpl_LevinsonDurbin() 78 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1); in WebRtcSpl_LevinsonDurbin() 91 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1); in WebRtcSpl_LevinsonDurbin() 101 temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp); in WebRtcSpl_LevinsonDurbin() 104 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1); in WebRtcSpl_LevinsonDurbin() 127 temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4); in WebRtcSpl_LevinsonDurbin() [all …]
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D | division_operations.c | 119 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1); in WebRtcSpl_DivW32HiLow() 128 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1); in WebRtcSpl_DivW32HiLow() 133 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1); in WebRtcSpl_DivW32HiLow() 141 tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3); in WebRtcSpl_DivW32HiLow()
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D | lpc_to_refl_coef.c | 42 tmp32[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16) in WebRtcSpl_LpcToReflCoef() 43 - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1); in WebRtcSpl_LpcToReflCoef() 54 k16[m - 1] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15 in WebRtcSpl_LpcToReflCoef()
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D | splitting_filter.c | 134 half_in2[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k], 10); in WebRtcSpl_AnalysisQMF() 135 half_in1[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k + 1], 10); in WebRtcSpl_AnalysisQMF() 175 half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10); in WebRtcSpl_SynthesisQMF() 177 half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10); in WebRtcSpl_SynthesisQMF()
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D | spl_sqrt.c | 146 A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A in WebRtcSpl_Sqrt() 160 A = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_norm, 16); in WebRtcSpl_Sqrt()
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D | signal_processing_unittest.cc | 97 EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1)); in TEST_F()
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/external/webrtc/src/modules/audio_processing/agc/ |
D | digital_agc.c | 120 + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)limiterLvlX, 13), in WebRtcAgc_CalculateGainTable() 147 inLevel = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)diffGain, 14) - inLevel; // Q14 in WebRtcAgc_CalculateGainTable() 189 numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14 in WebRtcAgc_CalculateGainTable() 202 numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros) in WebRtcAgc_CalculateGainTable() 217 tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14 in WebRtcAgc_CalculateGainTable() 229 tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16) in WebRtcAgc_CalculateGainTable() 239 tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart; in WebRtcAgc_CalculateGainTable() 242 tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2; in WebRtcAgc_CalculateGainTable() 250 gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart) in WebRtcAgc_CalculateGainTable() 465 tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF); in WebRtcAgc_ProcessDigital() [all …]
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D | analog_agc.c | 458 micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale); in WebRtcAgc_VirtualMic() 770 inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale); in WebRtcAgc_ProcessAnalog() 1071 tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14); in WebRtcAgc_ProcessAnalog() 1131 tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14); in WebRtcAgc_ProcessAnalog() 1579 maxLevel = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale); in WebRtcAgc_Init() 1580 minLevel = WEBRTC_SPL_LSHIFT_W32(minLevel, stt->scale); in WebRtcAgc_Init()
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/external/webrtc/src/common_audio/vad/ |
D | vad_filterbank.c | 87 int32_t state32 = WEBRTC_SPL_LSHIFT_W32((int32_t) (*filter_state), 16); // Q31 in WebRtcVad_Allpass() 93 in32 = WEBRTC_SPL_LSHIFT_W32(((int32_t) (*in_vector)), 14); in WebRtcVad_Allpass() 95 state32 = WEBRTC_SPL_LSHIFT_W32(state32, 1); in WebRtcVad_Allpass()
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D | vad_core.c | 450 tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29 in WebRtcVad_GmmProbability() 463 tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); in WebRtcVad_GmmProbability()
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/external/webrtc/src/modules/audio_processing/aecm/ |
D | aecm_core.c | 350 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32( in WebRtcAecm_InitEchoPathCore() 411 j = WEBRTC_SPL_LSHIFT_W32(i, 1); in InverseFFTAndWindowC() 434 j = WEBRTC_SPL_LSHIFT_W32(i, 1); in InverseFFTAndWindowC() 537 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32( in ResetAdaptiveChannelC() 539 aecm->channelAdapt32[i + 1] = WEBRTC_SPL_LSHIFT_W32( in ResetAdaptiveChannelC() 541 aecm->channelAdapt32[i + 2] = WEBRTC_SPL_LSHIFT_W32( in ResetAdaptiveChannelC() 543 aecm->channelAdapt32[i + 3] = WEBRTC_SPL_LSHIFT_W32( in ResetAdaptiveChannelC() 546 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16); in ResetAdaptiveChannelC() 1954 outLShift32 = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise); in ComfortNoise() 2016 aecm->noiseEst[i] = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise); in ComfortNoise()
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D | aecm_core_neon.c | 293 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32( in ResetAdaptiveChannelNeon()
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/external/webrtc/src/modules/audio_processing/ns/ |
D | nsx_core.c | 464 tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16); in UpdateNoiseEstimate() 717 tmp32no1 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(inst->minNorm - inst->stages), 11); in WebRtcNsx_CalcParametricNoiseEstimate() 1173 logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17 in WebRtcNsx_ComputeSpectralFlatness() 1181 currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart); in WebRtcNsx_ComputeSpectralFlatness() 1494 invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart) in WebRtcNsx_SpeechNoiseProb() 1511 tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inst->priorNonSpeechProb, 8); // Q22 in WebRtcNsx_SpeechNoiseProb() 1717 tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)sum_log_i_square, 4); in WebRtcNsx_DataAnalysis() 1730 tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9 in WebRtcNsx_DataAnalysis() 1747 tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)net_norm, 11); // Q11 in WebRtcNsx_DataAnalysis() 2264 tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)magnU16[i], nShifts) in WebRtcNsx_ProcessCore()
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D | nsx_core_neon.c | 89 tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16); in UpdateNoiseEstimateNeon()
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/external/webrtc/src/modules/audio_processing/ |
D | high_pass_filter_impl.cc | 87 WEBRTC_SPL_LSHIFT_W32(static_cast<WebRtc_Word32>(y[0]), 13)) << 2); in Filter()
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/external/webrtc/src/common_audio/signal_processing/include/ |
D | signal_processing_library.h | 151 #define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c)) macro
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