| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| D | cb_update_best_index.c | 43 shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax)); in WebRtcIlbcfix_CbUpdateBestIndex()
47 shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh); in WebRtcIlbcfix_CbUpdateBestIndex()
65 scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp); in WebRtcIlbcfix_CbUpdateBestIndex()
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| D | do_plc.c | 90 corrLen = (size_t)WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3)); in WebRtcIlbcfix_DoThePlc()
123 tmp1 = WEBRTC_SPL_MIN(31, (shiftMax<<1)+shift3-(shift1<<1)-shift2); in WebRtcIlbcfix_DoThePlc()
127 tmp2 = WEBRTC_SPL_MIN(31, (shift1<<1)+shift2-(shiftMax<<1)-shift3); in WebRtcIlbcfix_DoThePlc()
219 pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */ in WebRtcIlbcfix_DoThePlc()
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| D | enhancer_interface.c | 137 stop = WEBRTC_SPL_MIN(47, lagmax[i]) + 2; in WebRtcIlbcfix_EnhancerInterface()
162 sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]); in WebRtcIlbcfix_EnhancerInterface()
167 sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]); in WebRtcIlbcfix_EnhancerInterface()
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| D | encode.c | 298 for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++) in WebRtcIlbcfix_EncodeImpl()
320 end_count = WEBRTC_SPL_MIN (Nfor, (size_t)2); in WebRtcIlbcfix_EncodeImpl()
324 start_count = WEBRTC_SPL_MIN (Nfor, (size_t)2); in WebRtcIlbcfix_EncodeImpl()
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| D | gain_quant.c | 98 loc=WEBRTC_SPL_MIN(loc, (cblen-1)); in WebRtcIlbcfix_GainQuant()
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| D | cb_search_core.c | 98 tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr)); in WebRtcIlbcfix_CbSearchCore()
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| D | xcorr_coef.c | 106 scalediff=WEBRTC_SPL_MIN(scalediff,31); in WebRtcIlbcfix_XcorrCoef()
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| D | cb_search.c | 248 WEBRTC_SPL_MIN(39, (eInd + 20)), scale); in WebRtcIlbcfix_CbSearch()
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| /external/webrtc/webrtc/modules/audio_processing/aecm/ |
| D | echo_control_mobile.c | 340 = WEBRTC_SPL_MIN((3 * aecm->sum in WebRtcAecm_Process()
349 aecm->bufSizeStart = WEBRTC_SPL_MIN((3 * aecm->msInSndCardBuf in WebRtcAecm_Process()
639 nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp); in WebRtcAecm_DelayComp()
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| /external/webrtc/webrtc/modules/audio_processing/ns/ |
| D | nsx_core.c | 1113 minPause = WEBRTC_SPL_MIN(minPause, inst->avgMagnPause[i]); in WebRtcNsx_ComputeSpectralDifference()
1163 avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1); in WebRtcNsx_ComputeSpectralDifference()
1613 q_domain_to_use = WEBRTC_SPL_MIN((int)qNoise, inst->minNorm - inst->stages); in WebRtcNsx_ProcessCore()
1756 postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11 in WebRtcNsx_ProcessCore()
1770 tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11 in WebRtcNsx_ProcessCore()
1825 inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF, in WebRtcNsx_ProcessCore()
1978 norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1)); in WebRtcNsx_ProcessCore()
1984 curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11 in WebRtcNsx_ProcessCore()
2077 gainModHB = WEBRTC_SPL_MIN(avgProbSpeechHB, 3607); in WebRtcNsx_ProcessCore()
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| D | nsx_core_c.c | 150 normTmp = WEBRTC_SPL_MIN(20 - inst->stages, in WebRtcNsx_SpeechNoiseProb()
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| D | nsx_core_mips.c | 185 normTmp = WEBRTC_SPL_MIN(20 - inst->stages, in WebRtcNsx_SpeechNoiseProb()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| D | pitch_estimator.c | 291 npkq = WEBRTC_SPL_MIN(peaks_indq, 4); in WebRtcIsacfix_InitialPitch()
365 npkq = WEBRTC_SPL_MIN(peaks_indq, 4); in WebRtcIsacfix_InitialPitch()
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| /external/webrtc/webrtc/common_audio/signal_processing/ |
| D | min_max_operations.c | 76 maximum = WEBRTC_SPL_MIN(maximum, WEBRTC_SPL_WORD32_MAX); in WebRtcSpl_MaxAbsValueW32C()
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| D | min_max_operations_neon.c | 119 maximum = WEBRTC_SPL_MIN(maximum, WEBRTC_SPL_WORD32_MAX); in WebRtcSpl_MaxAbsValueW32Neon()
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| D | signal_processing_unittest.cc | 34 EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B)); in TEST_F()
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| /external/webrtc/webrtc/modules/audio_processing/aec/ |
| D | echo_cancellation.c | 661 WEBRTC_SPL_MIN((3 * aecpc->sum * aecpc->rate_factor * 8) / in ProcessNormal()
671 aecpc->bufSizeStart = WEBRTC_SPL_MIN( in ProcessNormal()
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| D | aec_core.c | 1127 hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]); in EchoSuppression()
1147 WEBRTC_SPL_MIN(aec->hNlFbLocalMin + 0.0008f / aec->mult, 1); in EchoSuppression()
1148 aec->hNlXdAvgMin = WEBRTC_SPL_MIN(aec->hNlXdAvgMin + 0.0006f / aec->mult, 1); in EchoSuppression()
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| /external/webrtc/webrtc/modules/audio_processing/agc/legacy/ |
| D | analog_agc.c | 540 *inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax); in WebRtcAgc_ZeroCtrl()
1131 *outMicLevel = WEBRTC_SPL_MIN(stt->micVol, stt->maxAnalog) >> stt->scale; in WebRtcAgc_ProcessAnalog()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| D | isac.c | 60 int16_t lim30MsPayloadBytes = WEBRTC_SPL_MIN( in UpdatePayloadSizeLimit()
63 int16_t lim60MsPayloadBytes = WEBRTC_SPL_MIN( in UpdatePayloadSizeLimit()
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| /external/webrtc/webrtc/common_audio/signal_processing/include/ |
| D | signal_processing_library.h | 30 #define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value macro
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