Searched refs:WEBRTC_SPL_DIV (Results 1 – 9 of 9) sorted by relevance
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
D | bandwidth_estimator.c | 300 bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 + in WebRtcIsacfix_UpdateUplinkBwImpl() 856 …if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_L… in WebRtcIsacfix_GetMinBytes() 858 inv_Q12 = WEBRTC_SPL_DIV(4096, WEBRTC_SPL_MUL(BURST_LEN, FrameSamples)); in WebRtcIsacfix_GetMinBytes() 862 inv_Q12 = WEBRTC_SPL_DIV(4096, FrameSamples); in WebRtcIsacfix_GetMinBytes() 897 …if (WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, FS8), FrameSamples) > (WEBRTC_SPL_MUL(517, BottleNec… in WebRtcIsacfix_GetMinBytes() 900 State->ExceedAgo -= WEBRTC_SPL_DIV(BURST_INTERVAL, BURST_LEN - 1); in WebRtcIsacfix_GetMinBytes() 924 …TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck); … in WebRtcIsacfix_GetMinBytes() 953 …TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 100… in WebRtcIsacfix_UpdateRateModel()
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D | decode.c | 79 frame_mode = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */ in WebRtcIsacfix_DecodeImpl() 80 …processed_samples = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either … in WebRtcIsacfix_DecodeImpl()
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D | arith_routines_logist.c | 286 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1); in WebRtcIsacfix_DecLogisticMulti2() 290 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1); in WebRtcIsacfix_DecLogisticMulti2()
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D | entropy_coding.c | 319 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1); in CalcRootInvArSpec() 323 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1); in CalcRootInvArSpec() 337 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1); in CalcRootInvArSpec() 341 newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1); in CalcRootInvArSpec()
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/external/webrtc/src/modules/audio_processing/agc/ |
D | analog_agc.c | 211 targetGainIdx = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32, tmp16); in WebRtcAgc_AddMic() 1074 volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32, in WebRtcAgc_ProcessAnalog() 1134 volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32, in WebRtcAgc_ProcessAnalog()
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D | digital_agc.c | 213 y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14 in WebRtcAgc_CalculateGainTable()
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/external/webrtc/src/common_audio/signal_processing/ |
D | signal_processing_unittest.cc | 55 EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b)); in TEST_F()
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/external/webrtc/src/common_audio/signal_processing/include/ |
D | signal_processing_library.h | 79 #define WEBRTC_SPL_DIV(a, b) \ macro
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/external/webrtc/src/modules/audio_processing/ns/ |
D | nsx_core.c | 1513 nonSpeechProbFinal[i] = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32no1, in WebRtcNsx_SpeechNoiseProb() 1820 energyRatio = (WebRtc_Word16)WEBRTC_SPL_DIV(energyOut in WebRtcNsx_DataSynthesis()
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