| /external/webrtc/src/common_audio/signal_processing/ |
| levinson_durbin.c | 46 temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
50 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
55 temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16)
56 + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31
68 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
78 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
91 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
101 temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
104 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
127 temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4)
[all...] |
| lpc_to_refl_coef.c | 42 tmp32[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16)
43 - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
54 k16[m - 1] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15
|
| division_operations.c | 119 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
128 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
133 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1);
141 tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
|
| splitting_filter.c | 134 half_in2[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k], 10);
135 half_in1[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k + 1], 10);
175 half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
177 half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
|
| spl_sqrt.c | 146 A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
160 A = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_norm, 16);
|
| signal_processing_unittest.cc | 97 EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
|
| /external/webrtc/src/modules/audio_processing/agc/ |
| digital_agc.c | 120 + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)limiterLvlX, 13),
147 inLevel = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)diffGain, 14) - inLevel; // Q14
189 numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
202 numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
217 tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
229 tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
239 tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
242 tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
250 gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
465 tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF)
[all...] |
| analog_agc.c | 458 micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
770 inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
[all...] |
| /external/webrtc/src/common_audio/vad/ |
| vad_filterbank.c | 87 int32_t state32 = WEBRTC_SPL_LSHIFT_W32((int32_t) (*filter_state), 16); // Q31
93 in32 = WEBRTC_SPL_LSHIFT_W32(((int32_t) (*in_vector)), 14);
95 state32 = WEBRTC_SPL_LSHIFT_W32(state32, 1);
|
| vad_core.c | 450 tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
463 tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
|
| /external/webrtc/src/modules/audio_processing/ |
| high_pass_filter_impl.cc | 87 WEBRTC_SPL_LSHIFT_W32(static_cast<WebRtc_Word32>(y[0]), 13)) << 2);
|
| /external/webrtc/src/modules/audio_processing/aecm/ |
| aecm_core.c | 350 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
411 j = WEBRTC_SPL_LSHIFT_W32(i, 1);
434 j = WEBRTC_SPL_LSHIFT_W32(i, 1);
537 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
539 aecm->channelAdapt32[i + 1] = WEBRTC_SPL_LSHIFT_W32(
541 aecm->channelAdapt32[i + 2] = WEBRTC_SPL_LSHIFT_W32(
543 aecm->channelAdapt32[i + 3] = WEBRTC_SPL_LSHIFT_W32(
546 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
[all...] |
| aecm_core_neon.c | 281 // aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)
293 aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
|
| /external/webrtc/src/modules/audio_processing/ns/ |
| nsx_core.c | 464 tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
[all...] |
| nsx_core_neon.c | 64 // tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
89 tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
|
| /external/webrtc/src/common_audio/signal_processing/include/ |
| signal_processing_library.h | 151 #define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c))
[all...] |
