| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.c | 203 recRtpRate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
218 recRtpRate = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
295 bweStr->recBwInv = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recBwInv, (WebRtc_Word32)reductionFactor);
389 tempUpper = WEBRTC_SPL_MUL(tempUpper, numBytesInv);
390 tempLower = WEBRTC_SPL_MUL(tempLower, numBytesInv);
427 arrTimeProj = WEBRTC_SPL_MUL((WebRtc_Word32)8000, recBwAvgInv);
431 arrTimeProj = WEBRTC_SPL_MUL(((WebRtc_Word32)pksize + HEADER_SIZE), arrTimeProj);
450 bweStr->recJitter = WEBRTC_SPL_MUL(weight, WEBRTC_SPL_LSHIFT_W32(arrTimeNoiseAbs, 5))
451 + WEBRTC_SPL_MUL(1024 - weight, bweStr->recJitter);
463 bweStr->recJitterShortTermAbs = WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32(arrTimeNoiseAbs, 3))
[all...] |
| entropy_coding.c | 154 sum += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], diff[n]) + 256, 9);
161 sum += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], summ[n]) + 256, 9);
181 sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
182 sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
183 CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
199 sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
201 CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
209 CurveQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], CorrQ11[k+1]) + 2, 2);
225 diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
229 diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2)
[all...] |
| arith_routines_logist.c | 74 ind = WEBRTC_SPL_MUL(5, qtmp1 - kHistEdges[0]);
|
| lattice.c | 22 ((WebRtc_Word32)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
|
| isacfix.c | [all...] |
| /external/webrtc/src/common_audio/signal_processing/ |
| get_scaling_square.c | 35 t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
|
| auto_correlation.c | 54 int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); // # of bits to normalize smax
|
| signal_processing_unittest.cc | 42 EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
43 EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
|
| /external/webrtc/src/modules/audio_processing/agc/ |
| analog_agc.c | 652 tmp32 = WEBRTC_SPL_MUL(1126, *inMicLevel);
853 stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 3), 7);
958 stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, RXX_BUFFER_LEN);
977 stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
[all...] |
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| entropy_coding.c | 107 sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
109 sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6),
111 CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
128 sum += WEBRTC_SPL_MUL(ARCoefQ12[n - k], ARCoefQ12[n]); /* Q24 */
130 CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round,
139 CurveQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(
158 diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(
164 diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(
[all...] |
| /external/webrtc/src/modules/audio_processing/ns/ |
| nsx_core.c | [all...] |
| /external/webrtc/src/common_audio/signal_processing/include/ |
| signal_processing_library.h | 63 #define WEBRTC_SPL_MUL(a, b) \
[all...] |
