/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
hp_input.c | 36 int32_t tmpW32; 46 tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */ 47 tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */ 48 tmpW32 = (tmpW32>>15); 49 tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */ 50 tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */ 51 tmpW32 = (tmpW32<<1); 53 tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] * [all...] |
hp_output.c | 36 int32_t tmpW32; 46 tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */ 47 tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */ 48 tmpW32 = (tmpW32>>15); 49 tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */ 50 tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */ 51 tmpW32 = (tmpW32<<1); 53 tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] * [all...] |
lsf_to_poly.c | 32 int32_t tmpW32; 73 tmpW32 = (*f1ptr) + (*f2ptr); 74 (*a1ptr) = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13); 76 tmpW32 = (*f1ptr) - (*f2ptr); 77 (*a2ptr) = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
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lsf_to_lsp.c | 35 int32_t tmpW32; 56 tmpW32 = WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kCosDerivative[k], diff); 57 lsp[i] = WebRtcIlbcfix_kCos[k]+(int16_t)(WEBRTC_SPL_RSHIFT_W32(tmpW32, 12));
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get_lsp_poly.c | 44 int32_t tmpW32; 70 tmpW32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(high, (*lspPtr)), 2) + 74 (*fPtr) -= tmpW32;
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poly_to_lsp.c | 37 int32_t tmpW32; 123 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(x, y, (19-shifts)); 126 y = (int16_t)(tmpW32&0xFFFF); 131 /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */ 132 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(ylow, y, 10); 133 xint = xlow-(int16_t)(tmpW32&0xFFFF);
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do_plc.c | 48 int32_t tmpW32, tmp2W32; 212 tmpW32 = (int32_t)WebRtcIlbcfix_kPlcPitchFact[ind] + 215 pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
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cb_search.c | 65 int32_t t32, tmpW32; 367 tmpW32 = ((int32_t)(gains[1]-1))<<1; 387 if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
pitch_filter_c.c | 34 int32_t tmpW32 = 0; 38 tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQpos2[*index2 + j], coefficient[j]); 42 tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104); 43 tmpW32 += 8192; 44 tmpW16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14); 53 tmpW32 = 0; 57 tmpW32 += WEBRTC_SPL_MUL_16_16(inputState[j], kDampFilter[j]); 61 tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208) [all...] |
pitch_filter.c | 76 int32_t tmpW32; 133 tmpW32 = WEBRTC_SPL_LSHIFT_W32(indW32, 7); 134 tmpW32 -= curLagQ7; 135 frcQQ = WEBRTC_SPL_RSHIFT_W32(tmpW32, 4); 175 int32_t tmpW32, tmp2W32, csum1QQ, esumxQQ; 223 tmpW32 = 0; 225 tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos3QQ + m], fracoeffQQ[m]); 231 tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32); 232 tmpW32 += 8192; 233 tmpW16 = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14) [all...] |
pitch_filter_armv6.S | 62 @ r2: tmpW32
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
pitch_filter_c.c | 34 WebRtc_Word32 tmpW32 = 0; 38 tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQpos2[*index2 + j], coefficient[j]); 42 tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104); 43 tmpW32 += 8192; 44 tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14); 53 tmpW32 = 0; 57 tmpW32 += WEBRTC_SPL_MUL_16_16(inputState[j], kDampFilter[j]); 61 tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208) [all...] |
pitch_filter.c | 87 WebRtc_Word32 tmpW32; 144 tmpW32 = WEBRTC_SPL_LSHIFT_W32(indW32, 7); 145 tmpW32 -= curLagQ7; 146 frcQQ = WEBRTC_SPL_RSHIFT_W32(tmpW32, 4); 186 WebRtc_Word32 tmpW32, tmp2W32, csum1QQ, esumxQQ; 234 tmpW32 = 0; 236 tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos3QQ + m], fracoeffQQ[m]); 242 tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32); 243 tmpW32 += 8192; 244 tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14) [all...] |
pitch_filter_armv6.S | 65 @ r2: tmpW32
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
division_operations.c | 103 int32_t tmpW32; 108 // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30) 109 tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1) 111 // tmpW32 = den * approx 113 tmpW32 = (int32_t)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx)) 115 // Store tmpW32 in hi and low format 116 tmp_hi = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16); 117 tmp_low = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32 [all...] |
vector_scaling_operations.c | 110 int32_t tmpW32; 119 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts); 120 (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
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/external/webrtc/src/common_audio/signal_processing/ |
division_operations.c | 104 WebRtc_Word32 tmpW32; 109 // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30) 110 tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1) 112 // tmpW32 = den * approx 114 tmpW32 = (WebRtc_Word32)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx)) 116 // Store tmpW32 in hi and low format 117 tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16); 118 tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32 [all...] |
vector_scaling_operations.c | 118 WebRtc_Word32 tmpW32; 127 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts); 128 (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
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