| /external/chromium_org/third_party/webrtc/common_audio/vad/ |
| vad_gmm.c | 46 inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2);
48 // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6);
57 *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inv_std2, tmp16, 10);
62 tmp32 = WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9);
70 tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(kLog2Exp, (int16_t) tmp32, 12);
|
| vad_sp.c | 41 WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], *signal_in, 14));
44 WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], tmp16_1, 12);
48 WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], *signal_in, 14));
51 WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], tmp16_2, 12);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| fft.c | 70 /* Uses 16x16 mul, without rounding, which is faster. Uses WEBRTC_SPL_MUL_16_16_RSFT */
148 RexQx[k1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, akpQx, 14) -
149 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, bkpQx, 14); // 6 non-mul + 2 mul cycles, i.e. 8 cycles (6+2*7=20 cycles if 16x32mul)
150 RexQx[k2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) -
151 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjpQx, 14);
152 RexQx[k3] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, akmQx, 14) -
153 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, bkmQx, 14);
154 ImxQx[k1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, akpQx, 14) +
155 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, bkpQx, 14);
156 ImxQx[k2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajpQx, 14)
[all...] |
| pitch_estimator_c.c | 39 ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[n],
42 csum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)x[n],
63 ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[k - 1],
66 ysum32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)in[PITCH_CORR_LEN2 + k - 1],
|
| pitch_filter.c | 96 gainsQ12[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
102 if ((WEBRTC_SPL_MUL_16_16_RSFT(lagsQ7[0], 3, 1) < oldLagQ7) ||
103 (lagsQ7[0] > WEBRTC_SPL_MUL_16_16_RSFT(oldLagQ7, 3, 1))) {
117 gaindeltaQ12 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
182 if ((WEBRTC_SPL_MUL_16_16_RSFT(lagsQ7[0], 3, 1) < oldLagQ7) ||
183 (lagsQ7[0] > WEBRTC_SPL_MUL_16_16_RSFT(oldLagQ7, 3, 1))) {
|
| decode.c | 132 PitchGains_Q12[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
138 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
139 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
178 tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(AvgPitchGain_Q12, 29, 0); // Q18
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
| interpolate_samples.c | 40 *tmpPtr++ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp2],*ppo, 15) +
41 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp1], *ppi, 15);
|
| lsf_to_lsp.c | 39 freq = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(lsf[i], 20861, 15);
|
| chebyshev.c | 54 WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15)), 2);
69 WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15), 1);
|
| do_plc.c | 95 crossSquareMax = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross, -shiftMax),
106 crossSquare = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1),
158 denom=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp1, tmp2, 16); /* denom in Q(scale1+scale2-16) */
213 WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kPlcPfSlope[ind], (max_perSquare-WebRtcIlbcfix_kPlcPerSqr[ind]), 11);
257 tot_gain=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(31130, use_gain, 15); /* 0.95*use_gain */
259 tot_gain=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(29491, use_gain, 15); /* 0.9*use_gain */
265 PLCresidual[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tot_gain,
273 energy += WEBRTC_SPL_MUL_16_16_RSFT(PLCresidual[i],
|
| window32_w32.c | 57 temp = (temp + (WEBRTC_SPL_MUL_16_16_RSFT(x_hi, y_low, 14)));
59 z[i] = (temp + (WEBRTC_SPL_MUL_16_16_RSFT(x_low, y_hi, 14)));
|
| cb_update_best_index.c | 68 gainW32 = WEBRTC_SPL_MUL_16_16_RSFT(
|
| get_lsp_poly.c | 71 WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(low, (*lspPtr), 15), 2);
|
| lsp_to_lsf.c | 71 tmp = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAcosDerivative[k],diff, 11);
|
| enhancer_interface.c | 152 corr16[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(corr16[i],
164 WEBRTC_SPL_MUL_16_16_RSFT(corr16[i], en16[ind], sh)) {
169 if (WEBRTC_SPL_MUL_16_16_RSFT(corr16[ind], en16[i], sh) <
301 (*tmpW16ptr)=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
324 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT((*enh_bufPtr1), win, 14);
325 *enh_bufPtr1 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
|
| poly_to_lsp.c | 123 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(x, y, (19-shifts));
132 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(ylow, y, 10);
|
| cb_mem_energy_augmentation.c | 48 WEBRTC_SPL_MUL_16_16_RSFT(*ppe, *ppe, scale);
|
| /external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
| energy.c | 30 en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling);
|
| refl_coef_to_lpc.c | 45 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT((*aptr2), (*kptr), 15);
|
| vector_scaling_operations.c | 100 (*outptr++) = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
119 tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
140 (*outptr++) = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
141 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
|
| ilbc_specific_functions.c | 35 (*outptr++) = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
50 (*outptr++) = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
|
| complex_fft.c | 222 tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
223 - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
226 (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
227 + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ns/ |
| nsx_core_c.c | 63 tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((int16_t)frac32, 5412, 12);
104 tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
136 tmp16no2 += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
207 inst->priorNonSpeechProb += (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
236 tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/cng/ |
| webrtc_cng.c | 347 inst->enc_reflCoefs[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
349 inst->enc_reflCoefs[i] += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
509 tmp3 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp1, Beta, 15);
510 tmp3 += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp2, BetaC, 15);
518 inst->dec_used_reflCoefs[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
520 inst->dec_used_reflCoefs[i] += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
541 temp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
545 En = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/aecm/ |
| aecm_core_c.c | 76 fft[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
80 fft[PART_LEN + i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
128 tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(ifft_out[PART_LEN + i],
255 tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(max_value, alpha, 15);
256 tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(min_value, beta, 15);
573 hnl[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], hnl[i], 14);
621 hnl[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14);
758 noiseRShift16[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16,
772 tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15);
775 uReal[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i]
[all...] |
