| /* |
| * G.729, G729 Annex D postfilter |
| * Copyright (c) 2008 Vladimir Voroshilov |
| * |
| * This file is part of FFmpeg. |
| * |
| * FFmpeg is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public |
| * License as published by the Free Software Foundation; either |
| * version 2.1 of the License, or (at your option) any later version. |
| * |
| * FFmpeg is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with FFmpeg; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| #include <inttypes.h> |
| #include <limits.h> |
| |
| #include "avcodec.h" |
| #include "g729.h" |
| #include "acelp_pitch_delay.h" |
| #include "g729postfilter.h" |
| #include "celp_math.h" |
| #include "acelp_filters.h" |
| #include "acelp_vectors.h" |
| #include "celp_filters.h" |
| |
| #define FRAC_BITS 15 |
| #include "mathops.h" |
| |
| /** |
| * short interpolation filter (of length 33, according to spec) |
| * for computing signal with non-integer delay |
| */ |
| static const int16_t ff_g729_interp_filt_short[(ANALYZED_FRAC_DELAYS+1)*SHORT_INT_FILT_LEN] = { |
| 0, 31650, 28469, 23705, 18050, 12266, 7041, 2873, |
| 0, -1597, -2147, -1992, -1492, -933, -484, -188, |
| }; |
| |
| /** |
| * long interpolation filter (of length 129, according to spec) |
| * for computing signal with non-integer delay |
| */ |
| static const int16_t ff_g729_interp_filt_long[(ANALYZED_FRAC_DELAYS+1)*LONG_INT_FILT_LEN] = { |
| 0, 31915, 29436, 25569, 20676, 15206, 9639, 4439, |
| 0, -3390, -5579, -6549, -6414, -5392, -3773, -1874, |
| 0, 1595, 2727, 3303, 3319, 2850, 2030, 1023, |
| 0, -887, -1527, -1860, -1876, -1614, -1150, -579, |
| 0, 501, 859, 1041, 1044, 892, 631, 315, |
| 0, -266, -453, -543, -538, -455, -317, -156, |
| 0, 130, 218, 258, 253, 212, 147, 72, |
| 0, -59, -101, -122, -123, -106, -77, -40, |
| }; |
| |
| /** |
| * formant_pp_factor_num_pow[i] = FORMANT_PP_FACTOR_NUM^(i+1) |
| */ |
| static const int16_t formant_pp_factor_num_pow[10]= { |
| /* (0.15) */ |
| 18022, 9912, 5451, 2998, 1649, 907, 499, 274, 151, 83 |
| }; |
| |
| /** |
| * formant_pp_factor_den_pow[i] = FORMANT_PP_FACTOR_DEN^(i+1) |
| */ |
| static const int16_t formant_pp_factor_den_pow[10] = { |
| /* (0.15) */ |
| 22938, 16057, 11240, 7868, 5508, 3856, 2699, 1889, 1322, 925 |
| }; |
| |
| /** |
| * \brief Residual signal calculation (4.2.1 if G.729) |
| * \param out [out] output data filtered through A(z/FORMANT_PP_FACTOR_NUM) |
| * \param filter_coeffs (3.12) A(z/FORMANT_PP_FACTOR_NUM) filter coefficients |
| * \param in input speech data to process |
| * \param subframe_size size of one subframe |
| * |
| * \note in buffer must contain 10 items of previous speech data before top of the buffer |
| * \remark It is safe to pass the same buffer for input and output. |
| */ |
| static void residual_filter(int16_t* out, const int16_t* filter_coeffs, const int16_t* in, |
| int subframe_size) |
| { |
| int i, n; |
| |
| for (n = subframe_size - 1; n >= 0; n--) { |
| int sum = 0x800; |
| for (i = 0; i < 10; i++) |
| sum += filter_coeffs[i] * in[n - i - 1]; |
| |
| out[n] = in[n] + (sum >> 12); |
| } |
| } |
| |
| /** |
| * \brief long-term postfilter (4.2.1) |
| * \param dsp initialized DSP context |
| * \param pitch_delay_int integer part of the pitch delay in the first subframe |
| * \param residual filtering input data |
| * \param residual_filt [out] speech signal with applied A(z/FORMANT_PP_FACTOR_NUM) filter |
| * \param subframe_size size of subframe |
| * |
| * \return 0 if long-term prediction gain is less than 3dB, 1 - otherwise |
| */ |
| static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int, |
| const int16_t* residual, int16_t *residual_filt, |
| int subframe_size) |
| { |
| int i, k, tmp, tmp2; |
| int sum; |
| int L_temp0; |
| int L_temp1; |
| int64_t L64_temp0; |
| int64_t L64_temp1; |
| int16_t shift; |
| int corr_int_num, corr_int_den; |
| |
| int ener; |
| int16_t sh_ener; |
| |
| int16_t gain_num,gain_den; //selected signal's gain numerator and denominator |
| int16_t sh_gain_num, sh_gain_den; |
| int gain_num_square; |
| |
| int16_t gain_long_num,gain_long_den; //filtered through long interpolation filter signal's gain numerator and denominator |
| int16_t sh_gain_long_num, sh_gain_long_den; |
| |
| int16_t best_delay_int, best_delay_frac; |
| |
| int16_t delayed_signal_offset; |
| int lt_filt_factor_a, lt_filt_factor_b; |
| |
| int16_t * selected_signal; |
| const int16_t * selected_signal_const; //Necessary to avoid compiler warning |
| |
| int16_t sig_scaled[SUBFRAME_SIZE + RES_PREV_DATA_SIZE]; |
| int16_t delayed_signal[ANALYZED_FRAC_DELAYS][SUBFRAME_SIZE+1]; |
| int corr_den[ANALYZED_FRAC_DELAYS][2]; |
| |
| tmp = 0; |
| for(i=0; i<subframe_size + RES_PREV_DATA_SIZE; i++) |
| tmp |= FFABS(residual[i]); |
| |
| if(!tmp) |
| shift = 3; |
| else |
| shift = av_log2(tmp) - 11; |
| |
| if (shift > 0) |
| for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++) |
| sig_scaled[i] = residual[i] >> shift; |
| else |
| for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++) |
| sig_scaled[i] = residual[i] << -shift; |
| |
| /* Start of best delay searching code */ |
| gain_num = 0; |
| |
| ener = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE, |
| sig_scaled + RES_PREV_DATA_SIZE, |
| subframe_size); |
| if (ener) { |
| sh_ener = av_log2(ener) - 14; |
| sh_ener = FFMAX(sh_ener, 0); |
| ener >>= sh_ener; |
| /* Search for best pitch delay. |
| |
| sum{ r(n) * r(k,n) ] }^2 |
| R'(k)^2 := ------------------------- |
| sum{ r(k,n) * r(k,n) } |
| |
| |
| R(T) := sum{ r(n) * r(n-T) ] } |
| |
| |
| where |
| r(n-T) is integer delayed signal with delay T |
| r(k,n) is non-integer delayed signal with integer delay best_delay |
| and fractional delay k */ |
| |
| /* Find integer delay best_delay which maximizes correlation R(T). |
| |
| This is also equals to numerator of R'(0), |
| since the fine search (second step) is done with 1/8 |
| precision around best_delay. */ |
| corr_int_num = 0; |
| best_delay_int = pitch_delay_int - 1; |
| for (i = pitch_delay_int - 1; i <= pitch_delay_int + 1; i++) { |
| sum = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE, |
| sig_scaled + RES_PREV_DATA_SIZE - i, |
| subframe_size); |
| if (sum > corr_int_num) { |
| corr_int_num = sum; |
| best_delay_int = i; |
| } |
| } |
| if (corr_int_num) { |
| /* Compute denominator of pseudo-normalized correlation R'(0). */ |
| corr_int_den = adsp->scalarproduct_int16(sig_scaled - best_delay_int + RES_PREV_DATA_SIZE, |
| sig_scaled - best_delay_int + RES_PREV_DATA_SIZE, |
| subframe_size); |
| |
| /* Compute signals with non-integer delay k (with 1/8 precision), |
| where k is in [0;6] range. |
| Entire delay is qual to best_delay+(k+1)/8 |
| This is archieved by applying an interpolation filter of |
| legth 33 to source signal. */ |
| for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) { |
| ff_acelp_interpolate(&delayed_signal[k][0], |
| &sig_scaled[RES_PREV_DATA_SIZE - best_delay_int], |
| ff_g729_interp_filt_short, |
| ANALYZED_FRAC_DELAYS+1, |
| 8 - k - 1, |
| SHORT_INT_FILT_LEN, |
| subframe_size + 1); |
| } |
| |
| /* Compute denominator of pseudo-normalized correlation R'(k). |
| |
| corr_den[k][0] is square root of R'(k) denominator, for int(T) == int(T0) |
| corr_den[k][1] is square root of R'(k) denominator, for int(T) == int(T0)+1 |
| |
| Also compute maximum value of above denominators over all k. */ |
| tmp = corr_int_den; |
| for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) { |
| sum = adsp->scalarproduct_int16(&delayed_signal[k][1], |
| &delayed_signal[k][1], |
| subframe_size - 1); |
| corr_den[k][0] = sum + delayed_signal[k][0 ] * delayed_signal[k][0 ]; |
| corr_den[k][1] = sum + delayed_signal[k][subframe_size] * delayed_signal[k][subframe_size]; |
| |
| tmp = FFMAX3(tmp, corr_den[k][0], corr_den[k][1]); |
| } |
| |
| sh_gain_den = av_log2(tmp) - 14; |
| if (sh_gain_den >= 0) { |
| |
| sh_gain_num = FFMAX(sh_gain_den, sh_ener); |
| /* Loop through all k and find delay that maximizes |
| R'(k) correlation. |
| Search is done in [int(T0)-1; intT(0)+1] range |
| with 1/8 precision. */ |
| delayed_signal_offset = 1; |
| best_delay_frac = 0; |
| gain_den = corr_int_den >> sh_gain_den; |
| gain_num = corr_int_num >> sh_gain_num; |
| gain_num_square = gain_num * gain_num; |
| for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) { |
| for (i = 0; i < 2; i++) { |
| int16_t gain_num_short, gain_den_short; |
| int gain_num_short_square; |
| /* Compute numerator of pseudo-normalized |
| correlation R'(k). */ |
| sum = adsp->scalarproduct_int16(&delayed_signal[k][i], |
| sig_scaled + RES_PREV_DATA_SIZE, |
| subframe_size); |
| gain_num_short = FFMAX(sum >> sh_gain_num, 0); |
| |
| /* |
| gain_num_short_square gain_num_square |
| R'(T)^2 = -----------------------, max R'(T)^2= -------------- |
| den gain_den |
| */ |
| gain_num_short_square = gain_num_short * gain_num_short; |
| gain_den_short = corr_den[k][i] >> sh_gain_den; |
| |
| tmp = MULL(gain_num_short_square, gain_den, FRAC_BITS); |
| tmp2 = MULL(gain_num_square, gain_den_short, FRAC_BITS); |
| |
| // R'(T)^2 > max R'(T)^2 |
| if (tmp > tmp2) { |
| gain_num = gain_num_short; |
| gain_den = gain_den_short; |
| gain_num_square = gain_num_short_square; |
| delayed_signal_offset = i; |
| best_delay_frac = k + 1; |
| } |
| } |
| } |
| |
| /* |
| R'(T)^2 |
| 2 * --------- < 1 |
| R(0) |
| */ |
| L64_temp0 = (int64_t)gain_num_square << ((sh_gain_num << 1) + 1); |
| L64_temp1 = ((int64_t)gain_den * ener) << (sh_gain_den + sh_ener); |
| if (L64_temp0 < L64_temp1) |
| gain_num = 0; |
| } // if(sh_gain_den >= 0) |
| } // if(corr_int_num) |
| } // if(ener) |
| /* End of best delay searching code */ |
| |
| if (!gain_num) { |
| memcpy(residual_filt, residual + RES_PREV_DATA_SIZE, subframe_size * sizeof(int16_t)); |
| |
| /* Long-term prediction gain is less than 3dB. Long-term postfilter is disabled. */ |
| return 0; |
| } |
| if (best_delay_frac) { |
| /* Recompute delayed signal with an interpolation filter of length 129. */ |
| ff_acelp_interpolate(residual_filt, |
| &sig_scaled[RES_PREV_DATA_SIZE - best_delay_int + delayed_signal_offset], |
| ff_g729_interp_filt_long, |
| ANALYZED_FRAC_DELAYS + 1, |
| 8 - best_delay_frac, |
| LONG_INT_FILT_LEN, |
| subframe_size + 1); |
| /* Compute R'(k) correlation's numerator. */ |
| sum = adsp->scalarproduct_int16(residual_filt, |
| sig_scaled + RES_PREV_DATA_SIZE, |
| subframe_size); |
| |
| if (sum < 0) { |
| gain_long_num = 0; |
| sh_gain_long_num = 0; |
| } else { |
| tmp = av_log2(sum) - 14; |
| tmp = FFMAX(tmp, 0); |
| sum >>= tmp; |
| gain_long_num = sum; |
| sh_gain_long_num = tmp; |
| } |
| |
| /* Compute R'(k) correlation's denominator. */ |
| sum = adsp->scalarproduct_int16(residual_filt, residual_filt, subframe_size); |
| |
| tmp = av_log2(sum) - 14; |
| tmp = FFMAX(tmp, 0); |
| sum >>= tmp; |
| gain_long_den = sum; |
| sh_gain_long_den = tmp; |
| |
| /* Select between original and delayed signal. |
| Delayed signal will be selected if it increases R'(k) |
| correlation. */ |
| L_temp0 = gain_num * gain_num; |
| L_temp0 = MULL(L_temp0, gain_long_den, FRAC_BITS); |
| |
| L_temp1 = gain_long_num * gain_long_num; |
| L_temp1 = MULL(L_temp1, gain_den, FRAC_BITS); |
| |
| tmp = ((sh_gain_long_num - sh_gain_num) << 1) - (sh_gain_long_den - sh_gain_den); |
| if (tmp > 0) |
| L_temp0 >>= tmp; |
| else |
| L_temp1 >>= -tmp; |
| |
| /* Check if longer filter increases the values of R'(k). */ |
| if (L_temp1 > L_temp0) { |
| /* Select long filter. */ |
| selected_signal = residual_filt; |
| gain_num = gain_long_num; |
| gain_den = gain_long_den; |
| sh_gain_num = sh_gain_long_num; |
| sh_gain_den = sh_gain_long_den; |
| } else |
| /* Select short filter. */ |
| selected_signal = &delayed_signal[best_delay_frac-1][delayed_signal_offset]; |
| |
| /* Rescale selected signal to original value. */ |
| if (shift > 0) |
| for (i = 0; i < subframe_size; i++) |
| selected_signal[i] <<= shift; |
| else |
| for (i = 0; i < subframe_size; i++) |
| selected_signal[i] >>= -shift; |
| |
| /* necessary to avoid compiler warning */ |
| selected_signal_const = selected_signal; |
| } // if(best_delay_frac) |
| else |
| selected_signal_const = residual + RES_PREV_DATA_SIZE - (best_delay_int + 1 - delayed_signal_offset); |
| #ifdef G729_BITEXACT |
| tmp = sh_gain_num - sh_gain_den; |
| if (tmp > 0) |
| gain_den >>= tmp; |
| else |
| gain_num >>= -tmp; |
| |
| if (gain_num > gain_den) |
| lt_filt_factor_a = MIN_LT_FILT_FACTOR_A; |
| else { |
| gain_num >>= 2; |
| gain_den >>= 1; |
| lt_filt_factor_a = (gain_den << 15) / (gain_den + gain_num); |
| } |
| #else |
| L64_temp0 = (((int64_t)gain_num) << sh_gain_num) >> 1; |
| L64_temp1 = ((int64_t)gain_den) << sh_gain_den; |
| lt_filt_factor_a = FFMAX((L64_temp1 << 15) / (L64_temp1 + L64_temp0), MIN_LT_FILT_FACTOR_A); |
| #endif |
| |
| /* Filter through selected filter. */ |
| lt_filt_factor_b = 32767 - lt_filt_factor_a + 1; |
| |
| ff_acelp_weighted_vector_sum(residual_filt, residual + RES_PREV_DATA_SIZE, |
| selected_signal_const, |
| lt_filt_factor_a, lt_filt_factor_b, |
| 1<<14, 15, subframe_size); |
| |
| // Long-term prediction gain is larger than 3dB. |
| return 1; |
| } |
| |
| /** |
| * \brief Calculate reflection coefficient for tilt compensation filter (4.2.3). |
| * \param dsp initialized DSP context |
| * \param lp_gn (3.12) coefficients of A(z/FORMANT_PP_FACTOR_NUM) filter |
| * \param lp_gd (3.12) coefficients of A(z/FORMANT_PP_FACTOR_DEN) filter |
| * \param speech speech to update |
| * \param subframe_size size of subframe |
| * |
| * \return (3.12) reflection coefficient |
| * |
| * \remark The routine also calculates the gain term for the short-term |
| * filter (gf) and multiplies the speech data by 1/gf. |
| * |
| * \note All members of lp_gn, except 10-19 must be equal to zero. |
| */ |
| static int16_t get_tilt_comp(AudioDSPContext *adsp, int16_t *lp_gn, |
| const int16_t *lp_gd, int16_t* speech, |
| int subframe_size) |
| { |
| int rh1,rh0; // (3.12) |
| int temp; |
| int i; |
| int gain_term; |
| |
| lp_gn[10] = 4096; //1.0 in (3.12) |
| |
| /* Apply 1/A(z/FORMANT_PP_FACTOR_DEN) filter to hf. */ |
| ff_celp_lp_synthesis_filter(lp_gn + 11, lp_gd + 1, lp_gn + 11, 22, 10, 0, 0, 0x800); |
| /* Now lp_gn (starting with 10) contains impulse response |
| of A(z/FORMANT_PP_FACTOR_NUM)/A(z/FORMANT_PP_FACTOR_DEN) filter. */ |
| |
| rh0 = adsp->scalarproduct_int16(lp_gn + 10, lp_gn + 10, 20); |
| rh1 = adsp->scalarproduct_int16(lp_gn + 10, lp_gn + 11, 20); |
| |
| /* downscale to avoid overflow */ |
| temp = av_log2(rh0) - 14; |
| if (temp > 0) { |
| rh0 >>= temp; |
| rh1 >>= temp; |
| } |
| |
| if (FFABS(rh1) > rh0 || !rh0) |
| return 0; |
| |
| gain_term = 0; |
| for (i = 0; i < 20; i++) |
| gain_term += FFABS(lp_gn[i + 10]); |
| gain_term >>= 2; // (3.12) -> (5.10) |
| |
| if (gain_term > 0x400) { // 1.0 in (5.10) |
| temp = 0x2000000 / gain_term; // 1.0/gain_term in (0.15) |
| for (i = 0; i < subframe_size; i++) |
| speech[i] = (speech[i] * temp + 0x4000) >> 15; |
| } |
| |
| return -(rh1 << 15) / rh0; |
| } |
| |
| /** |
| * \brief Apply tilt compensation filter (4.2.3). |
| * \param res_pst [in/out] residual signal (partially filtered) |
| * \param k1 (3.12) reflection coefficient |
| * \param subframe_size size of subframe |
| * \param ht_prev_data previous data for 4.2.3, equation 86 |
| * |
| * \return new value for ht_prev_data |
| */ |
| static int16_t apply_tilt_comp(int16_t* out, int16_t* res_pst, int refl_coeff, |
| int subframe_size, int16_t ht_prev_data) |
| { |
| int tmp, tmp2; |
| int i; |
| int gt, ga; |
| int fact, sh_fact; |
| |
| if (refl_coeff > 0) { |
| gt = (refl_coeff * G729_TILT_FACTOR_PLUS + 0x4000) >> 15; |
| fact = 0x4000; // 0.5 in (0.15) |
| sh_fact = 15; |
| } else { |
| gt = (refl_coeff * G729_TILT_FACTOR_MINUS + 0x4000) >> 15; |
| fact = 0x800; // 0.5 in (3.12) |
| sh_fact = 12; |
| } |
| ga = (fact << 15) / av_clip_int16(32768 - FFABS(gt)); |
| gt >>= 1; |
| |
| /* Apply tilt compensation filter to signal. */ |
| tmp = res_pst[subframe_size - 1]; |
| |
| for (i = subframe_size - 1; i >= 1; i--) { |
| tmp2 = (res_pst[i] << 15) + ((gt * res_pst[i-1]) << 1); |
| tmp2 = (tmp2 + 0x4000) >> 15; |
| |
| tmp2 = (tmp2 * ga * 2 + fact) >> sh_fact; |
| out[i] = tmp2; |
| } |
| tmp2 = (res_pst[0] << 15) + ((gt * ht_prev_data) << 1); |
| tmp2 = (tmp2 + 0x4000) >> 15; |
| tmp2 = (tmp2 * ga * 2 + fact) >> sh_fact; |
| out[0] = tmp2; |
| |
| return tmp; |
| } |
| |
| void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voicing, |
| const int16_t *lp_filter_coeffs, int pitch_delay_int, |
| int16_t* residual, int16_t* res_filter_data, |
| int16_t* pos_filter_data, int16_t *speech, int subframe_size) |
| { |
| int16_t residual_filt_buf[SUBFRAME_SIZE+11]; |
| int16_t lp_gn[33]; // (3.12) |
| int16_t lp_gd[11]; // (3.12) |
| int tilt_comp_coeff; |
| int i; |
| |
| /* Zero-filling is necessary for tilt-compensation filter. */ |
| memset(lp_gn, 0, 33 * sizeof(int16_t)); |
| |
| /* Calculate A(z/FORMANT_PP_FACTOR_NUM) filter coefficients. */ |
| for (i = 0; i < 10; i++) |
| lp_gn[i + 11] = (lp_filter_coeffs[i + 1] * formant_pp_factor_num_pow[i] + 0x4000) >> 15; |
| |
| /* Calculate A(z/FORMANT_PP_FACTOR_DEN) filter coefficients. */ |
| for (i = 0; i < 10; i++) |
| lp_gd[i + 1] = (lp_filter_coeffs[i + 1] * formant_pp_factor_den_pow[i] + 0x4000) >> 15; |
| |
| /* residual signal calculation (one-half of short-term postfilter) */ |
| memcpy(speech - 10, res_filter_data, 10 * sizeof(int16_t)); |
| residual_filter(residual + RES_PREV_DATA_SIZE, lp_gn + 11, speech, subframe_size); |
| /* Save data to use it in the next subframe. */ |
| memcpy(res_filter_data, speech + subframe_size - 10, 10 * sizeof(int16_t)); |
| |
| /* long-term filter. If long-term prediction gain is larger than 3dB (returned value is |
| nonzero) then declare current subframe as periodic. */ |
| i = long_term_filter(adsp, pitch_delay_int, |
| residual, residual_filt_buf + 10, |
| subframe_size); |
| *voicing = FFMAX(*voicing, i); |
| |
| /* shift residual for using in next subframe */ |
| memmove(residual, residual + subframe_size, RES_PREV_DATA_SIZE * sizeof(int16_t)); |
| |
| /* short-term filter tilt compensation */ |
| tilt_comp_coeff = get_tilt_comp(adsp, lp_gn, lp_gd, residual_filt_buf + 10, subframe_size); |
| |
| /* Apply second half of short-term postfilter: 1/A(z/FORMANT_PP_FACTOR_DEN) */ |
| ff_celp_lp_synthesis_filter(pos_filter_data + 10, lp_gd + 1, |
| residual_filt_buf + 10, |
| subframe_size, 10, 0, 0, 0x800); |
| memcpy(pos_filter_data, pos_filter_data + subframe_size, 10 * sizeof(int16_t)); |
| |
| *ht_prev_data = apply_tilt_comp(speech, pos_filter_data + 10, tilt_comp_coeff, |
| subframe_size, *ht_prev_data); |
| } |
| |
| /** |
| * \brief Adaptive gain control (4.2.4) |
| * \param gain_before gain of speech before applying postfilters |
| * \param gain_after gain of speech after applying postfilters |
| * \param speech [in/out] signal buffer |
| * \param subframe_size length of subframe |
| * \param gain_prev (3.12) previous value of gain coefficient |
| * |
| * \return (3.12) last value of gain coefficient |
| */ |
| int16_t ff_g729_adaptive_gain_control(int gain_before, int gain_after, int16_t *speech, |
| int subframe_size, int16_t gain_prev) |
| { |
| int gain; // (3.12) |
| int n; |
| int exp_before, exp_after; |
| |
| if(!gain_after && gain_before) |
| return 0; |
| |
| if (gain_before) { |
| |
| exp_before = 14 - av_log2(gain_before); |
| gain_before = bidir_sal(gain_before, exp_before); |
| |
| exp_after = 14 - av_log2(gain_after); |
| gain_after = bidir_sal(gain_after, exp_after); |
| |
| if (gain_before < gain_after) { |
| gain = (gain_before << 15) / gain_after; |
| gain = bidir_sal(gain, exp_after - exp_before - 1); |
| } else { |
| gain = ((gain_before - gain_after) << 14) / gain_after + 0x4000; |
| gain = bidir_sal(gain, exp_after - exp_before); |
| } |
| gain = (gain * G729_AGC_FAC1 + 0x4000) >> 15; // gain * (1-0.9875) |
| } else |
| gain = 0; |
| |
| for (n = 0; n < subframe_size; n++) { |
| // gain_prev = gain + 0.9875 * gain_prev |
| gain_prev = (G729_AGC_FACTOR * gain_prev + 0x4000) >> 15; |
| gain_prev = av_clip_int16(gain + gain_prev); |
| speech[n] = av_clip_int16((speech[n] * gain_prev + 0x2000) >> 14); |
| } |
| return gain_prev; |
| } |
