/* * 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 #include #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 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] = (unsigned)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 + RES_PREV_DATA_SIZE - best_delay_int, sig_scaled + RES_PREV_DATA_SIZE - best_delay_int, 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) * 2) - (sh_gain_long_den - sh_gain_den); if (tmp > 0) L_temp0 >>= tmp; else L_temp1 >>= FFMIN(-tmp, 31); /* 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] *= 1 << 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 * (1 << 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 = 0x2000; // 0.5 in (0.15) sh_fact = 14; } else { gt = (refl_coeff * G729_TILT_FACTOR_MINUS + 0x4000) >> 15; fact = 0x400; // 0.5 in (3.12) sh_fact = 11; } ga = (fact << 16) / 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 = (gt * res_pst[i-1]) * 2 + 0x4000; tmp2 = res_pst[i] + (tmp2 >> 15); tmp2 = (tmp2 * ga + fact) >> sh_fact; out[i] = tmp2; } tmp2 = (gt * ht_prev_data) * 2 + 0x4000; tmp2 = res_pst[0] + (tmp2 >> 15); tmp2 = (tmp2 * ga + 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) { unsigned 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 = FFMIN(gain, 32767); 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; }