bf807a5e87
* qatar/master: (29 commits) sbrdsp.asm: convert all instructions to float/SSE ones. dv: cosmetics. dv: check buffer size before reading profile. Revert "AAC SBR: group some writes." udp: Print an error message if bind fails cook: extend channel uncoupling tables so the full bit range is covered. roqvideo: cosmetics. roqvideo: convert to bytestream2 API. dca: don't use av_clip_uintp2(). wmall: fix build with -DDEBUG enabled. smc: port to bytestream2 API. AAC SBR: group some writes. dsputil: remove shift parameter from scalarproduct_int16 SBR DSP: unroll sum_square rv34: remove dead code in intra availability check rv34: clean a bit availability checks. v4l2: update documentation tgq: convert to bytestream2 API. parser: remove forward declaration of MpegEncContext dca: prevent accessing static arrays with invalid indexes. ... Conflicts: doc/indevs.texi libavcodec/Makefile libavcodec/dca.c libavcodec/dvdata.c libavcodec/eatgq.c libavcodec/mmvideo.c libavcodec/roqvideodec.c libavcodec/smc.c libswscale/output.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
611 lines
23 KiB
C
611 lines
23 KiB
C
/*
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* G.729, G729 Annex D postfilter
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* Copyright (c) 2008 Vladimir Voroshilov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <inttypes.h>
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#include <limits.h>
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#include "avcodec.h"
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#include "g729.h"
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#include "acelp_pitch_delay.h"
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#include "g729postfilter.h"
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#include "celp_math.h"
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#include "acelp_filters.h"
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#include "acelp_vectors.h"
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#include "celp_filters.h"
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#define FRAC_BITS 15
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#include "mathops.h"
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/**
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* short interpolation filter (of length 33, according to spec)
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* for computing signal with non-integer delay
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*/
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static const int16_t ff_g729_interp_filt_short[(ANALYZED_FRAC_DELAYS+1)*SHORT_INT_FILT_LEN] = {
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0, 31650, 28469, 23705, 18050, 12266, 7041, 2873,
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0, -1597, -2147, -1992, -1492, -933, -484, -188,
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};
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/**
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* long interpolation filter (of length 129, according to spec)
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* for computing signal with non-integer delay
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*/
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static const int16_t ff_g729_interp_filt_long[(ANALYZED_FRAC_DELAYS+1)*LONG_INT_FILT_LEN] = {
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0, 31915, 29436, 25569, 20676, 15206, 9639, 4439,
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0, -3390, -5579, -6549, -6414, -5392, -3773, -1874,
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0, 1595, 2727, 3303, 3319, 2850, 2030, 1023,
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0, -887, -1527, -1860, -1876, -1614, -1150, -579,
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0, 501, 859, 1041, 1044, 892, 631, 315,
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0, -266, -453, -543, -538, -455, -317, -156,
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0, 130, 218, 258, 253, 212, 147, 72,
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0, -59, -101, -122, -123, -106, -77, -40,
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};
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/**
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* formant_pp_factor_num_pow[i] = FORMANT_PP_FACTOR_NUM^(i+1)
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*/
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static const int16_t formant_pp_factor_num_pow[10]= {
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/* (0.15) */
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18022, 9912, 5451, 2998, 1649, 907, 499, 274, 151, 83
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};
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/**
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* formant_pp_factor_den_pow[i] = FORMANT_PP_FACTOR_DEN^(i+1)
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*/
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static const int16_t formant_pp_factor_den_pow[10] = {
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/* (0.15) */
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22938, 16057, 11240, 7868, 5508, 3856, 2699, 1889, 1322, 925
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};
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/**
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* \brief Residual signal calculation (4.2.1 if G.729)
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* \param out [out] output data filtered through A(z/FORMANT_PP_FACTOR_NUM)
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* \param filter_coeffs (3.12) A(z/FORMANT_PP_FACTOR_NUM) filter coefficients
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* \param in input speech data to process
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* \param subframe_size size of one subframe
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*
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* \note in buffer must contain 10 items of previous speech data before top of the buffer
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* \remark It is safe to pass the same buffer for input and output.
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*/
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static void residual_filter(int16_t* out, const int16_t* filter_coeffs, const int16_t* in,
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int subframe_size)
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{
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int i, n;
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for (n = subframe_size - 1; n >= 0; n--) {
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int sum = 0x800;
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for (i = 0; i < 10; i++)
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sum += filter_coeffs[i] * in[n - i - 1];
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out[n] = in[n] + (sum >> 12);
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}
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}
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/**
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* \brief long-term postfilter (4.2.1)
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* \param dsp initialized DSP context
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* \param pitch_delay_int integer part of the pitch delay in the first subframe
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* \param residual filtering input data
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* \param residual_filt [out] speech signal with applied A(z/FORMANT_PP_FACTOR_NUM) filter
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* \param subframe_size size of subframe
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*
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* \return 0 if long-term prediction gain is less than 3dB, 1 - otherwise
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*/
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static int16_t long_term_filter(DSPContext *dsp, int pitch_delay_int,
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const int16_t* residual, int16_t *residual_filt,
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int subframe_size)
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{
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int i, k, tmp, tmp2;
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int sum;
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int L_temp0;
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int L_temp1;
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int64_t L64_temp0;
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int64_t L64_temp1;
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int16_t shift;
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int corr_int_num, corr_int_den;
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int ener;
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int16_t sh_ener;
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int16_t gain_num,gain_den; //selected signal's gain numerator and denominator
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int16_t sh_gain_num, sh_gain_den;
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int gain_num_square;
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int16_t gain_long_num,gain_long_den; //filtered through long interpolation filter signal's gain numerator and denominator
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int16_t sh_gain_long_num, sh_gain_long_den;
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int16_t best_delay_int, best_delay_frac;
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int16_t delayed_signal_offset;
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int lt_filt_factor_a, lt_filt_factor_b;
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int16_t * selected_signal;
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const int16_t * selected_signal_const; //Necessary to avoid compiler warning
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int16_t sig_scaled[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
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int16_t delayed_signal[ANALYZED_FRAC_DELAYS][SUBFRAME_SIZE+1];
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int corr_den[ANALYZED_FRAC_DELAYS][2];
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tmp = 0;
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for(i=0; i<subframe_size + RES_PREV_DATA_SIZE; i++)
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tmp |= FFABS(residual[i]);
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if(!tmp)
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shift = 3;
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else
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shift = av_log2(tmp) - 11;
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if (shift > 0)
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for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
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sig_scaled[i] = residual[i] >> shift;
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else
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for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
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sig_scaled[i] = residual[i] << -shift;
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/* Start of best delay searching code */
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gain_num = 0;
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ener = dsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE,
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sig_scaled + RES_PREV_DATA_SIZE,
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subframe_size);
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if (ener) {
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sh_ener = FFMAX(av_log2(ener) - 14, 0);
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ener >>= sh_ener;
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/* Search for best pitch delay.
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sum{ r(n) * r(k,n) ] }^2
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R'(k)^2 := -------------------------
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sum{ r(k,n) * r(k,n) }
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R(T) := sum{ r(n) * r(n-T) ] }
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where
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r(n-T) is integer delayed signal with delay T
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r(k,n) is non-integer delayed signal with integer delay best_delay
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and fractional delay k */
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/* Find integer delay best_delay which maximizes correlation R(T).
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This is also equals to numerator of R'(0),
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since the fine search (second step) is done with 1/8
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precision around best_delay. */
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corr_int_num = 0;
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best_delay_int = pitch_delay_int - 1;
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for (i = pitch_delay_int - 1; i <= pitch_delay_int + 1; i++) {
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sum = dsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE,
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sig_scaled + RES_PREV_DATA_SIZE - i,
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subframe_size);
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if (sum > corr_int_num) {
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corr_int_num = sum;
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best_delay_int = i;
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}
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}
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if (corr_int_num) {
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/* Compute denominator of pseudo-normalized correlation R'(0). */
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corr_int_den = dsp->scalarproduct_int16(sig_scaled - best_delay_int + RES_PREV_DATA_SIZE,
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sig_scaled - best_delay_int + RES_PREV_DATA_SIZE,
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subframe_size);
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/* Compute signals with non-integer delay k (with 1/8 precision),
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where k is in [0;6] range.
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Entire delay is qual to best_delay+(k+1)/8
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This is archieved by applying an interpolation filter of
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legth 33 to source signal. */
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for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
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ff_acelp_interpolate(&delayed_signal[k][0],
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&sig_scaled[RES_PREV_DATA_SIZE - best_delay_int],
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ff_g729_interp_filt_short,
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ANALYZED_FRAC_DELAYS+1,
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8 - k - 1,
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SHORT_INT_FILT_LEN,
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subframe_size + 1);
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}
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/* Compute denominator of pseudo-normalized correlation R'(k).
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corr_den[k][0] is square root of R'(k) denominator, for int(T) == int(T0)
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corr_den[k][1] is square root of R'(k) denominator, for int(T) == int(T0)+1
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Also compute maximum value of above denominators over all k. */
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tmp = corr_int_den;
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for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
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sum = dsp->scalarproduct_int16(&delayed_signal[k][1],
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&delayed_signal[k][1],
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subframe_size - 1);
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corr_den[k][0] = sum + delayed_signal[k][0 ] * delayed_signal[k][0 ];
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corr_den[k][1] = sum + delayed_signal[k][subframe_size] * delayed_signal[k][subframe_size];
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tmp = FFMAX3(tmp, corr_den[k][0], corr_den[k][1]);
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}
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sh_gain_den = av_log2(tmp) - 14;
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if (sh_gain_den >= 0) {
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sh_gain_num = FFMAX(sh_gain_den, sh_ener);
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/* Loop through all k and find delay that maximizes
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R'(k) correlation.
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Search is done in [int(T0)-1; intT(0)+1] range
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with 1/8 precision. */
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delayed_signal_offset = 1;
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best_delay_frac = 0;
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gain_den = corr_int_den >> sh_gain_den;
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gain_num = corr_int_num >> sh_gain_num;
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gain_num_square = gain_num * gain_num;
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for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
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for (i = 0; i < 2; i++) {
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int16_t gain_num_short, gain_den_short;
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int gain_num_short_square;
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/* Compute numerator of pseudo-normalized
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correlation R'(k). */
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sum = dsp->scalarproduct_int16(&delayed_signal[k][i],
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sig_scaled + RES_PREV_DATA_SIZE,
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subframe_size);
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gain_num_short = FFMAX(sum >> sh_gain_num, 0);
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/*
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gain_num_short_square gain_num_square
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R'(T)^2 = -----------------------, max R'(T)^2= --------------
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den gain_den
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*/
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gain_num_short_square = gain_num_short * gain_num_short;
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gain_den_short = corr_den[k][i] >> sh_gain_den;
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tmp = MULL(gain_num_short_square, gain_den, FRAC_BITS);
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tmp2 = MULL(gain_num_square, gain_den_short, FRAC_BITS);
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// R'(T)^2 > max R'(T)^2
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if (tmp > tmp2) {
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gain_num = gain_num_short;
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gain_den = gain_den_short;
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gain_num_square = gain_num_short_square;
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delayed_signal_offset = i;
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best_delay_frac = k + 1;
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}
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}
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}
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/*
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R'(T)^2
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2 * --------- < 1
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R(0)
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*/
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L64_temp0 = (int64_t)gain_num_square << ((sh_gain_num << 1) + 1);
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L64_temp1 = ((int64_t)gain_den * ener) << (sh_gain_den + sh_ener);
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if (L64_temp0 < L64_temp1)
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gain_num = 0;
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} // if(sh_gain_den >= 0)
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} // if(corr_int_num)
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} // if(ener)
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/* End of best delay searching code */
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if (!gain_num) {
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memcpy(residual_filt, residual + RES_PREV_DATA_SIZE, subframe_size * sizeof(int16_t));
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/* Long-term prediction gain is less than 3dB. Long-term postfilter is disabled. */
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return 0;
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}
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if (best_delay_frac) {
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/* Recompute delayed signal with an interpolation filter of length 129. */
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ff_acelp_interpolate(residual_filt,
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&sig_scaled[RES_PREV_DATA_SIZE - best_delay_int + delayed_signal_offset],
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ff_g729_interp_filt_long,
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ANALYZED_FRAC_DELAYS + 1,
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8 - best_delay_frac,
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LONG_INT_FILT_LEN,
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subframe_size + 1);
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/* Compute R'(k) correlation's numerator. */
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sum = dsp->scalarproduct_int16(residual_filt,
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sig_scaled + RES_PREV_DATA_SIZE,
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subframe_size);
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if (sum < 0) {
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gain_long_num = 0;
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sh_gain_long_num = 0;
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} else {
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tmp = FFMAX(av_log2(sum) - 14, 0);
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sum >>= tmp;
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gain_long_num = sum;
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sh_gain_long_num = tmp;
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}
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/* Compute R'(k) correlation's denominator. */
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sum = dsp->scalarproduct_int16(residual_filt, residual_filt, subframe_size);
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tmp = FFMAX(av_log2(sum) - 14, 0);
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sum >>= tmp;
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gain_long_den = sum;
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sh_gain_long_den = tmp;
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/* Select between original and delayed signal.
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Delayed signal will be selected if it increases R'(k)
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correlation. */
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L_temp0 = gain_num * gain_num;
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L_temp0 = MULL(L_temp0, gain_long_den, FRAC_BITS);
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L_temp1 = gain_long_num * gain_long_num;
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L_temp1 = MULL(L_temp1, gain_den, FRAC_BITS);
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tmp = ((sh_gain_long_num - sh_gain_num) << 1) - (sh_gain_long_den - sh_gain_den);
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if (tmp > 0)
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L_temp0 >>= tmp;
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else
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L_temp1 >>= -tmp;
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/* Check if longer filter increases the values of R'(k). */
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if (L_temp1 > L_temp0) {
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/* Select long filter. */
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selected_signal = residual_filt;
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gain_num = gain_long_num;
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gain_den = gain_long_den;
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sh_gain_num = sh_gain_long_num;
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sh_gain_den = sh_gain_long_den;
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} else
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/* Select short filter. */
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selected_signal = &delayed_signal[best_delay_frac-1][delayed_signal_offset];
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/* Rescale selected signal to original value. */
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if (shift > 0)
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for (i = 0; i < subframe_size; i++)
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selected_signal[i] <<= shift;
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else
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for (i = 0; i < subframe_size; i++)
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selected_signal[i] >>= -shift;
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/* necessary to avoid compiler warning */
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selected_signal_const = selected_signal;
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} // if(best_delay_frac)
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else
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selected_signal_const = residual + RES_PREV_DATA_SIZE - (best_delay_int + 1 - delayed_signal_offset);
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#ifdef G729_BITEXACT
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tmp = sh_gain_num - sh_gain_den;
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if (tmp > 0)
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gain_den >>= tmp;
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else
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gain_num >>= -tmp;
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if (gain_num > gain_den)
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lt_filt_factor_a = MIN_LT_FILT_FACTOR_A;
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else {
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gain_num >>= 2;
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gain_den >>= 1;
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lt_filt_factor_a = (gain_den << 15) / (gain_den + gain_num);
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}
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#else
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L64_temp0 = ((int64_t)gain_num) << (sh_gain_num - 1);
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L64_temp1 = ((int64_t)gain_den) << sh_gain_den;
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lt_filt_factor_a = FFMAX((L64_temp1 << 15) / (L64_temp1 + L64_temp0), MIN_LT_FILT_FACTOR_A);
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#endif
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/* Filter through selected filter. */
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lt_filt_factor_b = 32767 - lt_filt_factor_a + 1;
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ff_acelp_weighted_vector_sum(residual_filt, residual + RES_PREV_DATA_SIZE,
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selected_signal_const,
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lt_filt_factor_a, lt_filt_factor_b,
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1<<14, 15, subframe_size);
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// Long-term prediction gain is larger than 3dB.
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return 1;
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}
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/**
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* \brief Calculate reflection coefficient for tilt compensation filter (4.2.3).
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* \param dsp initialized DSP context
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* \param lp_gn (3.12) coefficients of A(z/FORMANT_PP_FACTOR_NUM) filter
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* \param lp_gd (3.12) coefficients of A(z/FORMANT_PP_FACTOR_DEN) filter
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* \param speech speech to update
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* \param subframe_size size of subframe
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*
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* \return (3.12) reflection coefficient
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*
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* \remark The routine also calculates the gain term for the short-term
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* filter (gf) and multiplies the speech data by 1/gf.
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*
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* \note All members of lp_gn, except 10-19 must be equal to zero.
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*/
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static int16_t get_tilt_comp(DSPContext *dsp, int16_t *lp_gn,
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const int16_t *lp_gd, int16_t* speech,
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int subframe_size)
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|
{
|
|
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 = dsp->scalarproduct_int16(lp_gn + 10, lp_gn + 10, 20);
|
|
rh1 = dsp->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(DSPContext *dsp, 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. */
|
|
*voicing = FFMAX(*voicing, long_term_filter(dsp, pitch_delay_int,
|
|
residual, residual_filt_buf + 10,
|
|
subframe_size));
|
|
|
|
/* 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(dsp, 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;
|
|
}
|