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FFmpeg/libavcodec/g729postfilter.c
Michael Niedermayer 6e3244fb09 avcodec/g729postfilter: Avoid function calls in FFMAX() arguments 
This avoid double calling functions

Found-by: Muhammad Faiz <mfcc64@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2015-11-10 15:44:46 +01:00

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/*
* 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;
}
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