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fea1f42e5f
FFmpeg/libavcodec/g729postfilter.c
Michael Niedermayer 82d4c7b95e avcodec/g729postfilter: Clip gain before scaling with AGC_FAC1 
The fixed point integer reference specifies the multiplication used
to have 16bit input and clips so we need to clip the input
The floating point implementation does not seem to do that.

Fixes: signed integer overflow: 6317568 * 410 cannot be represented in type 'int'
Fixes: 20492/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_G729_fuzzer-5700189272932352

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-05-11 00:45:09 +02:00

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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] = (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 >>= -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] *= 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)
{
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 = av_clip_int16(gain);
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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