1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavcodec/aacpsdsp_template.c
Michael Niedermayer 0181b202cc avcodec/aacpsdsp_template: Fix undefined integer overflow in ps_add_squares_c()
Fixes runtime error: signed integer overflow: 1997494407 + 613252359 cannot be represented in type 'int'
Fixes: 2014/clusterfuzz-testcase-minimized-5186337030275072

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-08-27 19:40:28 +02:00

234 lines
8.1 KiB
C

/*
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*
* Note: Rounding-to-nearest used unless otherwise stated
*
*/
#include <stdint.h>
#include "config.h"
#include "libavutil/attributes.h"
#include "aacpsdsp.h"
static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
{
int i;
for (i = 0; i < n; i++)
dst[i] += (UINTFLOAT)AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
}
static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
int n)
{
int i;
for (i = 0; i < n; i++) {
dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
}
}
static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
const INTFLOAT (*filter)[8][2],
ptrdiff_t stride, int n)
{
int i, j;
for (i = 0; i < n; i++) {
INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
for (j = 0; j < 6; j++) {
INTFLOAT in0_re = in[j][0];
INTFLOAT in0_im = in[j][1];
INTFLOAT in1_re = in[12-j][0];
INTFLOAT in1_im = in[12-j][1];
sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
(INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
(INT64FLOAT)filter[i][j][1] * (in0_re - in1_re);
}
#if USE_FIXED
out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
#else
out[i * stride][0] = sum_re;
out[i * stride][1] = sum_im;
#endif /* USE_FIXED */
}
}
static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
int i, int len)
{
int j;
for (; i < 64; i++) {
for (j = 0; j < len; j++) {
out[i][j][0] = L[0][j][i];
out[i][j][1] = L[1][j][i];
}
}
}
static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
INTFLOAT (*in)[32][2],
int i, int len)
{
int n;
for (; i < 64; i++) {
for (n = 0; n < len; n++) {
out[0][n][i] = in[i][n][0];
out[1][n][i] = in[i][n][1];
}
}
}
static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
const INTFLOAT *transient_gain,
INTFLOAT g_decay_slope,
int len)
{
static const INTFLOAT a[] = { Q31(0.65143905753106f),
Q31(0.56471812200776f),
Q31(0.48954165955695f) };
INTFLOAT ag[PS_AP_LINKS];
int m, n;
for (m = 0; m < PS_AP_LINKS; m++)
ag[m] = AAC_MUL30(a[m], g_decay_slope);
for (n = 0; n < len; n++) {
INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
for (m = 0; m < PS_AP_LINKS; m++) {
INTFLOAT a_re = AAC_MUL31(ag[m], in_re);
INTFLOAT a_im = AAC_MUL31(ag[m], in_im);
INTFLOAT link_delay_re = ap_delay[m][n+2-m][0];
INTFLOAT link_delay_im = ap_delay[m][n+2-m][1];
INTFLOAT fractional_delay_re = Q_fract[m][0];
INTFLOAT fractional_delay_im = Q_fract[m][1];
INTFLOAT apd_re = in_re;
INTFLOAT apd_im = in_im;
in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
link_delay_im, fractional_delay_im);
in_re -= a_re;
in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
link_delay_im, fractional_delay_re);
in_im -= a_im;
ap_delay[m][n+5][0] = apd_re + AAC_MUL31(ag[m], in_re);
ap_delay[m][n+5][1] = apd_im + AAC_MUL31(ag[m], in_im);
}
out[n][0] = AAC_MUL16(transient_gain[n], in_re);
out[n][1] = AAC_MUL16(transient_gain[n], in_im);
}
}
static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len)
{
INTFLOAT h0 = h[0][0];
INTFLOAT h1 = h[0][1];
INTFLOAT h2 = h[0][2];
INTFLOAT h3 = h[0][3];
INTFLOAT hs0 = h_step[0][0];
INTFLOAT hs1 = h_step[0][1];
INTFLOAT hs2 = h_step[0][2];
INTFLOAT hs3 = h_step[0][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
INTFLOAT l_re = l[n][0];
INTFLOAT l_im = l[n][1];
INTFLOAT r_re = r[n][0];
INTFLOAT r_im = r[n][1];
h0 += hs0;
h1 += hs1;
h2 += hs2;
h3 += hs3;
l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
}
}
static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
INTFLOAT h[2][4], INTFLOAT h_step[2][4],
int len)
{
INTFLOAT h00 = h[0][0], h10 = h[1][0];
INTFLOAT h01 = h[0][1], h11 = h[1][1];
INTFLOAT h02 = h[0][2], h12 = h[1][2];
INTFLOAT h03 = h[0][3], h13 = h[1][3];
INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
int n;
for (n = 0; n < len; n++) {
//l is s, r is d
INTFLOAT l_re = l[n][0];
INTFLOAT l_im = l[n][1];
INTFLOAT r_re = r[n][0];
INTFLOAT r_im = r[n][1];
h00 += hs00;
h01 += hs01;
h02 += hs02;
h03 += hs03;
h10 += hs10;
h11 += hs11;
h12 += hs12;
h13 += hs13;
l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
}
}
av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
{
s->add_squares = ps_add_squares_c;
s->mul_pair_single = ps_mul_pair_single_c;
s->hybrid_analysis = ps_hybrid_analysis_c;
s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
s->decorrelate = ps_decorrelate_c;
s->stereo_interpolate[0] = ps_stereo_interpolate_c;
s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
#if !USE_FIXED
if (ARCH_ARM)
ff_psdsp_init_arm(s);
if (ARCH_AARCH64)
ff_psdsp_init_aarch64(s);
if (ARCH_MIPS)
ff_psdsp_init_mips(s);
if (ARCH_X86)
ff_psdsp_init_x86(s);
#endif /* !USE_FIXED */
}