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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/ac3dec_fixed.c
Michael Niedermayer 8e30502abe avcodec/ac3dec_fixed: Fix several invalid left shifts in scale_coefs()
Fixes: left shift of negative value -14336
Fixes: 20298/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AC3_FIXED_fuzzer-5675484201615360

Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2020-04-16 00:56:49 +02:00

199 lines
6.6 KiB
C

/*
* Copyright (c) 2012
* MIPS Technologies, Inc., California.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* Author: Stanislav Ocovaj (socovaj@mips.com)
*
* AC3 fixed-point decoder for MIPS platforms
*
* 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
*/
#define FFT_FLOAT 0
#define USE_FIXED 1
#define FFT_FIXED_32 1
#include "ac3dec.h"
static const int end_freq_inv_tab[8] =
{
50529027, 44278013, 39403370, 32292987, 27356480, 23729101, 20951060, 18755316
};
static void scale_coefs (
int32_t *dst,
const int32_t *src,
int dynrng,
int len)
{
int i, shift;
unsigned mul, round;
int temp, temp1, temp2, temp3, temp4, temp5, temp6, temp7;
mul = (dynrng & 0x1f) + 0x20;
shift = 4 - (sign_extend(dynrng, 9) >> 5);
if (shift > 0 ) {
round = 1 << (shift-1);
for (i=0; i<len; i+=8) {
temp = src[i] * mul;
temp1 = src[i+1] * mul;
temp = temp + round;
temp2 = src[i+2] * mul;
temp1 = temp1 + round;
dst[i] = temp >> shift;
temp3 = src[i+3] * mul;
temp2 = temp2 + round;
dst[i+1] = temp1 >> shift;
temp4 = src[i + 4] * mul;
temp3 = temp3 + round;
dst[i+2] = temp2 >> shift;
temp5 = src[i+5] * mul;
temp4 = temp4 + round;
dst[i+3] = temp3 >> shift;
temp6 = src[i+6] * mul;
dst[i+4] = temp4 >> shift;
temp5 = temp5 + round;
temp7 = src[i+7] * mul;
temp6 = temp6 + round;
dst[i+5] = temp5 >> shift;
temp7 = temp7 + round;
dst[i+6] = temp6 >> shift;
dst[i+7] = temp7 >> shift;
}
} else {
shift = -shift;
mul <<= shift;
for (i=0; i<len; i+=8) {
temp = src[i] * mul;
temp1 = src[i+1] * mul;
temp2 = src[i+2] * mul;
dst[i] = temp;
temp3 = src[i+3] * mul;
dst[i+1] = temp1;
temp4 = src[i + 4] * mul;
dst[i+2] = temp2;
temp5 = src[i+5] * mul;
dst[i+3] = temp3;
temp6 = src[i+6] * mul;
dst[i+4] = temp4;
temp7 = src[i+7] * mul;
dst[i+5] = temp5;
dst[i+6] = temp6;
dst[i+7] = temp7;
}
}
}
/**
* Downmix samples from original signal to stereo or mono (this is for 16-bit samples
* and fixed point decoder - original (for 32-bit samples) is in ac3dsp.c).
*/
static void ac3_downmix_c_fixed16(int16_t **samples, int16_t **matrix,
int out_ch, int in_ch, int len)
{
int i, j;
int v0, v1;
if (out_ch == 2) {
for (i = 0; i < len; i++) {
v0 = v1 = 0;
for (j = 0; j < in_ch; j++) {
v0 += samples[j][i] * matrix[0][j];
v1 += samples[j][i] * matrix[1][j];
}
samples[0][i] = (v0+2048)>>12;
samples[1][i] = (v1+2048)>>12;
}
} else if (out_ch == 1) {
for (i = 0; i < len; i++) {
v0 = 0;
for (j = 0; j < in_ch; j++)
v0 += samples[j][i] * matrix[0][j];
samples[0][i] = (v0+2048)>>12;
}
}
}
#include "eac3dec.c"
#include "ac3dec.c"
static const AVOption options[] = {
{ "cons_noisegen", "enable consistent noise generation", OFFSET(consistent_noise_generation), AV_OPT_TYPE_BOOL, {.i64 = 0 }, 0, 1, PAR },
{ "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
{ "heavy_compr", "enable heavy dynamic range compression", OFFSET(heavy_compression), AV_OPT_TYPE_BOOL, {.i64 = 0 }, 0, 1, PAR },
{ NULL},
};
static const AVClass ac3_decoder_class = {
.class_name = "Fixed-Point AC-3 Decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVCodec ff_ac3_fixed_decoder = {
.name = "ac3_fixed",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AC3,
.priv_data_size = sizeof (AC3DecodeContext),
.init = ac3_decode_init,
.close = ac3_decode_end,
.decode = ac3_decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.priv_class = &ac3_decoder_class,
};