mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-28 20:53:54 +02:00
9cef555bc5
regression since 18b2ecc247
Fixes: assertion failure
Fixes: 62164/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_WAVARC_fuzzer-6280165808013312
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
893 lines
26 KiB
C
893 lines
26 KiB
C
/*
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* WavArc audio decoder
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* Copyright (c) 2023 Paul B Mahol
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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 "libavutil/internal.h"
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "codec_internal.h"
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#include "decode.h"
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#include "get_bits.h"
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#include "bytestream.h"
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#include "mathops.h"
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#include "unary.h"
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typedef struct WavArcContext {
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AVClass *av_class;
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GetBitContext gb;
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int shift;
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int nb_samples;
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int offset;
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int align;
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int eof;
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int skip;
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uint8_t *bitstream;
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int64_t max_framesize;
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int bitstream_size;
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int bitstream_index;
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int pred[2][70];
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int filter[2][70];
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int samples[2][640];
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uint8_t model[256];
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uint16_t freqs[257];
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uint16_t ac_value;
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uint16_t ac_low;
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uint16_t ac_high;
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uint16_t range_high;
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uint16_t range_low;
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uint16_t freq_range;
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int ac_pred[70];
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int ac_out[570];
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} WavArcContext;
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static av_cold int wavarc_init(AVCodecContext *avctx)
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{
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WavArcContext *s = avctx->priv_data;
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if (avctx->extradata_size < 52)
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return AVERROR_INVALIDDATA;
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if (AV_RL32(avctx->extradata + 16) != MKTAG('R','I','F','F'))
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return AVERROR_INVALIDDATA;
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if (AV_RL32(avctx->extradata + 24) != MKTAG('W','A','V','E'))
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return AVERROR_INVALIDDATA;
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if (AV_RL32(avctx->extradata + 28) != MKTAG('f','m','t',' '))
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return AVERROR_INVALIDDATA;
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if (AV_RL16(avctx->extradata + 38) != 1 &&
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AV_RL16(avctx->extradata + 38) != 2)
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return AVERROR_INVALIDDATA;
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av_channel_layout_uninit(&avctx->ch_layout);
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av_channel_layout_default(&avctx->ch_layout, AV_RL16(avctx->extradata + 38));
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avctx->sample_rate = AV_RL32(avctx->extradata + 40);
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s->align = avctx->ch_layout.nb_channels;
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switch (AV_RL16(avctx->extradata + 50)) {
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case 8: avctx->sample_fmt = AV_SAMPLE_FMT_U8P; break;
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case 16: s->align *= 2;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break;
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}
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s->shift = 0;
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switch (avctx->codec_tag) {
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case MKTAG('0','C','P','Y'):
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s->nb_samples = 640;
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s->offset = 0;
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break;
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case MKTAG('1','D','I','F'):
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s->nb_samples = 256;
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s->offset = 4;
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break;
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case MKTAG('2','S','L','P'):
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case MKTAG('3','N','L','P'):
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case MKTAG('4','A','L','P'):
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case MKTAG('5','E','L','P'):
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s->nb_samples = 570;
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s->offset = 70;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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s->max_framesize = s->nb_samples * 16;
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s->bitstream = av_calloc(s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE, sizeof(*s->bitstream));
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if (!s->bitstream)
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return AVERROR(ENOMEM);
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return 0;
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}
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static unsigned get_urice(GetBitContext *gb, int k)
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{
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unsigned x = get_unary(gb, 1, get_bits_left(gb));
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unsigned y = get_bits_long(gb, k);
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unsigned z = (x << k) | y;
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return z;
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}
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static int get_srice(GetBitContext *gb, int k)
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{
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unsigned z = get_urice(gb, k);
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return (z & 1) ? ~((int)(z >> 1)) : z >> 1;
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}
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static void do_stereo(WavArcContext *s, int ch, int correlated, int len)
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{
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const int nb_samples = s->nb_samples;
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const int shift = s->shift;
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if (ch == 0) {
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if (correlated) {
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for (int n = 0; n < len; n++) {
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s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
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s->samples[1][n] = s->pred[1][n] >> shift;
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}
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} else {
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for (int n = 0; n < len; n++) {
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s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
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s->samples[1][n] = s->pred[0][n] >> shift;
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}
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}
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} else {
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if (correlated) {
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for (int n = 0; n < nb_samples; n++)
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s->samples[1][n + len] += (unsigned)s->samples[0][n + len];
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}
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for (int n = 0; n < len; n++) {
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s->pred[0][n] = s->samples[1][nb_samples + n];
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s->pred[1][n] = s->pred[0][n] - (unsigned)s->samples[0][nb_samples + n];
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}
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}
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}
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static int decode_0cpy(AVCodecContext *avctx,
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WavArcContext *s, GetBitContext *gb)
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{
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const int bits = s->align * 8;
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s->nb_samples = FFMIN(640, get_bits_left(gb) / bits);
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switch (avctx->sample_fmt) {
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case AV_SAMPLE_FMT_U8P:
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for (int n = 0; n < s->nb_samples; n++) {
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
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s->samples[ch][n] = get_bits(gb, 8) - 0x80;
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}
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break;
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case AV_SAMPLE_FMT_S16P:
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for (int n = 0; n < s->nb_samples; n++) {
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for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
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s->samples[ch][n] = sign_extend(av_bswap16(get_bits(gb, 16)), 16);
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}
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break;
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}
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return 0;
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}
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static int decode_1dif(AVCodecContext *avctx,
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WavArcContext *s, GetBitContext *gb)
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{
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int ch, finished, fill, correlated;
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ch = 0;
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finished = 0;
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while (!finished) {
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int *samples = s->samples[ch];
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int k, block_type;
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if (get_bits_left(gb) <= 0)
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return AVERROR_INVALIDDATA;
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block_type = get_urice(gb, 1);
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if (block_type < 4 && block_type >= 0) {
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k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
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k = get_urice(gb, k) + 1;
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if (k >= 32)
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return AVERROR_INVALIDDATA;
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}
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switch (block_type) {
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case 8:
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s->eof = 1;
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return AVERROR_EOF;
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case 7:
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s->nb_samples = get_bits(gb, 8);
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continue;
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case 6:
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s->shift = get_urice(gb, 2);
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if ((unsigned)s->shift > 31) {
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s->shift = 0;
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return AVERROR_INVALIDDATA;
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}
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continue;
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case 5:
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if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
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fill = (int8_t)get_bits(gb, 8);
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fill -= 0x80;
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} else {
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fill = (int16_t)get_bits(gb, 16);
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fill -= 0x8000;
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}
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = fill;
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finished = 1;
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break;
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case 4:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = 0;
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finished = 1;
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break;
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case 3:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k) + (samples[n + 3] - (unsigned)samples[n + 2]) * 3 +
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samples[n + 1];
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finished = 1;
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break;
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case 2:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k) + (samples[n + 3] * 2U - samples[n + 2]);
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finished = 1;
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break;
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case 1:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k) + (unsigned)samples[n + 3];
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finished = 1;
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break;
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case 0:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 4] = get_srice(gb, k);
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finished = 1;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
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if (ch == 0)
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correlated = get_bits1(gb);
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finished = ch != 0;
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do_stereo(s, ch, correlated, 4);
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ch = 1;
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}
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}
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if (avctx->ch_layout.nb_channels == 1) {
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for (int n = 0; n < 4; n++)
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s->samples[0][n] = s->samples[0][s->nb_samples + n];
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}
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return 0;
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}
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static int decode_2slp(AVCodecContext *avctx,
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WavArcContext *s, GetBitContext *gb)
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{
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int ch, finished, fill, correlated, order;
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ch = 0;
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finished = 0;
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while (!finished) {
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int *samples = s->samples[ch];
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int k, block_type;
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if (get_bits_left(gb) <= 0)
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return AVERROR_INVALIDDATA;
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block_type = get_urice(gb, 1);
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if (block_type < 5 && block_type >= 0) {
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k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
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k = get_urice(gb, k) + 1;
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if (k >= 32)
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return AVERROR_INVALIDDATA;
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}
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switch (block_type) {
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case 9:
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s->eof = 1;
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return AVERROR_EOF;
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case 8:
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s->nb_samples = get_urice(gb, 8);
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if (s->nb_samples > 570U) {
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s->nb_samples = 570;
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return AVERROR_INVALIDDATA;
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}
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continue;
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case 7:
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s->shift = get_urice(gb, 2);
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if ((unsigned)s->shift > 31) {
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s->shift = 0;
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return AVERROR_INVALIDDATA;
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}
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continue;
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case 6:
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if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
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fill = (int8_t)get_bits(gb, 8);
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fill -= 0x80;
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} else {
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fill = (int16_t)get_bits(gb, 16);
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fill -= 0x8000;
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}
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = fill;
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finished = 1;
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break;
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case 5:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = 0;
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finished = 1;
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break;
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case 4:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k) + (samples[n + 69] - (unsigned)samples[n + 68]) * 3 +
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samples[n + 67];
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finished = 1;
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break;
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case 3:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k) + (samples[n + 69] * 2U - samples[n + 68]);
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finished = 1;
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break;
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case 2:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k);
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finished = 1;
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break;
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case 1:
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for (int n = 0; n < s->nb_samples; n++)
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samples[n + 70] = get_srice(gb, k) + (unsigned)samples[n + 69];
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finished = 1;
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break;
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case 0:
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order = get_urice(gb, 2);
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if ((unsigned)order > FF_ARRAY_ELEMS(s->filter[ch]))
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return AVERROR_INVALIDDATA;
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for (int o = 0; o < order; o++)
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s->filter[ch][o] = get_srice(gb, 2);
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for (int n = 0; n < s->nb_samples; n++) {
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int sum = 15;
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for (int o = 0; o < order; o++)
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sum += s->filter[ch][o] * (unsigned)samples[n + 70 - o - 1];
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samples[n + 70] = get_srice(gb, k) + (sum >> 4);
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}
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finished = 1;
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break;
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default:
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return AVERROR_INVALIDDATA;
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}
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if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
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if (ch == 0)
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correlated = get_bits1(gb);
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finished = ch != 0;
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do_stereo(s, ch, correlated, 70);
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ch = 1;
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}
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}
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if (avctx->ch_layout.nb_channels == 1) {
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for (int n = 0; n < 70; n++)
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s->samples[0][n] = s->samples[0][s->nb_samples + n];
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}
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return 0;
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}
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static int ac_init(AVCodecContext *avctx,
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WavArcContext *s, GetBitContext *gb)
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{
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s->ac_low = 0;
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s->ac_high = 0xffffu;
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s->ac_value = get_bits(gb, 16);
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s->freq_range = s->freqs[256];
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if (!s->freq_range)
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return AVERROR_INVALIDDATA;
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return 0;
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}
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static uint16_t ac_get_prob(WavArcContext *s)
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{
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return ((s->freq_range - 1) + (s->ac_value - s->ac_low) * s->freq_range) /
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((s->ac_high - s->ac_low) + 1U);
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}
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static uint8_t ac_map_symbol(WavArcContext *s, uint16_t prob)
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{
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int idx = 255;
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while (prob < s->freqs[idx])
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idx--;
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s->range_high = s->freqs[idx + 1];
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s->range_low = s->freqs[idx];
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return idx;
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}
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static int ac_normalize(AVCodecContext *avctx, WavArcContext *s, GetBitContext *gb)
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{
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int range;
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if (s->ac_high < s->ac_low)
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goto fail;
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range = (s->ac_high - s->ac_low) + 1;
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s->ac_high = (range * s->range_high) / s->freq_range + s->ac_low - 1;
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s->ac_low += (range * s->range_low) / s->freq_range;
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if (s->ac_high < s->ac_low)
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goto fail;
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for (;;) {
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if ((s->ac_high & 0x8000) != (s->ac_low & 0x8000)) {
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if (((s->ac_low & 0x4000) == 0) || ((s->ac_high & 0x4000) != 0))
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return 0;
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s->ac_value ^= 0x4000;
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s->ac_low &= 0x3fff;
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s->ac_high |= 0x4000;
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}
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s->ac_low = s->ac_low * 2;
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s->ac_high = s->ac_high * 2 | 1;
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if (s->ac_high < s->ac_low)
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goto fail;
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if (get_bits_left(gb) <= 0) {
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av_log(avctx, AV_LOG_ERROR, "overread in arithmetic coder\n");
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goto fail;
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}
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s->ac_value = s->ac_value * 2 + get_bits1(gb);
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if (s->ac_low > s->ac_value || s->ac_high < s->ac_value)
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goto fail;
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}
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fail:
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av_log(avctx, AV_LOG_ERROR, "invalid state\n");
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return AVERROR_INVALIDDATA;
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}
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static void ac_init_model(WavArcContext *s)
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{
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memset(s->freqs, 0, sizeof(s->freqs));
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for (int n = 0; n < 256; n++)
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s->freqs[n+1] = s->model[n] + s->freqs[n];
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}
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static int ac_read_model(AVCodecContext *avctx,
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WavArcContext *s,
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GetBitContext *gb)
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{
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unsigned start, end;
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memset(s->model, 0, sizeof(s->model));
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start = get_bits(gb, 8);
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end = get_bits(gb, 8);
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for (;;) {
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while (start <= end) {
|
|
if (get_bits_left(gb) < 8)
|
|
return AVERROR_INVALIDDATA;
|
|
s->model[start++] = get_bits(gb, 8);
|
|
}
|
|
|
|
if (get_bits_left(gb) < 8)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
start = get_bits(gb, 8);
|
|
if (!start)
|
|
break;
|
|
|
|
end = get_bits(gb, 8);
|
|
}
|
|
|
|
ac_init_model(s);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int decode_5elp(AVCodecContext *avctx,
|
|
WavArcContext *s, GetBitContext *gb)
|
|
{
|
|
int ch, finished, fill, correlated, order = 0;
|
|
|
|
ch = 0;
|
|
finished = 0;
|
|
while (!finished) {
|
|
int *samples = s->samples[ch];
|
|
int *ac_pred = s->ac_pred;
|
|
int *ac_out = s->ac_out;
|
|
int k, block_type;
|
|
|
|
if (get_bits_left(gb) <= 0)
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
memset(s->ac_out, 0, sizeof(s->ac_out));
|
|
|
|
block_type = get_urice(gb, 1);
|
|
av_log(avctx, AV_LOG_DEBUG, "block_type : %d\n", block_type);
|
|
|
|
if (block_type >= 0 && block_type <= 7) {
|
|
k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
|
|
k = get_urice(gb, k) + 1;
|
|
if (k >= 32)
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (block_type <= 2 || block_type == 6 || block_type == 13 ||
|
|
block_type == 14 || block_type == 15 || block_type == 19) {
|
|
order = get_urice(gb, 2);
|
|
if ((unsigned)order > FF_ARRAY_ELEMS(s->filter[ch]))
|
|
return AVERROR_INVALIDDATA;
|
|
for (int o = 0; o < order; o++)
|
|
s->filter[ch][o] = get_srice(gb, 2);
|
|
}
|
|
|
|
if (block_type >= 0 && block_type <= 7) {
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] = get_srice(gb, k);
|
|
} else {
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] = 0;
|
|
}
|
|
|
|
if (block_type >= 13 && block_type <= 20) {
|
|
const int ac_size = get_bits(gb, 12);
|
|
const int ac_pos = get_bits_count(gb);
|
|
GetBitContext ac_gb = *gb;
|
|
int ret;
|
|
|
|
skip_bits_long(gb, ac_size);
|
|
ret = ac_read_model(avctx, s, &ac_gb);
|
|
if (ret < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "bad arithmetic model\n");
|
|
return ret;
|
|
}
|
|
|
|
ret = ac_init(avctx, s, &ac_gb);
|
|
if (ret < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "cannot init arithmetic decoder\n");
|
|
return ret;
|
|
}
|
|
|
|
for (int n = 0; n < s->nb_samples; n++) {
|
|
uint16_t prob = ac_get_prob(s);
|
|
int ac = ac_map_symbol(s, prob);
|
|
ac_out[n] = ac - 0x80;
|
|
if ((ret = ac_normalize(avctx, s, &ac_gb)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
if (get_bits_count(&ac_gb) != ac_pos + ac_size) {
|
|
av_log(avctx, AV_LOG_DEBUG, "over/under-read in arithmetic coder: %d\n",
|
|
ac_pos + ac_size - get_bits_count(&ac_gb));
|
|
}
|
|
}
|
|
|
|
switch (block_type) {
|
|
case 12:
|
|
s->eof = 1;
|
|
return AVERROR_EOF;
|
|
case 11:
|
|
s->nb_samples = get_urice(gb, 8);
|
|
if (s->nb_samples > 570U) {
|
|
s->nb_samples = 570;
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
continue;
|
|
case 10:
|
|
s->shift = get_urice(gb, 2);
|
|
if ((unsigned)s->shift > 31) {
|
|
s->shift = 0;
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
continue;
|
|
case 9:
|
|
if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
|
|
fill = (int8_t)get_bits(gb, 8);
|
|
fill -= 0x80;
|
|
} else {
|
|
fill = (int16_t)get_bits(gb, 16);
|
|
fill -= 0x8000;
|
|
}
|
|
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] = fill;
|
|
finished = 1;
|
|
break;
|
|
case 8:
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] = 0;
|
|
finished = 1;
|
|
break;
|
|
case 20:
|
|
case 7:
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += ac_out[n] + samples[n + 69] * 3U - samples[n + 68] * 3U + samples[n + 67];
|
|
finished = 1;
|
|
break;
|
|
case 19:
|
|
case 6:
|
|
for (int n = 0; n < 70; n++) {
|
|
ac_pred[n] = samples[n];
|
|
samples[n] = 0;
|
|
}
|
|
|
|
for (int n = 0; n < s->nb_samples; n++) {
|
|
int sum = 15;
|
|
|
|
for (int o = 0; o < order; o++)
|
|
sum += s->filter[ch][o] * (unsigned)samples[n + 70 - o - 1];
|
|
|
|
samples[n + 70] += ac_out[n] + (sum >> 4);
|
|
}
|
|
|
|
for (int n = 0; n < 70; n++)
|
|
samples[n] = ac_pred[n];
|
|
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += ac_out[n] + samples[n + 69] * 3U - samples[n + 68] * 3U + samples[n + 67];
|
|
|
|
finished = 1;
|
|
break;
|
|
case 18:
|
|
case 5:
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += ac_out[n] + samples[n + 69] * 2U - samples[n + 68];
|
|
finished = 1;
|
|
break;
|
|
case 17:
|
|
case 4:
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += ac_out[n];
|
|
finished = 1;
|
|
break;
|
|
case 16:
|
|
case 3:
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += ac_out[n] + (unsigned)samples[n + 69];
|
|
finished = 1;
|
|
break;
|
|
case 15:
|
|
case 2:
|
|
for (int n = 0; n < 70; n++) {
|
|
ac_pred[n] = samples[n];
|
|
samples[n] = 0;
|
|
}
|
|
|
|
for (int n = 0; n < s->nb_samples; n++) {
|
|
int sum = 15;
|
|
|
|
for (int o = 0; o < order; o++)
|
|
sum += s->filter[ch][o] * (unsigned)samples[n + 70 - o - 1];
|
|
|
|
samples[n + 70] += ac_out[n] + (sum >> 4);
|
|
}
|
|
|
|
for (int n = 0; n < 70; n++)
|
|
samples[n] = ac_pred[n];
|
|
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += samples[n + 69] * 2U - samples[n + 68];
|
|
|
|
finished = 1;
|
|
break;
|
|
case 14:
|
|
case 1:
|
|
for (int n = 0; n < 70; n++) {
|
|
ac_pred[n] = samples[n];
|
|
samples[n] = 0;
|
|
}
|
|
|
|
for (int n = 0; n < s->nb_samples; n++) {
|
|
int sum = 15;
|
|
|
|
for (int o = 0; o < order; o++)
|
|
sum += s->filter[ch][o] * (unsigned)samples[n + 70 - o - 1];
|
|
|
|
samples[n + 70] += (unsigned)ac_out[n] + (sum >> 4);
|
|
}
|
|
|
|
for (int n = 0; n < 70; n++)
|
|
samples[n] = ac_pred[n];
|
|
|
|
for (int n = 0; n < s->nb_samples; n++)
|
|
samples[n + 70] += (unsigned)samples[n + 69];
|
|
|
|
finished = 1;
|
|
break;
|
|
case 13:
|
|
case 0:
|
|
for (int n = 0; n < s->nb_samples; n++) {
|
|
int sum = 15;
|
|
|
|
for (int o = 0; o < order; o++)
|
|
sum += s->filter[ch][o] * (unsigned)samples[n + 70 - o - 1];
|
|
|
|
samples[n + 70] += (unsigned)ac_out[n] + (sum >> 4);
|
|
}
|
|
finished = 1;
|
|
break;
|
|
default:
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
|
|
if (ch == 0)
|
|
correlated = get_bits1(gb);
|
|
finished = ch != 0;
|
|
do_stereo(s, ch, correlated, 70);
|
|
ch = 1;
|
|
}
|
|
}
|
|
|
|
if (avctx->ch_layout.nb_channels == 1) {
|
|
for (int n = 0; n < 70; n++)
|
|
s->samples[0][n] = s->samples[0][s->nb_samples + n];
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int wavarc_decode(AVCodecContext *avctx, AVFrame *frame,
|
|
int *got_frame_ptr, AVPacket *pkt)
|
|
{
|
|
WavArcContext *s = avctx->priv_data;
|
|
GetBitContext *gb = &s->gb;
|
|
int buf_size, input_buf_size;
|
|
const uint8_t *buf;
|
|
int ret, n;
|
|
|
|
if ((!pkt->size && !s->bitstream_size) || s->nb_samples == 0 || s->eof) {
|
|
*got_frame_ptr = 0;
|
|
return pkt->size;
|
|
}
|
|
|
|
buf_size = FFMIN(pkt->size, s->max_framesize - s->bitstream_size);
|
|
input_buf_size = buf_size;
|
|
if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > s->max_framesize) {
|
|
memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
|
|
s->bitstream_index = 0;
|
|
}
|
|
if (pkt->data)
|
|
memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
|
|
buf = &s->bitstream[s->bitstream_index];
|
|
buf_size += s->bitstream_size;
|
|
s->bitstream_size = buf_size;
|
|
if (buf_size < s->max_framesize && pkt->data) {
|
|
*got_frame_ptr = 0;
|
|
return input_buf_size;
|
|
}
|
|
|
|
if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
|
|
goto fail;
|
|
skip_bits(gb, s->skip);
|
|
|
|
switch (avctx->codec_tag) {
|
|
case MKTAG('0','C','P','Y'):
|
|
ret = decode_0cpy(avctx, s, gb);
|
|
break;
|
|
case MKTAG('1','D','I','F'):
|
|
ret = decode_1dif(avctx, s, gb);
|
|
break;
|
|
case MKTAG('2','S','L','P'):
|
|
case MKTAG('3','N','L','P'):
|
|
case MKTAG('4','A','L','P'):
|
|
ret = decode_2slp(avctx, s, gb);
|
|
break;
|
|
case MKTAG('5','E','L','P'):
|
|
ret = decode_5elp(avctx, s, gb);
|
|
break;
|
|
default:
|
|
ret = AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
|
|
n = get_bits_count(gb) / 8;
|
|
|
|
if (n > buf_size) {
|
|
fail:
|
|
s->bitstream_size = 0;
|
|
s->bitstream_index = 0;
|
|
if (ret == AVERROR_EOF)
|
|
return 0;
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
frame->nb_samples = s->nb_samples;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
goto fail;
|
|
|
|
switch (avctx->sample_fmt) {
|
|
case AV_SAMPLE_FMT_U8P:
|
|
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
|
|
uint8_t *dst = (uint8_t *)frame->extended_data[ch];
|
|
const int *src = s->samples[ch] + s->offset;
|
|
|
|
for (int n = 0; n < frame->nb_samples; n++)
|
|
dst[n] = src[n] * (1U << s->shift) + 0x80U;
|
|
}
|
|
break;
|
|
case AV_SAMPLE_FMT_S16P:
|
|
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
|
|
int16_t *dst = (int16_t *)frame->extended_data[ch];
|
|
const int *src = s->samples[ch] + s->offset;
|
|
|
|
for (int n = 0; n < frame->nb_samples; n++)
|
|
dst[n] = src[n] * (1U << s->shift);
|
|
}
|
|
break;
|
|
}
|
|
|
|
*got_frame_ptr = 1;
|
|
|
|
if (s->bitstream_size) {
|
|
s->bitstream_index += n;
|
|
s->bitstream_size -= n;
|
|
return input_buf_size;
|
|
}
|
|
|
|
return n;
|
|
}
|
|
|
|
static av_cold int wavarc_close(AVCodecContext *avctx)
|
|
{
|
|
WavArcContext *s = avctx->priv_data;
|
|
|
|
av_freep(&s->bitstream);
|
|
s->bitstream_size = 0;
|
|
|
|
return 0;
|
|
}
|
|
|
|
const FFCodec ff_wavarc_decoder = {
|
|
.p.name = "wavarc",
|
|
CODEC_LONG_NAME("Waveform Archiver"),
|
|
.p.type = AVMEDIA_TYPE_AUDIO,
|
|
.p.id = AV_CODEC_ID_WAVARC,
|
|
.priv_data_size = sizeof(WavArcContext),
|
|
.init = wavarc_init,
|
|
FF_CODEC_DECODE_CB(wavarc_decode),
|
|
.close = wavarc_close,
|
|
.p.capabilities = AV_CODEC_CAP_DR1 |
|
|
#if FF_API_SUBFRAMES
|
|
AV_CODEC_CAP_SUBFRAMES |
|
|
#endif
|
|
AV_CODEC_CAP_DELAY,
|
|
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
|
|
AV_SAMPLE_FMT_S16P,
|
|
AV_SAMPLE_FMT_NONE },
|
|
};
|