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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavcodec/wavarc.c
Michael Niedermayer 5f5a1ccd04
avcodec/wavarc: Check that nb_samples is not negative
It is currently probably not possible for it to be negative as
the needed 2Mb input buf size is not achievable. But it is more
robust to check for it too.
If it would become negative than code like
s->samples[0][n] = s->samples[0][s->nb_samples + n];
would crash

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2023-08-23 22:52:26 +02:00

869 lines
25 KiB
C

/*
* WavArc audio decoder
* Copyright (c) 2023 Paul B Mahol
*
* 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 "libavutil/internal.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "decode.h"
#include "get_bits.h"
#include "bytestream.h"
#include "mathops.h"
#include "unary.h"
typedef struct WavArcContext {
AVClass *av_class;
GetBitContext gb;
int shift;
int nb_samples;
int offset;
int align;
int eof;
int skip;
uint8_t *bitstream;
int64_t max_framesize;
int bitstream_size;
int bitstream_index;
int pred[2][70];
int filter[2][70];
int samples[2][640];
uint8_t model[256];
uint16_t freqs[257];
uint16_t ac_value;
uint16_t ac_low;
uint16_t ac_high;
uint16_t range_high;
uint16_t range_low;
uint16_t freq_range;
int ac_pred[70];
int ac_out[570];
} WavArcContext;
static av_cold int wavarc_init(AVCodecContext *avctx)
{
WavArcContext *s = avctx->priv_data;
if (avctx->extradata_size < 52)
return AVERROR_INVALIDDATA;
if (AV_RL32(avctx->extradata + 16) != MKTAG('R','I','F','F'))
return AVERROR_INVALIDDATA;
if (AV_RL32(avctx->extradata + 24) != MKTAG('W','A','V','E'))
return AVERROR_INVALIDDATA;
if (AV_RL32(avctx->extradata + 28) != MKTAG('f','m','t',' '))
return AVERROR_INVALIDDATA;
if (AV_RL16(avctx->extradata + 38) != 1 &&
AV_RL16(avctx->extradata + 38) != 2)
return AVERROR_INVALIDDATA;
av_channel_layout_uninit(&avctx->ch_layout);
av_channel_layout_default(&avctx->ch_layout, AV_RL16(avctx->extradata + 38));
avctx->sample_rate = AV_RL32(avctx->extradata + 40);
s->align = avctx->ch_layout.nb_channels;
switch (AV_RL16(avctx->extradata + 50)) {
case 8: avctx->sample_fmt = AV_SAMPLE_FMT_U8P; break;
case 16: s->align *= 2;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break;
}
s->shift = 0;
switch (avctx->codec_tag) {
case MKTAG('0','C','P','Y'):
s->nb_samples = 640;
s->offset = 0;
break;
case MKTAG('1','D','I','F'):
s->nb_samples = 256;
s->offset = 4;
break;
case MKTAG('2','S','L','P'):
case MKTAG('3','N','L','P'):
case MKTAG('4','A','L','P'):
case MKTAG('5','E','L','P'):
s->nb_samples = 570;
s->offset = 70;
break;
default:
return AVERROR_INVALIDDATA;
}
s->max_framesize = s->nb_samples * 16;
s->bitstream = av_calloc(s->max_framesize, sizeof(*s->bitstream));
if (!s->bitstream)
return AVERROR(ENOMEM);
return 0;
}
static unsigned get_urice(GetBitContext *gb, int k)
{
unsigned x = get_unary(gb, 1, get_bits_left(gb));
unsigned y = get_bits_long(gb, k);
unsigned z = (x << k) | y;
return z;
}
static int get_srice(GetBitContext *gb, int k)
{
unsigned z = get_urice(gb, k);
return (z & 1) ? ~((int)(z >> 1)) : z >> 1;
}
static void do_stereo(WavArcContext *s, int ch, int correlated, int len)
{
const int nb_samples = s->nb_samples;
const int shift = s->shift;
if (ch == 0) {
if (correlated) {
for (int n = 0; n < len; n++) {
s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
s->samples[1][n] = s->pred[1][n] >> shift;
}
} else {
for (int n = 0; n < len; n++) {
s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
s->samples[1][n] = s->pred[0][n] >> shift;
}
}
} else {
if (correlated) {
for (int n = 0; n < nb_samples; n++)
s->samples[1][n + len] += s->samples[0][n + len];
}
for (int n = 0; n < len; n++) {
s->pred[0][n] = s->samples[1][nb_samples + n];
s->pred[1][n] = s->pred[0][n] - s->samples[0][nb_samples + n];
}
}
}
static int decode_0cpy(AVCodecContext *avctx,
WavArcContext *s, GetBitContext *gb)
{
const int bits = s->align * 8;
s->nb_samples = FFMIN(640, get_bits_left(gb) / bits);
switch (avctx->sample_fmt) {
case AV_SAMPLE_FMT_U8P:
for (int n = 0; n < s->nb_samples; n++) {
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
s->samples[ch][n] = get_bits(gb, 8) - 0x80;
}
break;
case AV_SAMPLE_FMT_S16P:
for (int n = 0; n < s->nb_samples; n++) {
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
s->samples[ch][n] = sign_extend(av_bswap16(get_bits(gb, 16)), 16);
}
break;
}
return 0;
}
static int decode_1dif(AVCodecContext *avctx,
WavArcContext *s, GetBitContext *gb)
{
int ch, finished, fill, correlated;
ch = 0;
finished = 0;
while (!finished) {
int *samples = s->samples[ch];
int k, block_type;
if (get_bits_left(gb) <= 0)
return AVERROR_INVALIDDATA;
block_type = get_urice(gb, 1);
if (block_type < 4 && block_type >= 0) {
k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
k = get_urice(gb, k) + 1;
if (k >= 32)
return AVERROR_INVALIDDATA;
}
switch (block_type) {
case 8:
s->eof = 1;
return AVERROR_EOF;
case 7:
s->nb_samples = get_bits(gb, 8);
continue;
case 6:
s->shift = get_urice(gb, 2);
if ((unsigned)s->shift > 31) {
s->shift = 0;
return AVERROR_INVALIDDATA;
}
continue;
case 5:
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 + 4] = fill;
finished = 1;
break;
case 4:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 4] = 0;
finished = 1;
break;
case 3:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 4] = get_srice(gb, k) + (samples[n + 3] - samples[n + 2]) * 3 +
samples[n + 1];
finished = 1;
break;
case 2:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 4] = get_srice(gb, k) + (samples[n + 3] * 2 - samples[n + 2]);
finished = 1;
break;
case 1:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 4] = get_srice(gb, k) + samples[n + 3];
finished = 1;
break;
case 0:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 4] = get_srice(gb, k);
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, 4);
ch = 1;
}
}
if (avctx->ch_layout.nb_channels == 1) {
for (int n = 0; n < 4; n++)
s->samples[0][n] = s->samples[0][s->nb_samples + n];
}
return 0;
}
static int decode_2slp(AVCodecContext *avctx,
WavArcContext *s, GetBitContext *gb)
{
int ch, finished, fill, correlated, order;
ch = 0;
finished = 0;
while (!finished) {
int *samples = s->samples[ch];
int k, block_type;
if (get_bits_left(gb) <= 0)
return AVERROR_INVALIDDATA;
block_type = get_urice(gb, 1);
if (block_type < 5 && block_type >= 0) {
k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
k = get_urice(gb, k) + 1;
if (k >= 32)
return AVERROR_INVALIDDATA;
}
switch (block_type) {
case 9:
s->eof = 1;
return AVERROR_EOF;
case 8:
s->nb_samples = get_urice(gb, 8);
if (s->nb_samples > 570U) {
s->nb_samples = 570;
return AVERROR_INVALIDDATA;
}
continue;
case 7:
s->shift = get_urice(gb, 2);
if ((unsigned)s->shift > 31) {
s->shift = 0;
return AVERROR_INVALIDDATA;
}
continue;
case 6:
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 5:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 70] = 0;
finished = 1;
break;
case 4:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 70] = get_srice(gb, k) + (samples[n + 69] - samples[n + 68]) * 3 +
samples[n + 67];
finished = 1;
break;
case 3:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 70] = get_srice(gb, k) + (samples[n + 69] * 2 - samples[n + 68]);
finished = 1;
break;
case 2:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 70] = get_srice(gb, k);
finished = 1;
break;
case 1:
for (int n = 0; n < s->nb_samples; n++)
samples[n + 70] = get_srice(gb, k) + samples[n + 69];
finished = 1;
break;
case 0:
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);
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] = get_srice(gb, k) + (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 ac_init(AVCodecContext *avctx,
WavArcContext *s, GetBitContext *gb)
{
s->ac_low = 0;
s->ac_high = 0xffffu;
s->ac_value = get_bits(gb, 16);
s->freq_range = s->freqs[256];
if (!s->freq_range)
return AVERROR_INVALIDDATA;
return 0;
}
static unsigned ac_get_prob(WavArcContext *s)
{
return ((s->freq_range - 1) + ((unsigned)s->ac_value - (unsigned)s->ac_low) *
(unsigned)s->freq_range) / (((unsigned)s->ac_high - (unsigned)s->ac_low) + 1) & 0xffffu;
}
static unsigned ac_map_symbol(WavArcContext *s, unsigned prob)
{
int idx = 255;
while (prob < s->freqs[idx])
idx--;
s->range_high = s->freqs[idx + 1];
s->range_low = s->freqs[idx];
return idx;
}
static int ac_normalize(AVCodecContext *avctx, WavArcContext *s, GetBitContext *gb)
{
int range;
range = (unsigned)s->ac_high - (unsigned)s->ac_low + 1;
s->ac_high = (int16_t)((range * (unsigned)s->range_high) / (unsigned)s->freq_range) + -1 + s->ac_low;
s->ac_low += (int16_t)((range * (unsigned)s->range_low) / (unsigned)s->freq_range);
for (;;) {
if ((s->ac_high & 0x8000) != (s->ac_low & 0x8000)) {
if ((s->ac_low & 0x4000) == 0 || (s->ac_high & 0x4000) != 0)
return 0;
s->ac_value ^= 0x4000;
s->ac_low &= 0x3fff;
s->ac_high |= 0x4000;
}
s->ac_low = s->ac_low << 1;
s->ac_high = s->ac_high * 2 + 1;
if (s->ac_high < s->ac_low) {
av_log(avctx, AV_LOG_ERROR, "invalid state\n");
return AVERROR_INVALIDDATA;
}
if (get_bits_left(gb) <= 0) {
av_log(avctx, AV_LOG_ERROR, "overread in arithmetic coder\n");
return AVERROR_INVALIDDATA;
}
s->ac_value = s->ac_value * 2 + get_bits1(gb);
}
}
static void ac_init_model(WavArcContext *s)
{
memset(s->freqs, 0, sizeof(s->freqs));
for (int n = 0; n < 256; n++)
s->freqs[n+1] = s->model[n] + s->freqs[n];
}
static int ac_read_model(AVCodecContext *avctx,
WavArcContext *s,
GetBitContext *gb)
{
unsigned start, end;
memset(s->model, 0, sizeof(s->model));
start = get_bits(gb, 8);
end = get_bits(gb, 8);
for (;;) {
while (start <= end)
s->model[start++] = get_bits(gb, 8);
if (get_bits_left(gb) < 8)
return AVERROR_INVALIDDATA;
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 (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++) {
unsigned prob = ac_get_prob(s);
prob = ac_map_symbol(s, prob);
ac_out[n] = prob - 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] * 3 - samples[n + 68] * 3 + samples[n + 67];
finished = 1;
break;
case 19:
case 6:
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);
}
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] * 2 - 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] + 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] * 2 - 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] += 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];
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] += 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 },
};