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FFmpeg/libavformat/mca.c

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/*
* MCA demuxer
* Copyright (c) 2020 Zixing Liu
*
* 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/intreadwrite.h"
#include "avformat.h"
#include "avio_internal.h"
#include "internal.h"
typedef struct MCADemuxContext {
uint32_t block_count;
uint16_t block_size;
uint32_t current_block;
uint32_t data_start;
uint32_t samples_per_block;
} MCADemuxContext;
static int probe(const AVProbeData *p)
{
if (AV_RL32(p->buf) == MKTAG('M', 'A', 'D', 'P') &&
AV_RL16(p->buf + 4) <= 0x5)
return AVPROBE_SCORE_MAX / 3 * 2;
return 0;
}
static int read_header(AVFormatContext *s)
{
AVStream *st;
MCADemuxContext *m = s->priv_data;
AVCodecParameters *par;
int64_t file_size = avio_size(s->pb);
uint16_t version = 0;
uint32_t header_size, data_size, data_offset, loop_start, loop_end,
nb_samples, nb_metadata, coef_offset = 0;
int ch, ret;
int64_t ret_size;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
par = st->codecpar;
par->codec_type = AVMEDIA_TYPE_AUDIO;
// parse file headers
avio_skip(s->pb, 0x4); // skip the file magic
version = avio_rl16(s->pb);
avio_skip(s->pb, 0x2); // padding
par->channels = avio_r8(s->pb);
avio_skip(s->pb, 0x1); // padding
m->block_size = avio_rl16(s->pb);
nb_samples = avio_rl32(s->pb);
par->sample_rate = avio_rl32(s->pb);
loop_start = avio_rl32(s->pb);
loop_end = avio_rl32(s->pb);
header_size = avio_rl32(s->pb);
data_size = avio_rl32(s->pb);
avio_skip(s->pb, 0x4);
nb_metadata = avio_rl16(s->pb);
avio_skip(s->pb, 0x2); // unknown u16 field
// samples per frame = 14; frame size = 8 (2^3)
m->samples_per_block = (m->block_size * 14) >> 3;
if (m->samples_per_block < 1)
return AVERROR_INVALIDDATA;
m->block_count = nb_samples / m->samples_per_block;
st->duration = nb_samples;
// sanity checks
if (!par->channels || par->sample_rate <= 0
|| loop_start > loop_end || m->block_count < 1)
return AVERROR_INVALIDDATA;
if ((ret = av_dict_set_int(&s->metadata, "loop_start",
av_rescale(loop_start, AV_TIME_BASE,
par->sample_rate), 0)) < 0)
return ret;
if ((ret = av_dict_set_int(&s->metadata, "loop_end",
av_rescale(loop_end, AV_TIME_BASE,
par->sample_rate), 0)) < 0)
return ret;
if ((32 + 4 + m->block_size) > (INT_MAX / par->channels) ||
(32 + 4 + m->block_size) * par->channels > INT_MAX - 8)
return AVERROR_INVALIDDATA;
avpriv_set_pts_info(st, 64, 1, par->sample_rate);
if (version <= 4) {
// version <= 4 needs to use the file size to calculate the offsets
if (file_size < 0) {
return AVERROR(EIO);
}
if (file_size - data_size > UINT32_MAX)
return AVERROR_INVALIDDATA;
m->data_start = file_size - data_size;
if (version <= 3) {
nb_metadata = 0;
// header_size is not available or incorrect in older versions
header_size = m->data_start;
}
} else if (version == 5) {
// read data_start location from the header
if (0x30 * par->channels + 0x4 > header_size)
return AVERROR_INVALIDDATA;
data_offset = header_size - 0x30 * par->channels - 0x4;
if ((ret_size = avio_seek(s->pb, data_offset, SEEK_SET)) < 0)
return ret_size;
m->data_start = avio_rl32(s->pb);
// check if the metadata is reasonable
if (file_size > 0 && (int64_t)m->data_start + data_size > file_size) {
// the header is broken beyond repair
if ((int64_t)header_size + data_size > file_size) {
av_log(s, AV_LOG_ERROR,
"MCA metadata corrupted, unable to determine the data offset.\n");
return AVERROR_INVALIDDATA;
}
// recover the data_start information from the data size
av_log(s, AV_LOG_WARNING,
"Incorrect header size found in metadata, "
"header size approximated from the data size\n");
if (file_size - data_offset > UINT32_MAX)
return AVERROR_INVALIDDATA;
m->data_start = file_size - data_size;
}
} else {
avpriv_request_sample(s, "version %d", version);
return AVERROR_PATCHWELCOME;
}
// coefficient alignment = 0x30; metadata size = 0x14
if (0x30 * par->channels + nb_metadata * 0x14 > header_size)
return AVERROR_INVALIDDATA;
coef_offset =
header_size - 0x30 * par->channels + nb_metadata * 0x14;
st->start_time = 0;
par->codec_id = AV_CODEC_ID_ADPCM_THP_LE;
ret = ff_alloc_extradata(st->codecpar, 32 * par->channels);
if (ret < 0)
return ret;
if ((ret_size = avio_seek(s->pb, coef_offset, SEEK_SET)) < 0)
return ret_size;
for (ch = 0; ch < par->channels; ch++) {
if ((ret = ffio_read_size(s->pb, par->extradata + ch * 32, 32)) < 0)
return ret;
// 0x30 (alignment) - 0x20 (actual size, 32) = 0x10 (padding)
avio_skip(s->pb, 0x10);
}
// seek to the beginning of the adpcm data
// there are some files where the adpcm audio data is not immediately after the header
if ((ret_size = avio_seek(s->pb, m->data_start, SEEK_SET)) < 0)
return ret_size;
return 0;
}
static int read_packet(AVFormatContext *s, AVPacket *pkt)
{
AVCodecParameters *par = s->streams[0]->codecpar;
MCADemuxContext *m = s->priv_data;
uint16_t size = m->block_size;
uint32_t samples = m->samples_per_block;
int ret = 0;
if (avio_feof(s->pb))
return AVERROR_EOF;
m->current_block++;
if (m->current_block > m->block_count)
return AVERROR_EOF;
if ((ret = av_get_packet(s->pb, pkt, size * par->channels)) < 0)
return ret;
pkt->duration = samples;
pkt->stream_index = 0;
return 0;
}
static int read_seek(AVFormatContext *s, int stream_index,
int64_t timestamp, int flags)
{
AVStream *st = s->streams[stream_index];
MCADemuxContext *m = s->priv_data;
int64_t ret = 0;
if (timestamp < 0)
timestamp = 0;
timestamp /= m->samples_per_block;
if (timestamp >= m->block_count)
timestamp = m->block_count - 1;
ret = avio_seek(s->pb, m->data_start + timestamp * m->block_size *
st->codecpar->channels, SEEK_SET);
if (ret < 0)
return ret;
m->current_block = timestamp;
ff_update_cur_dts(s, st, timestamp * m->samples_per_block);
return 0;
}
AVInputFormat ff_mca_demuxer = {
.name = "mca",
.long_name = NULL_IF_CONFIG_SMALL("MCA Audio Format"),
.priv_data_size = sizeof(MCADemuxContext),
.read_probe = probe,
.read_header = read_header,
.read_packet = read_packet,
.read_seek = read_seek,
.extensions = "mca",
};