librempeg/libavformat/dsfdec.c
Andreas Rheinhardt fed0282508 avformat: Avoid allocation for AVFormatInternal
Do this by allocating AVFormatContext together with the data that is
currently in AVFormatInternal; or rather: Put AVFormatContext at the
beginning of a new structure called FFFormatContext (which encompasses
more than just the internal fields and is a proper context in its own
right, hence the name) and remove AVFormatInternal altogether.

The biggest simplifications occured in avformat_alloc_context(), where
one can now simply call avformat_free_context() in case of errors.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-17 04:58:34 +02:00

212 lines
6.5 KiB
C

/*
* DSD Stream File (DSF) demuxer
* Copyright (c) 2014 Peter Ross
*
* 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/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#include "id3v2.h"
typedef struct {
uint64_t data_end;
uint64_t audio_size;
uint64_t data_size;
} DSFContext;
static int dsf_probe(const AVProbeData *p)
{
if (p->buf_size < 12 || memcmp(p->buf, "DSD ", 4) || AV_RL64(p->buf + 4) != 28)
return 0;
return AVPROBE_SCORE_MAX;
}
static const uint64_t dsf_channel_layout[] = {
0,
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_QUAD,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
};
static void read_id3(AVFormatContext *s, uint64_t id3pos)
{
ID3v2ExtraMeta *id3v2_extra_meta;
if (avio_seek(s->pb, id3pos, SEEK_SET) < 0)
return;
ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta, 0);
if (id3v2_extra_meta) {
ff_id3v2_parse_apic(s, id3v2_extra_meta);
ff_id3v2_parse_chapters(s, id3v2_extra_meta);
}
ff_id3v2_free_extra_meta(&id3v2_extra_meta);
}
static int dsf_read_header(AVFormatContext *s)
{
DSFContext *dsf = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st;
uint64_t id3pos;
unsigned int channel_type;
avio_skip(pb, 4);
if (avio_rl64(pb) != 28)
return AVERROR_INVALIDDATA;
/* create primary stream before any id3 coverart streams */
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avio_skip(pb, 8);
id3pos = avio_rl64(pb);
if (pb->seekable & AVIO_SEEKABLE_NORMAL) {
read_id3(s, id3pos);
avio_seek(pb, 28, SEEK_SET);
}
/* fmt chunk */
if (avio_rl32(pb) != MKTAG('f', 'm', 't', ' ') || avio_rl64(pb) != 52)
return AVERROR_INVALIDDATA;
if (avio_rl32(pb) != 1) {
avpriv_request_sample(s, "unknown format version");
return AVERROR_INVALIDDATA;
}
if (avio_rl32(pb)) {
avpriv_request_sample(s, "unknown format id");
return AVERROR_INVALIDDATA;
}
channel_type = avio_rl32(pb);
if (channel_type < FF_ARRAY_ELEMS(dsf_channel_layout))
st->codecpar->channel_layout = dsf_channel_layout[channel_type];
if (!st->codecpar->channel_layout)
avpriv_request_sample(s, "channel type %i", channel_type);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->channels = avio_rl32(pb);
st->codecpar->sample_rate = avio_rl32(pb) / 8;
if (st->codecpar->channels <= 0)
return AVERROR_INVALIDDATA;
switch(avio_rl32(pb)) {
case 1: st->codecpar->codec_id = AV_CODEC_ID_DSD_LSBF_PLANAR; break;
case 8: st->codecpar->codec_id = AV_CODEC_ID_DSD_MSBF_PLANAR; break;
default:
avpriv_request_sample(s, "unknown most significant bit");
return AVERROR_INVALIDDATA;
}
dsf->audio_size = avio_rl64(pb) / 8 * st->codecpar->channels;
st->codecpar->block_align = avio_rl32(pb);
if (st->codecpar->block_align > INT_MAX / st->codecpar->channels || st->codecpar->block_align <= 0) {
avpriv_request_sample(s, "block_align invalid");
return AVERROR_INVALIDDATA;
}
st->codecpar->block_align *= st->codecpar->channels;
st->codecpar->bit_rate = st->codecpar->channels * 8LL * st->codecpar->sample_rate;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
avio_skip(pb, 4);
/* data chunk */
dsf->data_end = avio_tell(pb);
if (avio_rl32(pb) != MKTAG('d', 'a', 't', 'a'))
return AVERROR_INVALIDDATA;
dsf->data_size = avio_rl64(pb) - 12;
dsf->data_end += dsf->data_size + 12;
return 0;
}
static int dsf_read_packet(AVFormatContext *s, AVPacket *pkt)
{
FFFormatContext *const si = ffformatcontext(s);
DSFContext *dsf = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st = s->streams[0];
int64_t pos = avio_tell(pb);
int ret;
if (pos >= dsf->data_end)
return AVERROR_EOF;
if (dsf->data_size > dsf->audio_size) {
int last_packet = pos == (dsf->data_end - st->codecpar->block_align);
if (last_packet) {
int64_t data_pos = pos - si->data_offset;
int64_t packet_size = dsf->audio_size - data_pos;
int64_t skip_size = dsf->data_size - data_pos - packet_size;
uint8_t *dst;
int ch, ret;
if (packet_size <= 0 || skip_size <= 0)
return AVERROR_INVALIDDATA;
if ((ret = av_new_packet(pkt, packet_size)) < 0)
return ret;
dst = pkt->data;
for (ch = 0; ch < st->codecpar->channels; ch++) {
ret = avio_read(pb, dst, packet_size / st->codecpar->channels);
if (ret < packet_size / st->codecpar->channels)
return AVERROR_EOF;
dst += ret;
avio_skip(pb, skip_size / st->codecpar->channels);
}
pkt->pos = pos;
pkt->stream_index = 0;
pkt->pts = (pos - si->data_offset) / st->codecpar->channels;
pkt->duration = packet_size / st->codecpar->channels;
return 0;
}
}
ret = av_get_packet(pb, pkt, FFMIN(dsf->data_end - pos, st->codecpar->block_align));
if (ret < 0)
return ret;
pkt->stream_index = 0;
pkt->pts = (pos - si->data_offset) / st->codecpar->channels;
pkt->duration = st->codecpar->block_align / st->codecpar->channels;
return 0;
}
const AVInputFormat ff_dsf_demuxer = {
.name = "dsf",
.long_name = NULL_IF_CONFIG_SMALL("DSD Stream File (DSF)"),
.priv_data_size = sizeof(DSFContext),
.read_probe = dsf_probe,
.read_header = dsf_read_header,
.read_packet = dsf_read_packet,
.flags = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK,
};