ffmpeg/libavformat/dsfdec.c

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/*
* 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 "demux.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 AVChannelLayout dsf_channel_layout[] = {
{ .order = AV_CHANNEL_ORDER_UNSPEC },
AV_CHANNEL_LAYOUT_MONO,
AV_CHANNEL_LAYOUT_STEREO,
AV_CHANNEL_LAYOUT_SURROUND,
AV_CHANNEL_LAYOUT_QUAD,
AV_CHANNEL_LAYOUT_4POINT0,
AV_CHANNEL_LAYOUT_5POINT0_BACK,
AV_CHANNEL_LAYOUT_5POINT1_BACK,
};
static void read_id3(AVFormatContext *s, uint64_t id3pos)
{
avformat/id3v2: Don't reverse the order of id3v2 APICs When parsing ID3v2 tags, special (non-text) metadata is not applied directly and unconditionally; instead it is stored in a linked list in which elements are prepended. When traversing the list to add APICs (or private tags) at the end, the order is reversed. The same also happens for chapters and therefore the chapter parsing code already reverses the chapters. This commit changes this: By keeping pointers to both head and tail of the linked list one can preserve the order of the entries and remove the reordering code for chapters. Only the pointer to head will be exported: No current caller uses a nonempty list, so exporting both head and tail is unnecessary. This removes the functionality to combine the lists of special metadata read from different ID3v2 tags, but that doesn't make really much sense anyway (and would be trivial to implement if desired) and allows to remove the now unnecessary initializations performed by the callers. The FATE-reference for the id3v2-priv test had to be updated because the order of the tags read into the dict is reversed; for id3v2-priv-remux only the md5 and not the ffprobe output of the remuxed file changes because the order of the private tags has up until now been reversed twice. The references for the aiff/mp3 cover-art tests needed to be updated, because the order of the attached pics is reversed upon reading. It is still not correct, because the muxers write the pics in the order in which they arrive at the muxer instead of the order given by pkt->stream_index. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-04-12 23:24:03 +00:00
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;
int channels;
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->ch_layout = dsf_channel_layout[channel_type];
if (!st->codecpar->ch_layout.nb_channels)
avpriv_request_sample(s, "channel type %i", channel_type);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
channels = avio_rl32(pb);
if (!st->codecpar->ch_layout.nb_channels) {
st->codecpar->ch_layout.nb_channels = channels;
} else if (channels != st->codecpar->ch_layout.nb_channels) {
av_log(s, AV_LOG_ERROR, "Channel count mismatch\n");
return AVERROR(EINVAL);
}
st->codecpar->sample_rate = avio_rl32(pb) / 8;
if (st->codecpar->ch_layout.nb_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->ch_layout.nb_channels;
st->codecpar->block_align = avio_rl32(pb);
if (st->codecpar->block_align > INT_MAX / st->codecpar->ch_layout.nb_channels ||
st->codecpar->block_align <= 0) {
avpriv_request_sample(s, "block_align invalid");
return AVERROR_INVALIDDATA;
}
st->codecpar->block_align *= st->codecpar->ch_layout.nb_channels;
st->codecpar->bit_rate = st->codecpar->ch_layout.nb_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 channels = st->codecpar->ch_layout.nb_channels;
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->ch_layout.nb_channels; ch++) {
ret = avio_read(pb, dst, packet_size / st->codecpar->ch_layout.nb_channels);
if (ret < packet_size / st->codecpar->ch_layout.nb_channels)
return AVERROR_EOF;
dst += ret;
avio_skip(pb, skip_size / st->codecpar->ch_layout.nb_channels);
}
pkt->pos = pos;
pkt->stream_index = 0;
pkt->pts = (pos - si->data_offset) / channels;
pkt->duration = packet_size / 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) / channels;
pkt->duration = st->codecpar->block_align / channels;
return 0;
}
const FFInputFormat ff_dsf_demuxer = {
.p.name = "dsf",
.p.long_name = NULL_IF_CONFIG_SMALL("DSD Stream File (DSF)"),
.p.flags = AVFMT_GENERIC_INDEX | AVFMT_NO_BYTE_SEEK,
.priv_data_size = sizeof(DSFContext),
.read_probe = dsf_probe,
.read_header = dsf_read_header,
.read_packet = dsf_read_packet,
};