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FFmpeg/libavformat/aacdec.c
Andreas Rheinhardt 8068f2fcf3 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>
2021-04-18 02:24:44 +02:00

220 lines
6.1 KiB
C

/*
* raw ADTS AAC demuxer
* Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2009 Robert Swain ( rob opendot cl )
*
* 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/avassert.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "avio_internal.h"
#include "internal.h"
#include "id3v1.h"
#include "id3v2.h"
#include "apetag.h"
#define ADTS_HEADER_SIZE 7
static int adts_aac_probe(const AVProbeData *p)
{
int max_frames = 0, first_frames = 0;
int fsize, frames;
const uint8_t *buf0 = p->buf;
const uint8_t *buf2;
const uint8_t *buf;
const uint8_t *end = buf0 + p->buf_size - 7;
buf = buf0;
for (; buf < end; buf = buf2 + 1) {
buf2 = buf;
for (frames = 0; buf2 < end; frames++) {
uint32_t header = AV_RB16(buf2);
if ((header & 0xFFF6) != 0xFFF0) {
if (buf != buf0) {
// Found something that isn't an ADTS header, starting
// from a position other than the start of the buffer.
// Discard the count we've accumulated so far since it
// probably was a false positive.
frames = 0;
}
break;
}
fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;
if (fsize < 7)
break;
fsize = FFMIN(fsize, end - buf2);
buf2 += fsize;
}
max_frames = FFMAX(max_frames, frames);
if (buf == buf0)
first_frames = frames;
}
if (first_frames >= 3)
return AVPROBE_SCORE_EXTENSION + 1;
else if (max_frames > 100)
return AVPROBE_SCORE_EXTENSION;
else if (max_frames >= 3)
return AVPROBE_SCORE_EXTENSION / 2;
else if (first_frames >= 1)
return 1;
else
return 0;
}
static int adts_aac_resync(AVFormatContext *s)
{
uint16_t state;
// skip data until an ADTS frame is found
state = avio_r8(s->pb);
while (!avio_feof(s->pb) && avio_tell(s->pb) < s->probesize) {
state = (state << 8) | avio_r8(s->pb);
if ((state >> 4) != 0xFFF)
continue;
avio_seek(s->pb, -2, SEEK_CUR);
break;
}
if (s->pb->eof_reached)
return AVERROR_EOF;
if ((state >> 4) != 0xFFF)
return AVERROR_INVALIDDATA;
return 0;
}
static int adts_aac_read_header(AVFormatContext *s)
{
AVStream *st;
int ret;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = s->iformat->raw_codec_id;
st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
ff_id3v1_read(s);
if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&
!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {
int64_t cur = avio_tell(s->pb);
ff_ape_parse_tag(s);
avio_seek(s->pb, cur, SEEK_SET);
}
ret = adts_aac_resync(s);
if (ret < 0)
return ret;
// LCM of all possible ADTS sample rates
avpriv_set_pts_info(st, 64, 1, 28224000);
return 0;
}
static int handle_id3(AVFormatContext *s, AVPacket *pkt)
{
AVDictionary *metadata = NULL;
AVIOContext ioctx;
ID3v2ExtraMeta *id3v2_extra_meta;
int ret;
ret = av_append_packet(s->pb, pkt, ff_id3v2_tag_len(pkt->data) - pkt->size);
if (ret < 0) {
return ret;
}
ffio_init_context(&ioctx, pkt->data, pkt->size, 0, NULL, NULL, NULL, NULL);
ff_id3v2_read_dict(&ioctx, &metadata, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta);
if ((ret = ff_id3v2_parse_priv_dict(&metadata, id3v2_extra_meta)) < 0)
goto error;
if (metadata) {
if ((ret = av_dict_copy(&s->metadata, metadata, 0)) < 0)
goto error;
s->event_flags |= AVFMT_EVENT_FLAG_METADATA_UPDATED;
}
error:
av_packet_unref(pkt);
ff_id3v2_free_extra_meta(&id3v2_extra_meta);
av_dict_free(&metadata);
return ret;
}
static int adts_aac_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, fsize;
retry:
ret = av_get_packet(s->pb, pkt, ADTS_HEADER_SIZE);
if (ret < 0)
return ret;
if (ret < ADTS_HEADER_SIZE) {
return AVERROR(EIO);
}
if ((AV_RB16(pkt->data) >> 4) != 0xfff) {
// Parse all the ID3 headers between frames
int append = ID3v2_HEADER_SIZE - ADTS_HEADER_SIZE;
av_assert2(append > 0);
ret = av_append_packet(s->pb, pkt, append);
if (ret != append) {
return AVERROR(EIO);
}
if (!ff_id3v2_match(pkt->data, ID3v2_DEFAULT_MAGIC)) {
av_packet_unref(pkt);
ret = adts_aac_resync(s);
} else
ret = handle_id3(s, pkt);
if (ret < 0)
return ret;
goto retry;
}
fsize = (AV_RB32(pkt->data + 3) >> 13) & 0x1FFF;
if (fsize < ADTS_HEADER_SIZE) {
return AVERROR_INVALIDDATA;
}
ret = av_append_packet(s->pb, pkt, fsize - pkt->size);
return ret;
}
AVInputFormat ff_aac_demuxer = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
.read_probe = adts_aac_probe,
.read_header = adts_aac_read_header,
.read_packet = adts_aac_read_packet,
.flags = AVFMT_GENERIC_INDEX,
.extensions = "aac",
.mime_type = "audio/aac,audio/aacp,audio/x-aac",
.raw_codec_id = AV_CODEC_ID_AAC,
};