ffmpeg/libavformat/aacdec.c

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/*
* 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 "demux.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;
int64_t start_pos = avio_tell(s->pb);
// skip data until an ADTS frame is found
state = avio_r8(s->pb);
while (!avio_feof(s->pb) &&
(avio_tell(s->pb) - start_pos) < 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);
lavf: replace AVStream.codec with AVStream.codecpar Currently, AVStream contains an embedded AVCodecContext instance, which is used by demuxers to export stream parameters to the caller and by muxers to receive stream parameters from the caller. It is also used internally as the codec context that is passed to parsers. In addition, it is also widely used by the callers as the decoding (when demuxer) or encoding (when muxing) context, though this has been officially discouraged since Libav 11. There are multiple important problems with this approach: - the fields in AVCodecContext are in general one of * stream parameters * codec options * codec state However, it's not clear which ones are which. It is consequently unclear which fields are a demuxer allowed to set or a muxer allowed to read. This leads to erratic behaviour depending on whether decoding or encoding is being performed or not (and whether it uses the AVStream embedded codec context). - various synchronization issues arising from the fact that the same context is used by several different APIs (muxers/demuxers, parsers, bitstream filters and encoders/decoders) simultaneously, with there being no clear rules for who can modify what and the different processes being typically delayed with respect to each other. - avformat_find_stream_info() making it necessary to support opening and closing a single codec context multiple times, thus complicating the semantics of freeing various allocated objects in the codec context. Those problems are resolved by replacing the AVStream embedded codec context with a newly added AVCodecParameters instance, which stores only the stream parameters exported by the demuxers or read by the muxers.
2014-06-18 18:42:52 +00:00
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_CODEC_ID_AAC;
ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
ff_id3v1_read(s);
if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&
!av_dict_count(s->metadata)) {
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;
FFIOContext pb;
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;
int ret;
ret = av_append_packet(s->pb, pkt, ff_id3v2_tag_len(pkt->data) - pkt->size);
if (ret < 0)
return ret;
ffio_init_read_context(&pb, pkt->data, pkt->size);
ff_id3v2_read_dict(&pb.pub, &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;
}
const FFInputFormat ff_aac_demuxer = {
.p.name = "aac",
.p.long_name = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
.p.flags = AVFMT_GENERIC_INDEX,
.p.extensions = "aac",
.p.mime_type = "audio/aac,audio/aacp,audio/x-aac",
.read_probe = adts_aac_probe,
.read_header = adts_aac_read_header,
.read_packet = adts_aac_read_packet,
.raw_codec_id = AV_CODEC_ID_AAC,
};