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FFmpeg/libavformat/aacdec.c
James Almer b9c5fdf602 avformat: move AVStream.{parser,need_parsing} to AVStreamInternal
Those are private fields, no reason to have them exposed in a public
header.

Signed-off-by: James Almer <jamrial@gmail.com>
2021-05-07 09:27:21 -03:00

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->internal->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;
}
const 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,
};