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c20577806f
Before adts_aac_resync would always bail out after probesize amount of bytes had been progressed from the start of the input. Now just query the current position when entering resync, and at most advance probesize amount of data from that start position. Fixes #9433
222 lines
6.1 KiB
C
222 lines
6.1 KiB
C
/*
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* raw ADTS AAC demuxer
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* Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2009 Robert Swain ( rob opendot cl )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "avio_internal.h"
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#include "internal.h"
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#include "id3v1.h"
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#include "id3v2.h"
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#include "apetag.h"
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#define ADTS_HEADER_SIZE 7
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static int adts_aac_probe(const AVProbeData *p)
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{
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int max_frames = 0, first_frames = 0;
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int fsize, frames;
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const uint8_t *buf0 = p->buf;
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const uint8_t *buf2;
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const uint8_t *buf;
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const uint8_t *end = buf0 + p->buf_size - 7;
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buf = buf0;
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for (; buf < end; buf = buf2 + 1) {
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buf2 = buf;
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for (frames = 0; buf2 < end; frames++) {
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uint32_t header = AV_RB16(buf2);
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if ((header & 0xFFF6) != 0xFFF0) {
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if (buf != buf0) {
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// Found something that isn't an ADTS header, starting
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// from a position other than the start of the buffer.
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// Discard the count we've accumulated so far since it
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// probably was a false positive.
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frames = 0;
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}
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break;
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}
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fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;
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if (fsize < 7)
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break;
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fsize = FFMIN(fsize, end - buf2);
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buf2 += fsize;
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}
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max_frames = FFMAX(max_frames, frames);
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if (buf == buf0)
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first_frames = frames;
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}
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if (first_frames >= 3)
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return AVPROBE_SCORE_EXTENSION + 1;
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else if (max_frames > 100)
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return AVPROBE_SCORE_EXTENSION;
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else if (max_frames >= 3)
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return AVPROBE_SCORE_EXTENSION / 2;
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else if (first_frames >= 1)
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return 1;
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else
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return 0;
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}
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static int adts_aac_resync(AVFormatContext *s)
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{
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uint16_t state;
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int64_t start_pos = avio_tell(s->pb);
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// skip data until an ADTS frame is found
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state = avio_r8(s->pb);
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while (!avio_feof(s->pb) &&
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(avio_tell(s->pb) - start_pos) < s->probesize) {
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state = (state << 8) | avio_r8(s->pb);
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if ((state >> 4) != 0xFFF)
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continue;
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avio_seek(s->pb, -2, SEEK_CUR);
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break;
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}
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if (s->pb->eof_reached)
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return AVERROR_EOF;
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if ((state >> 4) != 0xFFF)
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return AVERROR_INVALIDDATA;
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return 0;
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}
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static int adts_aac_read_header(AVFormatContext *s)
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{
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AVStream *st;
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int ret;
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = s->iformat->raw_codec_id;
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ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
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ff_id3v1_read(s);
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if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&
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!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {
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int64_t cur = avio_tell(s->pb);
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ff_ape_parse_tag(s);
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avio_seek(s->pb, cur, SEEK_SET);
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}
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ret = adts_aac_resync(s);
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if (ret < 0)
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return ret;
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// LCM of all possible ADTS sample rates
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avpriv_set_pts_info(st, 64, 1, 28224000);
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return 0;
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}
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static int handle_id3(AVFormatContext *s, AVPacket *pkt)
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{
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AVDictionary *metadata = NULL;
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FFIOContext pb;
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ID3v2ExtraMeta *id3v2_extra_meta;
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int ret;
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ret = av_append_packet(s->pb, pkt, ff_id3v2_tag_len(pkt->data) - pkt->size);
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if (ret < 0) {
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return ret;
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}
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ffio_init_context(&pb, pkt->data, pkt->size, 0, NULL, NULL, NULL, NULL);
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ff_id3v2_read_dict(&pb.pub, &metadata, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta);
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if ((ret = ff_id3v2_parse_priv_dict(&metadata, id3v2_extra_meta)) < 0)
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goto error;
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if (metadata) {
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if ((ret = av_dict_copy(&s->metadata, metadata, 0)) < 0)
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goto error;
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s->event_flags |= AVFMT_EVENT_FLAG_METADATA_UPDATED;
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}
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error:
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av_packet_unref(pkt);
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ff_id3v2_free_extra_meta(&id3v2_extra_meta);
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av_dict_free(&metadata);
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return ret;
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}
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static int adts_aac_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret, fsize;
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retry:
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ret = av_get_packet(s->pb, pkt, ADTS_HEADER_SIZE);
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if (ret < 0)
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return ret;
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if (ret < ADTS_HEADER_SIZE) {
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return AVERROR(EIO);
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}
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if ((AV_RB16(pkt->data) >> 4) != 0xfff) {
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// Parse all the ID3 headers between frames
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int append = ID3v2_HEADER_SIZE - ADTS_HEADER_SIZE;
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av_assert2(append > 0);
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ret = av_append_packet(s->pb, pkt, append);
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if (ret != append) {
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return AVERROR(EIO);
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}
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if (!ff_id3v2_match(pkt->data, ID3v2_DEFAULT_MAGIC)) {
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av_packet_unref(pkt);
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ret = adts_aac_resync(s);
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} else
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ret = handle_id3(s, pkt);
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if (ret < 0)
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return ret;
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goto retry;
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}
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fsize = (AV_RB32(pkt->data + 3) >> 13) & 0x1FFF;
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if (fsize < ADTS_HEADER_SIZE) {
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return AVERROR_INVALIDDATA;
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}
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ret = av_append_packet(s->pb, pkt, fsize - pkt->size);
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return ret;
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}
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const AVInputFormat ff_aac_demuxer = {
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.name = "aac",
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.long_name = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
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.read_probe = adts_aac_probe,
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.read_header = adts_aac_read_header,
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.read_packet = adts_aac_read_packet,
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.flags = AVFMT_GENERIC_INDEX,
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.extensions = "aac",
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.mime_type = "audio/aac,audio/aacp,audio/x-aac",
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.raw_codec_id = AV_CODEC_ID_AAC,
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};
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