mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
830b5cc35e
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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 = AV_CODEC_ID_AAC;
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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_read_context(&pb, pkt->data, pkt->size);
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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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