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78accb876c
* qatar/master: ffmpeg: fix some indentation ffmpeg: fix operation with --disable-avfilter simple_idct: remove disabled code motion_est: remove disabled code vc1: remove disabled code fate: separate lavf-mxf_d10 test from lavf-mxf cabac: Move code only used in the cabac test program to cabac.c. ffplay: warn that -pix_fmt is no longer working, suggest alternative ffplay: warn that -s is no longer working, suggest alternative lavf: rename enc variable in utils.c:has_codec_parameters() lavf: use designated initialisers for all (de)muxers. wav: remove a use of deprecated AV_METADATA_ macro rmdec: remove useless ap parameter from rm_read_header_old() dct-test: remove write-only variable des: fix #if conditional around P_shuffle Use LOCAL_ALIGNED in ff_check_alignment() Conflicts: ffmpeg.c libavformat/avidec.c libavformat/matroskaenc.c libavformat/mp3enc.c libavformat/oggenc.c libavformat/utils.c tests/ref/lavf/mxf Merged-by: Michael Niedermayer <michaelni@gmx.at>
206 lines
6.7 KiB
C
206 lines
6.7 KiB
C
/*
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* Sony OpenMG (OMA) demuxer
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*
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* Copyright (c) 2008 Maxim Poliakovski
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* 2008 Benjamin Larsson
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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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/**
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* @file
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* This is a demuxer for Sony OpenMG Music files
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*
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* Known file extensions: ".oma", "aa3"
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* The format of such files consists of three parts:
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* - "ea3" header carrying overall info and metadata. Except for starting with
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* "ea" instead of "ID", it's an ID3v2 header.
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* - "EA3" header is a Sony-specific header containing information about
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* the OpenMG file: codec type (usually ATRAC, can also be MP3 or WMA),
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* codec specific info (packet size, sample rate, channels and so on)
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* and DRM related info (file encryption, content id).
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* - Sound data organized in packets follow the EA3 header
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* (can be encrypted using the Sony DRM!).
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*
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* LIMITATIONS: This version supports only plain (unencrypted) OMA files.
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* If any DRM-protected (encrypted) file is encountered you will get the
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* corresponding error message. Try to remove the encryption using any
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* Sony software (for example SonicStage).
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* CODEC SUPPORT: Only ATRAC3 codec is currently supported!
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*/
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#include "avformat.h"
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#include "libavutil/intreadwrite.h"
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#include "pcm.h"
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#include "riff.h"
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#include "id3v2.h"
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#define EA3_HEADER_SIZE 96
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enum {
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OMA_CODECID_ATRAC3 = 0,
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OMA_CODECID_ATRAC3P = 1,
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OMA_CODECID_MP3 = 3,
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OMA_CODECID_LPCM = 4,
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OMA_CODECID_WMA = 5,
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};
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static const AVCodecTag codec_oma_tags[] = {
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{ CODEC_ID_ATRAC3, OMA_CODECID_ATRAC3 },
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{ CODEC_ID_ATRAC3P, OMA_CODECID_ATRAC3P },
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{ CODEC_ID_MP3, OMA_CODECID_MP3 },
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};
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#define ID3v2_EA3_MAGIC "ea3"
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static int oma_read_header(AVFormatContext *s,
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AVFormatParameters *ap)
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{
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static const uint16_t srate_tab[6] = {320,441,480,882,960,0};
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int ret, framesize, jsflag, samplerate;
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uint32_t codec_params;
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int16_t eid;
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uint8_t buf[EA3_HEADER_SIZE];
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uint8_t *edata;
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AVStream *st;
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ff_id3v2_read(s, ID3v2_EA3_MAGIC);
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ret = avio_read(s->pb, buf, EA3_HEADER_SIZE);
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if (ret < EA3_HEADER_SIZE)
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return -1;
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if (memcmp(buf, ((const uint8_t[]){'E', 'A', '3'}),3) || buf[4] != 0 || buf[5] != EA3_HEADER_SIZE) {
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av_log(s, AV_LOG_ERROR, "Couldn't find the EA3 header !\n");
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return -1;
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}
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eid = AV_RB16(&buf[6]);
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if (eid != -1 && eid != -128) {
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av_log(s, AV_LOG_ERROR, "Encrypted file! Eid: %d\n", eid);
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return -1;
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}
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codec_params = AV_RB24(&buf[33]);
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st = av_new_stream(s, 0);
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if (!st)
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return AVERROR(ENOMEM);
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st->start_time = 0;
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_tag = buf[32];
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st->codec->codec_id = ff_codec_get_id(codec_oma_tags, st->codec->codec_tag);
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switch (buf[32]) {
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case OMA_CODECID_ATRAC3:
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samplerate = srate_tab[(codec_params >> 13) & 7]*100;
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if (samplerate != 44100)
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av_log_ask_for_sample(s, "Unsupported sample rate: %d\n",
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samplerate);
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framesize = (codec_params & 0x3FF) * 8;
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jsflag = (codec_params >> 17) & 1; /* get stereo coding mode, 1 for joint-stereo */
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st->codec->channels = 2;
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st->codec->sample_rate = samplerate;
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st->codec->bit_rate = st->codec->sample_rate * framesize * 8 / 1024;
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/* fake the atrac3 extradata (wav format, makes stream copy to wav work) */
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st->codec->extradata_size = 14;
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edata = av_mallocz(14 + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!edata)
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return AVERROR(ENOMEM);
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st->codec->extradata = edata;
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AV_WL16(&edata[0], 1); // always 1
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AV_WL32(&edata[2], samplerate); // samples rate
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AV_WL16(&edata[6], jsflag); // coding mode
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AV_WL16(&edata[8], jsflag); // coding mode
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AV_WL16(&edata[10], 1); // always 1
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// AV_WL16(&edata[12], 0); // always 0
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av_set_pts_info(st, 64, 1, st->codec->sample_rate);
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break;
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case OMA_CODECID_ATRAC3P:
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st->codec->channels = (codec_params >> 10) & 7;
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framesize = ((codec_params & 0x3FF) * 8) + 8;
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st->codec->sample_rate = srate_tab[(codec_params >> 13) & 7]*100;
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st->codec->bit_rate = st->codec->sample_rate * framesize * 8 / 1024;
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av_set_pts_info(st, 64, 1, st->codec->sample_rate);
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av_log(s, AV_LOG_ERROR, "Unsupported codec ATRAC3+!\n");
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break;
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case OMA_CODECID_MP3:
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st->need_parsing = AVSTREAM_PARSE_FULL;
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framesize = 1024;
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break;
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default:
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av_log(s, AV_LOG_ERROR, "Unsupported codec %d!\n",buf[32]);
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return -1;
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}
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st->codec->block_align = framesize;
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return 0;
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}
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static int oma_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret = av_get_packet(s->pb, pkt, s->streams[0]->codec->block_align);
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pkt->stream_index = 0;
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if (ret <= 0)
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return AVERROR(EIO);
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return ret;
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}
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static int oma_read_probe(AVProbeData *p)
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{
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const uint8_t *buf;
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unsigned tag_len = 0;
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buf = p->buf;
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/* version must be 3 and flags byte zero */
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if (ff_id3v2_match(buf, ID3v2_EA3_MAGIC) && buf[3] == 3 && !buf[4])
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tag_len = ff_id3v2_tag_len(buf);
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// This check cannot overflow as tag_len has at most 28 bits
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if (p->buf_size < tag_len + 5)
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return 0;
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buf += tag_len;
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if (!memcmp(buf, "EA3", 3) && !buf[4] && buf[5] == EA3_HEADER_SIZE)
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return AVPROBE_SCORE_MAX;
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else
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return 0;
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}
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AVInputFormat ff_oma_demuxer = {
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.name = "oma",
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.long_name = NULL_IF_CONFIG_SMALL("Sony OpenMG audio"),
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.read_probe = oma_read_probe,
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.read_header = oma_read_header,
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.read_packet = oma_read_packet,
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.read_seek = pcm_read_seek,
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.flags= AVFMT_GENERIC_INDEX,
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.extensions = "oma,aa3",
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.codec_tag= (const AVCodecTag* const []){codec_oma_tags, 0},
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};
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