03b078721c
* commit '97bf7c03b1338a867da52c159a2afecbdedcfa88': doc: git-howto: Leave reviewers time to react before pushing patches Include libavutil/channel_layout.h instead of libavutil/audioconvert.h lavu: rename audioconvert.* to channel_layout.* and deprecate audioconvert.h Conflicts: doc/APIchanges doc/examples/decoding_encoding.c doc/git-howto.texi ffmpeg_filter.c libavcodec/flacdec.c libavcodec/imc.c libavcodec/mpegaudiodec.c libavcodec/utils.c libavfilter/asrc_anullsrc.c libavfilter/audio.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/avfiltergraph.c libavfilter/buffer.c libavutil/Makefile libavutil/audioconvert.h libavutil/channel_layout.c libavutil/version.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
106 lines
3.1 KiB
C
106 lines
3.1 KiB
C
/*
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* MD STUDIO audio demuxer
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*
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* Copyright (c) 2009 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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#include "libavutil/channel_layout.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "pcm.h"
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#define AT1_SU_SIZE 212
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static int aea_read_probe(AVProbeData *p)
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{
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if (p->buf_size <= 2048+212)
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return 0;
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/* Magic is '00 08 00 00' in Little Endian*/
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if (AV_RL32(p->buf)==0x800) {
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int bsm_s, bsm_e, inb_s, inb_e, ch;
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ch = p->buf[264];
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bsm_s = p->buf[2048];
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inb_s = p->buf[2048+1];
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inb_e = p->buf[2048+210];
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bsm_e = p->buf[2048+211];
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if (ch != 1 && ch != 2)
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return 0;
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/* Check so that the redundant bsm bytes and info bytes are valid
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* the block size mode bytes have to be the same
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* the info bytes have to be the same
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*/
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if (bsm_s == bsm_e && inb_s == inb_e)
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return AVPROBE_SCORE_MAX / 4 + 1;
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}
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return 0;
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}
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static int aea_read_header(AVFormatContext *s)
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{
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AVStream *st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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/* Parse the amount of channels and skip to pos 2048(0x800) */
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avio_skip(s->pb, 264);
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st->codec->channels = avio_r8(s->pb);
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avio_skip(s->pb, 1783);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = AV_CODEC_ID_ATRAC1;
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st->codec->sample_rate = 44100;
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st->codec->bit_rate = 292000;
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if (st->codec->channels != 1 && st->codec->channels != 2) {
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av_log(s,AV_LOG_ERROR,"Channels %d not supported!\n",st->codec->channels);
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return -1;
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}
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st->codec->channel_layout = (st->codec->channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
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st->codec->block_align = AT1_SU_SIZE * st->codec->channels;
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return 0;
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}
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static int aea_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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AVInputFormat ff_aea_demuxer = {
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.name = "aea",
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.long_name = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
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.read_probe = aea_read_probe,
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.read_header = aea_read_header,
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.read_packet = aea_read_packet,
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.read_seek = ff_pcm_read_seek,
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.flags = AVFMT_GENERIC_INDEX,
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.extensions = "aea",
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};
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