9d76cf0b18
* qatar/master: rtpdec: Templatize the code for different g726 bitrate variants rv40: move loop filter to rv34dsp context lavf: make av_set_pts_info private. rtpdec: Add support for G726 audio rtpdec: Add an init function that can do custom codec context initialization avconv: make copy_tb on by default. matroskadec: don't set codec timebase. rmdec: don't set codec timebase. avconv: compute next_pts from input packet duration when possible. lavf: estimate frame duration from r_frame_rate. avconv: update InputStream.pts in the streamcopy case. Conflicts: avconv.c libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/oss_audio.c libavdevice/v4l.c libavdevice/v4l2.c libavdevice/vfwcap.c libavdevice/x11grab.c libavformat/au.c libavformat/eacdata.c libavformat/flvdec.c libavformat/mpegts.c libavformat/mxfenc.c libavformat/rtpdec_g726.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
153 lines
4.1 KiB
C
153 lines
4.1 KiB
C
/*
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* Sierra SOL demuxer
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* Copyright Konstantin Shishkov
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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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* Based on documents from Game Audio Player and own research
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*/
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "internal.h"
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#include "pcm.h"
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/* if we don't know the size in advance */
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#define AU_UNKNOWN_SIZE ((uint32_t)(~0))
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static int sol_probe(AVProbeData *p)
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{
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/* check file header */
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uint16_t magic = AV_RL32(p->buf);
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if ((magic == 0x0B8D || magic == 0x0C0D || magic == 0x0C8D) &&
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p->buf[2] == 'S' && p->buf[3] == 'O' &&
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p->buf[4] == 'L' && p->buf[5] == 0)
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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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#define SOL_DPCM 1
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#define SOL_16BIT 4
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#define SOL_STEREO 16
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static enum CodecID sol_codec_id(int magic, int type)
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{
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if (magic == 0x0B8D)
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{
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if (type & SOL_DPCM) return CODEC_ID_SOL_DPCM;
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else return CODEC_ID_PCM_U8;
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}
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if (type & SOL_DPCM)
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{
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if (type & SOL_16BIT) return CODEC_ID_SOL_DPCM;
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else if (magic == 0x0C8D) return CODEC_ID_SOL_DPCM;
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else return CODEC_ID_SOL_DPCM;
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}
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if (type & SOL_16BIT) return CODEC_ID_PCM_S16LE;
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return CODEC_ID_PCM_U8;
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}
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static int sol_codec_type(int magic, int type)
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{
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if (magic == 0x0B8D) return 1;//SOL_DPCM_OLD;
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if (type & SOL_DPCM)
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{
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if (type & SOL_16BIT) return 3;//SOL_DPCM_NEW16;
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else if (magic == 0x0C8D) return 1;//SOL_DPCM_OLD;
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else return 2;//SOL_DPCM_NEW8;
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}
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return -1;
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}
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static int sol_channels(int magic, int type)
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{
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if (magic == 0x0B8D || !(type & SOL_STEREO)) return 1;
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return 2;
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}
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static int sol_read_header(AVFormatContext *s,
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AVFormatParameters *ap)
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{
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unsigned int magic,tag;
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AVIOContext *pb = s->pb;
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unsigned int id, channels, rate, type;
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enum CodecID codec;
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AVStream *st;
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/* check ".snd" header */
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magic = avio_rl16(pb);
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tag = avio_rl32(pb);
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if (tag != MKTAG('S', 'O', 'L', 0))
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return -1;
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rate = avio_rl16(pb);
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type = avio_r8(pb);
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avio_skip(pb, 4); /* size */
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if (magic != 0x0B8D)
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avio_r8(pb); /* newer SOLs contain padding byte */
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codec = sol_codec_id(magic, type);
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channels = sol_channels(magic, type);
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if (codec == CODEC_ID_SOL_DPCM)
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id = sol_codec_type(magic, type);
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else id = 0;
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/* now we are ready: build format streams */
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st = avformat_new_stream(s, NULL);
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if (!st)
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return -1;
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_tag = id;
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st->codec->codec_id = codec;
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st->codec->channels = channels;
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st->codec->sample_rate = rate;
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avpriv_set_pts_info(st, 64, 1, rate);
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return 0;
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}
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#define MAX_SIZE 4096
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static int sol_read_packet(AVFormatContext *s,
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AVPacket *pkt)
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{
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int ret;
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if (url_feof(s->pb))
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return AVERROR(EIO);
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ret= av_get_packet(s->pb, pkt, MAX_SIZE);
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if (ret < 0)
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return ret;
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pkt->stream_index = 0;
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/* note: we need to modify the packet size here to handle the last
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packet */
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pkt->size = ret;
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return 0;
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}
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AVInputFormat ff_sol_demuxer = {
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.name = "sol",
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.long_name = NULL_IF_CONFIG_SMALL("Sierra SOL format"),
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.read_probe = sol_probe,
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.read_header = sol_read_header,
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.read_packet = sol_read_packet,
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.read_seek = pcm_read_seek,
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};
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