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https://github.com/FFmpeg/FFmpeg.git
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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>
100 lines
2.9 KiB
C
100 lines
2.9 KiB
C
/*
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* RSO demuxer
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* Copyright (c) 2001 Fabrice Bellard (original AU code)
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* Copyright (c) 2010 Rafael Carre
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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/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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#include "riff.h"
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#include "rso.h"
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static int rso_read_header(AVFormatContext *s, AVFormatParameters *ap)
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{
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AVIOContext *pb = s->pb;
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int id, rate, bps;
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unsigned int size;
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enum CodecID codec;
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AVStream *st;
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id = avio_rb16(pb);
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size = avio_rb16(pb);
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rate = avio_rb16(pb);
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avio_rb16(pb); /* play mode ? (0x0000 = don't loop) */
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codec = ff_codec_get_id(ff_codec_rso_tags, id);
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if (codec == CODEC_ID_ADPCM_IMA_WAV) {
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av_log(s, AV_LOG_ERROR, "ADPCM in RSO not implemented\n");
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return AVERROR_PATCHWELCOME;
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}
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bps = av_get_bits_per_sample(codec);
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if (!bps) {
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av_log_ask_for_sample(s, "could not determine bits per sample\n");
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return AVERROR_INVALIDDATA;
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}
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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 AVERROR(ENOMEM);
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st->duration = (size * 8) / bps;
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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 = 1;
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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 BLOCK_SIZE 1024 /* in samples */
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static int rso_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int bps = av_get_bits_per_sample(s->streams[0]->codec->codec_id);
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int ret = av_get_packet(s->pb, pkt, BLOCK_SIZE * bps >> 3);
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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 packet */
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pkt->size = ret;
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return 0;
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}
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AVInputFormat ff_rso_demuxer = {
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.name = "rso",
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.long_name = NULL_IF_CONFIG_SMALL("Lego Mindstorms RSO format"),
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.extensions = "rso",
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.read_header = rso_read_header,
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.read_packet = rso_read_packet,
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.read_seek = pcm_read_seek,
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.codec_tag = (const AVCodecTag* const []){ff_codec_rso_tags, 0},
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};
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