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FFmpeg/libavformat/rsodec.c
Michael Niedermayer 9d76cf0b18 Merge remote-tracking branch 'qatar/master'
* 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>
2011-12-01 02:54:24 +01:00

100 lines
2.9 KiB
C

/*
* RSO demuxer
* Copyright (c) 2001 Fabrice Bellard (original AU code)
* Copyright (c) 2010 Rafael Carre
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#include "pcm.h"
#include "riff.h"
#include "rso.h"
static int rso_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
AVIOContext *pb = s->pb;
int id, rate, bps;
unsigned int size;
enum CodecID codec;
AVStream *st;
id = avio_rb16(pb);
size = avio_rb16(pb);
rate = avio_rb16(pb);
avio_rb16(pb); /* play mode ? (0x0000 = don't loop) */
codec = ff_codec_get_id(ff_codec_rso_tags, id);
if (codec == CODEC_ID_ADPCM_IMA_WAV) {
av_log(s, AV_LOG_ERROR, "ADPCM in RSO not implemented\n");
return AVERROR_PATCHWELCOME;
}
bps = av_get_bits_per_sample(codec);
if (!bps) {
av_log_ask_for_sample(s, "could not determine bits per sample\n");
return AVERROR_INVALIDDATA;
}
/* now we are ready: build format streams */
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->duration = (size * 8) / bps;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_tag = id;
st->codec->codec_id = codec;
st->codec->channels = 1;
st->codec->sample_rate = rate;
avpriv_set_pts_info(st, 64, 1, rate);
return 0;
}
#define BLOCK_SIZE 1024 /* in samples */
static int rso_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int bps = av_get_bits_per_sample(s->streams[0]->codec->codec_id);
int ret = av_get_packet(s->pb, pkt, BLOCK_SIZE * bps >> 3);
if (ret < 0)
return ret;
pkt->stream_index = 0;
/* note: we need to modify the packet size here to handle the last packet */
pkt->size = ret;
return 0;
}
AVInputFormat ff_rso_demuxer = {
.name = "rso",
.long_name = NULL_IF_CONFIG_SMALL("Lego Mindstorms RSO format"),
.extensions = "rso",
.read_header = rso_read_header,
.read_packet = rso_read_packet,
.read_seek = pcm_read_seek,
.codec_tag = (const AVCodecTag* const []){ff_codec_rso_tags, 0},
};