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a493f80a2c
The new object file is added to the SDP demuxer in the makefile, since it is needed in both the RTSP muxer and demuxer and in the SDP demuxer, due to the current code coupling. Originally committed as revision 25410 to svn://svn.ffmpeg.org/ffmpeg/trunk
83 lines
2.5 KiB
C
83 lines
2.5 KiB
C
/*
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* RTP muxer chaining code
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* Copyright (c) 2010 Martin Storsjo
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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 "avformat.h"
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#include "rtpenc_chain.h"
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AVFormatContext *ff_rtp_chain_mux_open(AVFormatContext *s, AVStream *st,
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URLContext *handle, int packet_size)
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{
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AVFormatContext *rtpctx;
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int ret;
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AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
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if (!rtp_format)
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return NULL;
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/* Allocate an AVFormatContext for each output stream */
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rtpctx = avformat_alloc_context();
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if (!rtpctx)
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return NULL;
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rtpctx->oformat = rtp_format;
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if (!av_new_stream(rtpctx, 0)) {
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av_free(rtpctx);
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return NULL;
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}
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/* Copy the max delay setting; the rtp muxer reads this. */
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rtpctx->max_delay = s->max_delay;
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/* Copy other stream parameters. */
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rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
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/* Set the synchronized start time. */
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rtpctx->start_time_realtime = s->start_time_realtime;
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/* Remove the local codec, link to the original codec
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* context instead, to give the rtp muxer access to
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* codec parameters. */
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av_free(rtpctx->streams[0]->codec);
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rtpctx->streams[0]->codec = st->codec;
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if (handle) {
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url_fdopen(&rtpctx->pb, handle);
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} else
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url_open_dyn_packet_buf(&rtpctx->pb, packet_size);
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ret = av_write_header(rtpctx);
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if (ret) {
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if (handle) {
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url_fclose(rtpctx->pb);
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} else {
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uint8_t *ptr;
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url_close_dyn_buf(rtpctx->pb, &ptr);
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av_free(ptr);
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}
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av_free(rtpctx->streams[0]);
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av_free(rtpctx);
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return NULL;
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}
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/* Copy the RTP AVStream timebase back to the original AVStream */
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st->time_base = rtpctx->streams[0]->time_base;
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return rtpctx;
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}
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