PeerStreamer

/*

 * RTP muxer chaining code

 * Copyright (c) 2010 Martin Storsjo

 *

 * 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 "avformat.h"

#include "rtpenc_chain.h"

AVFormatContext *ff_rtp_chain_mux_open(AVFormatContext *s, AVStream *st,

                                       URLContext *handle, int packet_size)

{

    AVFormatContext *rtpctx;

    int ret;

    AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);

    if (!rtp_format)

        return NULL;

    /* Allocate an AVFormatContext for each output stream */

    rtpctx = avformat_alloc_context();

    if (!rtpctx)

        return NULL;

    rtpctx->oformat = rtp_format;

    if (!av_new_stream(rtpctx, 0)) {

        av_free(rtpctx);

        return NULL;

    }

    /* Copy the max delay setting; the rtp muxer reads this. */

    rtpctx->max_delay = s->max_delay;

    /* Copy other stream parameters. */

    rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;

    /* Set the synchronized start time. */

    rtpctx->start_time_realtime = s->start_time_realtime;

    /* Remove the local codec, link to the original codec

     * context instead, to give the rtp muxer access to

     * codec parameters. */

    av_free(rtpctx->streams[0]->codec);

    rtpctx->streams[0]->codec = st->codec;

    if (handle) {

        url_fdopen(&rtpctx->pb, handle);

    } else

        url_open_dyn_packet_buf(&rtpctx->pb, packet_size);

    ret = av_write_header(rtpctx);

    if (ret) {

        if (handle) {

            url_fclose(rtpctx->pb);

        } else {

            uint8_t *ptr;

            url_close_dyn_buf(rtpctx->pb, &ptr);

            av_free(ptr);

        }

        av_free(rtpctx->streams[0]);

        av_free(rtpctx);

        return NULL;

    }

    /* Copy the RTP AVStream timebase back to the original AVStream */

    st->time_base = rtpctx->streams[0]->time_base;

    return rtpctx;

}