PeerStreamer

/*

 * RTSP muxer

 * 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 <sys/time.h>

#if HAVE_SYS_SELECT_H

#include <sys/select.h>

#endif

#include "network.h"

#include "rtsp.h"

#include "internal.h"

#include "libavutil/intreadwrite.h"

#include "libavutil/avstring.h"

#define SDP_MAX_SIZE 16384

int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)

{

    RTSPState *rt = s->priv_data;

    RTSPMessageHeader reply1, *reply = &reply1;

    int i;

    char *sdp;

    AVFormatContext sdp_ctx, *ctx_array[1];

    s->start_time_realtime = av_gettime();

    /* Announce the stream */

    sdp = av_mallocz(SDP_MAX_SIZE);

    if (sdp == NULL)

        return AVERROR(ENOMEM);

    /* We create the SDP based on the RTSP AVFormatContext where we

     * aren't allowed to change the filename field. (We create the SDP

     * based on the RTSP context since the contexts for the RTP streams

     * don't exist yet.) In order to specify a custom URL with the actual

     * peer IP instead of the originally specified hostname, we create

     * a temporary copy of the AVFormatContext, where the custom URL is set.

     *

     * FIXME: Create the SDP without copying the AVFormatContext.

     * This either requires setting up the RTP stream AVFormatContexts

     * already here (complicating things immensely) or getting a more

     * flexible SDP creation interface.

     */

    sdp_ctx = *s;

    ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),

                "rtsp", NULL, addr, -1, NULL);

    ctx_array[0] = &sdp_ctx;

    if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {

        av_free(sdp);

        return AVERROR_INVALIDDATA;

    }

    av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);

    ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,

                                  "Content-Type: application/sdp\r\n",

                                  reply, NULL, sdp, strlen(sdp));

    av_free(sdp);

    if (reply->status_code != RTSP_STATUS_OK)

        return AVERROR_INVALIDDATA;

    /* Set up the RTSPStreams for each AVStream */

    for (i = 0; i < s->nb_streams; i++) {

        RTSPStream *rtsp_st;

        AVStream *st = s->streams[i];

        rtsp_st = av_mallocz(sizeof(RTSPStream));

        if (!rtsp_st)

            return AVERROR(ENOMEM);

        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);

        st->priv_data = rtsp_st;

        rtsp_st->stream_index = i;

        av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));

        /* Note, this must match the relative uri set in the sdp content */

        av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),

                    "/streamid=%d", i);

    }

    return 0;

}

static int rtsp_write_record(AVFormatContext *s)

{

    RTSPState *rt = s->priv_data;

    RTSPMessageHeader reply1, *reply = &reply1;

    char cmd[1024];

    snprintf(cmd, sizeof(cmd),

             "Range: npt=%0.3f-\r\n",

             (double) 0);

    ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);

    if (reply->status_code != RTSP_STATUS_OK)

        return -1;

    rt->state = RTSP_STATE_STREAMING;

    return 0;

}

static int rtsp_write_header(AVFormatContext *s)

{

    int ret;

    ret = ff_rtsp_connect(s);

    if (ret)

        return ret;

    if (rtsp_write_record(s) < 0) {

        ff_rtsp_close_streams(s);

        ff_rtsp_close_connections(s);

        return AVERROR_INVALIDDATA;

    }

    return 0;

}

static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)

{

    RTSPState *rt = s->priv_data;

    AVFormatContext *rtpctx = rtsp_st->transport_priv;

    uint8_t *buf, *ptr;

    int size;

    uint8_t *interleave_header, *interleaved_packet;

    size = url_close_dyn_buf(rtpctx->pb, &buf);

    ptr = buf;

    while (size > 4) {

        uint32_t packet_len = AV_RB32(ptr);

        int id;

        /* The interleaving header is exactly 4 bytes, which happens to be

         * the same size as the packet length header from

         * url_open_dyn_packet_buf. So by writing the interleaving header

         * over these bytes, we get a consecutive interleaved packet

         * that can be written in one call. */

        interleaved_packet = interleave_header = ptr;

        ptr += 4;

        size -= 4;

        if (packet_len > size || packet_len < 2)

            break;

        if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)

            id = rtsp_st->interleaved_max; /* RTCP */

        else

            id = rtsp_st->interleaved_min; /* RTP */

        interleave_header[0] = '$';

        interleave_header[1] = id;

        AV_WB16(interleave_header + 2, packet_len);

        url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);

        ptr += packet_len;

        size -= packet_len;

    }

    av_free(buf);

    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);

    return 0;

}

static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)

{

    RTSPState *rt = s->priv_data;

    RTSPStream *rtsp_st;

    fd_set rfds;

    int n, tcp_fd;

    struct timeval tv;

    AVFormatContext *rtpctx;

    int ret;

    tcp_fd = url_get_file_handle(rt->rtsp_hd);

    while (1) {

        FD_ZERO(&rfds);

        FD_SET(tcp_fd, &rfds);

        tv.tv_sec = 0;

        tv.tv_usec = 0;

        n = select(tcp_fd + 1, &rfds, NULL, NULL, &tv);

        if (n <= 0)

            break;

        if (FD_ISSET(tcp_fd, &rfds)) {

            RTSPMessageHeader reply;

            /* Don't let ff_rtsp_read_reply handle interleaved packets,

             * since it would block and wait for an RTSP reply on the socket

             * (which may not be coming any time soon) if it handles

             * interleaved packets internally. */

            ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);

            if (ret < 0)

                return AVERROR(EPIPE);

            if (ret == 1)

                ff_rtsp_skip_packet(s);

            /* XXX: parse message */

            if (rt->state != RTSP_STATE_STREAMING)

                return AVERROR(EPIPE);

        }

    }

    if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)

        return AVERROR_INVALIDDATA;

    rtsp_st = rt->rtsp_streams[pkt->stream_index];

    rtpctx = rtsp_st->transport_priv;

    ret = ff_write_chained(rtpctx, 0, pkt, s);

    /* ff_write_chained does all the RTP packetization. If using TCP as

     * transport, rtpctx->pb is only a dyn_packet_buf that queues up the

     * packets, so we need to send them out on the TCP connection separately.

     */

    if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)

        ret = tcp_write_packet(s, rtsp_st);

    return ret;

}

static int rtsp_write_close(AVFormatContext *s)

{

    RTSPState *rt = s->priv_data;

    ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);

    ff_rtsp_close_streams(s);

    ff_rtsp_close_connections(s);

    ff_network_close();

    return 0;

}

AVOutputFormat rtsp_muxer = {

    "rtsp",

    NULL_IF_CONFIG_SMALL("RTSP output format"),

    NULL,

    NULL,

    sizeof(RTSPState),

    CODEC_ID_AAC,

    CODEC_ID_MPEG4,

    rtsp_write_header,

    rtsp_write_packet,

    rtsp_write_close,

    .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,

};