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  1. /*
  2.  * RTSP muxer
  3.  * Copyright (c) 2010 Martin Storsjo
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. #include "avformat.h"
  23.  
  24. #if HAVE_POLL_H
  25. #include <poll.h>
  26. #endif
  27. #include "network.h"
  28. #include "os_support.h"
  29. #include "rtsp.h"
  30. #include "internal.h"
  31. #include "avio_internal.h"
  32. #include "libavutil/intreadwrite.h"
  33. #include "libavutil/avstring.h"
  34. #include "libavutil/time.h"
  35. #include "url.h"
  36.  
  37. #define SDP_MAX_SIZE 16384
  38.  
  39. static const AVClass rtsp_muxer_class = {
  40.     .class_name = "RTSP muxer",
  41.     .item_name  = av_default_item_name,
  42.     .option     = ff_rtsp_options,
  43.     .version    = LIBAVUTIL_VERSION_INT,
  44. };
  45.  
  46. int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
  47. {
  48.     RTSPState *rt = s->priv_data;
  49.     RTSPMessageHeader reply1, *reply = &reply1;
  50.     int i;
  51.     char *sdp;
  52.     AVFormatContext sdp_ctx, *ctx_array[1];
  53.  
  54.     s->start_time_realtime = av_gettime();
  55.  
  56.     /* Announce the stream */
  57.     sdp = av_mallocz(SDP_MAX_SIZE);
  58.     if (sdp == NULL)
  59.         return AVERROR(ENOMEM);
  60.     /* We create the SDP based on the RTSP AVFormatContext where we
  61.      * aren't allowed to change the filename field. (We create the SDP
  62.      * based on the RTSP context since the contexts for the RTP streams
  63.      * don't exist yet.) In order to specify a custom URL with the actual
  64.      * peer IP instead of the originally specified hostname, we create
  65.      * a temporary copy of the AVFormatContext, where the custom URL is set.
  66.      *
  67.      * FIXME: Create the SDP without copying the AVFormatContext.
  68.      * This either requires setting up the RTP stream AVFormatContexts
  69.      * already here (complicating things immensely) or getting a more
  70.      * flexible SDP creation interface.
  71.      */
  72.     sdp_ctx = *s;
  73.     ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
  74.                 "rtsp", NULL, addr, -1, NULL);
  75.     ctx_array[0] = &sdp_ctx;
  76.     if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
  77.         av_free(sdp);
  78.         return AVERROR_INVALIDDATA;
  79.     }
  80.     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
  81.     ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
  82.                                   "Content-Type: application/sdp\r\n",
  83.                                   reply, NULL, sdp, strlen(sdp));
  84.     av_free(sdp);
  85.     if (reply->status_code != RTSP_STATUS_OK)
  86.         return AVERROR_INVALIDDATA;
  87.  
  88.     /* Set up the RTSPStreams for each AVStream */
  89.     for (i = 0; i < s->nb_streams; i++) {
  90.         RTSPStream *rtsp_st;
  91.  
  92.         rtsp_st = av_mallocz(sizeof(RTSPStream));
  93.         if (!rtsp_st)
  94.             return AVERROR(ENOMEM);
  95.         dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
  96.  
  97.         rtsp_st->stream_index = i;
  98.  
  99.         av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
  100.         /* Note, this must match the relative uri set in the sdp content */
  101.         av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
  102.                     "/streamid=%d", i);
  103.     }
  104.  
  105.     return 0;
  106. }
  107.  
  108. static int rtsp_write_record(AVFormatContext *s)
  109. {
  110.     RTSPState *rt = s->priv_data;
  111.     RTSPMessageHeader reply1, *reply = &reply1;
  112.     char cmd[1024];
  113.  
  114.     snprintf(cmd, sizeof(cmd),
  115.              "Range: npt=0.000-\r\n");
  116.     ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
  117.     if (reply->status_code != RTSP_STATUS_OK)
  118.         return -1;
  119.     rt->state = RTSP_STATE_STREAMING;
  120.     return 0;
  121. }
  122.  
  123. static int rtsp_write_header(AVFormatContext *s)
  124. {
  125.     int ret;
  126.  
  127.     ret = ff_rtsp_connect(s);
  128.     if (ret)
  129.         return ret;
  130.  
  131.     if (rtsp_write_record(s) < 0) {
  132.         ff_rtsp_close_streams(s);
  133.         ff_rtsp_close_connections(s);
  134.         return AVERROR_INVALIDDATA;
  135.     }
  136.     return 0;
  137. }
  138.  
  139. static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
  140. {
  141.     RTSPState *rt = s->priv_data;
  142.     AVFormatContext *rtpctx = rtsp_st->transport_priv;
  143.     uint8_t *buf, *ptr;
  144.     int size;
  145.     uint8_t *interleave_header, *interleaved_packet;
  146.  
  147.     size = avio_close_dyn_buf(rtpctx->pb, &buf);
  148.     rtpctx->pb = NULL;
  149.     ptr = buf;
  150.     while (size > 4) {
  151.         uint32_t packet_len = AV_RB32(ptr);
  152.         int id;
  153.         /* The interleaving header is exactly 4 bytes, which happens to be
  154.          * the same size as the packet length header from
  155.          * ffio_open_dyn_packet_buf. So by writing the interleaving header
  156.          * over these bytes, we get a consecutive interleaved packet
  157.          * that can be written in one call. */
  158.         interleaved_packet = interleave_header = ptr;
  159.         ptr += 4;
  160.         size -= 4;
  161.         if (packet_len > size || packet_len < 2)
  162.             break;
  163.         if (RTP_PT_IS_RTCP(ptr[1]))
  164.             id = rtsp_st->interleaved_max; /* RTCP */
  165.         else
  166.             id = rtsp_st->interleaved_min; /* RTP */
  167.         interleave_header[0] = '$';
  168.         interleave_header[1] = id;
  169.         AV_WB16(interleave_header + 2, packet_len);
  170.         ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
  171.         ptr += packet_len;
  172.         size -= packet_len;
  173.     }
  174.     av_free(buf);
  175.     return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
  176. }
  177.  
  178. static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
  179. {
  180.     RTSPState *rt = s->priv_data;
  181.     RTSPStream *rtsp_st;
  182.     int n;
  183.     struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
  184.     AVFormatContext *rtpctx;
  185.     int ret;
  186.  
  187.     while (1) {
  188.         n = poll(&p, 1, 0);
  189.         if (n <= 0)
  190.             break;
  191.         if (p.revents & POLLIN) {
  192.             RTSPMessageHeader reply;
  193.  
  194.             /* Don't let ff_rtsp_read_reply handle interleaved packets,
  195.              * since it would block and wait for an RTSP reply on the socket
  196.              * (which may not be coming any time soon) if it handles
  197.              * interleaved packets internally. */
  198.             ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
  199.             if (ret < 0)
  200.                 return AVERROR(EPIPE);
  201.             if (ret == 1)
  202.                 ff_rtsp_skip_packet(s);
  203.             /* XXX: parse message */
  204.             if (rt->state != RTSP_STATE_STREAMING)
  205.                 return AVERROR(EPIPE);
  206.         }
  207.     }
  208.  
  209.     if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
  210.         return AVERROR_INVALIDDATA;
  211.     rtsp_st = rt->rtsp_streams[pkt->stream_index];
  212.     rtpctx = rtsp_st->transport_priv;
  213.  
  214.     ret = ff_write_chained(rtpctx, 0, pkt, s);
  215.     /* ff_write_chained does all the RTP packetization. If using TCP as
  216.      * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
  217.      * packets, so we need to send them out on the TCP connection separately.
  218.      */
  219.     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
  220.         ret = tcp_write_packet(s, rtsp_st);
  221.     return ret;
  222. }
  223.  
  224. static int rtsp_write_close(AVFormatContext *s)
  225. {
  226.     RTSPState *rt = s->priv_data;
  227.  
  228.     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
  229.  
  230.     ff_rtsp_close_streams(s);
  231.     ff_rtsp_close_connections(s);
  232.     ff_network_close();
  233.     return 0;
  234. }
  235.  
  236. AVOutputFormat ff_rtsp_muxer = {
  237.     .name              = "rtsp",
  238.     .long_name         = NULL_IF_CONFIG_SMALL("RTSP output"),
  239.     .priv_data_size    = sizeof(RTSPState),
  240.     .audio_codec       = AV_CODEC_ID_AAC,
  241.     .video_codec       = AV_CODEC_ID_MPEG4,
  242.     .write_header      = rtsp_write_header,
  243.     .write_packet      = rtsp_write_packet,
  244.     .write_trailer     = rtsp_write_close,
  245.     .flags             = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
  246.     .priv_class        = &rtsp_muxer_class,
  247. };
  248.