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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.     if (s->start_time_realtime == 0  ||  s->start_time_realtime == AV_NOPTS_VALUE)
  55.         s->start_time_realtime = av_gettime();
  56.  
  57.     /* Announce the stream */
  58.     sdp = av_mallocz(SDP_MAX_SIZE);
  59.     if (!sdp)
  60.         return AVERROR(ENOMEM);
  61.     /* We create the SDP based on the RTSP AVFormatContext where we
  62.      * aren't allowed to change the filename field. (We create the SDP
  63.      * based on the RTSP context since the contexts for the RTP streams
  64.      * don't exist yet.) In order to specify a custom URL with the actual
  65.      * peer IP instead of the originally specified hostname, we create
  66.      * a temporary copy of the AVFormatContext, where the custom URL is set.
  67.      *
  68.      * FIXME: Create the SDP without copying the AVFormatContext.
  69.      * This either requires setting up the RTP stream AVFormatContexts
  70.      * already here (complicating things immensely) or getting a more
  71.      * flexible SDP creation interface.
  72.      */
  73.     sdp_ctx = *s;
  74.     ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
  75.                 "rtsp", NULL, addr, -1, NULL);
  76.     ctx_array[0] = &sdp_ctx;
  77.     if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
  78.         av_free(sdp);
  79.         return AVERROR_INVALIDDATA;
  80.     }
  81.     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
  82.     ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
  83.                                   "Content-Type: application/sdp\r\n",
  84.                                   reply, NULL, sdp, strlen(sdp));
  85.     av_free(sdp);
  86.     if (reply->status_code != RTSP_STATUS_OK)
  87.         return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
  88.  
  89.     /* Set up the RTSPStreams for each AVStream */
  90.     for (i = 0; i < s->nb_streams; i++) {
  91.         RTSPStream *rtsp_st;
  92.  
  93.         rtsp_st = av_mallocz(sizeof(RTSPStream));
  94.         if (!rtsp_st)
  95.             return AVERROR(ENOMEM);
  96.         dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
  97.  
  98.         rtsp_st->stream_index = i;
  99.  
  100.         av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
  101.         /* Note, this must match the relative uri set in the sdp content */
  102.         av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
  103.                     "/streamid=%d", i);
  104.     }
  105.  
  106.     return 0;
  107. }
  108.  
  109. static int rtsp_write_record(AVFormatContext *s)
  110. {
  111.     RTSPState *rt = s->priv_data;
  112.     RTSPMessageHeader reply1, *reply = &reply1;
  113.     char cmd[1024];
  114.  
  115.     snprintf(cmd, sizeof(cmd),
  116.              "Range: npt=0.000-\r\n");
  117.     ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
  118.     if (reply->status_code != RTSP_STATUS_OK)
  119.         return ff_rtsp_averror(reply->status_code, -1);
  120.     rt->state = RTSP_STATE_STREAMING;
  121.     return 0;
  122. }
  123.  
  124. static int rtsp_write_header(AVFormatContext *s)
  125. {
  126.     int ret;
  127.  
  128.     ret = ff_rtsp_connect(s);
  129.     if (ret)
  130.         return ret;
  131.  
  132.     if (rtsp_write_record(s) < 0) {
  133.         ff_rtsp_close_streams(s);
  134.         ff_rtsp_close_connections(s);
  135.         return AVERROR_INVALIDDATA;
  136.     }
  137.     return 0;
  138. }
  139.  
  140. int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
  141. {
  142.     RTSPState *rt = s->priv_data;
  143.     AVFormatContext *rtpctx = rtsp_st->transport_priv;
  144.     uint8_t *buf, *ptr;
  145.     int size;
  146.     uint8_t *interleave_header, *interleaved_packet;
  147.  
  148.     size = avio_close_dyn_buf(rtpctx->pb, &buf);
  149.     rtpctx->pb = NULL;
  150.     ptr = buf;
  151.     while (size > 4) {
  152.         uint32_t packet_len = AV_RB32(ptr);
  153.         int id;
  154.         /* The interleaving header is exactly 4 bytes, which happens to be
  155.          * the same size as the packet length header from
  156.          * ffio_open_dyn_packet_buf. So by writing the interleaving header
  157.          * over these bytes, we get a consecutive interleaved packet
  158.          * that can be written in one call. */
  159.         interleaved_packet = interleave_header = ptr;
  160.         ptr += 4;
  161.         size -= 4;
  162.         if (packet_len > size || packet_len < 2)
  163.             break;
  164.         if (RTP_PT_IS_RTCP(ptr[1]))
  165.             id = rtsp_st->interleaved_max; /* RTCP */
  166.         else
  167.             id = rtsp_st->interleaved_min; /* RTP */
  168.         interleave_header[0] = '$';
  169.         interleave_header[1] = id;
  170.         AV_WB16(interleave_header + 2, packet_len);
  171.         ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
  172.         ptr += packet_len;
  173.         size -= packet_len;
  174.     }
  175.     av_free(buf);
  176.     return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
  177. }
  178.  
  179. static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
  180. {
  181.     RTSPState *rt = s->priv_data;
  182.     RTSPStream *rtsp_st;
  183.     int n;
  184.     struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
  185.     AVFormatContext *rtpctx;
  186.     int ret;
  187.  
  188.     while (1) {
  189.         n = poll(&p, 1, 0);
  190.         if (n <= 0)
  191.             break;
  192.         if (p.revents & POLLIN) {
  193.             RTSPMessageHeader reply;
  194.  
  195.             /* Don't let ff_rtsp_read_reply handle interleaved packets,
  196.              * since it would block and wait for an RTSP reply on the socket
  197.              * (which may not be coming any time soon) if it handles
  198.              * interleaved packets internally. */
  199.             ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
  200.             if (ret < 0)
  201.                 return AVERROR(EPIPE);
  202.             if (ret == 1)
  203.                 ff_rtsp_skip_packet(s);
  204.             /* XXX: parse message */
  205.             if (rt->state != RTSP_STATE_STREAMING)
  206.                 return AVERROR(EPIPE);
  207.         }
  208.     }
  209.  
  210.     if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
  211.         return AVERROR_INVALIDDATA;
  212.     rtsp_st = rt->rtsp_streams[pkt->stream_index];
  213.     rtpctx = rtsp_st->transport_priv;
  214.  
  215.     ret = ff_write_chained(rtpctx, 0, pkt, s, 0);
  216.     /* ff_write_chained does all the RTP packetization. If using TCP as
  217.      * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
  218.      * packets, so we need to send them out on the TCP connection separately.
  219.      */
  220.     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
  221.         ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
  222.     return ret;
  223. }
  224.  
  225. static int rtsp_write_close(AVFormatContext *s)
  226. {
  227.     RTSPState *rt = s->priv_data;
  228.  
  229.     // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
  230.     // Thus call this on all streams before doing the teardown. This is
  231.     // done within ff_rtsp_undo_setup.
  232.     ff_rtsp_undo_setup(s, 1);
  233.  
  234.     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
  235.  
  236.     ff_rtsp_close_streams(s);
  237.     ff_rtsp_close_connections(s);
  238.     ff_network_close();
  239.     return 0;
  240. }
  241.  
  242. AVOutputFormat ff_rtsp_muxer = {
  243.     .name              = "rtsp",
  244.     .long_name         = NULL_IF_CONFIG_SMALL("RTSP output"),
  245.     .priv_data_size    = sizeof(RTSPState),
  246.     .audio_codec       = AV_CODEC_ID_AAC,
  247.     .video_codec       = AV_CODEC_ID_MPEG4,
  248.     .write_header      = rtsp_write_header,
  249.     .write_packet      = rtsp_write_packet,
  250.     .write_trailer     = rtsp_write_close,
  251.     .flags             = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
  252.     .priv_class        = &rtsp_muxer_class,
  253. };
  254.