libavformat/rtspenc.c

   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     char url[1024];
  54
  55     if (s->start_time_realtime == 0  ||  s->start_time_realtime == AV_NOPTS_VALUE)
  56         s->start_time_realtime = av_gettime();
  57
  58     /* Announce the stream */
  59     sdp = av_mallocz(SDP_MAX_SIZE);
  60     if (!sdp)
  61         return AVERROR(ENOMEM);
  62     /* We create the SDP based on the RTSP AVFormatContext where we
  63      * aren't allowed to change the filename field. (We create the SDP
  64      * based on the RTSP context since the contexts for the RTP streams
  65      * don't exist yet.) In order to specify a custom URL with the actual
  66      * peer IP instead of the originally specified hostname, we create
  67      * a temporary copy of the AVFormatContext, where the custom URL is set.
  68      *
  69      * FIXME: Create the SDP without copying the AVFormatContext.
  70      * This either requires setting up the RTP stream AVFormatContexts
  71      * already here (complicating things immensely) or getting a more
  72      * flexible SDP creation interface.
  73      */
  74     sdp_ctx = *s;
  75     sdp_ctx.url = url;
  76     ff_url_join(url, sizeof(url),
  77                 "rtsp", NULL, addr, -1, NULL);
  78     ctx_array[0] = &sdp_ctx;
  79     if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
  80         av_free(sdp);
  81         return AVERROR_INVALIDDATA;
  82     }
  83     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
  84     ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
  85                                   "Content-Type: application/sdp\r\n",
  86                                   reply, NULL, sdp, strlen(sdp));
  87     av_free(sdp);
  88     if (reply->status_code != RTSP_STATUS_OK)
  89         return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
  90
  91     /* Set up the RTSPStreams for each AVStream */
  92     for (i = 0; i < s->nb_streams; i++) {
  93         RTSPStream *rtsp_st;
  94
  95         rtsp_st = av_mallocz(sizeof(RTSPStream));
  96         if (!rtsp_st)
  97             return AVERROR(ENOMEM);
  98         dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
  99
 100         rtsp_st->stream_index = i;
 101
 102         av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
 103         /* Note, this must match the relative uri set in the sdp content */
 104         av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
 105                     "/streamid=%d", i);
 106     }
 107
 108     return 0;
 109 }
 110
 111 static int rtsp_write_record(AVFormatContext *s)
 112 {
 113     RTSPState *rt = s->priv_data;
 114     RTSPMessageHeader reply1, *reply = &reply1;
 115     char cmd[1024];
 116
 117     snprintf(cmd, sizeof(cmd),
 118              "Range: npt=0.000-\r\n");
 119     ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
 120     if (reply->status_code != RTSP_STATUS_OK)
 121         return ff_rtsp_averror(reply->status_code, -1);
 122     rt->state = RTSP_STATE_STREAMING;
 123     return 0;
 124 }
 125
 126 static int rtsp_write_header(AVFormatContext *s)
 127 {
 128     int ret;
 129
 130     ret = ff_rtsp_connect(s);
 131     if (ret)
 132         return ret;
 133
 134     if (rtsp_write_record(s) < 0) {
 135         ff_rtsp_close_streams(s);
 136         ff_rtsp_close_connections(s);
 137         return AVERROR_INVALIDDATA;
 138     }
 139     return 0;
 140 }
 141
 142 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
 143 {
 144     RTSPState *rt = s->priv_data;
 145     AVFormatContext *rtpctx = rtsp_st->transport_priv;
 146     uint8_t *buf, *ptr;
 147     int size;
 148     uint8_t *interleave_header, *interleaved_packet;
 149
 150     size = avio_close_dyn_buf(rtpctx->pb, &buf);
 151     rtpctx->pb = NULL;
 152     ptr = buf;
 153     while (size > 4) {
 154         uint32_t packet_len = AV_RB32(ptr);
 155         int id;
 156         /* The interleaving header is exactly 4 bytes, which happens to be
 157          * the same size as the packet length header from
 158          * ffio_open_dyn_packet_buf. So by writing the interleaving header
 159          * over these bytes, we get a consecutive interleaved packet
 160          * that can be written in one call. */
 161         interleaved_packet = interleave_header = ptr;
 162         ptr += 4;
 163         size -= 4;
 164         if (packet_len > size || packet_len < 2)
 165             break;
 166         if (RTP_PT_IS_RTCP(ptr[1]))
 167             id = rtsp_st->interleaved_max; /* RTCP */
 168         else
 169             id = rtsp_st->interleaved_min; /* RTP */
 170         interleave_header[0] = '$';
 171         interleave_header[1] = id;
 172         AV_WB16(interleave_header + 2, packet_len);
 173         ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
 174         ptr += packet_len;
 175         size -= packet_len;
 176     }
 177     av_free(buf);
 178     return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
 179 }
 180
 181 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
 182 {
 183     RTSPState *rt = s->priv_data;
 184     RTSPStream *rtsp_st;
 185     int n;
 186     struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
 187     AVFormatContext *rtpctx;
 188     int ret;
 189
 190     while (1) {
 191         n = poll(&p, 1, 0);
 192         if (n <= 0)
 193             break;
 194         if (p.revents & POLLIN) {
 195             RTSPMessageHeader reply;
 196
 197             /* Don't let ff_rtsp_read_reply handle interleaved packets,
 198              * since it would block and wait for an RTSP reply on the socket
 199              * (which may not be coming any time soon) if it handles
 200              * interleaved packets internally. */
 201             ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
 202             if (ret < 0)
 203                 return AVERROR(EPIPE);
 204             if (ret == 1)
 205                 ff_rtsp_skip_packet(s);
 206             /* XXX: parse message */
 207             if (rt->state != RTSP_STATE_STREAMING)
 208                 return AVERROR(EPIPE);
 209         }
 210     }
 211
 212     if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
 213         return AVERROR_INVALIDDATA;
 214     rtsp_st = rt->rtsp_streams[pkt->stream_index];
 215     rtpctx = rtsp_st->transport_priv;
 216
 217     ret = ff_write_chained(rtpctx, 0, pkt, s, 0);
 218     /* ff_write_chained does all the RTP packetization. If using TCP as
 219      * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
 220      * packets, so we need to send them out on the TCP connection separately.
 221      */
 222     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
 223         ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
 224     return ret;
 225 }
 226
 227 static int rtsp_write_close(AVFormatContext *s)
 228 {
 229     RTSPState *rt = s->priv_data;
 230
 231     // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
 232     // Thus call this on all streams before doing the teardown. This is
 233     // done within ff_rtsp_undo_setup.
 234     ff_rtsp_undo_setup(s, 1);
 235
 236     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
 237
 238     ff_rtsp_close_streams(s);
 239     ff_rtsp_close_connections(s);
 240     ff_network_close();
 241     return 0;
 242 }
 243
 244 AVOutputFormat ff_rtsp_muxer = {
 245     .name              = "rtsp",
 246     .long_name         = NULL_IF_CONFIG_SMALL("RTSP output"),
 247     .priv_data_size    = sizeof(RTSPState),
 248     .audio_codec       = AV_CODEC_ID_AAC,
 249     .video_codec       = AV_CODEC_ID_MPEG4,
 250     .write_header      = rtsp_write_header,
 251     .write_packet      = rtsp_write_packet,
 252     .write_trailer     = rtsp_write_close,
 253     .flags             = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
 254     .priv_class        = &rtsp_muxer_class,
 255 };