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 "mux.h"
  28 #include "network.h"
  29 #include "os_support.h"
  30 #include "rtsp.h"
  31 #include "internal.h"
  32 #include "avio_internal.h"
  33 #include "libavutil/intreadwrite.h"
  34 #include "libavutil/avstring.h"
  35 #include "libavutil/mem.h"
  36 #include "libavutil/time.h"
  37 #include "url.h"
  38
  39
  40 static const AVClass rtsp_muxer_class = {
  41     .class_name = "RTSP muxer",
  42     .item_name  = av_default_item_name,
  43     .option     = ff_rtsp_options,
  44     .version    = LIBAVUTIL_VERSION_INT,
  45 };
  46
  47 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
  48 {
  49     RTSPState *rt = s->priv_data;
  50     RTSPMessageHeader reply1, *reply = &reply1;
  51     int i;
  52     char *sdp;
  53     AVFormatContext sdp_ctx, *ctx_array[1];
  54     char url[MAX_URL_SIZE];
  55
  56     if (s->start_time_realtime == 0  ||  s->start_time_realtime == AV_NOPTS_VALUE)
  57         s->start_time_realtime = av_gettime();
  58
  59     /* Announce the stream */
  60     sdp = av_mallocz(SDP_MAX_SIZE);
  61     if (!sdp)
  62         return AVERROR(ENOMEM);
  63     /* We create the SDP based on the RTSP AVFormatContext where we
  64      * aren't allowed to change the filename field. (We create the SDP
  65      * based on the RTSP context since the contexts for the RTP streams
  66      * don't exist yet.) In order to specify a custom URL with the actual
  67      * peer IP instead of the originally specified hostname, we create
  68      * a temporary copy of the AVFormatContext, where the custom URL is set.
  69      *
  70      * FIXME: Create the SDP without copying the AVFormatContext.
  71      * This either requires setting up the RTP stream AVFormatContexts
  72      * already here (complicating things immensely) or getting a more
  73      * flexible SDP creation interface.
  74      */
  75     sdp_ctx = *s;
  76     sdp_ctx.url = url;
  77     ff_url_join(url, sizeof(url),
  78                 "rtsp", NULL, addr, -1, NULL);
  79     ctx_array[0] = &sdp_ctx;
  80     if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
  81         av_free(sdp);
  82         return AVERROR_INVALIDDATA;
  83     }
  84     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
  85     ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
  86                                   "Content-Type: application/sdp\r\n",
  87                                   reply, NULL, sdp, strlen(sdp));
  88     av_free(sdp);
  89     if (reply->status_code != RTSP_STATUS_OK)
  90         return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
  91
  92     /* Set up the RTSPStreams for each AVStream */
  93     for (i = 0; i < s->nb_streams; i++) {
  94         RTSPStream *rtsp_st;
  95
  96         rtsp_st = av_mallocz(sizeof(RTSPStream));
  97         if (!rtsp_st)
  98             return AVERROR(ENOMEM);
  99         dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
 100
 101         rtsp_st->stream_index = i;
 102
 103         av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
 104         /* Note, this must match the relative uri set in the sdp content */
 105         av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
 106                     "/streamid=%d", i);
 107     }
 108
 109     return 0;
 110 }
 111
 112 static int rtsp_write_record(AVFormatContext *s)
 113 {
 114     RTSPState *rt = s->priv_data;
 115     RTSPMessageHeader reply1, *reply = &reply1;
 116     char cmd[MAX_URL_SIZE];
 117
 118     snprintf(cmd, sizeof(cmd),
 119              "Range: npt=0.000-\r\n");
 120     ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
 121     if (reply->status_code != RTSP_STATUS_OK)
 122         return ff_rtsp_averror(reply->status_code, -1);
 123     rt->state = RTSP_STATE_STREAMING;
 124     return 0;
 125 }
 126
 127 static int rtsp_write_header(AVFormatContext *s)
 128 {
 129     int ret;
 130
 131     ret = ff_rtsp_connect(s);
 132     if (ret)
 133         return ret;
 134
 135     if (rtsp_write_record(s) < 0) {
 136         ff_rtsp_close_streams(s);
 137         ff_rtsp_close_connections(s);
 138         return AVERROR_INVALIDDATA;
 139     }
 140     return 0;
 141 }
 142
 143 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
 144 {
 145     RTSPState *rt = s->priv_data;
 146     AVFormatContext *rtpctx = rtsp_st->transport_priv;
 147     uint8_t *buf, *ptr;
 148     int size;
 149     uint8_t *interleave_header, *interleaved_packet;
 150
 151     size = avio_close_dyn_buf(rtpctx->pb, &buf);
 152     rtpctx->pb = NULL;
 153     ptr = buf;
 154     while (size > 4) {
 155         uint32_t packet_len = AV_RB32(ptr);
 156         int id;
 157         /* The interleaving header is exactly 4 bytes, which happens to be
 158          * the same size as the packet length header from
 159          * ffio_open_dyn_packet_buf. So by writing the interleaving header
 160          * over these bytes, we get a consecutive interleaved packet
 161          * that can be written in one call. */
 162         interleaved_packet = interleave_header = ptr;
 163         ptr += 4;
 164         size -= 4;
 165         if (packet_len > size || packet_len < 2)
 166             break;
 167         if (RTP_PT_IS_RTCP(ptr[1]))
 168             id = rtsp_st->interleaved_max; /* RTCP */
 169         else
 170             id = rtsp_st->interleaved_min; /* RTP */
 171         interleave_header[0] = '$';
 172         interleave_header[1] = id;
 173         AV_WB16(interleave_header + 2, packet_len);
 174         ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
 175         ptr += packet_len;
 176         size -= packet_len;
 177     }
 178     av_free(buf);
 179     return ffio_open_dyn_packet_buf(&rtpctx->pb, rt->pkt_size);
 180 }
 181
 182 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
 183 {
 184     RTSPState *rt = s->priv_data;
 185     RTSPStream *rtsp_st;
 186     int n;
 187     struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
 188     AVFormatContext *rtpctx;
 189     int ret;
 190
 191     while (1) {
 192         n = poll(&p, 1, 0);
 193         if (n <= 0)
 194             break;
 195         if (p.revents & POLLIN) {
 196             RTSPMessageHeader reply;
 197
 198             /* Don't let ff_rtsp_read_reply handle interleaved packets,
 199              * since it would block and wait for an RTSP reply on the socket
 200              * (which may not be coming any time soon) if it handles
 201              * interleaved packets internally. */
 202             ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
 203             if (ret < 0)
 204                 return AVERROR(EPIPE);
 205             if (ret == 1) {
 206                 ret = ff_rtsp_skip_packet(s);
 207                 if (ret < 0)
 208                     return ret;
 209             }
 210             /* XXX: parse message */
 211             if (rt->state != RTSP_STATE_STREAMING)
 212                 return AVERROR(EPIPE);
 213         }
 214     }
 215
 216     if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
 217         return AVERROR_INVALIDDATA;
 218     rtsp_st = rt->rtsp_streams[pkt->stream_index];
 219     rtpctx = rtsp_st->transport_priv;
 220
 221     ret = ff_write_chained(rtpctx, 0, pkt, s, 0);
 222     /* ff_write_chained does all the RTP packetization. If using TCP as
 223      * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
 224      * packets, so we need to send them out on the TCP connection separately.
 225      */
 226     if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
 227         ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
 228     return ret;
 229 }
 230
 231 static int rtsp_write_close(AVFormatContext *s)
 232 {
 233     RTSPState *rt = s->priv_data;
 234
 235     // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
 236     // Thus call this on all streams before doing the teardown. This is
 237     // done within ff_rtsp_undo_setup.
 238     ff_rtsp_undo_setup(s, 1);
 239
 240     ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
 241
 242     ff_rtsp_close_streams(s);
 243     ff_rtsp_close_connections(s);
 244     ff_network_close();
 245     return 0;
 246 }
 247
 248 const FFOutputFormat ff_rtsp_muxer = {
 249     .p.name            = "rtsp",
 250     .p.long_name       = NULL_IF_CONFIG_SMALL("RTSP output"),
 251     .priv_data_size    = sizeof(RTSPState),
 252     .p.audio_codec     = AV_CODEC_ID_AAC,
 253     .p.video_codec     = AV_CODEC_ID_MPEG4,
 254     .write_header      = rtsp_write_header,
 255     .write_packet      = rtsp_write_packet,
 256     .write_trailer     = rtsp_write_close,
 257     .p.flags           = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
 258     .p.priv_class      = &rtsp_muxer_class,
 259 };