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rtspenc.c
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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 
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 
56 
57  /* Announce the stream */
58  sdp = av_mallocz(SDP_MAX_SIZE);
59  if (sdp == NULL)
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 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 
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 -1;
121  return 0;
122 }
123 
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) {
135  return AVERROR_INVALIDDATA;
136  }
137  return 0;
138 }
139 
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);
177 }
178 
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)
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 
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 
239  return 0;
240 }
241 
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,
252  .priv_class = &rtsp_muxer_class,
253 };