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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 
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 
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;
120  return 0;
121 }
122 
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) {
134  return AVERROR_INVALIDDATA;
135  }
136  return 0;
137 }
138 
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);
176 }
177 
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)
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 = ff_rtsp_tcp_write_packet(s, rtsp_st);
221  return ret;
222 }
223 
225 {
226  RTSPState *rt = s->priv_data;
227 
228  // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
229  // Thus call this on all streams before doing the teardown. This is
230  // done within ff_rtsp_undo_setup.
231  ff_rtsp_undo_setup(s, 1);
232 
233  ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
234 
238  return 0;
239 }
240 
242  .name = "rtsp",
243  .long_name = NULL_IF_CONFIG_SMALL("RTSP output"),
244  .priv_data_size = sizeof(RTSPState),
245  .audio_codec = AV_CODEC_ID_AAC,
246  .video_codec = AV_CODEC_ID_MPEG4,
251  .priv_class = &rtsp_muxer_class,
252 };