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
38 static const AVClass rtsp_muxer_class = {
39 .class_name = "RTSP muxer",
40 .item_name = av_default_item_name,
41 .option = ff_rtsp_options,
42 .version = LIBAVUTIL_VERSION_INT,
43 };
44
ff_rtsp_setup_output_streams(AVFormatContext * s,const char * addr)45 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
46 {
47 RTSPState *rt = s->priv_data;
48 RTSPMessageHeader reply1, *reply = &reply1;
49 int i;
50 char *sdp;
51 AVFormatContext sdp_ctx, *ctx_array[1];
52 char url[MAX_URL_SIZE];
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 sdp_ctx.url = url;
75 ff_url_join(url, sizeof(url),
76 "rtsp", NULL, addr, -1, NULL);
77 ctx_array[0] = &sdp_ctx;
78 if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
79 av_free(sdp);
80 return AVERROR_INVALIDDATA;
81 }
82 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
83 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
84 "Content-Type: application/sdp\r\n",
85 reply, NULL, sdp, strlen(sdp));
86 av_free(sdp);
87 if (reply->status_code != RTSP_STATUS_OK)
88 return ff_rtsp_averror(reply->status_code, AVERROR_INVALIDDATA);
89
90 /* Set up the RTSPStreams for each AVStream */
91 for (i = 0; i < s->nb_streams; i++) {
92 RTSPStream *rtsp_st;
93
94 rtsp_st = av_mallocz(sizeof(RTSPStream));
95 if (!rtsp_st)
96 return AVERROR(ENOMEM);
97 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
98
99 rtsp_st->stream_index = i;
100
101 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
102 /* Note, this must match the relative uri set in the sdp content */
103 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
104 "/streamid=%d", i);
105 }
106
107 return 0;
108 }
109
rtsp_write_record(AVFormatContext * s)110 static int rtsp_write_record(AVFormatContext *s)
111 {
112 RTSPState *rt = s->priv_data;
113 RTSPMessageHeader reply1, *reply = &reply1;
114 char cmd[MAX_URL_SIZE];
115
116 snprintf(cmd, sizeof(cmd),
117 "Range: npt=0.000-\r\n");
118 ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
119 if (reply->status_code != RTSP_STATUS_OK)
120 return ff_rtsp_averror(reply->status_code, -1);
121 rt->state = RTSP_STATE_STREAMING;
122 return 0;
123 }
124
rtsp_write_header(AVFormatContext * s)125 static int rtsp_write_header(AVFormatContext *s)
126 {
127 int ret;
128
129 ret = ff_rtsp_connect(s);
130 if (ret)
131 return ret;
132
133 if (rtsp_write_record(s) < 0) {
134 ff_rtsp_close_streams(s);
135 ff_rtsp_close_connections(s);
136 return AVERROR_INVALIDDATA;
137 }
138 return 0;
139 }
140
ff_rtsp_tcp_write_packet(AVFormatContext * s,RTSPStream * rtsp_st)141 int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
142 {
143 RTSPState *rt = s->priv_data;
144 AVFormatContext *rtpctx = rtsp_st->transport_priv;
145 uint8_t *buf, *ptr;
146 int size;
147 uint8_t *interleave_header, *interleaved_packet;
148
149 size = avio_close_dyn_buf(rtpctx->pb, &buf);
150 rtpctx->pb = NULL;
151 ptr = buf;
152 while (size > 4) {
153 uint32_t packet_len = AV_RB32(ptr);
154 int id;
155 /* The interleaving header is exactly 4 bytes, which happens to be
156 * the same size as the packet length header from
157 * ffio_open_dyn_packet_buf. So by writing the interleaving header
158 * over these bytes, we get a consecutive interleaved packet
159 * that can be written in one call. */
160 interleaved_packet = interleave_header = ptr;
161 ptr += 4;
162 size -= 4;
163 if (packet_len > size || packet_len < 2)
164 break;
165 if (RTP_PT_IS_RTCP(ptr[1]))
166 id = rtsp_st->interleaved_max; /* RTCP */
167 else
168 id = rtsp_st->interleaved_min; /* RTP */
169 interleave_header[0] = '$';
170 interleave_header[1] = id;
171 AV_WB16(interleave_header + 2, packet_len);
172 ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
173 ptr += packet_len;
174 size -= packet_len;
175 }
176 av_free(buf);
177 return ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
178 }
179
rtsp_write_packet(AVFormatContext * s,AVPacket * pkt)180 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
181 {
182 RTSPState *rt = s->priv_data;
183 RTSPStream *rtsp_st;
184 int n;
185 struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
186 AVFormatContext *rtpctx;
187 int ret;
188
189 while (1) {
190 n = poll(&p, 1, 0);
191 if (n <= 0)
192 break;
193 if (p.revents & POLLIN) {
194 RTSPMessageHeader reply;
195
196 /* Don't let ff_rtsp_read_reply handle interleaved packets,
197 * since it would block and wait for an RTSP reply on the socket
198 * (which may not be coming any time soon) if it handles
199 * interleaved packets internally. */
200 ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
201 if (ret < 0)
202 return AVERROR(EPIPE);
203 if (ret == 1)
204 ff_rtsp_skip_packet(s);
205 /* XXX: parse message */
206 if (rt->state != RTSP_STATE_STREAMING)
207 return AVERROR(EPIPE);
208 }
209 }
210
211 if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
212 return AVERROR_INVALIDDATA;
213 rtsp_st = rt->rtsp_streams[pkt->stream_index];
214 rtpctx = rtsp_st->transport_priv;
215
216 ret = ff_write_chained(rtpctx, 0, pkt, s, 0);
217 /* ff_write_chained does all the RTP packetization. If using TCP as
218 * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
219 * packets, so we need to send them out on the TCP connection separately.
220 */
221 if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
222 ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
223 return ret;
224 }
225
rtsp_write_close(AVFormatContext * s)226 static int rtsp_write_close(AVFormatContext *s)
227 {
228 RTSPState *rt = s->priv_data;
229
230 // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
231 // Thus call this on all streams before doing the teardown. This is
232 // done within ff_rtsp_undo_setup.
233 ff_rtsp_undo_setup(s, 1);
234
235 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
236
237 ff_rtsp_close_streams(s);
238 ff_rtsp_close_connections(s);
239 ff_network_close();
240 return 0;
241 }
242
243 AVOutputFormat ff_rtsp_muxer = {
244 .name = "rtsp",
245 .long_name = NULL_IF_CONFIG_SMALL("RTSP output"),
246 .priv_data_size = sizeof(RTSPState),
247 .audio_codec = AV_CODEC_ID_AAC,
248 .video_codec = AV_CODEC_ID_MPEG4,
249 .write_header = rtsp_write_header,
250 .write_packet = rtsp_write_packet,
251 .write_trailer = rtsp_write_close,
252 .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
253 .priv_class = &rtsp_muxer_class,
254 };
255