FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavformat/rtspenc.c
Date: 2024-11-20 23:03:26
Exec Total Coverage
Lines: 0 103 0.0%
Functions: 0 6 0.0%
Branches: 0 54 0.0%

Line Branch Exec Source
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 };
260