3 * Copyright (c) 2010 Martin Storsjo
5 * This file is part of FFmpeg.
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.
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.
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
29 #include "os_support.h"
32 #include "avio_internal.h"
33 #include "libavutil/intreadwrite.h"
34 #include "libavutil/avstring.h"
36 #include "libavutil/opt.h"
39 #define SDP_MAX_SIZE 16384
41 static const AVOption options[] = {
42 FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
46 static const AVClass rtsp_muxer_class = {
47 .class_name = "RTSP muxer",
48 .item_name = av_default_item_name,
50 .version = LIBAVUTIL_VERSION_INT,
53 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
55 RTSPState *rt = s->priv_data;
56 RTSPMessageHeader reply1, *reply = &reply1;
59 AVFormatContext sdp_ctx, *ctx_array[1];
61 s->start_time_realtime = av_gettime();
63 /* Announce the stream */
64 sdp = av_mallocz(SDP_MAX_SIZE);
66 return AVERROR(ENOMEM);
67 /* We create the SDP based on the RTSP AVFormatContext where we
68 * aren't allowed to change the filename field. (We create the SDP
69 * based on the RTSP context since the contexts for the RTP streams
70 * don't exist yet.) In order to specify a custom URL with the actual
71 * peer IP instead of the originally specified hostname, we create
72 * a temporary copy of the AVFormatContext, where the custom URL is set.
74 * FIXME: Create the SDP without copying the AVFormatContext.
75 * This either requires setting up the RTP stream AVFormatContexts
76 * already here (complicating things immensely) or getting a more
77 * flexible SDP creation interface.
80 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
81 "rtsp", NULL, addr, -1, NULL);
82 ctx_array[0] = &sdp_ctx;
83 if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
85 return AVERROR_INVALIDDATA;
87 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
88 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
89 "Content-Type: application/sdp\r\n",
90 reply, NULL, sdp, strlen(sdp));
92 if (reply->status_code != RTSP_STATUS_OK)
93 return AVERROR_INVALIDDATA;
95 /* Set up the RTSPStreams for each AVStream */
96 for (i = 0; i < s->nb_streams; i++) {
99 rtsp_st = av_mallocz(sizeof(RTSPStream));
101 return AVERROR(ENOMEM);
102 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
104 rtsp_st->stream_index = i;
106 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
107 /* Note, this must match the relative uri set in the sdp content */
108 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
115 static int rtsp_write_record(AVFormatContext *s)
117 RTSPState *rt = s->priv_data;
118 RTSPMessageHeader reply1, *reply = &reply1;
121 snprintf(cmd, sizeof(cmd),
122 "Range: npt=0.000-\r\n");
123 ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
124 if (reply->status_code != RTSP_STATUS_OK)
126 rt->state = RTSP_STATE_STREAMING;
130 static int rtsp_write_header(AVFormatContext *s)
134 ret = ff_rtsp_connect(s);
138 if (rtsp_write_record(s) < 0) {
139 ff_rtsp_close_streams(s);
140 ff_rtsp_close_connections(s);
141 return AVERROR_INVALIDDATA;
146 static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
148 RTSPState *rt = s->priv_data;
149 AVFormatContext *rtpctx = rtsp_st->transport_priv;
152 uint8_t *interleave_header, *interleaved_packet;
154 size = avio_close_dyn_buf(rtpctx->pb, &buf);
157 uint32_t packet_len = AV_RB32(ptr);
159 /* The interleaving header is exactly 4 bytes, which happens to be
160 * the same size as the packet length header from
161 * ffio_open_dyn_packet_buf. So by writing the interleaving header
162 * over these bytes, we get a consecutive interleaved packet
163 * that can be written in one call. */
164 interleaved_packet = interleave_header = ptr;
167 if (packet_len > size || packet_len < 2)
169 if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
170 id = rtsp_st->interleaved_max; /* RTCP */
172 id = rtsp_st->interleaved_min; /* RTP */
173 interleave_header[0] = '$';
174 interleave_header[1] = id;
175 AV_WB16(interleave_header + 2, packet_len);
176 ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
181 ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
185 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
187 RTSPState *rt = s->priv_data;
190 struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
191 AVFormatContext *rtpctx;
198 if (p.revents & POLLIN) {
199 RTSPMessageHeader reply;
201 /* Don't let ff_rtsp_read_reply handle interleaved packets,
202 * since it would block and wait for an RTSP reply on the socket
203 * (which may not be coming any time soon) if it handles
204 * interleaved packets internally. */
205 ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
207 return AVERROR(EPIPE);
209 ff_rtsp_skip_packet(s);
210 /* XXX: parse message */
211 if (rt->state != RTSP_STATE_STREAMING)
212 return AVERROR(EPIPE);
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;
221 ret = ff_write_chained(rtpctx, 0, pkt, s);
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.
226 if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
227 ret = tcp_write_packet(s, rtsp_st);
231 static int rtsp_write_close(AVFormatContext *s)
233 RTSPState *rt = s->priv_data;
235 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
237 ff_rtsp_close_streams(s);
238 ff_rtsp_close_connections(s);
243 AVOutputFormat ff_rtsp_muxer = {
245 NULL_IF_CONFIG_SMALL("RTSP output format"),
254 .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
255 .priv_class = &rtsp_muxer_class,