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
28 #include "os_support.h"
31 #include "avio_internal.h"
32 #include "libavutil/intreadwrite.h"
33 #include "libavutil/avstring.h"
36 #define SDP_MAX_SIZE 16384
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,
45 int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
47 RTSPState *rt = s->priv_data;
48 RTSPMessageHeader reply1, *reply = &reply1;
51 AVFormatContext sdp_ctx, *ctx_array[1];
53 s->start_time_realtime = av_gettime();
55 /* Announce the stream */
56 sdp = av_mallocz(SDP_MAX_SIZE);
58 return AVERROR(ENOMEM);
59 /* We create the SDP based on the RTSP AVFormatContext where we
60 * aren't allowed to change the filename field. (We create the SDP
61 * based on the RTSP context since the contexts for the RTP streams
62 * don't exist yet.) In order to specify a custom URL with the actual
63 * peer IP instead of the originally specified hostname, we create
64 * a temporary copy of the AVFormatContext, where the custom URL is set.
66 * FIXME: Create the SDP without copying the AVFormatContext.
67 * This either requires setting up the RTP stream AVFormatContexts
68 * already here (complicating things immensely) or getting a more
69 * flexible SDP creation interface.
72 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
73 "rtsp", NULL, addr, -1, NULL);
74 ctx_array[0] = &sdp_ctx;
75 if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
77 return AVERROR_INVALIDDATA;
79 av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
80 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
81 "Content-Type: application/sdp\r\n",
82 reply, NULL, sdp, strlen(sdp));
84 if (reply->status_code != RTSP_STATUS_OK)
85 return AVERROR_INVALIDDATA;
87 /* Set up the RTSPStreams for each AVStream */
88 for (i = 0; i < s->nb_streams; i++) {
91 rtsp_st = av_mallocz(sizeof(RTSPStream));
93 return AVERROR(ENOMEM);
94 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
96 rtsp_st->stream_index = i;
98 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
99 /* Note, this must match the relative uri set in the sdp content */
100 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
107 static int rtsp_write_record(AVFormatContext *s)
109 RTSPState *rt = s->priv_data;
110 RTSPMessageHeader reply1, *reply = &reply1;
113 snprintf(cmd, sizeof(cmd),
114 "Range: npt=0.000-\r\n");
115 ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
116 if (reply->status_code != RTSP_STATUS_OK)
118 rt->state = RTSP_STATE_STREAMING;
122 static int rtsp_write_header(AVFormatContext *s)
126 ret = ff_rtsp_connect(s);
130 if (rtsp_write_record(s) < 0) {
131 ff_rtsp_close_streams(s);
132 ff_rtsp_close_connections(s);
133 return AVERROR_INVALIDDATA;
138 static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
140 RTSPState *rt = s->priv_data;
141 AVFormatContext *rtpctx = rtsp_st->transport_priv;
144 uint8_t *interleave_header, *interleaved_packet;
146 size = avio_close_dyn_buf(rtpctx->pb, &buf);
149 uint32_t packet_len = AV_RB32(ptr);
151 /* The interleaving header is exactly 4 bytes, which happens to be
152 * the same size as the packet length header from
153 * ffio_open_dyn_packet_buf. So by writing the interleaving header
154 * over these bytes, we get a consecutive interleaved packet
155 * that can be written in one call. */
156 interleaved_packet = interleave_header = ptr;
159 if (packet_len > size || packet_len < 2)
161 if (RTP_PT_IS_RTCP(ptr[1]))
162 id = rtsp_st->interleaved_max; /* RTCP */
164 id = rtsp_st->interleaved_min; /* RTP */
165 interleave_header[0] = '$';
166 interleave_header[1] = id;
167 AV_WB16(interleave_header + 2, packet_len);
168 ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
173 ffio_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
177 static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
179 RTSPState *rt = s->priv_data;
182 struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
183 AVFormatContext *rtpctx;
190 if (p.revents & POLLIN) {
191 RTSPMessageHeader reply;
193 /* Don't let ff_rtsp_read_reply handle interleaved packets,
194 * since it would block and wait for an RTSP reply on the socket
195 * (which may not be coming any time soon) if it handles
196 * interleaved packets internally. */
197 ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
199 return AVERROR(EPIPE);
201 ff_rtsp_skip_packet(s);
202 /* XXX: parse message */
203 if (rt->state != RTSP_STATE_STREAMING)
204 return AVERROR(EPIPE);
208 if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
209 return AVERROR_INVALIDDATA;
210 rtsp_st = rt->rtsp_streams[pkt->stream_index];
211 rtpctx = rtsp_st->transport_priv;
213 ret = ff_write_chained(rtpctx, 0, pkt, s);
214 /* ff_write_chained does all the RTP packetization. If using TCP as
215 * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
216 * packets, so we need to send them out on the TCP connection separately.
218 if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
219 ret = tcp_write_packet(s, rtsp_st);
223 static int rtsp_write_close(AVFormatContext *s)
225 RTSPState *rt = s->priv_data;
227 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
229 ff_rtsp_close_streams(s);
230 ff_rtsp_close_connections(s);
235 AVOutputFormat ff_rtsp_muxer = {
237 .long_name = NULL_IF_CONFIG_SMALL("RTSP output format"),
238 .priv_data_size = sizeof(RTSPState),
239 .audio_codec = CODEC_ID_AAC,
240 .video_codec = CODEC_ID_MPEG4,
241 .write_header = rtsp_write_header,
242 .write_packet = rtsp_write_packet,
243 .write_trailer = rtsp_write_close,
244 .flags = AVFMT_NOFILE | AVFMT_GLOBALHEADER,
245 .priv_class = &rtsp_muxer_class,