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"
34 #include "libavutil/time.h"
37 #define SDP_MAX_SIZE 16384
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
,
46 int ff_rtsp_setup_output_streams(AVFormatContext
*s
, const char *addr
)
48 RTSPState
*rt
= s
->priv_data
;
49 RTSPMessageHeader reply1
, *reply
= &reply1
;
52 AVFormatContext sdp_ctx
, *ctx_array
[1];
54 if (s
->start_time_realtime
== 0 || s
->start_time_realtime
== AV_NOPTS_VALUE
)
55 s
->start_time_realtime
= av_gettime();
57 /* Announce the stream */
58 sdp
= av_mallocz(SDP_MAX_SIZE
);
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.
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.
74 ff_url_join(sdp_ctx
.filename
, sizeof(sdp_ctx
.filename
),
75 "rtsp", NULL
, addr
, -1, NULL
);
76 ctx_array
[0] = &sdp_ctx
;
77 if (av_sdp_create(ctx_array
, 1, sdp
, SDP_MAX_SIZE
)) {
79 return AVERROR_INVALIDDATA
;
81 av_log(s
, AV_LOG_VERBOSE
, "SDP:\n%s\n", sdp
);
82 ff_rtsp_send_cmd_with_content(s
, "ANNOUNCE", rt
->control_uri
,
83 "Content-Type: application/sdp\r\n",
84 reply
, NULL
, sdp
, strlen(sdp
));
86 if (reply
->status_code
!= RTSP_STATUS_OK
)
87 return ff_rtsp_averror(reply
->status_code
, AVERROR_INVALIDDATA
);
89 /* Set up the RTSPStreams for each AVStream */
90 for (i
= 0; i
< s
->nb_streams
; i
++) {
93 rtsp_st
= av_mallocz(sizeof(RTSPStream
));
95 return AVERROR(ENOMEM
);
96 dynarray_add(&rt
->rtsp_streams
, &rt
->nb_rtsp_streams
, rtsp_st
);
98 rtsp_st
->stream_index
= i
;
100 av_strlcpy(rtsp_st
->control_url
, rt
->control_uri
, sizeof(rtsp_st
->control_url
));
101 /* Note, this must match the relative uri set in the sdp content */
102 av_strlcatf(rtsp_st
->control_url
, sizeof(rtsp_st
->control_url
),
109 static int rtsp_write_record(AVFormatContext
*s
)
111 RTSPState
*rt
= s
->priv_data
;
112 RTSPMessageHeader reply1
, *reply
= &reply1
;
115 snprintf(cmd
, sizeof(cmd
),
116 "Range: npt=0.000-\r\n");
117 ff_rtsp_send_cmd(s
, "RECORD", rt
->control_uri
, cmd
, reply
, NULL
);
118 if (reply
->status_code
!= RTSP_STATUS_OK
)
119 return ff_rtsp_averror(reply
->status_code
, -1);
120 rt
->state
= RTSP_STATE_STREAMING
;
124 static int rtsp_write_header(AVFormatContext
*s
)
128 ret
= ff_rtsp_connect(s
);
132 if (rtsp_write_record(s
) < 0) {
133 ff_rtsp_close_streams(s
);
134 ff_rtsp_close_connections(s
);
135 return AVERROR_INVALIDDATA
;
140 int ff_rtsp_tcp_write_packet(AVFormatContext
*s
, RTSPStream
*rtsp_st
)
142 RTSPState
*rt
= s
->priv_data
;
143 AVFormatContext
*rtpctx
= rtsp_st
->transport_priv
;
146 uint8_t *interleave_header
, *interleaved_packet
;
148 size
= avio_close_dyn_buf(rtpctx
->pb
, &buf
);
152 uint32_t packet_len
= AV_RB32(ptr
);
154 /* The interleaving header is exactly 4 bytes, which happens to be
155 * the same size as the packet length header from
156 * ffio_open_dyn_packet_buf. So by writing the interleaving header
157 * over these bytes, we get a consecutive interleaved packet
158 * that can be written in one call. */
159 interleaved_packet
= interleave_header
= ptr
;
162 if (packet_len
> size
|| packet_len
< 2)
164 if (RTP_PT_IS_RTCP(ptr
[1]))
165 id
= rtsp_st
->interleaved_max
; /* RTCP */
167 id
= rtsp_st
->interleaved_min
; /* RTP */
168 interleave_header
[0] = '$';
169 interleave_header
[1] = id
;
170 AV_WB16(interleave_header
+ 2, packet_len
);
171 ffurl_write(rt
->rtsp_hd_out
, interleaved_packet
, 4 + packet_len
);
176 return ffio_open_dyn_packet_buf(&rtpctx
->pb
, RTSP_TCP_MAX_PACKET_SIZE
);
179 static int rtsp_write_packet(AVFormatContext
*s
, AVPacket
*pkt
)
181 RTSPState
*rt
= s
->priv_data
;
184 struct pollfd p
= {ffurl_get_file_handle(rt
->rtsp_hd
), POLLIN
, 0};
185 AVFormatContext
*rtpctx
;
192 if (p
.revents
& POLLIN
) {
193 RTSPMessageHeader reply
;
195 /* Don't let ff_rtsp_read_reply handle interleaved packets,
196 * since it would block and wait for an RTSP reply on the socket
197 * (which may not be coming any time soon) if it handles
198 * interleaved packets internally. */
199 ret
= ff_rtsp_read_reply(s
, &reply
, NULL
, 1, NULL
);
201 return AVERROR(EPIPE
);
203 ff_rtsp_skip_packet(s
);
204 /* XXX: parse message */
205 if (rt
->state
!= RTSP_STATE_STREAMING
)
206 return AVERROR(EPIPE
);
210 if (pkt
->stream_index
< 0 || pkt
->stream_index
>= rt
->nb_rtsp_streams
)
211 return AVERROR_INVALIDDATA
;
212 rtsp_st
= rt
->rtsp_streams
[pkt
->stream_index
];
213 rtpctx
= rtsp_st
->transport_priv
;
215 ret
= ff_write_chained(rtpctx
, 0, pkt
, s
, 0);
216 /* ff_write_chained does all the RTP packetization. If using TCP as
217 * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
218 * packets, so we need to send them out on the TCP connection separately.
220 if (!ret
&& rt
->lower_transport
== RTSP_LOWER_TRANSPORT_TCP
)
221 ret
= ff_rtsp_tcp_write_packet(s
, rtsp_st
);
225 static int rtsp_write_close(AVFormatContext
*s
)
227 RTSPState
*rt
= s
->priv_data
;
229 // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
230 // Thus call this on all streams before doing the teardown. This is
231 // done within ff_rtsp_undo_setup.
232 ff_rtsp_undo_setup(s
, 1);
234 ff_rtsp_send_cmd_async(s
, "TEARDOWN", rt
->control_uri
, NULL
);
236 ff_rtsp_close_streams(s
);
237 ff_rtsp_close_connections(s
);
242 AVOutputFormat ff_rtsp_muxer
= {
244 .long_name
= NULL_IF_CONFIG_SMALL("RTSP output"),
245 .priv_data_size
= sizeof(RTSPState
),
246 .audio_codec
= AV_CODEC_ID_AAC
,
247 .video_codec
= AV_CODEC_ID_MPEG4
,
248 .write_header
= rtsp_write_header
,
249 .write_packet
= rtsp_write_packet
,
250 .write_trailer
= rtsp_write_close
,
251 .flags
= AVFMT_NOFILE
| AVFMT_GLOBALHEADER
,
252 .priv_class
= &rtsp_muxer_class
,