FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
rtspenc.c
Go to the documentation of this file.
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 #define SDP_MAX_SIZE 16384
38 
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,
44 };
45 
47 {
48  RTSPState *rt = s->priv_data;
49  RTSPMessageHeader reply1, *reply = &reply1;
50  int i;
51  char *sdp;
52  AVFormatContext sdp_ctx, *ctx_array[1];
53  char url[1024];
54 
57 
58  /* Announce the stream */
59  sdp = av_mallocz(SDP_MAX_SIZE);
60  if (!sdp)
61  return AVERROR(ENOMEM);
62  /* We create the SDP based on the RTSP AVFormatContext where we
63  * aren't allowed to change the filename field. (We create the SDP
64  * based on the RTSP context since the contexts for the RTP streams
65  * don't exist yet.) In order to specify a custom URL with the actual
66  * peer IP instead of the originally specified hostname, we create
67  * a temporary copy of the AVFormatContext, where the custom URL is set.
68  *
69  * FIXME: Create the SDP without copying the AVFormatContext.
70  * This either requires setting up the RTP stream AVFormatContexts
71  * already here (complicating things immensely) or getting a more
72  * flexible SDP creation interface.
73  */
74  sdp_ctx = *s;
75  sdp_ctx.url = url;
76  ff_url_join(url, sizeof(url),
77  "rtsp", NULL, addr, -1, NULL);
78  ctx_array[0] = &sdp_ctx;
79  if (av_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) {
80  av_free(sdp);
81  return AVERROR_INVALIDDATA;
82  }
83  av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
84  ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
85  "Content-Type: application/sdp\r\n",
86  reply, NULL, sdp, strlen(sdp));
87  av_free(sdp);
88  if (reply->status_code != RTSP_STATUS_OK)
90 
91  /* Set up the RTSPStreams for each AVStream */
92  for (i = 0; i < s->nb_streams; i++) {
93  RTSPStream *rtsp_st;
94 
95  rtsp_st = av_mallocz(sizeof(RTSPStream));
96  if (!rtsp_st)
97  return AVERROR(ENOMEM);
98  dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
99 
100  rtsp_st->stream_index = i;
101 
102  av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
103  /* Note, this must match the relative uri set in the sdp content */
104  av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
105  "/streamid=%d", i);
106  }
107 
108  return 0;
109 }
110 
112 {
113  RTSPState *rt = s->priv_data;
114  RTSPMessageHeader reply1, *reply = &reply1;
115  char cmd[1024];
116 
117  snprintf(cmd, sizeof(cmd),
118  "Range: npt=0.000-\r\n");
119  ff_rtsp_send_cmd(s, "RECORD", rt->control_uri, cmd, reply, NULL);
120  if (reply->status_code != RTSP_STATUS_OK)
121  return ff_rtsp_averror(reply->status_code, -1);
123  return 0;
124 }
125 
127 {
128  int ret;
129 
130  ret = ff_rtsp_connect(s);
131  if (ret)
132  return ret;
133 
134  if (rtsp_write_record(s) < 0) {
137  return AVERROR_INVALIDDATA;
138  }
139  return 0;
140 }
141 
143 {
144  RTSPState *rt = s->priv_data;
145  AVFormatContext *rtpctx = rtsp_st->transport_priv;
146  uint8_t *buf, *ptr;
147  int size;
148  uint8_t *interleave_header, *interleaved_packet;
149 
150  size = avio_close_dyn_buf(rtpctx->pb, &buf);
151  rtpctx->pb = NULL;
152  ptr = buf;
153  while (size > 4) {
154  uint32_t packet_len = AV_RB32(ptr);
155  int id;
156  /* The interleaving header is exactly 4 bytes, which happens to be
157  * the same size as the packet length header from
158  * ffio_open_dyn_packet_buf. So by writing the interleaving header
159  * over these bytes, we get a consecutive interleaved packet
160  * that can be written in one call. */
161  interleaved_packet = interleave_header = ptr;
162  ptr += 4;
163  size -= 4;
164  if (packet_len > size || packet_len < 2)
165  break;
166  if (RTP_PT_IS_RTCP(ptr[1]))
167  id = rtsp_st->interleaved_max; /* RTCP */
168  else
169  id = rtsp_st->interleaved_min; /* RTP */
170  interleave_header[0] = '$';
171  interleave_header[1] = id;
172  AV_WB16(interleave_header + 2, packet_len);
173  ffurl_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
174  ptr += packet_len;
175  size -= packet_len;
176  }
177  av_free(buf);
179 }
180 
182 {
183  RTSPState *rt = s->priv_data;
184  RTSPStream *rtsp_st;
185  int n;
186  struct pollfd p = {ffurl_get_file_handle(rt->rtsp_hd), POLLIN, 0};
187  AVFormatContext *rtpctx;
188  int ret;
189 
190  while (1) {
191  n = poll(&p, 1, 0);
192  if (n <= 0)
193  break;
194  if (p.revents & POLLIN) {
195  RTSPMessageHeader reply;
196 
197  /* Don't let ff_rtsp_read_reply handle interleaved packets,
198  * since it would block and wait for an RTSP reply on the socket
199  * (which may not be coming any time soon) if it handles
200  * interleaved packets internally. */
201  ret = ff_rtsp_read_reply(s, &reply, NULL, 1, NULL);
202  if (ret < 0)
203  return AVERROR(EPIPE);
204  if (ret == 1)
206  /* XXX: parse message */
207  if (rt->state != RTSP_STATE_STREAMING)
208  return AVERROR(EPIPE);
209  }
210  }
211 
212  if (pkt->stream_index < 0 || pkt->stream_index >= rt->nb_rtsp_streams)
213  return AVERROR_INVALIDDATA;
214  rtsp_st = rt->rtsp_streams[pkt->stream_index];
215  rtpctx = rtsp_st->transport_priv;
216 
217  ret = ff_write_chained(rtpctx, 0, pkt, s, 0);
218  /* ff_write_chained does all the RTP packetization. If using TCP as
219  * transport, rtpctx->pb is only a dyn_packet_buf that queues up the
220  * packets, so we need to send them out on the TCP connection separately.
221  */
222  if (!ret && rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP)
223  ret = ff_rtsp_tcp_write_packet(s, rtsp_st);
224  return ret;
225 }
226 
228 {
229  RTSPState *rt = s->priv_data;
230 
231  // If we want to send RTCP_BYE packets, these are sent by av_write_trailer.
232  // Thus call this on all streams before doing the teardown. This is
233  // done within ff_rtsp_undo_setup.
234  ff_rtsp_undo_setup(s, 1);
235 
236  ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
237 
241  return 0;
242 }
243 
245  .name = "rtsp",
246  .long_name = NULL_IF_CONFIG_SMALL("RTSP output"),
247  .priv_data_size = sizeof(RTSPState),
248  .audio_codec = AV_CODEC_ID_AAC,
249  .video_codec = AV_CODEC_ID_MPEG4,
254  .priv_class = &rtsp_muxer_class,
255 };
static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int unqueue)
Definition: ffmpeg.c:689
void ff_rtsp_skip_packet(AVFormatContext *s)
Skip a RTP/TCP interleaved packet.
#define NULL
Definition: coverity.c:32
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int64_t start_time_realtime
Start time of the stream in real world time, in microseconds since the Unix epoch (00:00 1st January ...
Definition: avformat.h:1604
int avio_close_dyn_buf(AVIOContext *s, uint8_t **pbuffer)
Return the written size and a pointer to the buffer.
Definition: aviobuf.c:1420
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
char control_uri[1024]
some MS RTSP streams contain a URL in the SDP that we need to use for all subsequent RTSP requests...
Definition: rtsp.h:317
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
Write size bytes from buf to the resource accessed by h.
Definition: avio.c:421
int ff_write_chained(AVFormatContext *dst, int dst_stream, AVPacket *pkt, AVFormatContext *src, int interleave)
Write a packet to another muxer than the one the user originally intended.
Definition: mux.c:1311
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
void ff_network_close(void)
Definition: network.c:115
initialized and sending/receiving data
Definition: rtsp.h:197
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
static AVPacket pkt
This describes the server response to each RTSP command.
Definition: rtsp.h:127
Format I/O context.
Definition: avformat.h:1351
int ff_rtsp_connect(AVFormatContext *s)
Connect to the RTSP server and set up the individual media streams.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
uint8_t
miscellaneous OS support macros and functions.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
URLContext * rtsp_hd_out
Additional output handle, used when input and output are done separately, eg for HTTP tunneling...
Definition: rtsp.h:328
Describe a single stream, as identified by a single m= line block in the SDP content.
Definition: rtsp.h:436
enum RTSPStatusCode status_code
response code from server
Definition: rtsp.h:131
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
int av_sdp_create(AVFormatContext *ac[], int n_files, char *buf, int size)
Generate an SDP for an RTP session.
Definition: sdp.c:841
ptrdiff_t size
Definition: opengl_enc.c:101
#define flags(name, subs,...)
Definition: cbs_h2645.c:263
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr)
Send a command to the RTSP server and wait for the reply.
#define AV_WB16(p, v)
Definition: intreadwrite.h:405
static int ff_rtsp_averror(enum RTSPStatusCode status_code, int default_averror)
Definition: rtspcodes.h:144
#define av_log(a,...)
Private data for the RTSP demuxer.
Definition: rtsp.h:218
AVOutputFormat ff_rtsp_muxer
Definition: rtspenc.c:244
#define AVERROR(e)
Definition: error.h:43
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
char * url
input or output URL.
Definition: avformat.h:1447
const AVOption ff_rtsp_options[]
Definition: rtsp.c:82
int ff_rtsp_tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
Send buffered packets over TCP.
Definition: rtspenc.c:142
URLContext * rtsp_hd
Definition: rtsp.h:220
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:83
struct RTSPStream ** rtsp_streams
streams in this session
Definition: rtsp.h:225
int stream_index
corresponding stream index, if any.
Definition: rtsp.h:441
static int rtsp_write_close(AVFormatContext *s)
Definition: rtspenc.c:227
static int rtsp_write_header(AVFormatContext *s)
Definition: rtspenc.c:126
unsigned int nb_streams
Number of elements in AVFormatContext.streams.
Definition: avformat.h:1407
#define dynarray_add(tab, nb_ptr, elem)
Definition: internal.h:198
int nb_rtsp_streams
number of items in the 'rtsp_streams' variable
Definition: rtsp.h:223
int ffio_open_dyn_packet_buf(AVIOContext **s, int max_packet_size)
Open a write only packetized memory stream with a maximum packet size of 'max_packet_size'.
Definition: aviobuf.c:1396
#define RTSP_TCP_MAX_PACKET_SIZE
Definition: rtsp.h:75
static int write_trailer(AVFormatContext *s1)
Definition: v4l2enc.c:94
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition: avformat.h:468
const char * name
Definition: avformat.h:507
static int rtsp_write_record(AVFormatContext *s)
Definition: rtspenc.c:111
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:684
int ffurl_get_file_handle(URLContext *h)
Return the file descriptor associated with this URL.
Definition: avio.c:626
#define SDP_MAX_SIZE
Definition: rtspenc.c:37
int64_t av_gettime(void)
Get the current time in microseconds.
Definition: time.c:39
int ff_url_join(char *str, int size, const char *proto, const char *authorization, const char *hostname, int port, const char *fmt,...)
Definition: url.c:36
enum RTSPLowerTransport lower_transport
the negotiated network layer transport protocol; e.g.
Definition: rtsp.h:262
void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)
Undo the effect of ff_rtsp_make_setup_request, close the transport_priv and rtp_handle fields...
Definition: rtsp.c:731
AVIOContext * pb
I/O context.
Definition: avformat.h:1393
void * buf
Definition: avisynth_c.h:690
Describe the class of an AVClass context structure.
Definition: log.h:67
size_t av_strlcatf(char *dst, size_t size, const char *fmt,...)
Definition: avstring.c:101
void ff_rtsp_close_streams(AVFormatContext *s)
Close and free all streams within the RTSP (de)muxer.
Definition: rtsp.c:765
#define snprintf
Definition: snprintf.h:34
#define RTP_PT_IS_RTCP(x)
Definition: rtp.h:110
enum RTSPClientState state
indicator of whether we are currently receiving data from the server.
Definition: rtsp.h:231
int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length)
Send a command to the RTSP server and wait for the reply.
Main libavformat public API header.
#define AVFMT_NOFILE
Demuxer will use avio_open, no opened file should be provided by the caller.
Definition: avformat.h:465
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data, const char *method)
Read a RTSP message from the server, or prepare to read data packets if we're reading data interleave...
#define av_free(p)
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers)
Send a command to the RTSP server without waiting for the reply.
TCP; interleaved in RTSP.
Definition: rtsp.h:39
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
Announce the stream to the server and set up the RTSPStream child objects for each media stream...
Definition: rtspenc.c:46
char control_url[1024]
url for this stream (from SDP)
Definition: rtsp.h:447
void * priv_data
Format private data.
Definition: avformat.h:1379
static void write_header(FFV1Context *f)
Definition: ffv1enc.c:337
static int rtsp_write_packet(AVFormatContext *s, AVPacket *pkt)
Definition: rtspenc.c:181
unbuffered private I/O API
int stream_index
Definition: avcodec.h:1433
int interleaved_max
Definition: rtsp.h:445
static const AVClass rtsp_muxer_class
Definition: rtspenc.c:39
enum AVCodecID id
int interleaved_min
interleave IDs; copies of RTSPTransportField->interleaved_min/max for the selected transport...
Definition: rtsp.h:445
This structure stores compressed data.
Definition: avcodec.h:1408
void ff_rtsp_close_connections(AVFormatContext *s)
Close all connection handles within the RTSP (de)muxer.
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
void * transport_priv
RTP/RDT parse context if input, RTP AVFormatContext if output.
Definition: rtsp.h:438