2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
77 * Last reviewed on 2006-08-18 (0.10.5)
86 #endif /* HAVE_UNISTD_H */
92 #include <gst/sdp/gstsdpmessage.h>
93 #include <gst/rtp/gstrtppayloads.h>
95 #include "gst/gst-i18n-plugin.h"
97 #include "gstrtspsrc.h"
99 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
100 #define GST_CAT_DEFAULT (rtspsrc_debug)
102 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
105 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
107 /* templates used internally */
108 static GstStaticPadTemplate anysrctemplate =
109 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
112 GST_STATIC_CAPS_ANY);
114 static GstStaticPadTemplate anysinktemplate =
115 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
118 GST_STATIC_CAPS_ANY);
126 enum _GstRtspSrcRtcpSyncMode
133 enum _GstRtspSrcBufferMode
141 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
143 gst_rtsp_src_buffer_mode_get_type (void)
145 static GType buffer_mode_type = 0;
146 static const GEnumValue buffer_modes[] = {
147 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
148 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
149 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
150 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
154 if (!buffer_mode_type) {
156 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
158 return buffer_mode_type;
161 #define DEFAULT_LOCATION NULL
162 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
163 #define DEFAULT_DEBUG FALSE
164 #define DEFAULT_RETRY 20
165 #define DEFAULT_TIMEOUT 5000000
166 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
167 #define DEFAULT_TCP_TIMEOUT 20000000
168 #define DEFAULT_LATENCY_MS 2000
169 #define DEFAULT_DROP_ON_LATENCY FALSE
170 #define DEFAULT_CONNECTION_SPEED 0
171 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
172 #define DEFAULT_DO_RTCP TRUE
173 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
174 #define DEFAULT_PROXY NULL
175 #define DEFAULT_RTP_BLOCKSIZE 0
176 #define DEFAULT_USER_ID NULL
177 #define DEFAULT_USER_PW NULL
178 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
179 #define DEFAULT_PORT_RANGE NULL
180 #define DEFAULT_SHORT_HEADER FALSE
181 #define DEFAULT_PROBATION 2
193 PROP_DROP_ON_LATENCY,
194 PROP_CONNECTION_SPEED,
197 PROP_DO_RTSP_KEEP_ALIVE,
204 PROP_UDP_BUFFER_SIZE,
210 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
212 gst_rtsp_nat_method_get_type (void)
214 static GType rtsp_nat_method_type = 0;
215 static const GEnumValue rtsp_nat_method[] = {
216 {GST_RTSP_NAT_NONE, "None", "none"},
217 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
221 if (!rtsp_nat_method_type) {
222 rtsp_nat_method_type =
223 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
225 return rtsp_nat_method_type;
228 static void gst_rtspsrc_finalize (GObject * object);
230 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
231 const GValue * value, GParamSpec * pspec);
232 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
233 GValue * value, GParamSpec * pspec);
235 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
236 gpointer iface_data);
238 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
241 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
242 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
244 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
246 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
247 GstStateChange transition);
248 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
249 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
251 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
252 GstRTSPMessage * response);
254 static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask);
255 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
256 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
258 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
259 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
261 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
262 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
263 gboolean only_close);
265 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
266 const gchar * uri, GError ** error);
267 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
269 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
270 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
271 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
272 GstRTSPStream * stream, GstEvent * event);
273 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
275 /* commands we send to out loop to notify it of events */
276 #define CMD_OPEN (1 << 0)
277 #define CMD_PLAY (1 << 1)
278 #define CMD_PAUSE (1 << 2)
279 #define CMD_CLOSE (1 << 3)
280 #define CMD_WAIT (1 << 4)
281 #define CMD_RECONNECT (1 << 5)
282 #define CMD_LOOP (1 << 6)
284 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
286 gchar *__txt = _gst_element_error_printf text; \
287 gst_element_post_message (GST_ELEMENT_CAST (el), \
288 gst_message_new_progress (GST_OBJECT_CAST (el), \
289 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
293 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
294 #define gst_rtspsrc_parent_class parent_class
295 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
296 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
299 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
301 GObjectClass *gobject_class;
302 GstElementClass *gstelement_class;
303 GstBinClass *gstbin_class;
305 gobject_class = (GObjectClass *) klass;
306 gstelement_class = (GstElementClass *) klass;
307 gstbin_class = (GstBinClass *) klass;
309 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
311 gobject_class->set_property = gst_rtspsrc_set_property;
312 gobject_class->get_property = gst_rtspsrc_get_property;
314 gobject_class->finalize = gst_rtspsrc_finalize;
316 g_object_class_install_property (gobject_class, PROP_LOCATION,
317 g_param_spec_string ("location", "RTSP Location",
318 "Location of the RTSP url to read",
319 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
321 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
322 g_param_spec_flags ("protocols", "Protocols",
323 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
324 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
326 g_object_class_install_property (gobject_class, PROP_DEBUG,
327 g_param_spec_boolean ("debug", "Debug",
328 "Dump request and response messages to stdout",
329 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
331 g_object_class_install_property (gobject_class, PROP_RETRY,
332 g_param_spec_uint ("retry", "Retry",
333 "Max number of retries when allocating RTP ports.",
334 0, G_MAXUINT16, DEFAULT_RETRY,
335 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
337 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
338 g_param_spec_uint64 ("timeout", "Timeout",
339 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
340 0, G_MAXUINT64, DEFAULT_TIMEOUT,
341 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
343 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
344 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
345 "Fail after timeout microseconds on TCP connections (0 = disabled)",
346 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
347 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
349 g_object_class_install_property (gobject_class, PROP_LATENCY,
350 g_param_spec_uint ("latency", "Buffer latency in ms",
351 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
352 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
354 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
355 g_param_spec_boolean ("drop-on-latency",
356 "Drop buffers when maximum latency is reached",
357 "Tells the jitterbuffer to never exceed the given latency in size",
358 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
360 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
361 g_param_spec_uint64 ("connection-speed", "Connection Speed",
362 "Network connection speed in kbps (0 = unknown)",
363 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
364 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
366 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
367 g_param_spec_enum ("nat-method", "NAT Method",
368 "Method to use for traversing firewalls and NAT",
369 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
370 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
373 * GstRTSPSrc::do-rtcp
375 * Enable RTCP support. Some old server don't like RTCP and then this property
376 * needs to be set to FALSE.
380 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
381 g_param_spec_boolean ("do-rtcp", "Do RTCP",
382 "Send RTCP packets, disable for old incompatible server.",
383 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
386 * GstRTSPSrc::do-rtsp-keep-alive
388 * Enable RTSP keep laive support. Some old server don't like RTSP
389 * keep alive and then this property needs to be set to FALSE.
393 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
394 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
395 "Send RTSP keep alive packets, disable for old incompatible server.",
396 DEFAULT_DO_RTSP_KEEP_ALIVE,
397 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
402 * Set the proxy parameters. This has to be a string of the format
403 * [http://][user:passwd@]host[:port].
407 g_object_class_install_property (gobject_class, PROP_PROXY,
408 g_param_spec_string ("proxy", "Proxy",
409 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
410 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
413 * GstRTSPSrc::rtp_blocksize
415 * RTP package size to suggest to server.
419 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
420 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
421 "RTP package size to suggest to server (0 = disabled)",
422 0, 65536, DEFAULT_RTP_BLOCKSIZE,
423 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
425 g_object_class_install_property (gobject_class,
427 g_param_spec_string ("user-id", "user-id",
428 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
429 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 g_object_class_install_property (gobject_class, PROP_USER_PW,
431 g_param_spec_string ("user-pw", "user-pw",
432 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
433 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
436 * GstRTSPSrc::buffer-mode:
438 * Control the buffering and timestamping mode used by the jitterbuffer.
442 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
443 g_param_spec_enum ("buffer-mode", "Buffer Mode",
444 "Control the buffering algorithm in use",
445 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
446 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
449 * GstRTSPSrc::port-range:
451 * Configure the client port numbers that can be used to recieve RTP and
456 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
457 g_param_spec_string ("port-range", "Port range",
458 "Client port range that can be used to receive RTP and RTCP data, "
459 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
460 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463 * GstRTSPSrc::udp-buffer-size:
465 * Size of the kernel UDP receive buffer in bytes.
469 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
470 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
471 "Size of the kernel UDP receive buffer in bytes, 0=default",
472 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
473 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
476 * GstRTSPSrc::short-header:
478 * Only send the basic RTSP headers for broken encoders.
482 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
483 g_param_spec_boolean ("short-header", "Short Header",
484 "Only send the basic RTSP headers for broken encoders",
485 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
487 g_object_class_install_property (gobject_class, PROP_PROBATION,
488 g_param_spec_uint ("probation", "Number of probations",
489 "Consecutive packet sequence numbers to accept the source",
490 0, G_MAXUINT, DEFAULT_PROBATION,
491 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
493 gstelement_class->send_event = gst_rtspsrc_send_event;
494 gstelement_class->change_state = gst_rtspsrc_change_state;
496 gst_element_class_add_pad_template (gstelement_class,
497 gst_static_pad_template_get (&rtptemplate));
499 gst_element_class_set_static_metadata (gstelement_class,
500 "RTSP packet receiver", "Source/Network",
501 "Receive data over the network via RTSP (RFC 2326)",
502 "Wim Taymans <wim@fluendo.com>, "
503 "Thijs Vermeir <thijs.vermeir@barco.com>, "
504 "Lutz Mueller <lutz@topfrose.de>");
506 gstbin_class->handle_message = gst_rtspsrc_handle_message;
508 gst_rtsp_ext_list_init ();
513 gst_rtspsrc_init (GstRTSPSrc * src)
515 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
516 src->protocols = DEFAULT_PROTOCOLS;
517 src->debug = DEFAULT_DEBUG;
518 src->retry = DEFAULT_RETRY;
519 src->udp_timeout = DEFAULT_TIMEOUT;
520 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
521 src->latency = DEFAULT_LATENCY_MS;
522 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
523 src->connection_speed = DEFAULT_CONNECTION_SPEED;
524 src->nat_method = DEFAULT_NAT_METHOD;
525 src->do_rtcp = DEFAULT_DO_RTCP;
526 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
527 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
528 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
529 src->user_id = g_strdup (DEFAULT_USER_ID);
530 src->user_pw = g_strdup (DEFAULT_USER_PW);
531 src->buffer_mode = DEFAULT_BUFFER_MODE;
532 src->client_port_range.min = 0;
533 src->client_port_range.max = 0;
534 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
535 src->short_header = DEFAULT_SHORT_HEADER;
536 src->probation = DEFAULT_PROBATION;
538 /* get a list of all extensions */
539 src->extensions = gst_rtsp_ext_list_get ();
541 /* connect to send signal */
542 gst_rtsp_ext_list_connect (src->extensions, "send",
543 (GCallback) gst_rtspsrc_send_cb, src);
545 /* protects the streaming thread in interleaved mode or the polling
546 * thread in UDP mode. */
547 g_rec_mutex_init (&src->stream_rec_lock);
549 /* protects our state changes from multiple invocations */
550 g_rec_mutex_init (&src->state_rec_lock);
552 src->state = GST_RTSP_STATE_INVALID;
554 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
558 gst_rtspsrc_finalize (GObject * object)
562 rtspsrc = GST_RTSPSRC (object);
564 gst_rtsp_ext_list_free (rtspsrc->extensions);
565 g_free (rtspsrc->conninfo.location);
566 gst_rtsp_url_free (rtspsrc->conninfo.url);
567 g_free (rtspsrc->conninfo.url_str);
568 g_free (rtspsrc->user_id);
569 g_free (rtspsrc->user_pw);
572 gst_sdp_message_free (rtspsrc->sdp);
577 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
578 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
580 G_OBJECT_CLASS (parent_class)->finalize (object);
583 /* a proxy string of the format [user:passwd@]host[:port] */
585 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
589 g_free (rtsp->proxy_user);
590 rtsp->proxy_user = NULL;
591 g_free (rtsp->proxy_passwd);
592 rtsp->proxy_passwd = NULL;
593 g_free (rtsp->proxy_host);
594 rtsp->proxy_host = NULL;
595 rtsp->proxy_port = 0;
602 /* we allow http:// in front but ignore it */
603 if (g_str_has_prefix (p, "http://"))
606 at = strchr (p, '@');
608 /* look for user:passwd */
609 col = strchr (proxy, ':');
610 if (col == NULL || col > at)
613 rtsp->proxy_user = g_strndup (p, col - p);
615 rtsp->proxy_passwd = g_strndup (col, at - col);
620 col = strchr (p, ':');
623 /* everything before the colon is the hostname */
624 rtsp->proxy_host = g_strndup (p, col - p);
626 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
628 rtsp->proxy_host = g_strdup (p);
629 rtsp->proxy_port = 8080;
635 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
637 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
638 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
641 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
643 rtspsrc->ptcp_timeout = NULL;
647 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
652 rtspsrc = GST_RTSPSRC (object);
656 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
657 g_value_get_string (value), NULL);
660 rtspsrc->protocols = g_value_get_flags (value);
663 rtspsrc->debug = g_value_get_boolean (value);
666 rtspsrc->retry = g_value_get_uint (value);
669 rtspsrc->udp_timeout = g_value_get_uint64 (value);
671 case PROP_TCP_TIMEOUT:
672 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
675 rtspsrc->latency = g_value_get_uint (value);
677 case PROP_DROP_ON_LATENCY:
678 rtspsrc->drop_on_latency = g_value_get_boolean (value);
680 case PROP_CONNECTION_SPEED:
681 rtspsrc->connection_speed = g_value_get_uint64 (value);
683 case PROP_NAT_METHOD:
684 rtspsrc->nat_method = g_value_get_enum (value);
687 rtspsrc->do_rtcp = g_value_get_boolean (value);
689 case PROP_DO_RTSP_KEEP_ALIVE:
690 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
693 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
695 case PROP_RTP_BLOCKSIZE:
696 rtspsrc->rtp_blocksize = g_value_get_uint (value);
699 if (rtspsrc->user_id)
700 g_free (rtspsrc->user_id);
701 rtspsrc->user_id = g_value_dup_string (value);
704 if (rtspsrc->user_pw)
705 g_free (rtspsrc->user_pw);
706 rtspsrc->user_pw = g_value_dup_string (value);
708 case PROP_BUFFER_MODE:
709 rtspsrc->buffer_mode = g_value_get_enum (value);
711 case PROP_PORT_RANGE:
715 str = g_value_get_string (value);
717 sscanf (str, "%u-%u",
718 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
720 rtspsrc->client_port_range.min = 0;
721 rtspsrc->client_port_range.max = 0;
725 case PROP_UDP_BUFFER_SIZE:
726 rtspsrc->udp_buffer_size = g_value_get_int (value);
728 case PROP_SHORT_HEADER:
729 rtspsrc->short_header = g_value_get_boolean (value);
732 rtspsrc->probation = g_value_get_uint (value);
735 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
741 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
746 rtspsrc = GST_RTSPSRC (object);
750 g_value_set_string (value, rtspsrc->conninfo.location);
753 g_value_set_flags (value, rtspsrc->protocols);
756 g_value_set_boolean (value, rtspsrc->debug);
759 g_value_set_uint (value, rtspsrc->retry);
762 g_value_set_uint64 (value, rtspsrc->udp_timeout);
764 case PROP_TCP_TIMEOUT:
768 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
769 rtspsrc->tcp_timeout.tv_usec;
770 g_value_set_uint64 (value, timeout);
774 g_value_set_uint (value, rtspsrc->latency);
776 case PROP_DROP_ON_LATENCY:
777 g_value_set_boolean (value, rtspsrc->drop_on_latency);
779 case PROP_CONNECTION_SPEED:
780 g_value_set_uint64 (value, rtspsrc->connection_speed);
782 case PROP_NAT_METHOD:
783 g_value_set_enum (value, rtspsrc->nat_method);
786 g_value_set_boolean (value, rtspsrc->do_rtcp);
788 case PROP_DO_RTSP_KEEP_ALIVE:
789 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
795 if (rtspsrc->proxy_host) {
797 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
801 g_value_take_string (value, str);
804 case PROP_RTP_BLOCKSIZE:
805 g_value_set_uint (value, rtspsrc->rtp_blocksize);
808 g_value_set_string (value, rtspsrc->user_id);
811 g_value_set_string (value, rtspsrc->user_pw);
813 case PROP_BUFFER_MODE:
814 g_value_set_enum (value, rtspsrc->buffer_mode);
816 case PROP_PORT_RANGE:
820 if (rtspsrc->client_port_range.min != 0) {
821 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
822 rtspsrc->client_port_range.max);
826 g_value_take_string (value, str);
829 case PROP_UDP_BUFFER_SIZE:
830 g_value_set_int (value, rtspsrc->udp_buffer_size);
832 case PROP_SHORT_HEADER:
833 g_value_set_boolean (value, rtspsrc->short_header);
836 g_value_set_uint (value, rtspsrc->probation);
839 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
845 find_stream_by_id (GstRTSPStream * stream, gint * id)
847 if (stream->id == *id)
854 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
856 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
863 find_stream_by_pt (GstRTSPStream * stream, gint * pt)
865 if (stream->pt == *pt)
872 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
874 GstElement *src = (GstElement *) a;
876 if (stream->udpsrc[0] == src)
878 if (stream->udpsrc[1] == src)
885 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
887 /* check qualified setup_url */
888 if (!strcmp (stream->conninfo.location, (gchar *) a))
890 /* check original control_url */
891 if (!strcmp (stream->control_url, (gchar *) a))
894 /* check if qualified setup_url ends with string */
895 if (g_str_has_suffix (stream->control_url, (gchar *) a))
901 static GstRTSPStream *
902 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
906 /* find and get stream */
907 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
908 return (GstRTSPStream *) lstream->data;
913 static const GstSDPBandwidth *
914 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
915 const GstSDPMedia * media, const gchar * type)
919 /* first look in the media specific section */
920 len = gst_sdp_media_bandwidths_len (media);
921 for (i = 0; i < len; i++) {
922 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
924 if (strcmp (bw->bwtype, type) == 0)
927 /* then look in the message specific section */
928 len = gst_sdp_message_bandwidths_len (sdp);
929 for (i = 0; i < len; i++) {
930 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
932 if (strcmp (bw->bwtype, type) == 0)
939 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
940 const GstSDPMedia * media, GstRTSPStream * stream)
942 const GstSDPBandwidth *bw;
944 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
945 stream->as_bandwidth = bw->bandwidth;
947 stream->as_bandwidth = -1;
949 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
950 stream->rr_bandwidth = bw->bandwidth;
952 stream->rr_bandwidth = -1;
954 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
955 stream->rs_bandwidth = bw->bandwidth;
957 stream->rs_bandwidth = -1;
961 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
962 const GstSDPConnection * conn)
964 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
967 if (conn->addrtype == NULL)
971 if (strcmp (conn->addrtype, "IP4") == 0)
972 stream->is_ipv6 = FALSE;
973 else if (strcmp (conn->addrtype, "IP6") == 0)
974 stream->is_ipv6 = TRUE;
979 g_free (stream->destination);
980 stream->destination = g_strdup (conn->address);
982 /* check for multicast */
983 stream->is_multicast =
984 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
986 stream->ttl = conn->ttl;
989 /* Go over the connections for a stream.
990 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
992 * - If we are dealing with a localhost address, we disable multicast
995 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
996 const GstSDPMedia * media, GstRTSPStream * stream)
998 const GstSDPConnection *conn;
1001 /* first look in the media specific section */
1002 len = gst_sdp_media_connections_len (media);
1003 for (i = 0; i < len; i++) {
1004 conn = gst_sdp_media_get_connection (media, i);
1006 gst_rtspsrc_do_stream_connection (src, stream, conn);
1008 /* then look in the message specific section */
1009 if ((conn = gst_sdp_message_get_connection (sdp))) {
1010 gst_rtspsrc_do_stream_connection (src, stream, conn);
1014 static GstRTSPStream *
1015 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1017 GstRTSPStream *stream;
1018 const gchar *control_url;
1019 const gchar *payload;
1020 const GstSDPMedia *media;
1022 /* get media, should not return NULL */
1023 media = gst_sdp_message_get_media (sdp, idx);
1027 stream = g_new0 (GstRTSPStream, 1);
1028 stream->parent = src;
1029 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1031 stream->last_ret = GST_FLOW_NOT_LINKED;
1032 stream->added = FALSE;
1033 stream->disabled = FALSE;
1034 stream->id = src->numstreams++;
1035 stream->eos = FALSE;
1036 stream->discont = TRUE;
1037 stream->seqbase = -1;
1038 stream->timebase = -1;
1040 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1041 * session manager to scale RTCP. */
1042 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1044 /* collect connection info */
1045 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1047 /* we must have a payload. No payload means we cannot create caps */
1048 /* FIXME, handle multiple formats. The problem here is that we just want to
1049 * take the first available format that we can handle but in order to do that
1050 * we need to scan for depayloader plugins. Scanning for payloader plugins is
1051 * also suboptimal because the user maybe just wants to save the raw stream
1052 * and then we don't care. */
1053 if ((payload = gst_sdp_media_get_format (media, 0))) {
1054 stream->pt = atoi (payload);
1056 stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
1058 GST_DEBUG ("mapping sdp session level attributes to caps");
1059 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
1060 GST_DEBUG ("mapping sdp media level attributes to caps");
1061 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
1063 if (stream->pt >= 96) {
1064 /* If we have a dynamic payload type, see if we have a stream with the
1065 * same payload number. If there is one, they are part of the same
1066 * container and we only need to add one pad. */
1067 if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
1068 stream->container = TRUE;
1069 GST_DEBUG ("found another stream with pt %d, marking as container",
1074 /* collect port number */
1075 stream->port = gst_sdp_media_get_port (media);
1077 /* get control url to construct the setup url. The setup url is used to
1078 * configure the transport of the stream and is used to identity the stream in
1079 * the RTP-Info header field returned from PLAY. */
1080 control_url = gst_sdp_media_get_attribute_val (media, "control");
1081 if (control_url == NULL)
1082 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1084 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1085 GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
1086 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1087 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1088 GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
1089 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1091 if (control_url != NULL) {
1092 stream->control_url = g_strdup (control_url);
1093 /* Build a fully qualified url using the content_base if any or by prefixing
1094 * the original request.
1095 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1096 * likely build a URL that the server will fail to understand, this is ok,
1097 * we will fail then. */
1098 if (g_str_has_prefix (control_url, "rtsp://"))
1099 stream->conninfo.location = g_strdup (control_url);
1104 if (g_strcmp0 (control_url, "*") == 0)
1108 base = src->control;
1109 else if (src->content_base)
1110 base = src->content_base;
1111 else if (src->conninfo.url_str)
1112 base = src->conninfo.url_str;
1116 /* check if the base ends or control starts with / */
1117 has_slash = g_str_has_prefix (control_url, "/");
1118 has_slash = has_slash || g_str_has_suffix (base, "/");
1120 /* concatenate the two strings, insert / when not present */
1121 stream->conninfo.location =
1122 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1125 GST_DEBUG_OBJECT (src, " setup: %s",
1126 GST_STR_NULL (stream->conninfo.location));
1128 /* we keep track of all streams */
1129 src->streams = g_list_append (src->streams, stream);
1137 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1141 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1144 gst_caps_unref (stream->caps);
1146 g_free (stream->destination);
1147 g_free (stream->control_url);
1148 g_free (stream->conninfo.location);
1150 for (i = 0; i < 2; i++) {
1151 if (stream->udpsrc[i]) {
1152 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1153 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1154 gst_object_unref (stream->udpsrc[i]);
1155 stream->udpsrc[i] = NULL;
1157 if (stream->channelpad[i]) {
1158 gst_object_unref (stream->channelpad[i]);
1159 stream->channelpad[i] = NULL;
1161 if (stream->udpsink[i]) {
1162 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1163 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1164 gst_object_unref (stream->udpsink[i]);
1165 stream->udpsink[i] = NULL;
1168 if (stream->fakesrc) {
1169 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1170 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1171 gst_object_unref (stream->fakesrc);
1172 stream->fakesrc = NULL;
1174 if (stream->srcpad) {
1175 gst_pad_set_active (stream->srcpad, FALSE);
1176 if (stream->added) {
1177 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1178 stream->added = FALSE;
1180 stream->srcpad = NULL;
1182 if (stream->rtcppad) {
1183 gst_object_unref (stream->rtcppad);
1184 stream->rtcppad = NULL;
1186 if (stream->session) {
1187 g_object_unref (stream->session);
1188 stream->session = NULL;
1194 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1198 GST_DEBUG_OBJECT (src, "cleanup");
1200 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1201 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1203 gst_rtspsrc_stream_free (src, stream);
1205 g_list_free (src->streams);
1206 src->streams = NULL;
1208 if (src->manager_sig_id) {
1209 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1210 src->manager_sig_id = 0;
1212 gst_element_set_state (src->manager, GST_STATE_NULL);
1213 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1214 src->manager = NULL;
1216 src->numstreams = 0;
1218 gst_structure_free (src->props);
1221 g_free (src->content_base);
1222 src->content_base = NULL;
1224 g_free (src->control);
1225 src->control = NULL;
1228 gst_rtsp_range_free (src->range);
1231 /* don't clear the SDP when it was used in the url */
1232 if (src->sdp && !src->from_sdp) {
1233 gst_sdp_message_free (src->sdp);
1238 #define PARSE_INT(p, del, res) \
1241 p = strstr (p, del); \
1251 #define PARSE_STRING(p, del, res) \
1254 p = strstr (p, del); \
1266 #define SKIP_SPACES(p) \
1267 while (*p && g_ascii_isspace (*p)) \
1272 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1275 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1276 gint * rate, gchar ** params)
1280 p = (gchar *) rtpmap;
1282 PARSE_INT (p, " ", *payload);
1290 PARSE_STRING (p, "/", *name);
1291 if (*name == NULL) {
1292 GST_DEBUG ("no rate, name %s", p);
1293 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1294 * streams seem to omit the rate. */
1301 p = strstr (p, "/");
1319 * Mapping SDP attributes to caps
1321 * prepend 'a-' to IANA registered sdp attributes names
1322 * (ie: not prefixed with 'x-') in order to avoid
1323 * collision with gstreamer standard caps properties names
1326 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1328 if (attributes->len > 0) {
1332 s = gst_caps_get_structure (caps, 0);
1334 for (i = 0; i < attributes->len; i++) {
1335 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1336 gchar *tofree, *key;
1340 /* skip some of the attribute we already handle */
1341 if (!strcmp (key, "fmtp"))
1343 if (!strcmp (key, "rtpmap"))
1345 if (!strcmp (key, "control"))
1347 if (!strcmp (key, "range"))
1350 /* string must be valid UTF8 */
1351 if (!g_utf8_validate (attr->value, -1, NULL))
1354 if (!g_str_has_prefix (key, "x-"))
1355 tofree = key = g_strdup_printf ("a-%s", key);
1359 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1360 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1367 * Mapping of caps to and from SDP fields:
1369 * m=<media> <UDP port> RTP/AVP <payload>
1370 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1371 * a=fmtp:<payload> <param>[=<value>];...
1374 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1377 const gchar *rtpmap;
1381 gchar *params = NULL;
1387 /* get and parse rtpmap */
1388 if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
1389 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1391 if (payload != pt) {
1392 /* we ignore the rtpmap if the payload type is different. */
1393 g_warning ("rtpmap of wrong payload type, ignoring");
1399 /* if we failed to parse the rtpmap for a dynamic payload type, we have an
1403 /* else we can ignore */
1404 g_warning ("error parsing rtpmap, ignoring");
1407 /* dynamic payloads need rtpmap or we fail */
1411 /* check if we have a rate, if not, we need to look up the rate from the
1412 * default rates based on the payload types. */
1414 const GstRTPPayloadInfo *info;
1416 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1417 /* dynamic types, use media and encoding_name */
1418 tmp = g_ascii_strdown (media->media, -1);
1419 info = gst_rtp_payload_info_for_name (tmp, name);
1422 /* static types, use payload type */
1423 info = gst_rtp_payload_info_for_pt (pt);
1427 if ((rate = info->clock_rate) == 0)
1430 /* we fail if we cannot find one */
1435 tmp = g_ascii_strdown (media->media, -1);
1436 caps = gst_caps_new_simple ("application/x-unknown",
1437 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1439 s = gst_caps_get_structure (caps, 0);
1441 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1443 /* encoding name must be upper case */
1445 tmp = g_ascii_strup (name, -1);
1446 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1450 /* params must be lower case */
1451 if (params != NULL) {
1452 tmp = g_ascii_strdown (params, -1);
1453 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1457 /* parse optional fmtp: field */
1458 if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
1464 /* p is now of the format <payload> <param>[=<value>];... */
1465 PARSE_INT (p, " ", payload);
1466 if (payload != -1 && payload == pt) {
1470 /* <param>[=<value>] are separated with ';' */
1471 pairs = g_strsplit (p, ";", 0);
1472 for (i = 0; pairs[i]; i++) {
1474 const gchar *val, *key;
1476 /* the key may not have a '=', the value can have other '='s */
1477 valpos = strstr (pairs[i], "=");
1479 /* we have a '=' and thus a value, remove the '=' with \0 */
1481 /* value is everything between '=' and ';'. We split the pairs at ;
1482 * boundaries so we can take the remainder of the value. Some servers
1483 * put spaces around the value which we strip off here. Alternatively
1484 * we could strip those spaces in the depayloaders should these spaces
1485 * actually carry any meaning in the future. */
1486 val = g_strstrip (valpos + 1);
1488 /* simple <param>;.. is translated into <param>=1;... */
1491 /* strip the key of spaces, convert key to lowercase but not the value. */
1492 key = g_strstrip (pairs[i]);
1493 if (strlen (key) > 1) {
1494 tmp = g_ascii_strdown (key, -1);
1495 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1507 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1512 g_warning ("rate unknown for payload type %d", pt);
1518 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1519 gint * rtpport, gint * rtcpport)
1522 GstStateChangeReturn ret;
1523 GstElement *udpsrc0, *udpsrc1;
1524 gint tmp_rtp, tmp_rtcp;
1528 src = stream->parent;
1534 /* Start at next port */
1535 tmp_rtp = src->next_port_num;
1537 if (stream->is_ipv6)
1538 host = "udp://[::0]";
1540 host = "udp://0.0.0.0";
1542 /* try to allocate 2 UDP ports, the RTP port should be an even
1543 * number and the RTCP port should be the next (uneven) port */
1546 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1547 tmp_rtp >= src->client_port_range.max)
1550 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
1551 if (udpsrc0 == NULL)
1552 goto no_udp_protocol;
1553 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1555 if (src->udp_buffer_size != 0)
1556 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1559 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1560 if (ret == GST_STATE_CHANGE_FAILURE) {
1562 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1565 if (++count > src->retry)
1568 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1569 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1570 gst_object_unref (udpsrc0);
1572 GST_DEBUG_OBJECT (src, "retry %d", count);
1575 goto no_udp_protocol;
1578 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1579 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1581 /* check if port is even */
1582 if ((tmp_rtp & 0x01) != 0) {
1583 /* port not even, close and allocate another */
1584 if (++count > src->retry)
1587 GST_DEBUG_OBJECT (src, "RTP port not even");
1589 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1590 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1591 gst_object_unref (udpsrc0);
1593 GST_DEBUG_OBJECT (src, "retry %d", count);
1598 /* allocate port+1 for RTCP now */
1599 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
1600 if (udpsrc1 == NULL)
1601 goto no_udp_rtcp_protocol;
1604 tmp_rtcp = tmp_rtp + 1;
1605 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
1608 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1610 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1611 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1612 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1613 if (ret == GST_STATE_CHANGE_FAILURE) {
1614 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
1616 if (++count > src->retry)
1619 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1620 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1621 gst_object_unref (udpsrc0);
1623 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
1624 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1625 gst_object_unref (udpsrc1);
1629 GST_DEBUG_OBJECT (src, "retry %d", count);
1633 /* all fine, do port check */
1634 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
1635 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
1637 /* this should not happen... */
1638 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
1641 /* we keep these elements, we configure all in configure_transport when the
1642 * server told us to really use the UDP ports. */
1643 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
1644 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
1646 /* keep track of next available port number when we have a range
1648 if (src->next_port_num != 0)
1649 src->next_port_num = tmp_rtcp + 1;
1656 GST_DEBUG_OBJECT (src, "could not get UDP source");
1661 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
1665 no_udp_rtcp_protocol:
1667 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
1672 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
1673 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
1679 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1680 gst_object_unref (udpsrc0);
1683 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1684 gst_object_unref (udpsrc1);
1691 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
1699 event = gst_event_new_flush_start ();
1700 GST_DEBUG_OBJECT (src, "start flush");
1702 state = GST_STATE_PAUSED;
1704 event = gst_event_new_flush_stop (FALSE);
1705 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
1708 state = GST_STATE_PLAYING;
1710 state = GST_STATE_PAUSED;
1712 gst_rtspsrc_push_event (src, event);
1713 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
1715 /* to manage jitterbuffer buffer mode */
1717 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
1719 /* make running time start start at 0 again */
1720 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1721 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1723 for (i = 0; i < 2; i++) {
1725 if (stream->udpsrc[i]) {
1726 gst_element_set_state (stream->udpsrc[i], state);
1732 static GstRTSPResult
1733 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
1734 GstRTSPMessage * message, GTimeVal * timeout)
1739 ret = gst_rtsp_connection_send (conn, message, timeout);
1741 ret = GST_RTSP_ERROR;
1746 static GstRTSPResult
1747 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
1748 GstRTSPMessage * message, GTimeVal * timeout)
1753 ret = gst_rtsp_connection_receive (conn, message, timeout);
1755 ret = GST_RTSP_ERROR;
1761 gst_rtspsrc_get_position (GstRTSPSrc * src)
1766 query = gst_query_new_position (GST_FORMAT_TIME);
1767 /* should be known somewhere down the stream (e.g. jitterbuffer) */
1768 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1769 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1773 if (stream->srcpad) {
1774 if (gst_pad_query (stream->srcpad, query)) {
1775 gst_query_parse_position (query, &fmt, &pos);
1776 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
1777 GST_TIME_ARGS (pos));
1778 src->last_pos = pos;
1788 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
1790 src->state = GST_RTSP_STATE_SEEKING;
1791 /* PLAY will add the range header now. */
1792 src->need_range = TRUE;
1798 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
1803 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
1805 gboolean flush, skip;
1808 GstSegment seeksegment = { 0, };
1812 GST_DEBUG_OBJECT (src, "doing seek with event");
1814 gst_event_parse_seek (event, &rate, &format, &flags,
1815 &cur_type, &cur, &stop_type, &stop);
1817 /* no negative rates yet */
1821 /* we need TIME format */
1822 if (format != src->segment.format)
1825 GST_DEBUG_OBJECT (src, "doing seek without event");
1827 cur_type = GST_SEEK_TYPE_SET;
1828 stop_type = GST_SEEK_TYPE_SET;
1831 /* get flush flag */
1832 flush = flags & GST_SEEK_FLAG_FLUSH;
1833 skip = flags & GST_SEEK_FLAG_SKIP;
1835 /* now we need to make sure the streaming thread is stopped. We do this by
1836 * either sending a FLUSH_START event downstream which will cause the
1837 * streaming thread to stop with a WRONG_STATE.
1838 * For a non-flushing seek we simply pause the task, which will happen as soon
1839 * as it completes one iteration (and thus might block when the sink is
1840 * blocking in preroll). */
1842 GST_DEBUG_OBJECT (src, "starting flush");
1843 gst_rtspsrc_flush (src, TRUE, FALSE);
1846 gst_task_pause (src->task);
1850 /* we should now be able to grab the streaming thread because we stopped it
1851 * with the above flush/pause code */
1852 GST_RTSP_STREAM_LOCK (src);
1854 GST_DEBUG_OBJECT (src, "stopped streaming");
1856 /* copy segment, we need this because we still need the old
1857 * segment when we close the current segment. */
1858 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
1860 /* configure the seek parameters in the seeksegment. We will then have the
1861 * right values in the segment to perform the seek */
1863 GST_DEBUG_OBJECT (src, "configuring seek");
1864 gst_segment_do_seek (&seeksegment, rate, format, flags,
1865 cur_type, cur, stop_type, stop, &update);
1868 /* figure out the last position we need to play. If it's configured (stop !=
1869 * -1), use that, else we play until the total duration of the file */
1870 if ((stop = seeksegment.stop) == -1)
1871 stop = seeksegment.duration;
1873 playing = (src->state == GST_RTSP_STATE_PLAYING);
1875 /* if we were playing, pause first */
1877 /* obtain current position in case seek fails */
1878 gst_rtspsrc_get_position (src);
1879 gst_rtspsrc_pause (src, FALSE);
1883 gst_rtspsrc_do_seek (src, &seeksegment);
1885 /* and continue playing */
1887 gst_rtspsrc_play (src, &seeksegment, FALSE);
1889 /* prepare for streaming again */
1891 /* if we started flush, we stop now */
1892 GST_DEBUG_OBJECT (src, "stopping flush");
1893 gst_rtspsrc_flush (src, FALSE, playing);
1896 /* now we did the seek and can activate the new segment values */
1897 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
1899 /* if we're doing a segment seek, post a SEGMENT_START message */
1900 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
1901 gst_element_post_message (GST_ELEMENT_CAST (src),
1902 gst_message_new_segment_start (GST_OBJECT_CAST (src),
1903 src->segment.format, src->segment.position));
1906 /* now create the newsegment */
1907 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
1908 " to %" G_GINT64_FORMAT, src->segment.position, stop);
1910 /* store the newsegment event so it can be sent from the streaming thread. */
1911 if (src->start_segment)
1912 gst_event_unref (src->start_segment);
1913 src->start_segment = gst_event_new_segment (&src->segment);
1916 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
1917 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1918 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1919 stream->discont = TRUE;
1922 GST_RTSP_STREAM_UNLOCK (src);
1929 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
1934 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
1940 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
1944 gboolean res = TRUE;
1947 src = GST_RTSPSRC_CAST (parent);
1949 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1950 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1952 switch (GST_EVENT_TYPE (event)) {
1953 case GST_EVENT_SEEK:
1954 res = gst_rtspsrc_perform_seek (src, event);
1958 case GST_EVENT_NAVIGATION:
1959 case GST_EVENT_LATENCY:
1967 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
1968 res = gst_pad_send_event (target, event);
1969 gst_object_unref (target);
1971 gst_event_unref (event);
1974 gst_event_unref (event);
1980 /* this is the final event function we receive on the internal source pad when
1981 * we deal with TCP connections */
1983 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
1988 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
1990 switch (GST_EVENT_TYPE (event)) {
1991 case GST_EVENT_SEEK:
1993 case GST_EVENT_NAVIGATION:
1994 case GST_EVENT_LATENCY:
1996 gst_event_unref (event);
2003 /* this is the final query function we receive on the internal source pad when
2004 * we deal with TCP connections */
2006 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2010 gboolean res = TRUE;
2012 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2014 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2015 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2017 switch (GST_QUERY_TYPE (query)) {
2018 case GST_QUERY_POSITION:
2023 case GST_QUERY_DURATION:
2027 gst_query_parse_duration (query, &format, NULL);
2030 case GST_FORMAT_TIME:
2031 gst_query_set_duration (query, format, src->segment.duration);
2039 case GST_QUERY_LATENCY:
2041 /* we are live with a min latency of 0 and unlimited max latency, this
2042 * result will be updated by the session manager if there is any. */
2043 gst_query_set_latency (query, TRUE, 0, -1);
2053 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2055 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2059 gboolean res = FALSE;
2061 src = GST_RTSPSRC_CAST (parent);
2063 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2064 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2066 switch (GST_QUERY_TYPE (query)) {
2067 case GST_QUERY_DURATION:
2071 gst_query_parse_duration (query, &format, NULL);
2074 case GST_FORMAT_TIME:
2075 gst_query_set_duration (query, format, src->segment.duration);
2083 case GST_QUERY_SEEKING:
2087 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2088 if (format == GST_FORMAT_TIME) {
2090 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2092 /* seeking without duration is unlikely */
2093 seekable = seekable && src->seekable && src->segment.duration &&
2094 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2096 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2097 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2098 src->segment.start, src->segment.stop);
2107 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2109 gst_query_set_uri (query, uri);
2117 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2119 /* forward the query to the proxy target pad */
2121 res = gst_pad_query (target, query);
2122 gst_object_unref (target);
2131 /* callback for RTCP messages to be sent to the server when operating in TCP
2133 static GstFlowReturn
2134 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2137 GstRTSPStream *stream;
2138 GstFlowReturn res = GST_FLOW_OK;
2143 GstRTSPMessage message = { 0 };
2144 GstRTSPConnection *conn;
2146 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2147 src = stream->parent;
2149 gst_buffer_map (buffer, &map, GST_MAP_READ);
2153 gst_rtsp_message_init_data (&message, stream->channel[1]);
2155 /* lend the body data to the message */
2156 gst_rtsp_message_take_body (&message, data, size);
2158 if (stream->conninfo.connection)
2159 conn = stream->conninfo.connection;
2161 conn = src->conninfo.connection;
2163 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2164 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2165 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2167 /* and steal it away again because we will free it when unreffing the
2169 gst_rtsp_message_steal_body (&message, &data, &size);
2170 gst_rtsp_message_unset (&message);
2172 gst_buffer_unmap (buffer, &map);
2173 gst_buffer_unref (buffer);
2178 static GstPadProbeReturn
2179 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2181 GstRTSPSrc *src = user_data;
2183 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2184 GST_DEBUG_PAD_NAME (pad));
2186 /* activate the streams */
2187 GST_OBJECT_LOCK (src);
2188 if (!src->need_activate)
2191 src->need_activate = FALSE;
2192 GST_OBJECT_UNLOCK (src);
2194 gst_rtspsrc_activate_streams (src);
2196 return GST_PAD_PROBE_OK;
2200 GST_OBJECT_UNLOCK (src);
2201 return GST_PAD_PROBE_OK;
2205 /* this callback is called when the session manager generated a new src pad with
2206 * payloaded RTP packets. We simply ghost the pad here. */
2208 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2211 GstPadTemplate *template;
2214 GstRTSPStream *stream;
2217 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2219 GST_RTSP_STATE_LOCK (src);
2221 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2222 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2223 goto unknown_stream;
2225 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
2227 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2229 goto unknown_stream;
2231 /* create a new pad we will use to stream to */
2232 template = gst_static_pad_template_get (&rtptemplate);
2233 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2234 gst_object_unref (template);
2237 stream->added = TRUE;
2238 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2239 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2240 gst_pad_set_active (stream->srcpad, TRUE);
2241 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2243 /* check if we added all streams */
2245 for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
2246 stream = (GstRTSPStream *) lstream->data;
2248 GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
2249 stream, stream->container, stream->disabled, stream->added);
2251 /* a container stream only needs one pad added. Also disabled streams don't
2253 if (!stream->container && !stream->disabled && !stream->added) {
2258 GST_RTSP_STATE_UNLOCK (src);
2261 GST_DEBUG_OBJECT (src, "We added all streams");
2262 /* when we get here, all stream are added and we can fire the no-more-pads
2264 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2272 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2273 GST_RTSP_STATE_UNLOCK (src);
2280 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2282 GstRTSPStream *stream;
2285 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2287 GST_RTSP_STATE_LOCK (src);
2288 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2290 goto unknown_stream;
2292 caps = stream->caps;
2294 gst_caps_ref (caps);
2295 GST_RTSP_STATE_UNLOCK (src);
2301 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2302 GST_RTSP_STATE_UNLOCK (src);
2308 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2310 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2316 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2322 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2328 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2330 GstRTSPSrc *src = stream->parent;
2332 GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
2334 gst_rtspsrc_do_stream_eos (src, stream);
2338 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2340 GstRTSPSrc *src = stream->parent;
2342 GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
2344 gst_rtspsrc_do_stream_eos (src, stream);
2348 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2350 GstRTSPStream *stream;
2352 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2354 /* get stream for session */
2355 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2357 gst_rtspsrc_do_stream_eos (src, stream);
2362 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2364 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2368 /* try to get and configure a manager */
2370 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2371 GstRTSPTransport * transport)
2373 const gchar *manager;
2375 GstStateChangeReturn ret;
2377 /* find a manager */
2378 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2382 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2384 /* configure the manager */
2385 if (src->manager == NULL) {
2386 GObjectClass *klass;
2389 if (!(src->manager = gst_element_factory_make (manager, NULL))) {
2391 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2395 goto use_no_manager;
2397 if (!(src->manager = gst_element_factory_make (manager, NULL)))
2398 goto manager_failed;
2401 /* we manage this element */
2402 gst_bin_add (GST_BIN_CAST (src), src->manager);
2404 GST_OBJECT_LOCK (src);
2405 target = GST_STATE_TARGET (src);
2406 GST_OBJECT_UNLOCK (src);
2408 ret = gst_element_set_state (src->manager, target);
2409 if (ret == GST_STATE_CHANGE_FAILURE)
2410 goto start_manager_failure;
2412 g_object_set (src->manager, "latency", src->latency, NULL);
2414 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2415 if (g_object_class_find_property (klass, "drop-on-latency")) {
2416 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
2420 if (g_object_class_find_property (klass, "buffer-mode")) {
2421 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2422 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2424 gboolean need_slave;
2426 const gchar *encoding;
2428 /* buffer mode pauses are handled by adding offsets to buffer times,
2429 * but some depayloaders may have a hard time syncing output times
2430 * with such input times, e.g. container ones, most notably ASF */
2431 /* TODO alternatives are having an event that indicates these shifts,
2432 * or having rtsp extensions provide suggestion on buffer mode */
2433 need_slave = stream->container;
2434 if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
2435 (encoding = gst_structure_get_string (s, "encoding-name")))
2436 need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
2437 GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d",
2439 /* valid duration implies not likely live pipeline,
2440 * so slaving in jitterbuffer does not make much sense
2441 * (and might mess things up due to bursts) */
2442 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2443 src->segment.duration && !need_slave) {
2444 GST_DEBUG_OBJECT (src, "selected buffer");
2445 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
2448 GST_DEBUG_OBJECT (src, "selected slave");
2449 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2454 /* connect to signals if we did not already do so */
2455 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2457 src->manager_sig_id =
2458 g_signal_connect (src->manager, "pad-added",
2459 (GCallback) new_manager_pad, src);
2460 src->manager_ptmap_id =
2461 g_signal_connect (src->manager, "request-pt-map",
2462 (GCallback) request_pt_map, src);
2464 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2468 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2469 * into a separate RTP session. */
2470 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2471 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2473 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2474 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2477 /* now configure the bandwidth in the manager */
2478 if (g_signal_lookup ("get-internal-session",
2479 G_OBJECT_TYPE (src->manager)) != 0) {
2480 GObject *rtpsession;
2482 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2485 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2487 stream->session = rtpsession;
2489 if (stream->as_bandwidth != -1) {
2490 GST_INFO_OBJECT (src, "setting AS: %f",
2491 (gdouble) (stream->as_bandwidth * 1000));
2492 g_object_set (rtpsession, "bandwidth",
2493 (gdouble) (stream->as_bandwidth * 1000), NULL);
2495 if (stream->rr_bandwidth != -1) {
2496 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2497 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2500 if (stream->rs_bandwidth != -1) {
2501 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2502 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2506 g_object_set (rtpsession, "probation", src->probation, NULL);
2508 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2510 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2512 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2514 g_signal_connect (rtpsession, "on-ssrc-active",
2515 (GCallback) on_ssrc_active, stream);
2526 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2531 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2534 start_manager_failure:
2536 GST_DEBUG_OBJECT (src, "could not start session manager");
2541 /* free the UDP sources allocated when negotiating a transport.
2542 * This function is called when the server negotiated to a transport where the
2543 * UDP sources are not needed anymore, such as TCP or multicast. */
2545 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
2549 for (i = 0; i < 2; i++) {
2550 if (stream->udpsrc[i]) {
2551 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
2552 gst_object_unref (stream->udpsrc[i]);
2553 stream->udpsrc[i] = NULL;
2558 /* for TCP, create pads to send and receive data to and from the manager and to
2559 * intercept various events and queries
2562 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
2563 GstRTSPTransport * transport, GstPad ** outpad)
2566 GstPadTemplate *template;
2567 GstPad *pad0, *pad1;
2569 /* configure for interleaved delivery, nothing needs to be done
2570 * here, the loop function will call the chain functions of the
2571 * session manager. */
2572 stream->channel[0] = transport->interleaved.min;
2573 stream->channel[1] = transport->interleaved.max;
2574 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
2575 stream->channel[0], stream->channel[1]);
2577 /* we can remove the allocated UDP ports now */
2578 gst_rtspsrc_stream_free_udp (stream);
2580 /* no session manager, send data to srcpad directly */
2581 if (!stream->channelpad[0]) {
2582 GST_DEBUG_OBJECT (src, "no manager, creating pad");
2584 /* create a new pad we will use to stream to */
2585 name = g_strdup_printf ("stream_%u", stream->id);
2586 template = gst_static_pad_template_get (&rtptemplate);
2587 stream->channelpad[0] = gst_pad_new_from_template (template, name);
2588 gst_object_unref (template);
2591 /* set caps and activate */
2592 gst_pad_use_fixed_caps (stream->channelpad[0]);
2593 gst_pad_set_active (stream->channelpad[0], TRUE);
2595 *outpad = gst_object_ref (stream->channelpad[0]);
2597 GST_DEBUG_OBJECT (src, "using manager source pad");
2599 template = gst_static_pad_template_get (&anysrctemplate);
2601 /* allocate pads for sending the channel data into the manager */
2602 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
2603 gst_pad_link (pad0, stream->channelpad[0]);
2604 gst_object_unref (stream->channelpad[0]);
2605 stream->channelpad[0] = pad0;
2606 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
2607 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
2608 gst_pad_set_element_private (pad0, src);
2609 gst_pad_set_active (pad0, TRUE);
2611 if (stream->channelpad[1]) {
2612 /* if we have a sinkpad for the other channel, create a pad and link to the
2614 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
2615 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
2616 gst_pad_link (pad1, stream->channelpad[1]);
2617 gst_object_unref (stream->channelpad[1]);
2618 stream->channelpad[1] = pad1;
2619 gst_pad_set_active (pad1, TRUE);
2621 gst_object_unref (template);
2623 /* setup RTCP transport back to the server if we have to. */
2624 if (src->manager && src->do_rtcp) {
2627 template = gst_static_pad_template_get (&anysinktemplate);
2629 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
2630 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
2631 gst_pad_set_element_private (stream->rtcppad, stream);
2632 gst_pad_set_active (stream->rtcppad, TRUE);
2634 /* get session RTCP pad */
2635 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2636 pad = gst_element_get_request_pad (src->manager, name);
2641 gst_pad_link (pad, stream->rtcppad);
2642 gst_object_unref (pad);
2645 gst_object_unref (template);
2651 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
2652 GstRTSPTransport * transport, const gchar ** destination, gint * min,
2653 gint * max, guint * ttl)
2655 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2657 if (!(*destination = transport->destination))
2658 *destination = stream->destination;
2661 /* transport first */
2662 *min = transport->port.min;
2663 *max = transport->port.max;
2664 if (*min == -1 && *max == -1) {
2665 /* then try from SDP */
2666 if (stream->port != 0) {
2667 *min = stream->port;
2668 *max = stream->port + 1;
2674 if (!(*ttl = transport->ttl))
2679 /* first take the source, then the endpoint to figure out where to send
2681 if (!(*destination = transport->source)) {
2682 if (src->conninfo.connection)
2683 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
2684 else if (stream->conninfo.connection)
2686 gst_rtsp_connection_get_ip (stream->conninfo.connection);
2690 /* for unicast we only expect the ports here */
2691 *min = transport->server_port.min;
2692 *max = transport->server_port.max;
2697 /* For multicast create UDP sources and join the multicast group. */
2699 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
2700 GstRTSPTransport * transport, GstPad ** outpad)
2703 const gchar *destination;
2706 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
2708 /* we can remove the allocated UDP ports now */
2709 gst_rtspsrc_stream_free_udp (stream);
2711 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
2714 /* we need a destination now */
2715 if (destination == NULL)
2716 goto no_destination;
2718 /* we really need ports now or we won't be able to receive anything at all */
2719 if (min == -1 && max == -1)
2722 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
2723 destination, min, max);
2725 /* creating UDP source for RTP */
2727 uri = g_strdup_printf ("udp://%s:%d", destination, min);
2729 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
2731 if (stream->udpsrc[0] == NULL)
2734 /* take ownership */
2735 gst_object_ref_sink (stream->udpsrc[0]);
2737 if (src->udp_buffer_size != 0)
2738 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
2739 src->udp_buffer_size, NULL);
2742 gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
2745 /* creating another UDP source for RTCP */
2747 uri = g_strdup_printf ("udp://%s:%d", destination, max);
2749 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
2751 if (stream->udpsrc[1] == NULL)
2754 /* take ownership */
2755 gst_object_ref_sink (stream->udpsrc[1]);
2757 gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
2764 GST_DEBUG_OBJECT (src, "no UDP source element found");
2769 GST_DEBUG_OBJECT (src, "no destination found");
2774 GST_DEBUG_OBJECT (src, "no ports found");
2779 /* configure the remainder of the UDP ports */
2781 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
2782 GstRTSPTransport * transport, GstPad ** outpad)
2784 /* we manage the UDP elements now. For unicast, the UDP sources where
2785 * allocated in the stream when we suggested a transport. */
2786 if (stream->udpsrc[0]) {
2787 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
2789 GST_DEBUG_OBJECT (src, "setting up UDP source");
2791 /* configure a timeout on the UDP port. When the timeout message is
2792 * posted, we assume UDP transport is not possible. We reconnect using TCP
2794 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
2797 /* get output pad of the UDP source. */
2798 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
2800 /* save it so we can unblock */
2801 stream->blockedpad = *outpad;
2803 /* configure pad block on the pad. As soon as there is dataflow on the
2804 * UDP source, we know that UDP is not blocked by a firewall and we can
2805 * configure all the streams to let the application autoplug decoders. */
2807 gst_pad_add_probe (stream->blockedpad,
2808 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
2810 if (stream->channelpad[0]) {
2811 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
2812 /* configure for UDP delivery, we need to connect the UDP pads to
2813 * the session plugin. */
2814 gst_pad_link (*outpad, stream->channelpad[0]);
2815 gst_object_unref (*outpad);
2817 /* we connected to pad-added signal to get pads from the manager */
2819 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
2824 if (stream->udpsrc[1]) {
2825 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
2827 if (stream->channelpad[1]) {
2830 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
2832 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
2833 gst_pad_link (pad, stream->channelpad[1]);
2834 gst_object_unref (pad);
2836 /* leave unlinked */
2842 /* configure the UDP sink back to the server for status reports */
2844 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
2845 GstRTSPStream * stream, GstRTSPTransport * transport)
2848 gint rtp_port, rtcp_port;
2849 gboolean do_rtp, do_rtcp;
2850 const gchar *destination;
2855 /* get transport info */
2856 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
2857 &rtp_port, &rtcp_port, &ttl);
2859 /* see what we need to do */
2860 do_rtp = (rtp_port != -1);
2861 /* it's possible that the server does not want us to send RTCP in which case
2863 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
2865 /* we need a destination when we have RTP or RTCP ports */
2866 if (destination == NULL && (do_rtp || do_rtcp))
2867 goto no_destination;
2869 /* try to construct the fakesrc to the RTP port of the server to open up any
2872 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
2875 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
2876 stream->udpsink[0] =
2877 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
2879 if (stream->udpsink[0] == NULL)
2880 goto no_sink_element;
2882 /* don't join multicast group, we will have the source socket do that */
2883 /* no sync or async state changes needed */
2884 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
2885 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2887 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2889 if (stream->udpsrc[0]) {
2890 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
2891 * so that NAT firewalls will open a hole for us */
2892 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
2893 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
2894 /* configure socket and make sure udpsink does not close it when shutting
2895 * down, it belongs to udpsrc after all. */
2896 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
2897 "close-socket", FALSE, NULL);
2898 g_object_unref (socket);
2901 /* the source for the dummy packets to open up NAT */
2902 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
2903 if (stream->fakesrc == NULL)
2904 goto no_fakesrc_element;
2906 /* random data in 5 buffers, a size of 200 bytes should be fine */
2907 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
2908 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
2910 /* we don't want to consider this a sink */
2911 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
2913 /* keep everything locked */
2914 gst_element_set_locked_state (stream->udpsink[0], TRUE);
2915 gst_element_set_locked_state (stream->fakesrc, TRUE);
2917 gst_object_ref (stream->udpsink[0]);
2918 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
2919 gst_object_ref (stream->fakesrc);
2920 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
2922 gst_element_link (stream->fakesrc, stream->udpsink[0]);
2925 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
2928 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
2929 stream->udpsink[1] =
2930 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
2932 if (stream->udpsink[1] == NULL)
2933 goto no_sink_element;
2935 /* don't join multicast group, we will have the source socket do that */
2936 /* no sync or async state changes needed */
2937 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
2938 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2940 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2942 if (stream->udpsrc[1]) {
2943 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
2944 * because some servers check the port number of where it sends RTCP to identify
2945 * the RTCP packets it receives */
2946 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
2947 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
2948 /* configure socket and make sure udpsink does not close it when shutting
2949 * down, it belongs to udpsrc after all. */
2950 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
2951 "close-socket", FALSE, NULL);
2952 g_object_unref (socket);
2955 /* we don't want to consider this a sink */
2956 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
2958 /* we keep this playing always */
2959 gst_element_set_locked_state (stream->udpsink[1], TRUE);
2960 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
2962 gst_object_ref (stream->udpsink[1]);
2963 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
2965 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
2967 /* get session RTCP pad */
2968 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2969 pad = gst_element_get_request_pad (src->manager, name);
2974 gst_pad_link (pad, stream->rtcppad);
2975 gst_object_unref (pad);
2984 GST_DEBUG_OBJECT (src, "no destination address specified");
2989 GST_DEBUG_OBJECT (src, "no UDP sink element found");
2994 GST_DEBUG_OBJECT (src, "no fakesrc element found");
2999 /* sets up all elements needed for streaming over the specified transport.
3000 * Does not yet expose the element pads, this will be done when there is actuall
3001 * dataflow detected, which might never happen when UDP is blocked in a
3002 * firewall, for example.
3005 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3006 GstRTSPTransport * transport)
3009 GstPad *outpad = NULL;
3010 GstPadTemplate *template;
3015 src = stream->parent;
3017 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3019 s = gst_caps_get_structure (stream->caps, 0);
3021 /* get the proper mime type for this stream now */
3022 if (gst_rtsp_transport_get_mime (transport->trans, &mime) < 0)
3023 goto unknown_transport;
3025 goto unknown_transport;
3027 /* configure the final mime type */
3028 GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
3029 gst_structure_set_name (s, mime);
3031 /* try to get and configure a manager, channelpad[0-1] will be configured with
3032 * the pads for the manager, or NULL when no manager is needed. */
3033 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3036 switch (transport->lower_transport) {
3037 case GST_RTSP_LOWER_TRANS_TCP:
3038 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3039 goto transport_failed;
3041 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3042 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3043 goto transport_failed;
3044 /* fallthrough, the rest is the same for UDP and MCAST */
3045 case GST_RTSP_LOWER_TRANS_UDP:
3046 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3047 goto transport_failed;
3048 /* configure udpsinks back to the server for RTCP messages and for the
3049 * dummy RTP messages to open NAT. */
3050 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3051 goto transport_failed;
3054 goto unknown_transport;
3058 GST_DEBUG_OBJECT (src, "creating ghostpad");
3060 gst_pad_use_fixed_caps (outpad);
3062 /* create ghostpad, don't add just yet, this will be done when we activate
3064 name = g_strdup_printf ("stream_%u", stream->id);
3065 template = gst_static_pad_template_get (&rtptemplate);
3066 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3067 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3068 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3069 gst_object_unref (template);
3072 gst_object_unref (outpad);
3074 /* mark pad as ok */
3075 stream->last_ret = GST_FLOW_OK;
3082 GST_DEBUG_OBJECT (src, "failed to configure transport");
3087 GST_DEBUG_OBJECT (src, "unknown transport");
3092 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3097 /* send a couple of dummy random packets on the receiver RTP port to the server,
3098 * this should make a firewall think we initiated the data transfer and
3099 * hopefully allow packets to go from the sender port to our RTP receiver port */
3101 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3105 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3108 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3109 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3111 if (stream->fakesrc && stream->udpsink[0]) {
3112 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3113 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3114 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3115 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3116 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3122 /* Adds the source pads of all configured streams to the element.
3123 * This code is performed when we detected dataflow.
3125 * We detect dataflow from either the _loop function or with pad probes on the
3129 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3133 GST_DEBUG_OBJECT (src, "activating streams");
3135 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3136 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3138 if (stream->udpsrc[0]) {
3139 /* remove timeout, we are streaming now and timeouts will be handled by
3140 * the session manager and jitter buffer */
3141 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3143 if (stream->srcpad) {
3144 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3145 gst_pad_set_active (stream->srcpad, TRUE);
3147 /* if we don't have a session manager, set the caps now. If we have a
3148 * session, we will get a notification of the pad and the caps. */
3149 if (!src->manager) {
3150 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3151 gst_pad_set_caps (stream->srcpad, stream->caps);
3154 if (!stream->added) {
3155 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3156 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3157 stream->added = TRUE;
3162 /* unblock all pads */
3163 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3164 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3166 if (stream->blockid) {
3167 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3168 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3169 stream->blockid = 0;
3177 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
3178 gboolean reset_manager)
3181 guint64 start, stop;
3182 gdouble play_speed, play_scale;
3184 GST_DEBUG_OBJECT (src, "configuring stream caps");
3186 start = segment->position;
3187 stop = segment->duration;
3188 play_speed = segment->rate;
3189 play_scale = segment->applied_rate;
3191 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3192 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3195 if ((caps = stream->caps)) {
3196 caps = gst_caps_make_writable (caps);
3198 if (stream->timebase != -1)
3199 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3200 (guint) stream->timebase, NULL);
3201 if (stream->seqbase != -1)
3202 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3203 (guint) stream->seqbase, NULL);
3204 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3206 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3207 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3208 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3210 stream->caps = caps;
3212 GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
3214 if (reset_manager && src->manager) {
3215 GST_DEBUG_OBJECT (src, "clear session");
3216 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3220 static GstFlowReturn
3221 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3226 /* store the value */
3227 stream->last_ret = ret;
3229 /* if it's success we can return the value right away */
3230 if (ret == GST_FLOW_OK)
3233 /* any other error that is not-linked can be returned right
3235 if (ret != GST_FLOW_NOT_LINKED)
3238 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3239 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3240 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3242 ret = ostream->last_ret;
3243 /* some other return value (must be SUCCESS but we can return
3244 * other values as well) */
3245 if (ret != GST_FLOW_NOT_LINKED)
3248 /* if we get here, all other pads were unlinked and we return
3249 * NOT_LINKED then */
3255 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3258 gboolean res = TRUE;
3260 /* only streams that have a connection to the outside world */
3261 if (stream->srcpad == NULL)
3264 if (stream->udpsrc[0]) {
3265 gst_event_ref (event);
3266 res = gst_element_send_event (stream->udpsrc[0], event);
3267 } else if (stream->channelpad[0]) {
3268 gst_event_ref (event);
3269 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3270 res = gst_pad_push_event (stream->channelpad[0], event);
3272 res = gst_pad_send_event (stream->channelpad[0], event);
3275 if (stream->udpsrc[1]) {
3276 gst_event_ref (event);
3277 res &= gst_element_send_event (stream->udpsrc[1], event);
3278 } else if (stream->channelpad[1]) {
3279 gst_event_ref (event);
3280 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3281 res &= gst_pad_push_event (stream->channelpad[1], event);
3283 res &= gst_pad_send_event (stream->channelpad[1], event);
3287 gst_event_unref (event);
3293 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
3296 gboolean res = TRUE;
3298 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3299 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3301 gst_event_ref (event);
3302 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
3304 gst_event_unref (event);
3309 static GstRTSPResult
3310 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3315 if (info->connection == NULL) {
3316 if (info->url == NULL) {
3317 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3318 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3322 /* create connection */
3323 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3324 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3325 goto could_not_create;
3328 g_free (info->url_str);
3329 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3331 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3333 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3334 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3336 if (src->proxy_host) {
3337 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3339 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3344 if (!info->connected) {
3347 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3348 ("Connecting to %s", info->location));
3349 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3351 gst_rtsp_connection_connect (info->connection,
3352 src->ptcp_timeout)) < 0)
3353 goto could_not_connect;
3355 info->connected = TRUE;
3362 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3367 gchar *str = gst_rtsp_strresult (res);
3368 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3374 gchar *str = gst_rtsp_strresult (res);
3375 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3381 static GstRTSPResult
3382 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3385 GST_RTSP_STATE_LOCK (src);
3386 if (info->connected) {
3387 GST_DEBUG_OBJECT (src, "closing connection...");
3388 gst_rtsp_connection_close (info->connection);
3389 info->connected = FALSE;
3391 if (free && info->connection) {
3392 /* free connection */
3393 GST_DEBUG_OBJECT (src, "freeing connection...");
3394 gst_rtsp_connection_free (info->connection);
3395 info->connection = NULL;
3397 GST_RTSP_STATE_UNLOCK (src);
3401 static GstRTSPResult
3402 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3407 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3408 gst_rtsp_conninfo_close (src, info, FALSE);
3409 res = gst_rtsp_conninfo_connect (src, info, async);
3415 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3419 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3420 GST_RTSP_STATE_LOCK (src);
3421 if (src->conninfo.connection) {
3422 GST_DEBUG_OBJECT (src, "connection flush");
3423 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3425 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3426 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3427 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3428 if (stream->conninfo.connection)
3429 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3431 GST_RTSP_STATE_UNLOCK (src);
3434 /* FIXME, handle server request, reply with OK, for now */
3435 static GstRTSPResult
3436 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3437 GstRTSPMessage * request)
3439 GstRTSPMessage response = { 0 };
3442 GST_DEBUG_OBJECT (src, "got server request message");
3445 gst_rtsp_message_dump (request);
3447 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3449 if (res == GST_RTSP_ENOTIMPL) {
3450 /* default implementation, send OK */
3452 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3457 GST_DEBUG_OBJECT (src, "replying with OK");
3460 gst_rtsp_message_dump (&response);
3462 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
3466 gst_rtsp_message_unset (&response);
3467 } else if (res == GST_RTSP_EEOF)
3475 gst_rtsp_message_unset (&response);
3480 /* send server keep-alive */
3481 static GstRTSPResult
3482 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
3484 GstRTSPMessage request = { 0 };
3486 GstRTSPMethod method;
3489 if (src->do_rtsp_keep_alive == FALSE) {
3490 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
3491 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
3495 GST_DEBUG_OBJECT (src, "creating server keep-alive");
3497 /* find a method to use for keep-alive */
3498 if (src->methods & GST_RTSP_GET_PARAMETER)
3499 method = GST_RTSP_GET_PARAMETER;
3501 method = GST_RTSP_OPTIONS;
3504 control = src->control;
3506 control = src->conninfo.url_str;
3508 if (control == NULL)
3511 res = gst_rtsp_message_init_request (&request, method, control);
3516 gst_rtsp_message_dump (&request);
3519 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
3524 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
3525 gst_rtsp_message_unset (&request);
3532 GST_WARNING_OBJECT (src, "no control url to send keepalive");
3537 gchar *str = gst_rtsp_strresult (res);
3539 gst_rtsp_message_unset (&request);
3540 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
3541 ("Could not send keep-alive. (%s)", str));
3547 static GstFlowReturn
3548 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
3550 GstRTSPMessage message = { 0 };
3553 GstRTSPStream *stream;
3554 GstPad *outpad = NULL;
3557 GstFlowReturn ret = GST_FLOW_OK;
3559 gboolean is_rtcp, have_data;
3561 /* here we are only interested in data messages */
3564 GTimeVal tv_timeout;
3566 /* get the next timeout interval */
3567 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3569 /* see if the timeout period expired */
3570 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
3571 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
3572 /* send keep-alive, only act on interrupt, a warning will be posted for
3574 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3576 /* get new timeout */
3577 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3580 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
3581 tv_timeout.tv_sec, tv_timeout.tv_usec);
3583 /* protect the connection with the connection lock so that we can see when
3584 * we are finished doing server communication */
3586 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3587 &message, src->ptcp_timeout);
3591 GST_DEBUG_OBJECT (src, "we received a server message");
3593 case GST_RTSP_EINTR:
3594 /* we got interrupted this means we need to stop */
3596 case GST_RTSP_ETIMEOUT:
3597 /* no reply, send keep alive */
3598 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3599 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3603 /* go EOS when the server closed the connection */
3609 switch (message.type) {
3610 case GST_RTSP_MESSAGE_REQUEST:
3611 /* server sends us a request message, handle it */
3613 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3615 if (res == GST_RTSP_EEOF)
3618 goto handle_request_failed;
3620 case GST_RTSP_MESSAGE_RESPONSE:
3621 /* we ignore response messages */
3622 GST_DEBUG_OBJECT (src, "ignoring response message");
3624 gst_rtsp_message_dump (&message);
3626 case GST_RTSP_MESSAGE_DATA:
3627 GST_DEBUG_OBJECT (src, "got data message");
3631 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3638 channel = message.type_data.data.channel;
3640 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
3642 goto unknown_stream;
3644 if (channel == stream->channel[0]) {
3645 outpad = stream->channelpad[0];
3647 } else if (channel == stream->channel[1]) {
3648 outpad = stream->channelpad[1];
3654 /* take a look at the body to figure out what we have */
3655 gst_rtsp_message_get_body (&message, &data, &size);
3657 goto invalid_length;
3659 /* channels are not correct on some servers, do extra check */
3660 if (data[1] >= 200 && data[1] <= 204) {
3661 /* hmm RTCP message switch to the RTCP pad of the same stream. */
3662 outpad = stream->channelpad[1];
3666 /* we have no clue what this is, just ignore then. */
3668 goto unknown_stream;
3670 /* take the message body for further processing */
3671 gst_rtsp_message_steal_body (&message, &data, &size);
3673 /* strip the trailing \0 */
3676 buf = gst_buffer_new ();
3677 gst_buffer_append_memory (buf,
3678 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
3680 /* don't need message anymore */
3681 gst_rtsp_message_unset (&message);
3683 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
3686 if (src->need_activate) {
3687 gst_rtspsrc_activate_streams (src);
3688 src->need_activate = FALSE;
3691 if (src->base_time == -1) {
3692 /* Take current running_time. This timestamp will be put on
3693 * the first buffer of each stream because we are a live source and so we
3694 * timestamp with the running_time. When we are dealing with TCP, we also
3695 * only timestamp the first buffer (using the DISCONT flag) because a server
3696 * typically bursts data, for which we don't want to compensate by speeding
3697 * up the media. The other timestamps will be interpollated from this one
3698 * using the RTP timestamps. */
3699 GST_OBJECT_LOCK (src);
3700 if (GST_ELEMENT_CLOCK (src)) {
3702 GstClockTime base_time;
3704 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
3705 base_time = GST_ELEMENT_CAST (src)->base_time;
3707 src->base_time = now - base_time;
3709 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
3710 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
3712 GST_OBJECT_UNLOCK (src);
3715 if (stream->discont && !is_rtcp) {
3716 /* mark first RTP buffer as discont */
3717 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
3718 stream->discont = FALSE;
3719 /* first buffer gets the timestamp, other buffers are not timestamped and
3720 * their presentation time will be interpollated from the rtp timestamps. */
3721 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
3722 GST_TIME_ARGS (src->base_time));
3724 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
3727 /* chain to the peer pad */
3728 if (GST_PAD_IS_SINK (outpad))
3729 ret = gst_pad_chain (outpad, buf);
3731 ret = gst_pad_push (outpad, buf);
3734 /* combine all stream flows for the data transport */
3735 ret = gst_rtspsrc_combine_flows (src, stream, ret);
3742 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
3743 gst_rtsp_message_unset (&message);
3748 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3749 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3750 ("The server closed the connection."));
3751 src->conninfo.connected = FALSE;
3752 gst_rtsp_message_unset (&message);
3753 return GST_FLOW_EOS;
3757 gst_rtsp_message_unset (&message);
3758 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3759 gst_rtspsrc_connection_flush (src, FALSE);
3760 return GST_FLOW_FLUSHING;
3764 gchar *str = gst_rtsp_strresult (res);
3766 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3767 ("Could not receive message. (%s)", str));
3770 gst_rtsp_message_unset (&message);
3771 return GST_FLOW_ERROR;
3773 handle_request_failed:
3775 gchar *str = gst_rtsp_strresult (res);
3777 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3778 ("Could not handle server message. (%s)", str));
3780 gst_rtsp_message_unset (&message);
3781 return GST_FLOW_ERROR;
3785 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3786 ("Short message received, ignoring."));
3787 gst_rtsp_message_unset (&message);
3792 static GstFlowReturn
3793 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
3796 GstRTSPMessage message = { 0 };
3800 GTimeVal tv_timeout;
3802 /* get the next timeout interval */
3803 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3805 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
3806 (gint) tv_timeout.tv_sec);
3808 gst_rtsp_message_unset (&message);
3810 /* we should continue reading the TCP socket because the server might
3811 * send us requests. When the session timeout expires, we need to send a
3812 * keep-alive request to keep the session open. */
3813 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3814 &message, &tv_timeout);
3818 GST_DEBUG_OBJECT (src, "we received a server message");
3820 case GST_RTSP_EINTR:
3821 /* we got interrupted, see what we have to do */
3823 case GST_RTSP_ETIMEOUT:
3824 /* send keep-alive, ignore the result, a warning will be posted. */
3825 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3826 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3830 /* server closed the connection. not very fatal for UDP, reconnect and
3831 * see what happens. */
3832 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3833 ("The server closed the connection."));
3835 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
3840 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
3842 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3843 ("Unhandled return value %d.", res));
3847 switch (message.type) {
3848 case GST_RTSP_MESSAGE_REQUEST:
3849 /* server sends us a request message, handle it */
3851 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3853 if (res == GST_RTSP_EEOF)
3856 goto handle_request_failed;
3858 case GST_RTSP_MESSAGE_RESPONSE:
3859 /* we ignore response and data messages */
3860 GST_DEBUG_OBJECT (src, "ignoring response message");
3862 gst_rtsp_message_dump (&message);
3863 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
3864 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
3865 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
3866 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
3867 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3874 case GST_RTSP_MESSAGE_DATA:
3875 /* we ignore response and data messages */
3876 GST_DEBUG_OBJECT (src, "ignoring data message");
3879 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3885 /* we get here when the connection got interrupted */
3888 gst_rtsp_message_unset (&message);
3889 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3890 gst_rtspsrc_connection_flush (src, FALSE);
3891 return GST_FLOW_FLUSHING;
3895 gchar *str = gst_rtsp_strresult (res);
3898 src->conninfo.connected = FALSE;
3899 if (res != GST_RTSP_EINTR) {
3900 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
3901 ("Could not connect to server. (%s)", str));
3903 ret = GST_FLOW_ERROR;
3905 ret = GST_FLOW_FLUSHING;
3911 gchar *str = gst_rtsp_strresult (res);
3913 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3914 ("Could not receive message. (%s)", str));
3916 return GST_FLOW_ERROR;
3918 handle_request_failed:
3920 gchar *str = gst_rtsp_strresult (res);
3923 gst_rtsp_message_unset (&message);
3924 if (res != GST_RTSP_EINTR) {
3925 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3926 ("Could not handle server message. (%s)", str));
3928 ret = GST_FLOW_ERROR;
3930 ret = GST_FLOW_FLUSHING;
3936 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3937 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3938 ("The server closed the connection."));
3939 src->conninfo.connected = FALSE;
3940 gst_rtsp_message_unset (&message);
3941 return GST_FLOW_EOS;
3945 static GstRTSPResult
3946 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
3948 GstRTSPResult res = GST_RTSP_OK;
3951 GST_DEBUG_OBJECT (src, "doing reconnect");
3953 GST_OBJECT_LOCK (src);
3954 /* only restart when the pads were not yet activated, else we were
3955 * streaming over UDP */
3956 restart = src->need_activate;
3957 GST_OBJECT_UNLOCK (src);
3959 /* no need to restart, we're done */
3963 /* we can try only TCP now */
3964 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
3966 /* close and cleanup our state */
3967 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
3970 /* see if we have TCP left to try. Also don't try TCP when we were configured
3972 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
3975 /* We post a warning message now to inform the user
3976 * that nothing happened. It's most likely a firewall thing. */
3977 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3978 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3979 "firewall is blocking it. Retrying using a TCP connection.",
3980 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3982 /* open new connection using tcp */
3983 if (gst_rtspsrc_open (src, async) < 0)
3986 /* start playback */
3987 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
3996 src->cur_protocols = 0;
3997 /* no transport possible, post an error and stop */
3998 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3999 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4000 "firewall is blocking it. No other protocols to try.",
4001 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4002 return GST_RTSP_ERROR;
4006 GST_DEBUG_OBJECT (src, "open failed");
4011 GST_DEBUG_OBJECT (src, "play failed");
4017 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
4021 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
4024 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
4027 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
4030 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
4038 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
4042 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
4045 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
4048 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
4051 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4059 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4063 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4066 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4069 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4072 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4080 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4084 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4087 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4090 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4093 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4101 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4103 if (ret == GST_RTSP_OK)
4104 gst_rtspsrc_loop_complete_cmd (src, cmd);
4105 else if (ret == GST_RTSP_EINTR)
4106 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4108 gst_rtspsrc_loop_error_cmd (src, cmd);
4112 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
4116 /* start new request */
4117 gst_rtspsrc_loop_start_cmd (src, cmd);
4119 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4121 GST_OBJECT_LOCK (src);
4122 old = src->pending_cmd;
4123 if (old != CMD_WAIT) {
4124 src->pending_cmd = CMD_WAIT;
4125 GST_OBJECT_UNLOCK (src);
4126 /* cancel previous request */
4127 gst_rtspsrc_loop_cancel_cmd (src, old);
4128 GST_OBJECT_LOCK (src);
4130 src->pending_cmd = cmd;
4131 /* interrupt if allowed */
4132 if (src->busy_cmd & mask) {
4133 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
4134 gst_rtspsrc_connection_flush (src, TRUE);
4136 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
4139 gst_task_start (src->task);
4140 GST_OBJECT_UNLOCK (src);
4144 gst_rtspsrc_loop (GstRTSPSrc * src)
4148 if (!src->conninfo.connection || !src->conninfo.connected)
4151 if (src->interleaved)
4152 ret = gst_rtspsrc_loop_interleaved (src);
4154 ret = gst_rtspsrc_loop_udp (src);
4156 if (ret != GST_FLOW_OK)
4164 GST_WARNING_OBJECT (src, "we are not connected");
4165 ret = GST_FLOW_FLUSHING;
4170 const gchar *reason = gst_flow_get_name (ret);
4172 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4173 src->running = FALSE;
4174 if (ret == GST_FLOW_EOS) {
4175 /* perform EOS logic */
4176 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4177 gst_element_post_message (GST_ELEMENT_CAST (src),
4178 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4179 src->segment.format, src->segment.position));
4180 gst_rtspsrc_push_event (src,
4181 gst_event_new_segment_done (src->segment.format,
4182 src->segment.position));
4184 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4186 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4187 /* for fatal errors we post an error message, post the error before the
4188 * EOS so the app knows about the error first. */
4189 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4190 ("Internal data flow error."),
4191 ("streaming task paused, reason %s (%d)", reason, ret));
4192 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4198 #ifndef GST_DISABLE_GST_DEBUG
4199 static const gchar *
4200 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4204 while (method != 0) {
4221 static const gchar *
4222 gst_rtspsrc_skip_lws (const gchar * s)
4224 while (g_ascii_isspace (*s))
4229 static const gchar *
4230 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4232 while (s > start && g_ascii_isspace (*(s - 1)))
4237 static const gchar *
4238 gst_rtspsrc_skip_commas (const gchar * s)
4240 /* The grammar allows for multiple commas */
4241 while (g_ascii_isspace (*s) || *s == ',')
4246 static const gchar *
4247 gst_rtspsrc_skip_item (const gchar * s)
4249 gboolean quoted = FALSE;
4250 const gchar *start = s;
4252 /* A list item ends at the last non-whitespace character
4253 * before a comma which is not inside a quoted-string. Or at
4254 * the end of the string.
4260 if (*s == '\\' && *(s + 1))
4269 return gst_rtspsrc_unskip_lws (s, start);
4273 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4277 src = quoted_string + 1;
4278 dst = quoted_string;
4279 while (*src && *src != '"') {
4280 if (*src == '\\' && *(src + 1))
4287 /* Extract the authentication tokens that the server provided for each method
4288 * into an array of structures and give those to the connection object.
4291 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4292 const gchar * header, gboolean * stale)
4294 GSList *list = NULL, *iter;
4296 gchar *item, *eq, *name_end, *value;
4298 g_return_if_fail (stale != NULL);
4300 gst_rtsp_connection_clear_auth_params (conn);
4303 /* Parse a header whose content is described by RFC2616 as
4304 * "#something", where "something" does not itself contain commas,
4305 * except as part of quoted-strings, into a list of allocated strings.
4307 header = gst_rtspsrc_skip_commas (header);
4309 end = gst_rtspsrc_skip_item (header);
4310 list = g_slist_prepend (list, g_strndup (header, end - header));
4311 header = gst_rtspsrc_skip_commas (end);
4316 list = g_slist_reverse (list);
4317 for (iter = list; iter; iter = iter->next) {
4320 eq = strchr (item, '=');
4322 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4323 if (name_end == item) {
4324 /* That's no good... */
4331 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4333 gst_rtsp_decode_quoted_string (value);
4337 if (item && (strcmp (item, "stale") == 0) &&
4338 value && (strcmp (value, "TRUE") == 0))
4340 gst_rtsp_connection_set_auth_param (conn, item, value);
4344 g_slist_free (list);
4347 /* Parse a WWW-Authenticate Response header and determine the
4348 * available authentication methods
4350 * This code should also cope with the fact that each WWW-Authenticate
4351 * header can contain multiple challenge methods + tokens
4353 * At the moment, for Basic auth, we just do a minimal check and don't
4354 * even parse out the realm */
4356 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4357 GstRTSPConnection * conn, gboolean * stale)
4361 g_return_if_fail (hdr != NULL);
4362 g_return_if_fail (methods != NULL);
4363 g_return_if_fail (stale != NULL);
4365 /* Skip whitespace at the start of the string */
4366 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4368 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4369 *methods |= GST_RTSP_AUTH_BASIC;
4370 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4371 *methods |= GST_RTSP_AUTH_DIGEST;
4372 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4377 * gst_rtspsrc_setup_auth:
4378 * @src: the rtsp source
4380 * Configure a username and password and auth method on the
4381 * connection object based on a response we received from the
4384 * Currently, this requires that a username and password were supplied
4385 * in the uri. In the future, they may be requested on demand by sending
4386 * a message up the bus.
4388 * Returns: TRUE if authentication information could be set up correctly.
4391 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4395 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4396 GstRTSPAuthMethod method;
4397 GstRTSPResult auth_result;
4399 GstRTSPConnection *conn;
4401 gboolean stale = FALSE;
4403 conn = src->conninfo.connection;
4405 /* Identify the available auth methods and see if any are supported */
4406 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4407 &hdr, 0) == GST_RTSP_OK) {
4408 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4411 if (avail_methods == GST_RTSP_AUTH_NONE)
4412 goto no_auth_available;
4414 /* For digest auth, if the response indicates that the session
4415 * data are stale, we just update them in the connection object and
4416 * return TRUE to retry the request */
4418 src->tried_url_auth = FALSE;
4420 url = gst_rtsp_connection_get_url (conn);
4422 /* Do we have username and password available? */
4423 if (url != NULL && !src->tried_url_auth && url->user != NULL
4424 && url->passwd != NULL) {
4427 src->tried_url_auth = TRUE;
4428 GST_DEBUG_OBJECT (src,
4429 "Attempting authentication using credentials from the URL");
4431 user = src->user_id;
4432 pass = src->user_pw;
4433 GST_DEBUG_OBJECT (src,
4434 "Attempting authentication using credentials from the properties");
4437 /* FIXME: If the url didn't contain username and password or we tried them
4438 * already, request a username and passwd from the application via some kind
4439 * of credentials request message */
4441 /* If we don't have a username and passwd at this point, bail out. */
4442 if (user == NULL || pass == NULL)
4445 /* Try to configure for each available authentication method, strongest to
4447 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
4448 /* Check if this method is available on the server */
4449 if ((method & avail_methods) == 0)
4452 /* Pass the credentials to the connection to try on the next request */
4453 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
4454 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
4455 * ignore it and end up retrying later */
4456 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
4457 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
4458 gst_rtsp_auth_method_to_string (method));
4463 if (method == GST_RTSP_AUTH_NONE)
4464 goto no_auth_available;
4470 /* Output an error indicating that we couldn't connect because there were
4471 * no supported authentication protocols */
4472 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4473 ("No supported authentication protocol was found"));
4478 /* We don't fire an error message, we just return FALSE and let the
4479 * normal NOT_AUTHORIZED error be propagated */
4484 static GstRTSPResult
4485 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4486 GstRTSPMessage * request, GstRTSPMessage * response,
4487 GstRTSPStatusCode * code)
4490 GstRTSPStatusCode thecode;
4491 gchar *content_base = NULL;
4495 if (!src->short_header)
4496 gst_rtsp_ext_list_before_send (src->extensions, request);
4498 GST_DEBUG_OBJECT (src, "sending message");
4501 gst_rtsp_message_dump (request);
4503 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
4507 gst_rtsp_connection_reset_timeout (conn);
4510 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
4515 gst_rtsp_message_dump (response);
4517 switch (response->type) {
4518 case GST_RTSP_MESSAGE_REQUEST:
4519 res = gst_rtspsrc_handle_request (src, conn, response);
4520 if (res == GST_RTSP_EEOF)
4523 goto handle_request_failed;
4525 case GST_RTSP_MESSAGE_RESPONSE:
4526 /* ok, a response is good */
4527 GST_DEBUG_OBJECT (src, "received response message");
4529 case GST_RTSP_MESSAGE_DATA:
4530 /* get next response */
4531 GST_DEBUG_OBJECT (src, "ignoring data response message");
4534 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4539 thecode = response->type_data.response.code;
4541 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
4543 /* if the caller wanted the result code, we store it. */
4547 /* If the request didn't succeed, bail out before doing any more */
4548 if (thecode != GST_RTSP_STS_OK)
4551 /* store new content base if any */
4552 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
4555 g_free (src->content_base);
4556 src->content_base = g_strdup (content_base);
4558 gst_rtsp_ext_list_after_send (src->extensions, request, response);
4565 gchar *str = gst_rtsp_strresult (res);
4567 if (res != GST_RTSP_EINTR) {
4568 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4569 ("Could not send message. (%s)", str));
4571 GST_WARNING_OBJECT (src, "send interrupted");
4580 GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
4583 /* if reconnect succeeds, try again */
4585 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
4589 /* only try once after reconnect, then fallthrough and error out */
4592 gchar *str = gst_rtsp_strresult (res);
4594 if (res != GST_RTSP_EINTR) {
4595 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4596 ("Could not receive message. (%s)", str));
4598 GST_WARNING_OBJECT (src, "receive interrupted");
4606 handle_request_failed:
4608 /* ERROR was posted */
4609 gst_rtsp_message_unset (response);
4614 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4615 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4616 ("The server closed the connection."));
4617 gst_rtsp_message_unset (response);
4624 * @src: the rtsp source
4625 * @conn: the connection to send on
4626 * @request: must point to a valid request
4627 * @response: must point to an empty #GstRTSPMessage
4628 * @code: an optional code result
4630 * send @request and retrieve the response in @response. optionally @code can be
4631 * non-NULL in which case it will contain the status code of the response.
4633 * If This function returns #GST_RTSP_OK, @response will contain a valid response
4634 * message that should be cleaned with gst_rtsp_message_unset() after usage.
4636 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
4637 * @response message) if the response code was not 200 (OK).
4639 * If the attempt results in an authentication failure, then this will attempt
4640 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
4643 * Returns: #GST_RTSP_OK if the processing was successful.
4645 static GstRTSPResult
4646 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4647 GstRTSPMessage * request, GstRTSPMessage * response,
4648 GstRTSPStatusCode * code)
4650 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
4651 GstRTSPResult res = GST_RTSP_ERROR;
4654 GstRTSPMethod method = GST_RTSP_INVALID;
4660 /* make sure we don't loop forever */
4664 /* save method so we can disable it when the server complains */
4665 method = request->type_data.request.method;
4668 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
4672 case GST_RTSP_STS_UNAUTHORIZED:
4673 if (gst_rtspsrc_setup_auth (src, response)) {
4674 /* Try the request/response again after configuring the auth info
4682 } while (retry == TRUE);
4684 /* If the user requested the code, let them handle errors, otherwise
4685 * post an error below */
4688 else if (int_code != GST_RTSP_STS_OK)
4689 goto error_response;
4696 GST_DEBUG_OBJECT (src, "got error %d", res);
4701 res = GST_RTSP_ERROR;
4703 switch (response->type_data.response.code) {
4704 case GST_RTSP_STS_NOT_FOUND:
4705 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
4706 response->type_data.response.reason));
4708 case GST_RTSP_STS_MOVED_PERMANENTLY:
4709 case GST_RTSP_STS_MOVE_TEMPORARILY:
4711 gchar *new_location;
4712 GstRTSPLowerTrans transports;
4714 GST_DEBUG_OBJECT (src, "got redirection");
4715 /* if we don't have a Location Header, we must error */
4716 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
4717 &new_location, 0) < 0)
4720 /* When we receive a redirect result, we go back to the INIT state after
4721 * parsing the new URI. The caller should do the needed steps to issue
4722 * a new setup when it detects this state change. */
4723 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
4725 /* save current transports */
4726 if (src->conninfo.url)
4727 transports = src->conninfo.url->transports;
4729 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
4731 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
4733 /* set old transports */
4734 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
4735 src->conninfo.url->transports = transports;
4737 src->need_redirect = TRUE;
4738 src->state = GST_RTSP_STATE_INIT;
4742 case GST_RTSP_STS_NOT_ACCEPTABLE:
4743 case GST_RTSP_STS_NOT_IMPLEMENTED:
4744 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
4745 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
4746 gst_rtsp_method_as_text (method));
4747 src->methods &= ~method;
4751 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4752 ("Got error response: %d (%s).", response->type_data.response.code,
4753 response->type_data.response.reason));
4756 /* if we return ERROR we should unset the response ourselves */
4757 if (res == GST_RTSP_ERROR)
4758 gst_rtsp_message_unset (response);
4764 static GstRTSPResult
4765 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
4766 GstRTSPMessage * response, GstRTSPSrc * src)
4768 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
4773 /* parse the response and collect all the supported methods. We need this
4774 * information so that we don't try to send an unsupported request to the
4778 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
4780 GstRTSPHeaderField field;
4786 /* reset supported methods */
4789 /* Try Allow Header first */
4790 field = GST_RTSP_HDR_ALLOW;
4793 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4794 if (indx == 0 && !respoptions) {
4795 /* if no Allow header was found then try the Public header... */
4796 field = GST_RTSP_HDR_PUBLIC;
4797 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4802 /* If we get here, the server gave a list of supported methods, parse
4803 * them here. The string is like:
4805 * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
4807 options = g_strsplit (respoptions, ",", 0);
4809 for (i = 0; options[i]; i++) {
4813 stripped = g_strstrip (options[i]);
4814 method = gst_rtsp_find_method (stripped);
4816 /* keep bitfield of supported methods */
4817 if (method != GST_RTSP_INVALID)
4818 src->methods |= method;
4820 g_strfreev (options);
4825 if (src->methods == 0) {
4826 /* neither Allow nor Public are required, assume the server supports
4827 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
4829 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
4830 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
4832 /* always assume PLAY, FIXME, extensions should be able to override
4834 src->methods |= GST_RTSP_PLAY;
4835 /* also assume it will support Range */
4836 src->seekable = TRUE;
4838 /* we need describe and setup */
4839 if (!(src->methods & GST_RTSP_DESCRIBE))
4841 if (!(src->methods & GST_RTSP_SETUP))
4849 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4850 ("Server does not support DESCRIBE."));
4855 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4856 ("Server does not support SETUP."));
4861 /* masks to be kept in sync with the hardcoded protocol order of preference
4863 static guint protocol_masks[] = {
4864 GST_RTSP_LOWER_TRANS_UDP,
4865 GST_RTSP_LOWER_TRANS_UDP_MCAST,
4866 GST_RTSP_LOWER_TRANS_TCP,
4870 static GstRTSPResult
4871 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
4872 GstRTSPLowerTrans protocols, gchar ** transports)
4876 gboolean add_udp_str;
4881 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
4886 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
4888 /* extension listed transports, use those */
4889 if (*transports != NULL)
4892 /* it's the default */
4893 add_udp_str = FALSE;
4895 /* the default RTSP transports */
4896 result = g_string_new ("");
4897 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
4898 GST_DEBUG_OBJECT (src, "adding UDP unicast");
4900 g_string_append (result, "RTP/AVP");
4902 g_string_append (result, "/UDP");
4903 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
4904 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
4905 GST_DEBUG_OBJECT (src, "adding UDP multicast");
4907 /* we don't have to allocate any UDP ports yet, if the selected transport
4908 * turns out to be multicast we can create them and join the multicast
4909 * group indicated in the transport reply */
4910 if (result->len > 0)
4911 g_string_append (result, ",");
4912 g_string_append (result, "RTP/AVP");
4914 g_string_append (result, "/UDP");
4915 g_string_append (result, ";multicast");
4916 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
4917 GST_DEBUG_OBJECT (src, "adding TCP");
4919 if (result->len > 0)
4920 g_string_append (result, ",");
4921 g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
4923 *transports = g_string_free (result, FALSE);
4925 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
4936 static GstRTSPResult
4937 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
4938 gint orig_rtpport, gint orig_rtcpport)
4941 gint nr_udp, nr_int;
4943 gint rtpport = 0, rtcpport = 0;
4946 src = stream->parent;
4948 /* find number of placeholders first */
4949 if (strstr (*transports, "%%i2"))
4951 else if (strstr (*transports, "%%i1"))
4956 if (strstr (*transports, "%%u2"))
4958 else if (strstr (*transports, "%%u1"))
4963 if (nr_udp == 0 && nr_int == 0)
4967 if (!orig_rtpport || !orig_rtcpport) {
4968 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
4971 rtpport = orig_rtpport;
4972 rtcpport = orig_rtcpport;
4976 str = g_string_new ("");
4978 while ((next = strstr (p, "%%"))) {
4979 g_string_append_len (str, p, next - p);
4980 if (next[2] == 'u') {
4982 g_string_append_printf (str, "%d", rtpport);
4983 else if (next[3] == '2')
4984 g_string_append_printf (str, "%d", rtcpport);
4986 if (next[2] == 'i') {
4988 g_string_append_printf (str, "%d", src->free_channel);
4989 else if (next[3] == '2')
4990 g_string_append_printf (str, "%d", src->free_channel + 1);
4995 /* append final part */
4996 g_string_append (str, p);
4998 g_free (*transports);
4999 *transports = g_string_free (str, FALSE);
5007 return GST_RTSP_ERROR;
5012 gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
5014 gboolean res = FALSE;
5018 const gchar *enc = NULL;
5020 s = gst_caps_get_structure (stream->caps, 0);
5021 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
5022 res = (strstr (enc, "-REAL") != NULL);
5028 /* Perform the SETUP request for all the streams.
5030 * We ask the server for a specific transport, which initially includes all the
5031 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
5032 * two local UDP ports that we send to the server.
5034 * Once the server replied with a transport, we configure the other streams
5035 * with the same transport.
5037 * This function will also configure the stream for the selected transport,
5038 * which basically means creating the pipeline.
5040 static GstRTSPResult
5041 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
5044 GstRTSPResult res = GST_RTSP_ERROR;
5045 GstRTSPMessage request = { 0 };
5046 GstRTSPMessage response = { 0 };
5047 GstRTSPStream *stream = NULL;
5048 GstRTSPLowerTrans protocols;
5049 GstRTSPStatusCode code;
5050 gboolean unsupported_real = FALSE;
5051 gint rtpport, rtcpport;
5055 if (src->conninfo.connection) {
5056 url = gst_rtsp_connection_get_url (src->conninfo.connection);
5057 /* we initially allow all configured lower transports. based on the URL
5058 * transports and the replies from the server we narrow them down. */
5059 protocols = url->transports & src->cur_protocols;
5062 protocols = src->cur_protocols;
5068 /* reset some state */
5069 src->free_channel = 0;
5070 src->interleaved = FALSE;
5071 src->need_activate = FALSE;
5072 /* keep track of next port number, 0 is random */
5073 src->next_port_num = src->client_port_range.min;
5074 rtpport = rtcpport = 0;
5076 if (G_UNLIKELY (src->streams == NULL))
5079 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5080 GstRTSPConnection *conn;
5085 stream = (GstRTSPStream *) walk->data;
5087 /* see if we need to configure this stream */
5088 if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
5089 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5091 stream->disabled = TRUE;
5095 /* merge/overwrite global caps */
5100 s = gst_caps_get_structure (stream->caps, 0);
5102 num = gst_structure_n_fields (src->props);
5103 for (j = 0; j < num; j++) {
5107 name = gst_structure_nth_field_name (src->props, j);
5108 val = gst_structure_get_value (src->props, name);
5109 gst_structure_set_value (s, name, val);
5111 GST_DEBUG_OBJECT (src, "copied %s", name);
5115 /* skip setup if we have no URL for it */
5116 if (stream->conninfo.location == NULL) {
5117 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5121 if (src->conninfo.connection == NULL) {
5122 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5123 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5126 conn = stream->conninfo.connection;
5128 conn = src->conninfo.connection;
5130 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5131 stream->conninfo.location);
5133 /* if we have a multicast connection, only suggest multicast from now on */
5134 if (stream->is_multicast)
5135 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5138 /* first selectable protocol */
5139 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5141 if (!protocol_masks[mask])
5145 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5146 protocol_masks[mask]);
5147 /* create a string with first transport in line */
5149 res = gst_rtspsrc_create_transports_string (src,
5150 protocols & protocol_masks[mask], &transports);
5151 if (res < 0 || transports == NULL)
5152 goto setup_transport_failed;
5154 if (strlen (transports) == 0) {
5155 g_free (transports);
5156 GST_DEBUG_OBJECT (src, "no transports found");
5161 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5163 /* replace placeholders with real values, this function will optionally
5164 * allocate UDP ports and other info needed to execute the setup request */
5165 res = gst_rtspsrc_prepare_transports (stream, &transports,
5166 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5168 g_free (transports);
5169 goto setup_transport_failed;
5172 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5174 /* create SETUP request */
5176 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5177 stream->conninfo.location);
5179 g_free (transports);
5180 goto create_request_failed;
5183 /* select transport, copy is made when adding to header so we can free it. */
5184 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5185 g_free (transports);
5187 /* if the user wants a non default RTP packet size we add the blocksize
5189 if (src->rtp_blocksize > 0) {
5190 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5191 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5196 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5199 /* handle the code ourselves */
5200 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5204 case GST_RTSP_STS_OK:
5206 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5207 gst_rtsp_message_unset (&request);
5208 gst_rtsp_message_unset (&response);
5209 /* cleanup of leftover transport */
5210 gst_rtspsrc_stream_free_udp (stream);
5211 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5212 * we might be in this case */
5213 if (stream->container && rtpport && rtcpport && !retry) {
5214 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5219 /* this transport did not go down well, but we may have others to try
5220 * that we did not send yet, try those and only give up then
5221 * but not without checking for lost cause/extension so we can
5222 * post a nicer/more useful error message later */
5223 if (!unsupported_real)
5224 unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
5225 /* select next available protocol, give up on this stream if none */
5227 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5229 if (!protocol_masks[mask] || unsupported_real)
5234 /* cleanup of leftover transport and move to the next stream */
5235 gst_rtspsrc_stream_free_udp (stream);
5236 goto response_error;
5239 /* parse response transport */
5241 gchar *resptrans = NULL;
5242 GstRTSPTransport transport = { 0 };
5244 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5247 gst_rtspsrc_stream_free_udp (stream);
5251 /* parse transport, go to next stream on parse error */
5252 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5253 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5257 /* update allowed transports for other streams. once the transport of
5258 * one stream has been determined, we make sure that all other streams
5259 * are configured in the same way */
5260 switch (transport.lower_transport) {
5261 case GST_RTSP_LOWER_TRANS_TCP:
5262 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5263 protocols = GST_RTSP_LOWER_TRANS_TCP;
5264 src->interleaved = TRUE;
5265 /* update free channels */
5267 MAX (transport.interleaved.min, src->free_channel);
5269 MAX (transport.interleaved.max, src->free_channel);
5270 src->free_channel++;
5272 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5273 /* only allow multicast for other streams */
5274 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5275 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5277 case GST_RTSP_LOWER_TRANS_UDP:
5278 /* only allow unicast for other streams */
5279 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5280 protocols = GST_RTSP_LOWER_TRANS_UDP;
5283 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5284 transport.lower_transport);
5288 if (!stream->container || (!src->interleaved && !retry)) {
5289 /* now configure the stream with the selected transport */
5290 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5291 GST_DEBUG_OBJECT (src,
5292 "could not configure stream %p transport, skipping stream",
5295 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5296 /* retain the first allocated UDP port pair */
5297 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5298 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5301 /* we need to activate at least one streams when we detect activity */
5302 src->need_activate = TRUE;
5304 /* clean up our transport struct */
5305 gst_rtsp_transport_init (&transport);
5306 /* clean up used RTSP messages */
5307 gst_rtsp_message_unset (&request);
5308 gst_rtsp_message_unset (&response);
5312 /* store the transport protocol that was configured */
5313 src->cur_protocols = protocols;
5315 gst_rtsp_ext_list_stream_select (src->extensions, url);
5317 /* if there is nothing to activate, error out */
5318 if (!src->need_activate)
5319 goto nothing_to_activate;
5326 /* no transport possible, post an error and stop */
5327 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5328 ("Could not connect to server, no protocols left"));
5329 return GST_RTSP_ERROR;
5333 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5334 ("SDP contains no streams"));
5335 return GST_RTSP_ERROR;
5337 create_request_failed:
5339 gchar *str = gst_rtsp_strresult (res);
5341 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5342 ("Could not create request. (%s)", str));
5346 setup_transport_failed:
5348 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5349 ("Could not setup transport."));
5350 res = GST_RTSP_ERROR;
5355 const gchar *str = gst_rtsp_status_as_text (code);
5357 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5358 ("Error (%d): %s", code, GST_STR_NULL (str)));
5359 res = GST_RTSP_ERROR;
5364 gchar *str = gst_rtsp_strresult (res);
5366 if (res != GST_RTSP_EINTR) {
5367 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5368 ("Could not send message. (%s)", str));
5370 GST_WARNING_OBJECT (src, "send interrupted");
5377 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5378 ("Server did not select transport."));
5379 res = GST_RTSP_ERROR;
5382 nothing_to_activate:
5384 /* none of the available error codes is really right .. */
5385 if (unsupported_real) {
5386 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5387 (_("No supported stream was found. You might need to install a "
5388 "GStreamer RTSP extension plugin for Real media streams.")),
5391 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5392 (_("No supported stream was found. You might need to allow "
5393 "more transport protocols or may otherwise be missing "
5394 "the right GStreamer RTSP extension plugin.")), (NULL));
5396 return GST_RTSP_ERROR;
5400 gst_rtsp_message_unset (&request);
5401 gst_rtsp_message_unset (&response);
5407 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
5408 GstSegment * segment)
5411 GstRTSPTimeRange *therange;
5414 gst_rtsp_range_free (src->range);
5416 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
5417 GST_DEBUG_OBJECT (src, "parsed range %s", range);
5418 src->range = therange;
5420 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
5422 gst_segment_init (segment, GST_FORMAT_TIME);
5426 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
5427 therange->min.type, therange->min.seconds, therange->max.type,
5428 therange->max.seconds);
5430 if (therange->min.type == GST_RTSP_TIME_NOW)
5432 else if (therange->min.type == GST_RTSP_TIME_END)
5435 seconds = therange->min.seconds * GST_SECOND;
5437 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
5438 GST_TIME_ARGS (seconds));
5440 /* we need to start playback without clipping from the position reported by
5442 segment->start = seconds;
5443 segment->position = seconds;
5445 if (therange->max.type == GST_RTSP_TIME_NOW)
5447 else if (therange->max.type == GST_RTSP_TIME_END)
5450 seconds = therange->max.seconds * GST_SECOND;
5452 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
5453 GST_TIME_ARGS (seconds));
5455 /* live (WMS) server might send overflowed large max as its idea of infinity,
5456 * compensate to prevent problems later on */
5457 if (seconds != -1 && seconds < 0) {
5459 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
5462 /* live (WMS) might send min == max, which is not worth recording */
5463 if (segment->duration == -1 && seconds == segment->start)
5466 /* don't change duration with unknown value, we might have a valid value
5467 * there that we want to keep. */
5469 segment->duration = seconds;
5474 /* must be called with the RTSP state lock */
5475 static GstRTSPResult
5476 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
5482 /* prepare global stream caps properties */
5484 gst_structure_remove_all_fields (src->props);
5486 src->props = gst_structure_new_empty ("RTSPProperties");
5489 gst_sdp_message_dump (sdp);
5491 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
5493 gst_segment_init (&src->segment, GST_FORMAT_TIME);
5495 /* parse range for duration reporting. */
5500 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
5504 /* keep track of the range and configure it in the segment */
5505 if (gst_rtspsrc_parse_range (src, range, &src->segment))
5509 /* try to find a global control attribute. Note that a '*' means that we should
5510 * do aggregate control with the current url (so we don't do anything and
5511 * leave the current connection as is) */
5513 const gchar *control;
5516 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
5517 if (control == NULL)
5520 /* only take fully qualified urls */
5521 if (g_str_has_prefix (control, "rtsp://"))
5525 g_free (src->conninfo.location);
5526 src->conninfo.location = g_strdup (control);
5527 /* make a connection for this, if there was a connection already, nothing
5529 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
5530 GST_ERROR_OBJECT (src, "could not connect");
5533 /* we need to keep the control url separate from the connection url because
5534 * the rules for constructing the media control url need it */
5535 g_free (src->control);
5536 src->control = g_strdup (control);
5539 /* create streams */
5540 n_streams = gst_sdp_message_medias_len (sdp);
5541 for (i = 0; i < n_streams; i++) {
5542 gst_rtspsrc_create_stream (src, sdp, i);
5545 src->state = GST_RTSP_STATE_INIT;
5548 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
5551 /* reset our state */
5552 src->need_range = TRUE;
5555 src->state = GST_RTSP_STATE_READY;
5562 GST_ERROR_OBJECT (src, "setup failed");
5563 gst_rtspsrc_cleanup (src);
5568 static GstRTSPResult
5569 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
5573 GstRTSPMessage request = { 0 };
5574 GstRTSPMessage response = { 0 };
5577 gchar *respcont = NULL;
5580 src->need_redirect = FALSE;
5582 /* can't continue without a valid url */
5583 if (G_UNLIKELY (src->conninfo.url == NULL)) {
5584 res = GST_RTSP_EINVAL;
5587 src->tried_url_auth = FALSE;
5589 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
5590 goto connect_failed;
5592 /* create OPTIONS */
5593 GST_DEBUG_OBJECT (src, "create options...");
5595 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
5596 src->conninfo.url_str);
5598 goto create_request_failed;
5601 GST_DEBUG_OBJECT (src, "send options...");
5604 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
5607 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5612 if (!gst_rtspsrc_parse_methods (src, &response))
5615 /* create DESCRIBE */
5616 GST_DEBUG_OBJECT (src, "create describe...");
5618 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
5619 src->conninfo.url_str);
5621 goto create_request_failed;
5623 /* we only accept SDP for now */
5624 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
5628 GST_DEBUG_OBJECT (src, "send describe...");
5631 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
5634 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5638 /* we only perform redirect for the describe, currently */
5639 if (src->need_redirect) {
5640 /* close connection, we don't have to send a TEARDOWN yet, ignore the
5642 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5644 gst_rtsp_message_unset (&request);
5645 gst_rtsp_message_unset (&response);
5651 /* it could be that the DESCRIBE method was not implemented */
5652 if (!src->methods & GST_RTSP_DESCRIBE)
5655 /* check if reply is SDP */
5656 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
5658 /* could not be set but since the request returned OK, we assume it
5659 * was SDP, else check it. */
5661 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
5662 goto wrong_content_type;
5665 /* get message body and parse as SDP */
5666 gst_rtsp_message_get_body (&response, &data, &size);
5667 if (data == NULL || size == 0)
5670 GST_DEBUG_OBJECT (src, "parse SDP...");
5671 gst_sdp_message_new (sdp);
5672 gst_sdp_message_parse_buffer (data, size, *sdp);
5674 /* clean up any messages */
5675 gst_rtsp_message_unset (&request);
5676 gst_rtsp_message_unset (&response);
5683 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
5684 ("No valid RTSP URL was provided"));
5689 gchar *str = gst_rtsp_strresult (res);
5691 if (res != GST_RTSP_EINTR) {
5692 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5693 ("Failed to connect. (%s)", str));
5695 GST_WARNING_OBJECT (src, "connect interrupted");
5700 create_request_failed:
5702 gchar *str = gst_rtsp_strresult (res);
5704 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5705 ("Could not create request. (%s)", str));
5711 /* Don't post a message - the rtsp_send method will have
5712 * taken care of it because we passed NULL for the response code */
5717 /* error was posted */
5718 res = GST_RTSP_ERROR;
5723 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5724 ("Server does not support SDP, got %s.", respcont));
5725 res = GST_RTSP_ERROR;
5730 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5731 ("Server can not provide an SDP."));
5732 res = GST_RTSP_ERROR;
5737 if (src->conninfo.connection) {
5738 GST_DEBUG_OBJECT (src, "free connection");
5739 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5741 gst_rtsp_message_unset (&request);
5742 gst_rtsp_message_unset (&response);
5747 static GstRTSPResult
5748 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
5753 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
5755 if (src->sdp == NULL) {
5756 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
5760 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
5765 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
5772 GST_WARNING_OBJECT (src, "can't get sdp");
5773 src->open_error = TRUE;
5778 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
5779 src->open_error = TRUE;
5784 static GstRTSPResult
5785 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
5787 GstRTSPMessage request = { 0 };
5788 GstRTSPMessage response = { 0 };
5789 GstRTSPResult res = GST_RTSP_OK;
5793 GST_DEBUG_OBJECT (src, "TEARDOWN...");
5795 if (src->state < GST_RTSP_STATE_READY) {
5796 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
5803 /* construct a control url */
5805 control = src->control;
5807 control = src->conninfo.url_str;
5809 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
5812 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5813 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5815 GstRTSPConnInfo *info;
5817 /* try aggregate control first but do non-aggregate control otherwise */
5819 setup_url = control;
5820 else if ((setup_url = stream->conninfo.location) == NULL)
5823 if (src->conninfo.connection) {
5824 info = &src->conninfo;
5825 } else if (stream->conninfo.connection) {
5826 info = &stream->conninfo;
5830 if (!info->connected)
5835 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
5837 goto create_request_failed;
5840 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
5843 gst_rtspsrc_send (src, info->connection, &request, &response,
5847 /* FIXME, parse result? */
5848 gst_rtsp_message_unset (&request);
5849 gst_rtsp_message_unset (&response);
5852 /* early exit when we did aggregate control */
5858 /* close connections */
5859 GST_DEBUG_OBJECT (src, "closing connection...");
5860 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5861 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5862 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5863 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
5867 gst_rtspsrc_cleanup (src);
5869 src->state = GST_RTSP_STATE_INVALID;
5872 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
5877 create_request_failed:
5879 gchar *str = gst_rtsp_strresult (res);
5881 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5882 ("Could not create request. (%s)", str));
5888 gchar *str = gst_rtsp_strresult (res);
5890 gst_rtsp_message_unset (&request);
5891 if (res != GST_RTSP_EINTR) {
5892 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5893 ("Could not send message. (%s)", str));
5895 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
5902 GST_DEBUG_OBJECT (src,
5903 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
5908 /* RTP-Info is of the format:
5910 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
5912 * rtptime corresponds to the timestamp for the NPT time given in the header
5913 * seqbase corresponds to the next sequence number we received. This number
5914 * indicates the first seqnum after the seek and should be used to discard
5915 * packets that are from before the seek.
5918 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
5923 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
5925 infos = g_strsplit (rtpinfo, ",", 0);
5926 for (i = 0; infos[i]; i++) {
5928 GstRTSPStream *stream;
5932 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
5934 /* init values, types of seqbase and timebase are bigger than needed so we
5935 * can store -1 as uninitialized values */
5940 /* parse url, find stream for url.
5941 * parse seq and rtptime. The seq number should be configured in the rtp
5942 * depayloader or session manager to detect gaps. Same for the rtptime, it
5943 * should be used to create an initial time newsegment. */
5944 fields = g_strsplit (infos[i], ";", 0);
5945 for (j = 0; fields[j]; j++) {
5946 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
5947 /* remove leading whitespace */
5948 fields[j] = g_strchug (fields[j]);
5949 if (g_str_has_prefix (fields[j], "url=")) {
5950 /* get the url and the stream */
5952 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
5953 } else if (g_str_has_prefix (fields[j], "seq=")) {
5954 seqbase = atoi (fields[j] + 4);
5955 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
5956 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
5959 g_strfreev (fields);
5960 /* now we need to store the values for the caps of the stream */
5961 if (stream != NULL) {
5962 GST_DEBUG_OBJECT (src,
5963 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
5964 stream, seqbase, timebase);
5966 /* we have a stream, configure detected params */
5967 stream->seqbase = seqbase;
5968 stream->timebase = timebase;
5977 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
5982 interval = strtoul (rtcp, NULL, 10);
5983 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
5988 interval *= GST_MSECOND;
5990 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5991 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5993 /* already (optionally) retrieved this when configuring manager */
5994 if (stream->session) {
5995 GObject *rtpsession = stream->session;
5997 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
5999 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
6003 /* now it happens that (Xenon) server sending this may also provide bogus
6004 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
6005 * and just use RTP-Info to sync */
6007 GObjectClass *klass;
6009 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
6010 if (g_object_class_find_property (klass, "rtcp-sync")) {
6011 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
6012 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
6018 gst_rtspsrc_get_float (const gchar * dstr)
6020 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6022 /* canonicalise floating point string so we can handle float strings
6023 * in the form "24.930" or "24,930" irrespective of the current locale */
6024 g_strlcpy (s, dstr, sizeof (s));
6025 g_strdelimit (s, ",", '.');
6026 return g_ascii_strtod (s, NULL);
6030 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
6032 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6034 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
6035 g_strlcpy (val_str, "now", sizeof (val_str));
6037 if (segment->position == 0) {
6038 g_strlcpy (val_str, "0", sizeof (val_str));
6040 g_ascii_dtostr (val_str, sizeof (val_str),
6041 ((gdouble) segment->position) / GST_SECOND);
6044 return g_strdup_printf ("npt=%s-", val_str);
6048 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
6050 stream->timebase = -1;
6051 stream->seqbase = -1;
6055 stream->caps = gst_caps_make_writable (stream->caps);
6056 s = gst_caps_get_structure (stream->caps, 0);
6057 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
6061 static GstRTSPResult
6062 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
6064 GstRTSPResult res = GST_RTSP_OK;
6066 if (src->state < GST_RTSP_STATE_READY) {
6067 res = GST_RTSP_ERROR;
6068 if (src->open_error) {
6069 GST_DEBUG_OBJECT (src, "the stream was in error");
6073 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
6075 if ((res = gst_rtspsrc_open (src, async)) < 0) {
6076 GST_DEBUG_OBJECT (src, "failed to open stream");
6085 static GstRTSPResult
6086 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6088 GstRTSPMessage request = { 0 };
6089 GstRTSPMessage response = { 0 };
6090 GstRTSPResult res = GST_RTSP_OK;
6096 GST_DEBUG_OBJECT (src, "PLAY...");
6098 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6101 if (!(src->methods & GST_RTSP_PLAY))
6104 if (src->state == GST_RTSP_STATE_PLAYING)
6107 if (!src->conninfo.connection || !src->conninfo.connected)
6110 /* send some dummy packets before we activate the receive in the
6112 gst_rtspsrc_send_dummy_packets (src);
6114 /* activate receive elements;
6115 * only in async case, since receive elements may not have been affected
6116 * by overall state change (e.g. not around yet),
6117 * do not mess with state in sync case (e.g. seeking) */
6119 gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
6121 /* construct a control url */
6123 control = src->control;
6125 control = src->conninfo.url_str;
6127 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6128 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6130 GstRTSPConnection *conn;
6132 /* try aggregate control first but do non-aggregate control otherwise */
6134 setup_url = control;
6135 else if ((setup_url = stream->conninfo.location) == NULL)
6138 if (src->conninfo.connection) {
6139 conn = src->conninfo.connection;
6140 } else if (stream->conninfo.connection) {
6141 conn = stream->conninfo.connection;
6147 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6149 goto create_request_failed;
6151 if (src->need_range) {
6152 hval = gen_range_header (src, segment);
6154 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
6158 if (segment->rate != 1.0) {
6159 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6161 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6163 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6165 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6169 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6171 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6174 /* seek may have silently failed as it is not supported */
6175 if (!(src->methods & GST_RTSP_PLAY)) {
6176 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6177 /* obviously it is supported as we made it here */
6178 src->methods |= GST_RTSP_PLAY;
6179 src->seekable = FALSE;
6180 /* but there is nothing to parse in the response,
6181 * so convey we have no idea and not to expect anything particular */
6182 clear_rtp_base (src, stream);
6186 /* need to do for all streams */
6187 for (run = src->streams; run; run = g_list_next (run))
6188 clear_rtp_base (src, (GstRTSPStream *) run->data);
6190 /* NOTE the above also disables npt based eos detection */
6191 /* and below forces position to 0,
6192 * which is visible feedback we lost the plot */
6193 segment->start = segment->position = src->last_pos;
6196 gst_rtsp_message_unset (&request);
6198 /* parse RTP npt field. This is the current position in the stream (Normal
6199 * Play Time) and should be put in the NEWSEGMENT position field. */
6200 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6202 gst_rtspsrc_parse_range (src, hval, segment);
6204 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6205 segment->rate = 1.0;
6207 /* parse Speed header. This is the intended playback rate of the stream
6208 * and should be put in the NEWSEGMENT rate field. */
6209 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6210 0) == GST_RTSP_OK) {
6211 segment->rate = gst_rtspsrc_get_float (hval);
6212 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6213 &hval, 0) == GST_RTSP_OK) {
6214 segment->rate = gst_rtspsrc_get_float (hval);
6217 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6218 * for the RTP packets. If this is not present, we assume all starts from 0...
6219 * This is info for the RTP session manager that we pass to it in caps. */
6221 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6222 &hval, hval_idx++) == GST_RTSP_OK)
6223 gst_rtspsrc_parse_rtpinfo (src, hval);
6225 /* some servers indicate RTCP parameters in PLAY response,
6226 * rather than properly in SDP */
6227 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6228 &hval, 0) == GST_RTSP_OK)
6229 gst_rtspsrc_handle_rtcp_interval (src, hval);
6231 gst_rtsp_message_unset (&response);
6233 /* early exit when we did aggregate control */
6237 /* configure the caps of the streams after we parsed all headers. Only reset
6238 * the manager object when we set a new Range header (we did a seek) */
6239 gst_rtspsrc_configure_caps (src, segment, src->need_range);
6241 /* set again when needed */
6242 src->need_range = FALSE;
6244 src->running = TRUE;
6245 src->base_time = -1;
6246 src->state = GST_RTSP_STATE_PLAYING;
6249 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6250 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6251 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6252 stream->discont = TRUE;
6257 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6264 GST_DEBUG_OBJECT (src, "failed to open stream");
6269 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6274 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6277 create_request_failed:
6279 gchar *str = gst_rtsp_strresult (res);
6281 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6282 ("Could not create request. (%s)", str));
6288 gchar *str = gst_rtsp_strresult (res);
6290 gst_rtsp_message_unset (&request);
6291 if (res != GST_RTSP_EINTR) {
6292 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6293 ("Could not send message. (%s)", str));
6295 GST_WARNING_OBJECT (src, "PLAY interrupted");
6302 static GstRTSPResult
6303 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
6305 GstRTSPResult res = GST_RTSP_OK;
6306 GstRTSPMessage request = { 0 };
6307 GstRTSPMessage response = { 0 };
6311 GST_DEBUG_OBJECT (src, "PAUSE...");
6313 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6316 if (!(src->methods & GST_RTSP_PAUSE))
6319 if (src->state == GST_RTSP_STATE_READY)
6322 if (!src->conninfo.connection || !src->conninfo.connected)
6325 /* construct a control url */
6327 control = src->control;
6329 control = src->conninfo.url_str;
6331 /* loop over the streams. We might exit the loop early when we could do an
6332 * aggregate control */
6333 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6334 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6335 GstRTSPConnection *conn;
6338 /* try aggregate control first but do non-aggregate control otherwise */
6340 setup_url = control;
6341 else if ((setup_url = stream->conninfo.location) == NULL)
6344 if (src->conninfo.connection) {
6345 conn = src->conninfo.connection;
6346 } else if (stream->conninfo.connection) {
6347 conn = stream->conninfo.connection;
6353 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
6354 ("Sending PAUSE request"));
6357 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
6359 goto create_request_failed;
6361 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6364 gst_rtsp_message_unset (&request);
6365 gst_rtsp_message_unset (&response);
6367 /* exit early when we did agregate control */
6373 src->state = GST_RTSP_STATE_READY;
6377 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
6384 GST_DEBUG_OBJECT (src, "failed to open stream");
6389 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
6394 GST_DEBUG_OBJECT (src, "we were already PAUSED");
6397 create_request_failed:
6399 gchar *str = gst_rtsp_strresult (res);
6401 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6402 ("Could not create request. (%s)", str));
6408 gchar *str = gst_rtsp_strresult (res);
6410 gst_rtsp_message_unset (&request);
6411 if (res != GST_RTSP_EINTR) {
6412 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6413 ("Could not send message. (%s)", str));
6415 GST_WARNING_OBJECT (src, "PAUSE interrupted");
6423 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
6425 GstRTSPSrc *rtspsrc;
6427 rtspsrc = GST_RTSPSRC (bin);
6429 switch (GST_MESSAGE_TYPE (message)) {
6430 case GST_MESSAGE_EOS:
6431 gst_message_unref (message);
6433 case GST_MESSAGE_ELEMENT:
6435 const GstStructure *s = gst_message_get_structure (message);
6437 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
6438 gboolean ignore_timeout;
6440 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
6442 GST_OBJECT_LOCK (rtspsrc);
6443 ignore_timeout = rtspsrc->ignore_timeout;
6444 rtspsrc->ignore_timeout = TRUE;
6445 GST_OBJECT_UNLOCK (rtspsrc);
6447 /* we only act on the first udp timeout message, others are irrelevant
6448 * and can be ignored. */
6449 if (!ignore_timeout)
6450 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
6452 gst_message_unref (message);
6455 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6458 case GST_MESSAGE_ERROR:
6461 GstRTSPStream *stream;
6464 udpsrc = GST_MESSAGE_SRC (message);
6466 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
6467 GST_ELEMENT_NAME (udpsrc));
6469 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
6473 /* we ignore the RTCP udpsrc */
6474 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
6477 /* if we get error messages from the udp sources, that's not a problem as
6478 * long as not all of them error out. We also don't really know what the
6479 * problem is, the message does not give enough detail... */
6480 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
6481 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
6482 if (ret != GST_FLOW_OK)
6486 gst_message_unref (message);
6490 /* fatal but not our message, forward */
6491 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6496 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6502 /* the thread where everything happens */
6504 gst_rtspsrc_thread (GstRTSPSrc * src)
6508 GST_OBJECT_LOCK (src);
6509 cmd = src->pending_cmd;
6510 if (cmd == CMD_PLAY || cmd == CMD_LOOP)
6511 src->pending_cmd = CMD_LOOP;
6513 src->pending_cmd = CMD_WAIT;
6514 GST_DEBUG_OBJECT (src, "got command %d", cmd);
6516 /* we got the message command, so ensure communication is possible again */
6517 gst_rtspsrc_connection_flush (src, FALSE);
6519 src->busy_cmd = cmd;
6520 GST_OBJECT_UNLOCK (src);
6524 gst_rtspsrc_open (src, TRUE);
6527 gst_rtspsrc_play (src, &src->segment, TRUE);
6530 gst_rtspsrc_pause (src, TRUE);
6533 gst_rtspsrc_close (src, TRUE, FALSE);
6536 gst_rtspsrc_loop (src);
6539 gst_rtspsrc_reconnect (src, FALSE);
6545 GST_OBJECT_LOCK (src);
6546 /* and go back to sleep */
6547 if (src->pending_cmd == CMD_WAIT) {
6549 gst_task_pause (src->task);
6552 src->busy_cmd = CMD_WAIT;
6553 GST_OBJECT_UNLOCK (src);
6557 gst_rtspsrc_start (GstRTSPSrc * src)
6559 GST_DEBUG_OBJECT (src, "starting");
6561 GST_OBJECT_LOCK (src);
6563 src->pending_cmd = CMD_WAIT;
6565 if (src->task == NULL) {
6566 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
6567 if (src->task == NULL)
6570 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
6572 GST_OBJECT_UNLOCK (src);
6579 GST_ERROR_OBJECT (src, "failed to create task");
6585 gst_rtspsrc_stop (GstRTSPSrc * src)
6589 GST_DEBUG_OBJECT (src, "stopping");
6591 /* also cancels pending task */
6592 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_CLOSE);
6594 GST_OBJECT_LOCK (src);
6595 if ((task = src->task)) {
6597 GST_OBJECT_UNLOCK (src);
6599 gst_task_stop (task);
6601 /* make sure it is not running */
6602 GST_RTSP_STREAM_LOCK (src);
6603 GST_RTSP_STREAM_UNLOCK (src);
6605 /* now wait for the task to finish */
6606 gst_task_join (task);
6608 /* and free the task */
6609 gst_object_unref (GST_OBJECT (task));
6611 GST_OBJECT_LOCK (src);
6613 GST_OBJECT_UNLOCK (src);
6615 /* ensure synchronously all is closed and clean */
6616 gst_rtspsrc_close (src, FALSE, TRUE);
6621 static GstStateChangeReturn
6622 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
6624 GstRTSPSrc *rtspsrc;
6625 GstStateChangeReturn ret;
6627 rtspsrc = GST_RTSPSRC (element);
6629 switch (transition) {
6630 case GST_STATE_CHANGE_NULL_TO_READY:
6631 if (!gst_rtspsrc_start (rtspsrc))
6634 case GST_STATE_CHANGE_READY_TO_PAUSED:
6635 /* init some state */
6636 rtspsrc->cur_protocols = rtspsrc->protocols;
6637 /* first attempt, don't ignore timeouts */
6638 rtspsrc->ignore_timeout = FALSE;
6639 rtspsrc->open_error = FALSE;
6640 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
6642 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6643 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6644 /* unblock the tcp tasks and make the loop waiting */
6645 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP);
6647 case GST_STATE_CHANGE_PAUSED_TO_READY:
6653 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
6654 if (ret == GST_STATE_CHANGE_FAILURE)
6657 switch (transition) {
6658 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6659 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
6661 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6662 /* send pause request and keep the idle task around */
6663 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
6664 ret = GST_STATE_CHANGE_NO_PREROLL;
6666 case GST_STATE_CHANGE_READY_TO_PAUSED:
6667 ret = GST_STATE_CHANGE_NO_PREROLL;
6669 case GST_STATE_CHANGE_PAUSED_TO_READY:
6670 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
6672 case GST_STATE_CHANGE_READY_TO_NULL:
6673 gst_rtspsrc_stop (rtspsrc);
6684 GST_DEBUG_OBJECT (rtspsrc, "start failed");
6685 return GST_STATE_CHANGE_FAILURE;
6690 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
6693 GstRTSPSrc *rtspsrc;
6695 rtspsrc = GST_RTSPSRC (element);
6697 if (GST_EVENT_IS_DOWNSTREAM (event)) {
6698 res = gst_rtspsrc_push_event (rtspsrc, event);
6700 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
6707 /*** GSTURIHANDLER INTERFACE *************************************************/
6710 gst_rtspsrc_uri_get_type (GType type)
6715 static const gchar *const *
6716 gst_rtspsrc_uri_get_protocols (GType type)
6718 static const gchar *protocols[] =
6719 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
6725 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
6727 GstRTSPSrc *src = GST_RTSPSRC (handler);
6729 /* FIXME: make thread-safe */
6730 return g_strdup (src->conninfo.location);
6734 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
6739 GstRTSPUrl *newurl = NULL;
6740 GstSDPMessage *sdp = NULL;
6742 src = GST_RTSPSRC (handler);
6744 /* same URI, we're fine */
6745 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
6748 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
6749 if ((res = gst_sdp_message_new (&sdp) < 0))
6752 GST_DEBUG_OBJECT (src, "parsing SDP message");
6753 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
6757 GST_DEBUG_OBJECT (src, "parsing URI");
6758 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
6762 /* if worked, free previous and store new url object along with the original
6764 GST_DEBUG_OBJECT (src, "configuring URI");
6765 g_free (src->conninfo.location);
6766 src->conninfo.location = g_strdup (uri);
6767 gst_rtsp_url_free (src->conninfo.url);
6768 src->conninfo.url = newurl;
6769 g_free (src->conninfo.url_str);
6771 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
6773 src->conninfo.url_str = NULL;
6776 gst_sdp_message_free (src->sdp);
6778 src->from_sdp = sdp != NULL;
6780 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
6781 GST_DEBUG_OBJECT (src, "request uri is: %s",
6782 GST_STR_NULL (src->conninfo.url_str));
6789 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
6794 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
6795 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6796 "Could not create SDP");
6801 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
6802 GST_STR_NULL (uri));
6803 gst_sdp_message_free (sdp);
6804 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6810 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
6811 GST_STR_NULL (uri), res);
6812 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6813 "Invalid RTSP URI");
6819 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
6821 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
6823 iface->get_type = gst_rtspsrc_uri_get_type;
6824 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
6825 iface->get_uri = gst_rtspsrc_uri_get_uri;
6826 iface->set_uri = gst_rtspsrc_uri_set_uri;