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., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, 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 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 */
93 #include <gst/sdp/gstsdpmessage.h>
94 #include <gst/rtp/gstrtppayloads.h>
96 #include "gst/gst-i18n-plugin.h"
98 #include "gstrtspsrc.h"
100 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
101 #define GST_CAT_DEFAULT (rtspsrc_debug)
103 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
106 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
108 /* templates used internally */
109 static GstStaticPadTemplate anysrctemplate =
110 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
113 GST_STATIC_CAPS_ANY);
115 static GstStaticPadTemplate anysinktemplate =
116 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
119 GST_STATIC_CAPS_ANY);
127 enum _GstRtspSrcRtcpSyncMode
134 enum _GstRtspSrcBufferMode
142 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
144 gst_rtsp_src_buffer_mode_get_type (void)
146 static GType buffer_mode_type = 0;
147 static const GEnumValue buffer_modes[] = {
148 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
149 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
150 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
151 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
155 if (!buffer_mode_type) {
157 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
159 return buffer_mode_type;
162 #define DEFAULT_LOCATION NULL
163 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
164 #define DEFAULT_DEBUG FALSE
165 #define DEFAULT_RETRY 20
166 #define DEFAULT_TIMEOUT 5000000
167 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
168 #define DEFAULT_TCP_TIMEOUT 20000000
169 #define DEFAULT_LATENCY_MS 2000
170 #define DEFAULT_CONNECTION_SPEED 0
171 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
172 #define DEFAULT_DO_RTCP TRUE
173 #define DEFAULT_PROXY NULL
174 #define DEFAULT_RTP_BLOCKSIZE 0
175 #define DEFAULT_USER_ID NULL
176 #define DEFAULT_USER_PW NULL
177 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
178 #define DEFAULT_PORT_RANGE NULL
179 #define DEFAULT_SHORT_HEADER FALSE
191 PROP_CONNECTION_SPEED,
200 PROP_UDP_BUFFER_SIZE,
205 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
207 gst_rtsp_nat_method_get_type (void)
209 static GType rtsp_nat_method_type = 0;
210 static const GEnumValue rtsp_nat_method[] = {
211 {GST_RTSP_NAT_NONE, "None", "none"},
212 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
216 if (!rtsp_nat_method_type) {
217 rtsp_nat_method_type =
218 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
220 return rtsp_nat_method_type;
223 static void gst_rtspsrc_finalize (GObject * object);
225 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
226 const GValue * value, GParamSpec * pspec);
227 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
228 GValue * value, GParamSpec * pspec);
230 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
231 gpointer iface_data);
233 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
236 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
237 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
239 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
241 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
242 GstStateChange transition);
243 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
244 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
246 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
247 GstRTSPMessage * response);
249 static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd);
250 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
251 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
253 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
254 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
256 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
258 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
259 gboolean only_close);
261 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
262 const gchar * uri, GError ** error);
264 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
265 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
266 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
267 GstRTSPStream * stream, GstEvent * event, gboolean source);
268 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
271 /* commands we send to out loop to notify it of events */
277 #define CMD_RECONNECT 5
280 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
282 gchar *__txt = _gst_element_error_printf text; \
283 gst_element_post_message (GST_ELEMENT_CAST (el), \
284 gst_message_new_progress (GST_OBJECT_CAST (el), \
285 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
289 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
290 #define gst_rtspsrc_parent_class parent_class
291 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
292 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
295 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
297 GObjectClass *gobject_class;
298 GstElementClass *gstelement_class;
299 GstBinClass *gstbin_class;
301 gobject_class = (GObjectClass *) klass;
302 gstelement_class = (GstElementClass *) klass;
303 gstbin_class = (GstBinClass *) klass;
305 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
307 gobject_class->set_property = gst_rtspsrc_set_property;
308 gobject_class->get_property = gst_rtspsrc_get_property;
310 gobject_class->finalize = gst_rtspsrc_finalize;
312 g_object_class_install_property (gobject_class, PROP_LOCATION,
313 g_param_spec_string ("location", "RTSP Location",
314 "Location of the RTSP url to read",
315 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
317 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
318 g_param_spec_flags ("protocols", "Protocols",
319 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
320 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
322 g_object_class_install_property (gobject_class, PROP_DEBUG,
323 g_param_spec_boolean ("debug", "Debug",
324 "Dump request and response messages to stdout",
325 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
327 g_object_class_install_property (gobject_class, PROP_RETRY,
328 g_param_spec_uint ("retry", "Retry",
329 "Max number of retries when allocating RTP ports.",
330 0, G_MAXUINT16, DEFAULT_RETRY,
331 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
333 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
334 g_param_spec_uint64 ("timeout", "Timeout",
335 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
336 0, G_MAXUINT64, DEFAULT_TIMEOUT,
337 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
339 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
340 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
341 "Fail after timeout microseconds on TCP connections (0 = disabled)",
342 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
343 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 g_object_class_install_property (gobject_class, PROP_LATENCY,
346 g_param_spec_uint ("latency", "Buffer latency in ms",
347 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
348 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
350 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
351 g_param_spec_uint64 ("connection-speed", "Connection Speed",
352 "Network connection speed in kbps (0 = unknown)",
353 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
354 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
356 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
357 g_param_spec_enum ("nat-method", "NAT Method",
358 "Method to use for traversing firewalls and NAT",
359 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
360 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
363 * GstRTSPSrc::do-rtcp
365 * Enable RTCP support. Some old server don't like RTCP and then this property
366 * needs to be set to FALSE.
370 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
371 g_param_spec_boolean ("do-rtcp", "Do RTCP",
372 "Send RTCP packets, disable for old incompatible server.",
373 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 * Set the proxy parameters. This has to be a string of the format
379 * [http://][user:passwd@]host[:port].
383 g_object_class_install_property (gobject_class, PROP_PROXY,
384 g_param_spec_string ("proxy", "Proxy",
385 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
386 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
389 * GstRTSPSrc::rtp_blocksize
391 * RTP package size to suggest to server.
395 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
396 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
397 "RTP package size to suggest to server (0 = disabled)",
398 0, 65536, DEFAULT_RTP_BLOCKSIZE,
399 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
401 g_object_class_install_property (gobject_class,
403 g_param_spec_string ("user-id", "user-id",
404 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
405 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 g_object_class_install_property (gobject_class, PROP_USER_PW,
407 g_param_spec_string ("user-pw", "user-pw",
408 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
409 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
412 * GstRTSPSrc::buffer-mode:
414 * Control the buffering and timestamping mode used by the jitterbuffer.
418 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
419 g_param_spec_enum ("buffer-mode", "Buffer Mode",
420 "Control the buffering algorithm in use",
421 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
422 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
425 * GstRTSPSrc::port-range:
427 * Configure the client port numbers that can be used to recieve RTP and
432 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
433 g_param_spec_string ("port-range", "Port range",
434 "Client port range that can be used to receive RTP and RTCP data, "
435 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
436 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
439 * GstRTSPSrc::udp-buffer-size:
441 * Size of the kernel UDP receive buffer in bytes.
445 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
446 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
447 "Size of the kernel UDP receive buffer in bytes, 0=default",
448 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
452 * GstRTSPSrc::short-header:
454 * Only send the basic RTSP headers for broken encoders.
458 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
459 g_param_spec_boolean ("short-header", "Short Header",
460 "Only send the basic RTSP headers for broken encoders",
461 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463 gstelement_class->send_event = gst_rtspsrc_send_event;
464 gstelement_class->change_state = gst_rtspsrc_change_state;
466 gst_element_class_add_pad_template (gstelement_class,
467 gst_static_pad_template_get (&rtptemplate));
469 gst_element_class_set_details_simple (gstelement_class,
470 "RTSP packet receiver", "Source/Network",
471 "Receive data over the network via RTSP (RFC 2326)",
472 "Wim Taymans <wim@fluendo.com>, "
473 "Thijs Vermeir <thijs.vermeir@barco.com>, "
474 "Lutz Mueller <lutz@topfrose.de>");
476 gstbin_class->handle_message = gst_rtspsrc_handle_message;
478 gst_rtsp_ext_list_init ();
483 gst_rtspsrc_init (GstRTSPSrc * src)
485 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
486 src->protocols = DEFAULT_PROTOCOLS;
487 src->debug = DEFAULT_DEBUG;
488 src->retry = DEFAULT_RETRY;
489 src->udp_timeout = DEFAULT_TIMEOUT;
490 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
491 src->latency = DEFAULT_LATENCY_MS;
492 src->connection_speed = DEFAULT_CONNECTION_SPEED;
493 src->nat_method = DEFAULT_NAT_METHOD;
494 src->do_rtcp = DEFAULT_DO_RTCP;
495 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
496 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
497 src->user_id = g_strdup (DEFAULT_USER_ID);
498 src->user_pw = g_strdup (DEFAULT_USER_PW);
499 src->buffer_mode = DEFAULT_BUFFER_MODE;
500 src->client_port_range.min = 0;
501 src->client_port_range.max = 0;
502 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
503 src->short_header = DEFAULT_SHORT_HEADER;
505 /* get a list of all extensions */
506 src->extensions = gst_rtsp_ext_list_get ();
508 /* connect to send signal */
509 gst_rtsp_ext_list_connect (src->extensions, "send",
510 (GCallback) gst_rtspsrc_send_cb, src);
512 /* protects the streaming thread in interleaved mode or the polling
513 * thread in UDP mode. */
514 g_rec_mutex_init (&src->stream_rec_lock);
516 /* protects our state changes from multiple invocations */
517 g_rec_mutex_init (&src->state_rec_lock);
519 src->state = GST_RTSP_STATE_INVALID;
521 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
525 gst_rtspsrc_finalize (GObject * object)
529 rtspsrc = GST_RTSPSRC (object);
531 gst_rtsp_ext_list_free (rtspsrc->extensions);
532 g_free (rtspsrc->conninfo.location);
533 gst_rtsp_url_free (rtspsrc->conninfo.url);
534 g_free (rtspsrc->conninfo.url_str);
535 g_free (rtspsrc->user_id);
536 g_free (rtspsrc->user_pw);
539 gst_sdp_message_free (rtspsrc->sdp);
544 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
545 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
547 G_OBJECT_CLASS (parent_class)->finalize (object);
550 /* a proxy string of the format [user:passwd@]host[:port] */
552 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
556 g_free (rtsp->proxy_user);
557 rtsp->proxy_user = NULL;
558 g_free (rtsp->proxy_passwd);
559 rtsp->proxy_passwd = NULL;
560 g_free (rtsp->proxy_host);
561 rtsp->proxy_host = NULL;
562 rtsp->proxy_port = 0;
569 /* we allow http:// in front but ignore it */
570 if (g_str_has_prefix (p, "http://"))
573 at = strchr (p, '@');
575 /* look for user:passwd */
576 col = strchr (proxy, ':');
577 if (col == NULL || col > at)
580 rtsp->proxy_user = g_strndup (p, col - p);
582 rtsp->proxy_passwd = g_strndup (col, at - col);
587 col = strchr (p, ':');
590 /* everything before the colon is the hostname */
591 rtsp->proxy_host = g_strndup (p, col - p);
593 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
595 rtsp->proxy_host = g_strdup (p);
596 rtsp->proxy_port = 8080;
602 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
604 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
605 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
608 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
610 rtspsrc->ptcp_timeout = NULL;
614 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
619 rtspsrc = GST_RTSPSRC (object);
623 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
624 g_value_get_string (value), NULL);
627 rtspsrc->protocols = g_value_get_flags (value);
630 rtspsrc->debug = g_value_get_boolean (value);
633 rtspsrc->retry = g_value_get_uint (value);
636 rtspsrc->udp_timeout = g_value_get_uint64 (value);
638 case PROP_TCP_TIMEOUT:
639 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
642 rtspsrc->latency = g_value_get_uint (value);
644 case PROP_CONNECTION_SPEED:
645 rtspsrc->connection_speed = g_value_get_uint64 (value);
647 case PROP_NAT_METHOD:
648 rtspsrc->nat_method = g_value_get_enum (value);
651 rtspsrc->do_rtcp = g_value_get_boolean (value);
654 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
656 case PROP_RTP_BLOCKSIZE:
657 rtspsrc->rtp_blocksize = g_value_get_uint (value);
660 if (rtspsrc->user_id)
661 g_free (rtspsrc->user_id);
662 rtspsrc->user_id = g_value_dup_string (value);
665 if (rtspsrc->user_pw)
666 g_free (rtspsrc->user_pw);
667 rtspsrc->user_pw = g_value_dup_string (value);
669 case PROP_BUFFER_MODE:
670 rtspsrc->buffer_mode = g_value_get_enum (value);
672 case PROP_PORT_RANGE:
676 str = g_value_get_string (value);
678 sscanf (str, "%u-%u",
679 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
681 rtspsrc->client_port_range.min = 0;
682 rtspsrc->client_port_range.max = 0;
686 case PROP_UDP_BUFFER_SIZE:
687 rtspsrc->udp_buffer_size = g_value_get_int (value);
689 case PROP_SHORT_HEADER:
690 rtspsrc->short_header = g_value_get_boolean (value);
693 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
699 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
704 rtspsrc = GST_RTSPSRC (object);
708 g_value_set_string (value, rtspsrc->conninfo.location);
711 g_value_set_flags (value, rtspsrc->protocols);
714 g_value_set_boolean (value, rtspsrc->debug);
717 g_value_set_uint (value, rtspsrc->retry);
720 g_value_set_uint64 (value, rtspsrc->udp_timeout);
722 case PROP_TCP_TIMEOUT:
726 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
727 rtspsrc->tcp_timeout.tv_usec;
728 g_value_set_uint64 (value, timeout);
732 g_value_set_uint (value, rtspsrc->latency);
734 case PROP_CONNECTION_SPEED:
735 g_value_set_uint64 (value, rtspsrc->connection_speed);
737 case PROP_NAT_METHOD:
738 g_value_set_enum (value, rtspsrc->nat_method);
741 g_value_set_boolean (value, rtspsrc->do_rtcp);
747 if (rtspsrc->proxy_host) {
749 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
753 g_value_take_string (value, str);
756 case PROP_RTP_BLOCKSIZE:
757 g_value_set_uint (value, rtspsrc->rtp_blocksize);
760 g_value_set_string (value, rtspsrc->user_id);
763 g_value_set_string (value, rtspsrc->user_pw);
765 case PROP_BUFFER_MODE:
766 g_value_set_enum (value, rtspsrc->buffer_mode);
768 case PROP_PORT_RANGE:
772 if (rtspsrc->client_port_range.min != 0) {
773 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
774 rtspsrc->client_port_range.max);
778 g_value_take_string (value, str);
781 case PROP_UDP_BUFFER_SIZE:
782 g_value_set_int (value, rtspsrc->udp_buffer_size);
784 case PROP_SHORT_HEADER:
785 g_value_set_boolean (value, rtspsrc->short_header);
788 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
794 find_stream_by_id (GstRTSPStream * stream, gint * id)
796 if (stream->id == *id)
803 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
805 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
812 find_stream_by_pt (GstRTSPStream * stream, gint * pt)
814 if (stream->pt == *pt)
821 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
823 GstElement *src = (GstElement *) a;
825 if (stream->udpsrc[0] == src)
827 if (stream->udpsrc[1] == src)
834 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
836 /* check qualified setup_url */
837 if (!strcmp (stream->conninfo.location, (gchar *) a))
839 /* check original control_url */
840 if (!strcmp (stream->control_url, (gchar *) a))
843 /* check if qualified setup_url ends with string */
844 if (g_str_has_suffix (stream->control_url, (gchar *) a))
850 static GstRTSPStream *
851 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
855 /* find and get stream */
856 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
857 return (GstRTSPStream *) lstream->data;
862 static const GstSDPBandwidth *
863 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
864 const GstSDPMedia * media, const gchar * type)
868 /* first look in the media specific section */
869 len = gst_sdp_media_bandwidths_len (media);
870 for (i = 0; i < len; i++) {
871 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
873 if (strcmp (bw->bwtype, type) == 0)
876 /* then look in the message specific section */
877 len = gst_sdp_message_bandwidths_len (sdp);
878 for (i = 0; i < len; i++) {
879 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
881 if (strcmp (bw->bwtype, type) == 0)
888 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
889 const GstSDPMedia * media, GstRTSPStream * stream)
891 const GstSDPBandwidth *bw;
893 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
894 stream->as_bandwidth = bw->bandwidth;
896 stream->as_bandwidth = -1;
898 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
899 stream->rr_bandwidth = bw->bandwidth;
901 stream->rr_bandwidth = -1;
903 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
904 stream->rs_bandwidth = bw->bandwidth;
906 stream->rs_bandwidth = -1;
910 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
911 const GstSDPConnection * conn)
913 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
916 if (conn->addrtype == NULL)
920 if (strcmp (conn->addrtype, "IP4") == 0)
921 stream->is_ipv6 = FALSE;
922 else if (strcmp (conn->addrtype, "IP6") == 0)
923 stream->is_ipv6 = TRUE;
928 g_free (stream->destination);
929 stream->destination = g_strdup (conn->address);
931 /* check for multicast */
932 stream->is_multicast =
933 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
935 stream->ttl = conn->ttl;
938 /* Go over the connections for a stream.
939 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
941 * - If we are dealing with a localhost address, we disable multicast
944 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
945 const GstSDPMedia * media, GstRTSPStream * stream)
947 const GstSDPConnection *conn;
950 /* first look in the media specific section */
951 len = gst_sdp_media_connections_len (media);
952 for (i = 0; i < len; i++) {
953 conn = gst_sdp_media_get_connection (media, i);
955 gst_rtspsrc_do_stream_connection (src, stream, conn);
957 /* then look in the message specific section */
958 if ((conn = gst_sdp_message_get_connection (sdp))) {
959 gst_rtspsrc_do_stream_connection (src, stream, conn);
963 static GstRTSPStream *
964 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
966 GstRTSPStream *stream;
967 const gchar *control_url;
968 const gchar *payload;
969 const GstSDPMedia *media;
971 /* get media, should not return NULL */
972 media = gst_sdp_message_get_media (sdp, idx);
976 stream = g_new0 (GstRTSPStream, 1);
977 stream->parent = src;
978 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
980 stream->last_ret = GST_FLOW_NOT_LINKED;
981 stream->added = FALSE;
982 stream->disabled = FALSE;
983 stream->id = src->numstreams++;
985 stream->discont = TRUE;
986 stream->seqbase = -1;
987 stream->timebase = -1;
989 /* collect bandwidth information for this steam. FIXME, configure in the RTP
990 * session manager to scale RTCP. */
991 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
993 /* collect connection info */
994 gst_rtspsrc_collect_connections (src, sdp, media, stream);
996 /* we must have a payload. No payload means we cannot create caps */
997 /* FIXME, handle multiple formats. The problem here is that we just want to
998 * take the first available format that we can handle but in order to do that
999 * we need to scan for depayloader plugins. Scanning for payloader plugins is
1000 * also suboptimal because the user maybe just wants to save the raw stream
1001 * and then we don't care. */
1002 if ((payload = gst_sdp_media_get_format (media, 0))) {
1003 stream->pt = atoi (payload);
1005 stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
1007 GST_DEBUG ("mapping sdp session level attributes to caps");
1008 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
1009 GST_DEBUG ("mapping sdp media level attributes to caps");
1010 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
1012 if (stream->pt >= 96) {
1013 /* If we have a dynamic payload type, see if we have a stream with the
1014 * same payload number. If there is one, they are part of the same
1015 * container and we only need to add one pad. */
1016 if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
1017 stream->container = TRUE;
1018 GST_DEBUG ("found another stream with pt %d, marking as container",
1023 /* collect port number */
1024 stream->port = gst_sdp_media_get_port (media);
1026 /* get control url to construct the setup url. The setup url is used to
1027 * configure the transport of the stream and is used to identity the stream in
1028 * the RTP-Info header field returned from PLAY. */
1029 control_url = gst_sdp_media_get_attribute_val (media, "control");
1030 if (control_url == NULL)
1031 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1033 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1034 GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
1035 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1036 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1037 GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
1038 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1040 if (control_url != NULL) {
1041 stream->control_url = g_strdup (control_url);
1042 /* Build a fully qualified url using the content_base if any or by prefixing
1043 * the original request.
1044 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1045 * likely build a URL that the server will fail to understand, this is ok,
1046 * we will fail then. */
1047 if (g_str_has_prefix (control_url, "rtsp://"))
1048 stream->conninfo.location = g_strdup (control_url);
1053 if (g_strcmp0 (control_url, "*") == 0)
1057 base = src->control;
1058 else if (src->content_base)
1059 base = src->content_base;
1060 else if (src->conninfo.url_str)
1061 base = src->conninfo.url_str;
1065 /* check if the base ends or control starts with / */
1066 has_slash = g_str_has_prefix (control_url, "/");
1067 has_slash = has_slash || g_str_has_suffix (base, "/");
1069 /* concatenate the two strings, insert / when not present */
1070 stream->conninfo.location =
1071 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1074 GST_DEBUG_OBJECT (src, " setup: %s",
1075 GST_STR_NULL (stream->conninfo.location));
1077 /* we keep track of all streams */
1078 src->streams = g_list_append (src->streams, stream);
1086 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1090 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1093 gst_caps_unref (stream->caps);
1095 g_free (stream->destination);
1096 g_free (stream->control_url);
1097 g_free (stream->conninfo.location);
1099 for (i = 0; i < 2; i++) {
1100 if (stream->udpsrc[i]) {
1101 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1102 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1103 gst_object_unref (stream->udpsrc[i]);
1104 stream->udpsrc[i] = NULL;
1106 if (stream->channelpad[i]) {
1107 gst_object_unref (stream->channelpad[i]);
1108 stream->channelpad[i] = NULL;
1110 if (stream->udpsink[i]) {
1111 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1112 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1113 gst_object_unref (stream->udpsink[i]);
1114 stream->udpsink[i] = NULL;
1117 if (stream->fakesrc) {
1118 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1119 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1120 gst_object_unref (stream->fakesrc);
1121 stream->fakesrc = NULL;
1123 if (stream->srcpad) {
1124 gst_pad_set_active (stream->srcpad, FALSE);
1125 if (stream->added) {
1126 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1127 stream->added = FALSE;
1129 stream->srcpad = NULL;
1131 if (stream->rtcppad) {
1132 gst_object_unref (stream->rtcppad);
1133 stream->rtcppad = NULL;
1135 if (stream->session) {
1136 g_object_unref (stream->session);
1137 stream->session = NULL;
1143 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1147 GST_DEBUG_OBJECT (src, "cleanup");
1149 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1150 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1152 gst_rtspsrc_stream_free (src, stream);
1154 g_list_free (src->streams);
1155 src->streams = NULL;
1157 if (src->manager_sig_id) {
1158 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1159 src->manager_sig_id = 0;
1161 gst_element_set_state (src->manager, GST_STATE_NULL);
1162 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1163 src->manager = NULL;
1165 src->numstreams = 0;
1167 gst_structure_free (src->props);
1170 g_free (src->content_base);
1171 src->content_base = NULL;
1173 g_free (src->control);
1174 src->control = NULL;
1177 gst_rtsp_range_free (src->range);
1180 /* don't clear the SDP when it was used in the url */
1181 if (src->sdp && !src->from_sdp) {
1182 gst_sdp_message_free (src->sdp);
1187 #define PARSE_INT(p, del, res) \
1190 p = strstr (p, del); \
1200 #define PARSE_STRING(p, del, res) \
1203 p = strstr (p, del); \
1215 #define SKIP_SPACES(p) \
1216 while (*p && g_ascii_isspace (*p)) \
1221 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1224 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1225 gint * rate, gchar ** params)
1229 p = (gchar *) rtpmap;
1231 PARSE_INT (p, " ", *payload);
1239 PARSE_STRING (p, "/", *name);
1240 if (*name == NULL) {
1241 GST_DEBUG ("no rate, name %s", p);
1242 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1243 * streams seem to omit the rate. */
1250 p = strstr (p, "/");
1268 * Mapping SDP attributes to caps
1270 * prepend 'a-' to IANA registered sdp attributes names
1271 * (ie: not prefixed with 'x-') in order to avoid
1272 * collision with gstreamer standard caps properties names
1275 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1277 if (attributes->len > 0) {
1281 s = gst_caps_get_structure (caps, 0);
1283 for (i = 0; i < attributes->len; i++) {
1284 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1285 gchar *tofree, *key;
1289 /* skip some of the attribute we already handle */
1290 if (!strcmp (key, "fmtp"))
1292 if (!strcmp (key, "rtpmap"))
1294 if (!strcmp (key, "control"))
1296 if (!strcmp (key, "range"))
1299 /* string must be valid UTF8 */
1300 if (!g_utf8_validate (attr->value, -1, NULL))
1303 if (!g_str_has_prefix (key, "x-"))
1304 tofree = key = g_strdup_printf ("a-%s", key);
1308 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1309 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1316 * Mapping of caps to and from SDP fields:
1318 * m=<media> <UDP port> RTP/AVP <payload>
1319 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1320 * a=fmtp:<payload> <param>[=<value>];...
1323 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1326 const gchar *rtpmap;
1330 gchar *params = NULL;
1336 /* get and parse rtpmap */
1337 if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
1338 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1340 if (payload != pt) {
1341 /* we ignore the rtpmap if the payload type is different. */
1342 g_warning ("rtpmap of wrong payload type, ignoring");
1348 /* if we failed to parse the rtpmap for a dynamic payload type, we have an
1352 /* else we can ignore */
1353 g_warning ("error parsing rtpmap, ignoring");
1356 /* dynamic payloads need rtpmap or we fail */
1360 /* check if we have a rate, if not, we need to look up the rate from the
1361 * default rates based on the payload types. */
1363 const GstRTPPayloadInfo *info;
1365 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1366 /* dynamic types, use media and encoding_name */
1367 tmp = g_ascii_strdown (media->media, -1);
1368 info = gst_rtp_payload_info_for_name (tmp, name);
1371 /* static types, use payload type */
1372 info = gst_rtp_payload_info_for_pt (pt);
1376 if ((rate = info->clock_rate) == 0)
1379 /* we fail if we cannot find one */
1384 tmp = g_ascii_strdown (media->media, -1);
1385 caps = gst_caps_new_simple ("application/x-unknown",
1386 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1388 s = gst_caps_get_structure (caps, 0);
1390 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1392 /* encoding name must be upper case */
1394 tmp = g_ascii_strup (name, -1);
1395 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1399 /* params must be lower case */
1400 if (params != NULL) {
1401 tmp = g_ascii_strdown (params, -1);
1402 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1406 /* parse optional fmtp: field */
1407 if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
1413 /* p is now of the format <payload> <param>[=<value>];... */
1414 PARSE_INT (p, " ", payload);
1415 if (payload != -1 && payload == pt) {
1419 /* <param>[=<value>] are separated with ';' */
1420 pairs = g_strsplit (p, ";", 0);
1421 for (i = 0; pairs[i]; i++) {
1423 const gchar *val, *key;
1425 /* the key may not have a '=', the value can have other '='s */
1426 valpos = strstr (pairs[i], "=");
1428 /* we have a '=' and thus a value, remove the '=' with \0 */
1430 /* value is everything between '=' and ';'. We split the pairs at ;
1431 * boundaries so we can take the remainder of the value. Some servers
1432 * put spaces around the value which we strip off here. Alternatively
1433 * we could strip those spaces in the depayloaders should these spaces
1434 * actually carry any meaning in the future. */
1435 val = g_strstrip (valpos + 1);
1437 /* simple <param>;.. is translated into <param>=1;... */
1440 /* strip the key of spaces, convert key to lowercase but not the value. */
1441 key = g_strstrip (pairs[i]);
1442 if (strlen (key) > 1) {
1443 tmp = g_ascii_strdown (key, -1);
1444 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1456 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1461 g_warning ("rate unknown for payload type %d", pt);
1467 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1468 gint * rtpport, gint * rtcpport)
1471 GstStateChangeReturn ret;
1472 GstElement *udpsrc0, *udpsrc1;
1473 gint tmp_rtp, tmp_rtcp;
1477 src = stream->parent;
1483 /* Start at next port */
1484 tmp_rtp = src->next_port_num;
1486 if (stream->is_ipv6)
1487 host = "udp://[::0]";
1489 host = "udp://0.0.0.0";
1491 /* try to allocate 2 UDP ports, the RTP port should be an even
1492 * number and the RTCP port should be the next (uneven) port */
1495 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1496 tmp_rtp >= src->client_port_range.max)
1499 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1500 if (udpsrc0 == NULL)
1501 goto no_udp_protocol;
1502 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1504 if (src->udp_buffer_size != 0)
1505 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1508 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1509 if (ret == GST_STATE_CHANGE_FAILURE) {
1511 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1514 if (++count > src->retry)
1517 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1518 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1519 gst_object_unref (udpsrc0);
1521 GST_DEBUG_OBJECT (src, "retry %d", count);
1524 goto no_udp_protocol;
1527 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1528 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1530 /* check if port is even */
1531 if ((tmp_rtp & 0x01) != 0) {
1532 /* port not even, close and allocate another */
1533 if (++count > src->retry)
1536 GST_DEBUG_OBJECT (src, "RTP port not even");
1538 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1539 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1540 gst_object_unref (udpsrc0);
1542 GST_DEBUG_OBJECT (src, "retry %d", count);
1547 /* allocate port+1 for RTCP now */
1548 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1549 if (udpsrc1 == NULL)
1550 goto no_udp_rtcp_protocol;
1553 tmp_rtcp = tmp_rtp + 1;
1554 if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
1557 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1559 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1560 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1561 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1562 if (ret == GST_STATE_CHANGE_FAILURE) {
1563 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
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, "free RTCP udpsrc");
1573 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1574 gst_object_unref (udpsrc1);
1578 GST_DEBUG_OBJECT (src, "retry %d", count);
1582 /* all fine, do port check */
1583 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
1584 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
1586 /* this should not happen... */
1587 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
1590 /* we keep these elements, we configure all in configure_transport when the
1591 * server told us to really use the UDP ports. */
1592 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
1593 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
1595 /* keep track of next available port number when we have a range
1597 if (src->next_port_num != 0)
1598 src->next_port_num = tmp_rtcp + 1;
1605 GST_DEBUG_OBJECT (src, "could not get UDP source");
1610 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
1614 no_udp_rtcp_protocol:
1616 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
1621 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
1622 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
1628 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1629 gst_object_unref (udpsrc0);
1632 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1633 gst_object_unref (udpsrc1);
1640 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
1647 GstClockTime base_time = GST_CLOCK_TIME_NONE;
1650 event = gst_event_new_flush_start ();
1651 GST_DEBUG_OBJECT (src, "start flush");
1653 state = GST_STATE_PAUSED;
1655 event = gst_event_new_flush_stop (TRUE);
1656 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
1659 state = GST_STATE_PLAYING;
1661 state = GST_STATE_PAUSED;
1662 clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
1664 base_time = gst_clock_get_time (clock);
1665 gst_object_unref (clock);
1668 gst_rtspsrc_push_event (src, event, FALSE);
1669 gst_rtspsrc_loop_send_cmd (src, cmd);
1671 /* set up manager before data-flow resumes */
1672 /* to manage jitterbuffer buffer mode */
1674 gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
1675 /* and to have base_time trickle further down,
1676 * e.g. to jitterbuffer for its timeout handling */
1677 if (base_time != -1)
1678 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
1681 /* make running time start start at 0 again */
1682 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1683 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1685 for (i = 0; i < 2; i++) {
1687 if (stream->udpsrc[i]) {
1688 if (base_time != -1)
1689 gst_element_set_base_time (stream->udpsrc[i], base_time);
1690 gst_element_set_state (stream->udpsrc[i], state);
1694 /* for tcp interleaved case */
1695 if (base_time != -1)
1696 gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
1699 static GstRTSPResult
1700 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
1701 GstRTSPMessage * message, GTimeVal * timeout)
1706 ret = gst_rtsp_connection_send (conn, message, timeout);
1708 ret = GST_RTSP_ERROR;
1713 static GstRTSPResult
1714 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
1715 GstRTSPMessage * message, GTimeVal * timeout)
1720 ret = gst_rtsp_connection_receive (conn, message, timeout);
1722 ret = GST_RTSP_ERROR;
1728 gst_rtspsrc_get_position (GstRTSPSrc * src)
1733 query = gst_query_new_position (GST_FORMAT_TIME);
1734 /* should be known somewhere down the stream (e.g. jitterbuffer) */
1735 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1736 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1740 if (stream->srcpad) {
1741 if (gst_pad_query (stream->srcpad, query)) {
1742 gst_query_parse_position (query, &fmt, &pos);
1743 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
1744 GST_TIME_ARGS (pos));
1745 src->last_pos = pos;
1755 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
1757 src->state = GST_RTSP_STATE_SEEKING;
1758 /* PLAY will add the range header now. */
1759 src->need_range = TRUE;
1765 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
1770 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
1772 gboolean flush, skip;
1775 GstSegment seeksegment = { 0, };
1779 GST_DEBUG_OBJECT (src, "doing seek with event");
1781 gst_event_parse_seek (event, &rate, &format, &flags,
1782 &cur_type, &cur, &stop_type, &stop);
1784 /* no negative rates yet */
1788 /* we need TIME format */
1789 if (format != src->segment.format)
1792 GST_DEBUG_OBJECT (src, "doing seek without event");
1794 cur_type = GST_SEEK_TYPE_SET;
1795 stop_type = GST_SEEK_TYPE_SET;
1798 /* get flush flag */
1799 flush = flags & GST_SEEK_FLAG_FLUSH;
1800 skip = flags & GST_SEEK_FLAG_SKIP;
1802 /* now we need to make sure the streaming thread is stopped. We do this by
1803 * either sending a FLUSH_START event downstream which will cause the
1804 * streaming thread to stop with a WRONG_STATE.
1805 * For a non-flushing seek we simply pause the task, which will happen as soon
1806 * as it completes one iteration (and thus might block when the sink is
1807 * blocking in preroll). */
1809 GST_DEBUG_OBJECT (src, "starting flush");
1810 gst_rtspsrc_flush (src, TRUE, FALSE);
1813 gst_task_pause (src->task);
1817 /* we should now be able to grab the streaming thread because we stopped it
1818 * with the above flush/pause code */
1819 GST_RTSP_STREAM_LOCK (src);
1821 GST_DEBUG_OBJECT (src, "stopped streaming");
1823 /* copy segment, we need this because we still need the old
1824 * segment when we close the current segment. */
1825 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
1827 /* configure the seek parameters in the seeksegment. We will then have the
1828 * right values in the segment to perform the seek */
1830 GST_DEBUG_OBJECT (src, "configuring seek");
1831 gst_segment_do_seek (&seeksegment, rate, format, flags,
1832 cur_type, cur, stop_type, stop, &update);
1835 /* figure out the last position we need to play. If it's configured (stop !=
1836 * -1), use that, else we play until the total duration of the file */
1837 if ((stop = seeksegment.stop) == -1)
1838 stop = seeksegment.duration;
1840 playing = (src->state == GST_RTSP_STATE_PLAYING);
1842 /* if we were playing, pause first */
1844 /* obtain current position in case seek fails */
1845 gst_rtspsrc_get_position (src);
1846 gst_rtspsrc_pause (src, FALSE, FALSE);
1849 gst_rtspsrc_do_seek (src, &seeksegment);
1851 /* and continue playing */
1853 gst_rtspsrc_play (src, &seeksegment, FALSE);
1855 /* prepare for streaming again */
1857 /* if we started flush, we stop now */
1858 GST_DEBUG_OBJECT (src, "stopping flush");
1859 gst_rtspsrc_flush (src, FALSE, playing);
1862 /* now we did the seek and can activate the new segment values */
1863 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
1865 /* if we're doing a segment seek, post a SEGMENT_START message */
1866 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
1867 gst_element_post_message (GST_ELEMENT_CAST (src),
1868 gst_message_new_segment_start (GST_OBJECT_CAST (src),
1869 src->segment.format, src->segment.position));
1872 /* now create the newsegment */
1873 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
1874 " to %" G_GINT64_FORMAT, src->segment.position, stop);
1876 /* store the newsegment event so it can be sent from the streaming thread. */
1877 if (src->start_segment)
1878 gst_event_unref (src->start_segment);
1879 src->start_segment = gst_event_new_segment (&src->segment);
1882 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
1883 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1884 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1885 stream->discont = TRUE;
1889 GST_RTSP_STREAM_UNLOCK (src);
1896 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
1901 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
1907 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
1911 gboolean res = TRUE;
1914 src = GST_RTSPSRC_CAST (parent);
1916 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1917 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1919 switch (GST_EVENT_TYPE (event)) {
1920 case GST_EVENT_SEEK:
1921 res = gst_rtspsrc_perform_seek (src, event);
1925 case GST_EVENT_NAVIGATION:
1926 case GST_EVENT_LATENCY:
1934 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
1935 res = gst_pad_send_event (target, event);
1936 gst_object_unref (target);
1938 gst_event_unref (event);
1941 gst_event_unref (event);
1947 /* this is the final event function we receive on the internal source pad when
1948 * we deal with TCP connections */
1950 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
1956 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
1958 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1959 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1961 switch (GST_EVENT_TYPE (event)) {
1962 case GST_EVENT_SEEK:
1964 case GST_EVENT_NAVIGATION:
1965 case GST_EVENT_LATENCY:
1967 gst_event_unref (event);
1974 /* this is the final query function we receive on the internal source pad when
1975 * we deal with TCP connections */
1977 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
1981 gboolean res = TRUE;
1983 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
1985 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
1986 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
1988 switch (GST_QUERY_TYPE (query)) {
1989 case GST_QUERY_POSITION:
1994 case GST_QUERY_DURATION:
1998 gst_query_parse_duration (query, &format, NULL);
2001 case GST_FORMAT_TIME:
2002 gst_query_set_duration (query, format, src->segment.duration);
2010 case GST_QUERY_LATENCY:
2012 /* we are live with a min latency of 0 and unlimited max latency, this
2013 * result will be updated by the session manager if there is any. */
2014 gst_query_set_latency (query, TRUE, 0, -1);
2024 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2026 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2030 gboolean res = FALSE;
2032 src = GST_RTSPSRC_CAST (parent);
2034 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2035 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2037 switch (GST_QUERY_TYPE (query)) {
2038 case GST_QUERY_DURATION:
2042 gst_query_parse_duration (query, &format, NULL);
2045 case GST_FORMAT_TIME:
2046 gst_query_set_duration (query, format, src->segment.duration);
2054 case GST_QUERY_SEEKING:
2058 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2059 if (format == GST_FORMAT_TIME) {
2061 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2063 /* seeking without duration is unlikely */
2064 seekable = seekable && src->seekable && src->segment.duration &&
2065 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2067 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2068 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2069 src->segment.start, src->segment.stop);
2076 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2078 /* forward the query to the proxy target pad */
2080 res = gst_pad_query (target, query);
2081 gst_object_unref (target);
2090 /* callback for RTCP messages to be sent to the server when operating in TCP
2092 static GstFlowReturn
2093 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2096 GstRTSPStream *stream;
2097 GstFlowReturn res = GST_FLOW_OK;
2102 GstRTSPMessage message = { 0 };
2103 GstRTSPConnection *conn;
2105 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2106 src = stream->parent;
2108 gst_buffer_map (buffer, &map, GST_MAP_READ);
2112 gst_rtsp_message_init_data (&message, stream->channel[1]);
2114 /* lend the body data to the message */
2115 gst_rtsp_message_take_body (&message, data, size);
2117 if (stream->conninfo.connection)
2118 conn = stream->conninfo.connection;
2120 conn = src->conninfo.connection;
2122 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2123 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2124 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2126 /* and steal it away again because we will free it when unreffing the
2128 gst_rtsp_message_steal_body (&message, &data, &size);
2129 gst_rtsp_message_unset (&message);
2131 gst_buffer_unmap (buffer, &map);
2132 gst_buffer_unref (buffer);
2137 static GstPadProbeReturn
2138 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2140 GstRTSPSrc *src = user_data;
2142 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2143 GST_DEBUG_PAD_NAME (pad));
2145 /* activate the streams */
2146 GST_OBJECT_LOCK (src);
2147 if (!src->need_activate)
2150 src->need_activate = FALSE;
2151 GST_OBJECT_UNLOCK (src);
2153 gst_rtspsrc_activate_streams (src);
2155 return GST_PAD_PROBE_OK;
2159 GST_OBJECT_UNLOCK (src);
2160 return GST_PAD_PROBE_OK;
2164 /* this callback is called when the session manager generated a new src pad with
2165 * payloaded RTP packets. We simply ghost the pad here. */
2167 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2170 GstPadTemplate *template;
2173 GstRTSPStream *stream;
2176 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2178 GST_RTSP_STATE_LOCK (src);
2180 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2181 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2182 goto unknown_stream;
2184 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
2186 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2188 goto unknown_stream;
2190 /* create a new pad we will use to stream to */
2191 template = gst_static_pad_template_get (&rtptemplate);
2192 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2193 gst_object_unref (template);
2196 stream->added = TRUE;
2197 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2198 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2199 gst_pad_set_active (stream->srcpad, TRUE);
2200 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2202 /* check if we added all streams */
2204 for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
2205 stream = (GstRTSPStream *) lstream->data;
2207 GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
2208 stream, stream->container, stream->disabled, stream->added);
2210 /* a container stream only needs one pad added. Also disabled streams don't
2212 if (!stream->container && !stream->disabled && !stream->added) {
2217 GST_RTSP_STATE_UNLOCK (src);
2220 GST_DEBUG_OBJECT (src, "We added all streams");
2221 /* when we get here, all stream are added and we can fire the no-more-pads
2223 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2231 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2232 GST_RTSP_STATE_UNLOCK (src);
2239 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2241 GstRTSPStream *stream;
2244 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2246 GST_RTSP_STATE_LOCK (src);
2247 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2249 goto unknown_stream;
2251 caps = stream->caps;
2253 gst_caps_ref (caps);
2254 GST_RTSP_STATE_UNLOCK (src);
2260 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2261 GST_RTSP_STATE_UNLOCK (src);
2267 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2269 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2275 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
2281 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2287 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2289 GstRTSPSrc *src = stream->parent;
2291 GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
2293 gst_rtspsrc_do_stream_eos (src, stream);
2297 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2299 GstRTSPSrc *src = stream->parent;
2301 GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
2303 gst_rtspsrc_do_stream_eos (src, stream);
2307 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2309 GstRTSPStream *stream;
2311 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2313 /* get stream for session */
2314 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2316 gst_rtspsrc_do_stream_eos (src, stream);
2321 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2323 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2327 /* try to get and configure a manager */
2329 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2330 GstRTSPTransport * transport)
2332 const gchar *manager;
2334 GstStateChangeReturn ret;
2336 /* find a manager */
2337 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2341 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2343 /* configure the manager */
2344 if (src->manager == NULL) {
2345 GObjectClass *klass;
2348 if (!(src->manager = gst_element_factory_make (manager, NULL))) {
2350 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2354 goto use_no_manager;
2356 if (!(src->manager = gst_element_factory_make (manager, NULL)))
2357 goto manager_failed;
2360 /* we manage this element */
2361 gst_bin_add (GST_BIN_CAST (src), src->manager);
2363 GST_OBJECT_LOCK (src);
2364 target = GST_STATE_TARGET (src);
2365 GST_OBJECT_UNLOCK (src);
2367 ret = gst_element_set_state (src->manager, target);
2368 if (ret == GST_STATE_CHANGE_FAILURE)
2369 goto start_manager_failure;
2371 g_object_set (src->manager, "latency", src->latency, NULL);
2373 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2374 if (g_object_class_find_property (klass, "buffer-mode")) {
2375 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2376 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2378 gboolean need_slave;
2380 const gchar *encoding;
2382 /* buffer mode pauses are handled by adding offsets to buffer times,
2383 * but some depayloaders may have a hard time syncing output times
2384 * with such input times, e.g. container ones, most notably ASF */
2385 /* TODO alternatives are having an event that indicates these shifts,
2386 * or having rtsp extensions provide suggestion on buffer mode */
2387 need_slave = stream->container;
2388 if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
2389 (encoding = gst_structure_get_string (s, "encoding-name")))
2390 need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
2391 GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d",
2393 /* valid duration implies not likely live pipeline,
2394 * so slaving in jitterbuffer does not make much sense
2395 * (and might mess things up due to bursts) */
2396 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2397 src->segment.duration && !need_slave) {
2398 GST_DEBUG_OBJECT (src, "selected buffer");
2399 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
2402 GST_DEBUG_OBJECT (src, "selected slave");
2403 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2408 /* connect to signals if we did not already do so */
2409 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2411 src->manager_sig_id =
2412 g_signal_connect (src->manager, "pad-added",
2413 (GCallback) new_manager_pad, src);
2414 src->manager_ptmap_id =
2415 g_signal_connect (src->manager, "request-pt-map",
2416 (GCallback) request_pt_map, src);
2418 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2422 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2423 * into a separate RTP session. */
2424 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2425 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2427 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2428 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2431 /* now configure the bandwidth in the manager */
2432 if (g_signal_lookup ("get-internal-session",
2433 G_OBJECT_TYPE (src->manager)) != 0) {
2434 GObject *rtpsession;
2436 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2439 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2441 stream->session = rtpsession;
2443 if (stream->as_bandwidth != -1) {
2444 GST_INFO_OBJECT (src, "setting AS: %f",
2445 (gdouble) (stream->as_bandwidth * 1000));
2446 g_object_set (rtpsession, "bandwidth",
2447 (gdouble) (stream->as_bandwidth * 1000), NULL);
2449 if (stream->rr_bandwidth != -1) {
2450 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2451 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2454 if (stream->rs_bandwidth != -1) {
2455 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2456 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2459 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2461 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2463 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2465 g_signal_connect (rtpsession, "on-ssrc-active",
2466 (GCallback) on_ssrc_active, stream);
2477 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2482 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2485 start_manager_failure:
2487 GST_DEBUG_OBJECT (src, "could not start session manager");
2492 /* free the UDP sources allocated when negotiating a transport.
2493 * This function is called when the server negotiated to a transport where the
2494 * UDP sources are not needed anymore, such as TCP or multicast. */
2496 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
2500 for (i = 0; i < 2; i++) {
2501 if (stream->udpsrc[i]) {
2502 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
2503 gst_object_unref (stream->udpsrc[i]);
2504 stream->udpsrc[i] = NULL;
2509 /* for TCP, create pads to send and receive data to and from the manager and to
2510 * intercept various events and queries
2513 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
2514 GstRTSPTransport * transport, GstPad ** outpad)
2517 GstPadTemplate *template;
2518 GstPad *pad0, *pad1;
2520 /* configure for interleaved delivery, nothing needs to be done
2521 * here, the loop function will call the chain functions of the
2522 * session manager. */
2523 stream->channel[0] = transport->interleaved.min;
2524 stream->channel[1] = transport->interleaved.max;
2525 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
2526 stream->channel[0], stream->channel[1]);
2528 /* we can remove the allocated UDP ports now */
2529 gst_rtspsrc_stream_free_udp (stream);
2531 /* no session manager, send data to srcpad directly */
2532 if (!stream->channelpad[0]) {
2533 GST_DEBUG_OBJECT (src, "no manager, creating pad");
2535 /* create a new pad we will use to stream to */
2536 name = g_strdup_printf ("stream_%u", stream->id);
2537 template = gst_static_pad_template_get (&rtptemplate);
2538 stream->channelpad[0] = gst_pad_new_from_template (template, name);
2539 gst_object_unref (template);
2542 /* set caps and activate */
2543 gst_pad_use_fixed_caps (stream->channelpad[0]);
2544 gst_pad_set_active (stream->channelpad[0], TRUE);
2546 *outpad = gst_object_ref (stream->channelpad[0]);
2548 GST_DEBUG_OBJECT (src, "using manager source pad");
2550 template = gst_static_pad_template_get (&anysrctemplate);
2552 /* allocate pads for sending the channel data into the manager */
2553 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
2554 gst_pad_link (pad0, stream->channelpad[0]);
2555 gst_object_unref (stream->channelpad[0]);
2556 stream->channelpad[0] = pad0;
2557 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
2558 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
2559 gst_pad_set_element_private (pad0, src);
2560 gst_pad_set_active (pad0, TRUE);
2562 if (stream->channelpad[1]) {
2563 /* if we have a sinkpad for the other channel, create a pad and link to the
2565 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
2566 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
2567 gst_pad_link (pad1, stream->channelpad[1]);
2568 gst_object_unref (stream->channelpad[1]);
2569 stream->channelpad[1] = pad1;
2570 gst_pad_set_active (pad1, TRUE);
2572 gst_object_unref (template);
2574 /* setup RTCP transport back to the server if we have to. */
2575 if (src->manager && src->do_rtcp) {
2578 template = gst_static_pad_template_get (&anysinktemplate);
2580 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
2581 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
2582 gst_pad_set_element_private (stream->rtcppad, stream);
2583 gst_pad_set_active (stream->rtcppad, TRUE);
2585 /* get session RTCP pad */
2586 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2587 pad = gst_element_get_request_pad (src->manager, name);
2592 gst_pad_link (pad, stream->rtcppad);
2593 gst_object_unref (pad);
2596 gst_object_unref (template);
2602 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
2603 GstRTSPTransport * transport, const gchar ** destination, gint * min,
2604 gint * max, guint * ttl)
2606 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2608 if (!(*destination = transport->destination))
2609 *destination = stream->destination;
2612 /* transport first */
2613 *min = transport->port.min;
2614 *max = transport->port.max;
2615 if (*min == -1 && *max == -1) {
2616 /* then try from SDP */
2617 if (stream->port != 0) {
2618 *min = stream->port;
2619 *max = stream->port + 1;
2625 if (!(*ttl = transport->ttl))
2630 /* first take the source, then the endpoint to figure out where to send
2632 if (!(*destination = transport->source)) {
2633 if (src->conninfo.connection)
2634 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
2635 else if (stream->conninfo.connection)
2637 gst_rtsp_connection_get_ip (stream->conninfo.connection);
2641 /* for unicast we only expect the ports here */
2642 *min = transport->server_port.min;
2643 *max = transport->server_port.max;
2648 /* For multicast create UDP sources and join the multicast group. */
2650 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
2651 GstRTSPTransport * transport, GstPad ** outpad)
2654 const gchar *destination;
2657 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
2659 /* we can remove the allocated UDP ports now */
2660 gst_rtspsrc_stream_free_udp (stream);
2662 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
2665 /* we need a destination now */
2666 if (destination == NULL)
2667 goto no_destination;
2669 /* we really need ports now or we won't be able to receive anything at all */
2670 if (min == -1 && max == -1)
2673 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
2674 destination, min, max);
2676 /* creating UDP source for RTP */
2678 uri = g_strdup_printf ("udp://%s:%d", destination, min);
2679 stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2681 if (stream->udpsrc[0] == NULL)
2684 /* take ownership */
2685 gst_object_ref_sink (stream->udpsrc[0]);
2688 gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
2691 /* creating another UDP source for RTCP */
2693 uri = g_strdup_printf ("udp://%s:%d", destination, max);
2694 stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2696 if (stream->udpsrc[1] == NULL)
2699 /* take ownership */
2700 gst_object_ref_sink (stream->udpsrc[1]);
2702 gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
2709 GST_DEBUG_OBJECT (src, "no UDP source element found");
2714 GST_DEBUG_OBJECT (src, "no destination found");
2719 GST_DEBUG_OBJECT (src, "no ports found");
2724 /* configure the remainder of the UDP ports */
2726 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
2727 GstRTSPTransport * transport, GstPad ** outpad)
2729 /* we manage the UDP elements now. For unicast, the UDP sources where
2730 * allocated in the stream when we suggested a transport. */
2731 if (stream->udpsrc[0]) {
2732 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
2734 GST_DEBUG_OBJECT (src, "setting up UDP source");
2736 /* configure a timeout on the UDP port. When the timeout message is
2737 * posted, we assume UDP transport is not possible. We reconnect using TCP
2739 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
2742 /* get output pad of the UDP source. */
2743 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
2745 /* save it so we can unblock */
2746 stream->blockedpad = *outpad;
2748 /* configure pad block on the pad. As soon as there is dataflow on the
2749 * UDP source, we know that UDP is not blocked by a firewall and we can
2750 * configure all the streams to let the application autoplug decoders. */
2752 gst_pad_add_probe (stream->blockedpad,
2753 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
2755 if (stream->channelpad[0]) {
2756 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
2757 /* configure for UDP delivery, we need to connect the UDP pads to
2758 * the session plugin. */
2759 gst_pad_link (*outpad, stream->channelpad[0]);
2760 gst_object_unref (*outpad);
2762 /* we connected to pad-added signal to get pads from the manager */
2764 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
2769 if (stream->udpsrc[1]) {
2770 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
2772 if (stream->channelpad[1]) {
2775 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
2777 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
2778 gst_pad_link (pad, stream->channelpad[1]);
2779 gst_object_unref (pad);
2781 /* leave unlinked */
2787 /* configure the UDP sink back to the server for status reports */
2789 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
2790 GstRTSPStream * stream, GstRTSPTransport * transport)
2793 gint rtp_port, rtcp_port;
2794 gboolean do_rtp, do_rtcp;
2795 const gchar *destination;
2800 /* get transport info */
2801 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
2802 &rtp_port, &rtcp_port, &ttl);
2804 /* see what we need to do */
2805 do_rtp = (rtp_port != -1);
2806 /* it's possible that the server does not want us to send RTCP in which case
2808 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
2810 /* we need a destination when we have RTP or RTCP ports */
2811 if (destination == NULL && (do_rtp || do_rtcp))
2812 goto no_destination;
2814 /* try to construct the fakesrc to the RTP port of the server to open up any
2817 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
2820 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
2821 stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2823 if (stream->udpsink[0] == NULL)
2824 goto no_sink_element;
2826 /* don't join multicast group, we will have the source socket do that */
2827 /* no sync or async state changes needed */
2828 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
2829 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2831 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2833 if (stream->udpsrc[0]) {
2834 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
2835 * so that NAT firewalls will open a hole for us */
2836 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
2837 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
2838 /* configure socket and make sure udpsink does not close it when shutting
2839 * down, it belongs to udpsrc after all. */
2840 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
2841 "close-socket", FALSE, NULL);
2842 g_object_unref (socket);
2845 /* the source for the dummy packets to open up NAT */
2846 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
2847 if (stream->fakesrc == NULL)
2848 goto no_fakesrc_element;
2850 /* random data in 5 buffers, a size of 200 bytes should be fine */
2851 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
2852 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
2854 /* we don't want to consider this a sink */
2855 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
2857 /* keep everything locked */
2858 gst_element_set_locked_state (stream->udpsink[0], TRUE);
2859 gst_element_set_locked_state (stream->fakesrc, TRUE);
2861 gst_object_ref (stream->udpsink[0]);
2862 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
2863 gst_object_ref (stream->fakesrc);
2864 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
2866 gst_element_link (stream->fakesrc, stream->udpsink[0]);
2869 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
2872 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
2873 stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2875 if (stream->udpsink[1] == NULL)
2876 goto no_sink_element;
2878 /* don't join multicast group, we will have the source socket do that */
2879 /* no sync or async state changes needed */
2880 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
2881 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2883 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2885 if (stream->udpsrc[1]) {
2886 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
2887 * because some servers check the port number of where it sends RTCP to identify
2888 * the RTCP packets it receives */
2889 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
2890 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
2891 /* configure socket and make sure udpsink does not close it when shutting
2892 * down, it belongs to udpsrc after all. */
2893 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
2894 "close-socket", FALSE, NULL);
2895 g_object_unref (socket);
2898 /* we don't want to consider this a sink */
2899 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
2901 /* we keep this playing always */
2902 gst_element_set_locked_state (stream->udpsink[1], TRUE);
2903 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
2905 gst_object_ref (stream->udpsink[1]);
2906 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
2908 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
2910 /* get session RTCP pad */
2911 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2912 pad = gst_element_get_request_pad (src->manager, name);
2917 gst_pad_link (pad, stream->rtcppad);
2918 gst_object_unref (pad);
2927 GST_DEBUG_OBJECT (src, "no destination address specified");
2932 GST_DEBUG_OBJECT (src, "no UDP sink element found");
2937 GST_DEBUG_OBJECT (src, "no fakesrc element found");
2942 /* sets up all elements needed for streaming over the specified transport.
2943 * Does not yet expose the element pads, this will be done when there is actuall
2944 * dataflow detected, which might never happen when UDP is blocked in a
2945 * firewall, for example.
2948 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
2949 GstRTSPTransport * transport)
2952 GstPad *outpad = NULL;
2953 GstPadTemplate *template;
2958 src = stream->parent;
2960 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
2962 s = gst_caps_get_structure (stream->caps, 0);
2964 /* get the proper mime type for this stream now */
2965 if (gst_rtsp_transport_get_mime (transport->trans, &mime) < 0)
2966 goto unknown_transport;
2968 goto unknown_transport;
2970 /* configure the final mime type */
2971 GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
2972 gst_structure_set_name (s, mime);
2974 /* try to get and configure a manager, channelpad[0-1] will be configured with
2975 * the pads for the manager, or NULL when no manager is needed. */
2976 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
2979 switch (transport->lower_transport) {
2980 case GST_RTSP_LOWER_TRANS_TCP:
2981 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
2982 goto transport_failed;
2984 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2985 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
2986 goto transport_failed;
2987 /* fallthrough, the rest is the same for UDP and MCAST */
2988 case GST_RTSP_LOWER_TRANS_UDP:
2989 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
2990 goto transport_failed;
2991 /* configure udpsinks back to the server for RTCP messages and for the
2992 * dummy RTP messages to open NAT. */
2993 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
2994 goto transport_failed;
2997 goto unknown_transport;
3001 GST_DEBUG_OBJECT (src, "creating ghostpad");
3003 gst_pad_use_fixed_caps (outpad);
3005 /* create ghostpad, don't add just yet, this will be done when we activate
3007 name = g_strdup_printf ("stream_%u", stream->id);
3008 template = gst_static_pad_template_get (&rtptemplate);
3009 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3010 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3011 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3012 gst_object_unref (template);
3015 gst_object_unref (outpad);
3017 /* mark pad as ok */
3018 stream->last_ret = GST_FLOW_OK;
3025 GST_DEBUG_OBJECT (src, "failed to configure transport");
3030 GST_DEBUG_OBJECT (src, "unknown transport");
3035 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3040 /* send a couple of dummy random packets on the receiver RTP port to the server,
3041 * this should make a firewall think we initiated the data transfer and
3042 * hopefully allow packets to go from the sender port to our RTP receiver port */
3044 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3048 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3051 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3052 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3054 if (stream->fakesrc && stream->udpsink[0]) {
3055 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3056 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3057 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3058 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3059 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3065 /* Adds the source pads of all configured streams to the element.
3066 * This code is performed when we detected dataflow.
3068 * We detect dataflow from either the _loop function or with pad probes on the
3072 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3076 GST_DEBUG_OBJECT (src, "activating streams");
3078 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3079 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3081 if (stream->udpsrc[0]) {
3082 /* remove timeout, we are streaming now and timeouts will be handled by
3083 * the session manager and jitter buffer */
3084 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3086 if (stream->srcpad) {
3087 /* if we don't have a session manager, set the caps now. If we have a
3088 * session, we will get a notification of the pad and the caps. */
3089 if (!src->manager) {
3090 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3091 gst_pad_set_caps (stream->srcpad, stream->caps);
3094 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3095 gst_pad_set_active (stream->srcpad, TRUE);
3097 if (!stream->added) {
3098 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3099 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3100 stream->added = TRUE;
3105 /* unblock all pads */
3106 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3107 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3109 if (stream->blockid) {
3110 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3111 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3112 stream->blockid = 0;
3120 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
3123 guint64 start, stop;
3124 gdouble play_speed, play_scale;
3126 GST_DEBUG_OBJECT (src, "configuring stream caps");
3128 start = segment->position;
3129 stop = segment->duration;
3130 play_speed = segment->rate;
3131 play_scale = segment->applied_rate;
3133 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3134 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3137 if ((caps = stream->caps)) {
3138 caps = gst_caps_make_writable (caps);
3140 if (stream->timebase != -1)
3141 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3142 (guint) stream->timebase, NULL);
3143 if (stream->seqbase != -1)
3144 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3145 (guint) stream->seqbase, NULL);
3146 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3148 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3149 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3150 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3152 stream->caps = caps;
3154 GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
3157 GST_DEBUG_OBJECT (src, "clear session");
3158 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3162 static GstFlowReturn
3163 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3168 /* store the value */
3169 stream->last_ret = ret;
3171 /* if it's success we can return the value right away */
3172 if (ret == GST_FLOW_OK)
3175 /* any other error that is not-linked can be returned right
3177 if (ret != GST_FLOW_NOT_LINKED)
3180 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3181 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3182 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3184 ret = ostream->last_ret;
3185 /* some other return value (must be SUCCESS but we can return
3186 * other values as well) */
3187 if (ret != GST_FLOW_NOT_LINKED)
3190 /* if we get here, all other pads were unlinked and we return
3191 * NOT_LINKED then */
3197 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3198 GstEvent * event, gboolean source)
3200 gboolean res = TRUE;
3202 /* only streams that have a connection to the outside world */
3203 if (stream->srcpad == NULL)
3206 if (source && stream->udpsrc[0]) {
3207 gst_event_ref (event);
3208 res = gst_element_send_event (stream->udpsrc[0], event);
3209 } else if (stream->channelpad[0]) {
3210 gst_event_ref (event);
3211 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3212 res = gst_pad_push_event (stream->channelpad[0], event);
3214 res = gst_pad_send_event (stream->channelpad[0], event);
3217 if (source && stream->udpsrc[1]) {
3218 gst_event_ref (event);
3219 res &= gst_element_send_event (stream->udpsrc[1], event);
3220 } else if (stream->channelpad[1]) {
3221 gst_event_ref (event);
3222 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3223 res &= gst_pad_push_event (stream->channelpad[1], event);
3225 res &= gst_pad_send_event (stream->channelpad[1], event);
3229 gst_event_unref (event);
3235 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
3238 gboolean res = TRUE;
3240 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3241 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3243 gst_event_ref (event);
3244 res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
3246 gst_event_unref (event);
3251 static GstRTSPResult
3252 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3257 if (info->connection == NULL) {
3258 if (info->url == NULL) {
3259 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3260 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3264 /* create connection */
3265 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3266 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3267 goto could_not_create;
3270 g_free (info->url_str);
3271 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3273 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3275 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3276 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3278 if (src->proxy_host) {
3279 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3281 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3286 if (!info->connected) {
3289 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3290 ("Connecting to %s", info->location));
3291 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3293 gst_rtsp_connection_connect (info->connection,
3294 src->ptcp_timeout)) < 0)
3295 goto could_not_connect;
3297 info->connected = TRUE;
3304 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3309 gchar *str = gst_rtsp_strresult (res);
3310 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3316 gchar *str = gst_rtsp_strresult (res);
3317 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3323 static GstRTSPResult
3324 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3327 if (info->connected) {
3328 GST_DEBUG_OBJECT (src, "closing connection...");
3329 gst_rtsp_connection_close (info->connection);
3330 info->connected = FALSE;
3332 if (free && info->connection) {
3333 /* free connection */
3334 GST_DEBUG_OBJECT (src, "freeing connection...");
3335 gst_rtsp_connection_free (info->connection);
3336 info->connection = NULL;
3341 static GstRTSPResult
3342 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3347 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3348 gst_rtsp_conninfo_close (src, info, FALSE);
3349 res = gst_rtsp_conninfo_connect (src, info, async);
3355 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3359 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3360 if (src->conninfo.connection) {
3361 GST_DEBUG_OBJECT (src, "connection flush");
3362 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3364 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3365 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3366 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3367 if (stream->conninfo.connection)
3368 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3372 /* FIXME, handle server request, reply with OK, for now */
3373 static GstRTSPResult
3374 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3375 GstRTSPMessage * request)
3377 GstRTSPMessage response = { 0 };
3380 GST_DEBUG_OBJECT (src, "got server request message");
3383 gst_rtsp_message_dump (request);
3385 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3387 if (res == GST_RTSP_ENOTIMPL) {
3388 /* default implementation, send OK */
3390 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3395 GST_DEBUG_OBJECT (src, "replying with OK");
3398 gst_rtsp_message_dump (&response);
3400 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
3404 gst_rtsp_message_unset (&response);
3405 } else if (res == GST_RTSP_EEOF)
3413 gst_rtsp_message_unset (&response);
3418 /* send server keep-alive */
3419 static GstRTSPResult
3420 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
3422 GstRTSPMessage request = { 0 };
3424 GstRTSPMethod method;
3427 GST_DEBUG_OBJECT (src, "creating server keep-alive");
3429 /* find a method to use for keep-alive */
3430 if (src->methods & GST_RTSP_GET_PARAMETER)
3431 method = GST_RTSP_GET_PARAMETER;
3433 method = GST_RTSP_OPTIONS;
3436 control = src->control;
3438 control = src->conninfo.url_str;
3440 if (control == NULL)
3443 res = gst_rtsp_message_init_request (&request, method, control);
3448 gst_rtsp_message_dump (&request);
3451 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
3456 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
3457 gst_rtsp_message_unset (&request);
3464 GST_WARNING_OBJECT (src, "no control url to send keepalive");
3469 gchar *str = gst_rtsp_strresult (res);
3471 gst_rtsp_message_unset (&request);
3472 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
3473 ("Could not send keep-alive. (%s)", str));
3479 static GstFlowReturn
3480 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
3482 GstRTSPMessage message = { 0 };
3485 GstRTSPStream *stream;
3486 GstPad *outpad = NULL;
3489 GstFlowReturn ret = GST_FLOW_OK;
3491 gboolean is_rtcp, have_data;
3493 /* here we are only interested in data messages */
3496 GTimeVal tv_timeout;
3498 /* get the next timeout interval */
3499 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3501 /* see if the timeout period expired */
3502 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
3503 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
3504 /* send keep-alive, only act on interrupt, a warning will be posted for
3506 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3508 /* get new timeout */
3509 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3512 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
3513 tv_timeout.tv_sec, tv_timeout.tv_usec);
3515 /* protect the connection with the connection lock so that we can see when
3516 * we are finished doing server communication */
3518 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3519 &message, src->ptcp_timeout);
3523 GST_DEBUG_OBJECT (src, "we received a server message");
3525 case GST_RTSP_EINTR:
3526 /* we got interrupted this means we need to stop */
3528 case GST_RTSP_ETIMEOUT:
3529 /* no reply, send keep alive */
3530 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3531 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3535 /* go EOS when the server closed the connection */
3541 switch (message.type) {
3542 case GST_RTSP_MESSAGE_REQUEST:
3543 /* server sends us a request message, handle it */
3545 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3547 if (res == GST_RTSP_EEOF)
3550 goto handle_request_failed;
3552 case GST_RTSP_MESSAGE_RESPONSE:
3553 /* we ignore response messages */
3554 GST_DEBUG_OBJECT (src, "ignoring response message");
3556 gst_rtsp_message_dump (&message);
3558 case GST_RTSP_MESSAGE_DATA:
3559 GST_DEBUG_OBJECT (src, "got data message");
3563 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3570 channel = message.type_data.data.channel;
3572 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
3574 goto unknown_stream;
3576 if (channel == stream->channel[0]) {
3577 outpad = stream->channelpad[0];
3579 } else if (channel == stream->channel[1]) {
3580 outpad = stream->channelpad[1];
3586 /* take a look at the body to figure out what we have */
3587 gst_rtsp_message_get_body (&message, &data, &size);
3589 goto invalid_length;
3591 /* channels are not correct on some servers, do extra check */
3592 if (data[1] >= 200 && data[1] <= 204) {
3593 /* hmm RTCP message switch to the RTCP pad of the same stream. */
3594 outpad = stream->channelpad[1];
3598 /* we have no clue what this is, just ignore then. */
3600 goto unknown_stream;
3602 /* take the message body for further processing */
3603 gst_rtsp_message_steal_body (&message, &data, &size);
3605 /* strip the trailing \0 */
3608 buf = gst_buffer_new ();
3609 gst_buffer_take_memory (buf, -1,
3610 gst_memory_new_wrapped (0, data, g_free, size, 0, size));
3612 /* don't need message anymore */
3613 gst_rtsp_message_unset (&message);
3615 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
3618 if (src->need_activate) {
3619 gst_rtspsrc_activate_streams (src);
3620 src->need_activate = FALSE;
3623 if (src->base_time == -1) {
3624 /* Take current running_time. This timestamp will be put on
3625 * the first buffer of each stream because we are a live source and so we
3626 * timestamp with the running_time. When we are dealing with TCP, we also
3627 * only timestamp the first buffer (using the DISCONT flag) because a server
3628 * typically bursts data, for which we don't want to compensate by speeding
3629 * up the media. The other timestamps will be interpollated from this one
3630 * using the RTP timestamps. */
3631 GST_OBJECT_LOCK (src);
3632 if (GST_ELEMENT_CLOCK (src)) {
3634 GstClockTime base_time;
3636 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
3637 base_time = GST_ELEMENT_CAST (src)->base_time;
3639 src->base_time = now - base_time;
3641 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
3642 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
3644 GST_OBJECT_UNLOCK (src);
3647 if (stream->discont && !is_rtcp) {
3648 /* mark first RTP buffer as discont */
3649 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
3650 stream->discont = FALSE;
3651 /* first buffer gets the timestamp, other buffers are not timestamped and
3652 * their presentation time will be interpollated from the rtp timestamps. */
3653 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
3654 GST_TIME_ARGS (src->base_time));
3656 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
3659 /* chain to the peer pad */
3660 if (GST_PAD_IS_SINK (outpad))
3661 ret = gst_pad_chain (outpad, buf);
3663 ret = gst_pad_push (outpad, buf);
3666 /* combine all stream flows for the data transport */
3667 ret = gst_rtspsrc_combine_flows (src, stream, ret);
3674 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
3675 gst_rtsp_message_unset (&message);
3680 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3681 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3682 ("The server closed the connection."));
3683 src->conninfo.connected = FALSE;
3684 gst_rtsp_message_unset (&message);
3685 return GST_FLOW_EOS;
3689 gst_rtsp_message_unset (&message);
3690 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3691 gst_rtspsrc_connection_flush (src, FALSE);
3692 return GST_FLOW_WRONG_STATE;
3696 gchar *str = gst_rtsp_strresult (res);
3698 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3699 ("Could not receive message. (%s)", str));
3702 gst_rtsp_message_unset (&message);
3703 return GST_FLOW_ERROR;
3705 handle_request_failed:
3707 gchar *str = gst_rtsp_strresult (res);
3709 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3710 ("Could not handle server message. (%s)", str));
3712 gst_rtsp_message_unset (&message);
3713 return GST_FLOW_ERROR;
3717 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3718 ("Short message received, ignoring."));
3719 gst_rtsp_message_unset (&message);
3724 static GstFlowReturn
3725 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
3728 GstRTSPMessage message = { 0 };
3732 GTimeVal tv_timeout;
3734 /* get the next timeout interval */
3735 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3737 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
3738 (gint) tv_timeout.tv_sec);
3740 gst_rtsp_message_unset (&message);
3742 /* we should continue reading the TCP socket because the server might
3743 * send us requests. When the session timeout expires, we need to send a
3744 * keep-alive request to keep the session open. */
3745 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3746 &message, &tv_timeout);
3750 GST_DEBUG_OBJECT (src, "we received a server message");
3752 case GST_RTSP_EINTR:
3753 /* we got interrupted, see what we have to do */
3755 case GST_RTSP_ETIMEOUT:
3756 /* send keep-alive, ignore the result, a warning will be posted. */
3757 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3758 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3762 /* server closed the connection. not very fatal for UDP, reconnect and
3763 * see what happens. */
3764 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3765 ("The server closed the connection."));
3767 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
3775 switch (message.type) {
3776 case GST_RTSP_MESSAGE_REQUEST:
3777 /* server sends us a request message, handle it */
3779 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3781 if (res == GST_RTSP_EEOF)
3784 goto handle_request_failed;
3786 case GST_RTSP_MESSAGE_RESPONSE:
3787 /* we ignore response and data messages */
3788 GST_DEBUG_OBJECT (src, "ignoring response message");
3790 gst_rtsp_message_dump (&message);
3791 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
3792 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
3793 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
3794 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
3795 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3802 case GST_RTSP_MESSAGE_DATA:
3803 /* we ignore response and data messages */
3804 GST_DEBUG_OBJECT (src, "ignoring data message");
3807 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3813 /* we get here when the connection got interrupted */
3816 gst_rtsp_message_unset (&message);
3817 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3818 gst_rtspsrc_connection_flush (src, FALSE);
3819 return GST_FLOW_WRONG_STATE;
3823 gchar *str = gst_rtsp_strresult (res);
3826 src->conninfo.connected = FALSE;
3827 if (res != GST_RTSP_EINTR) {
3828 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
3829 ("Could not connect to server. (%s)", str));
3831 ret = GST_FLOW_ERROR;
3833 ret = GST_FLOW_WRONG_STATE;
3839 gchar *str = gst_rtsp_strresult (res);
3841 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3842 ("Could not receive message. (%s)", str));
3844 return GST_FLOW_ERROR;
3846 handle_request_failed:
3848 gchar *str = gst_rtsp_strresult (res);
3851 gst_rtsp_message_unset (&message);
3852 if (res != GST_RTSP_EINTR) {
3853 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3854 ("Could not handle server message. (%s)", str));
3856 ret = GST_FLOW_ERROR;
3858 ret = GST_FLOW_WRONG_STATE;
3864 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3865 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3866 ("The server closed the connection."));
3867 src->conninfo.connected = FALSE;
3868 gst_rtsp_message_unset (&message);
3869 return GST_FLOW_EOS;
3873 static GstRTSPResult
3874 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
3876 GstRTSPResult res = GST_RTSP_OK;
3879 GST_DEBUG_OBJECT (src, "doing reconnect");
3881 GST_OBJECT_LOCK (src);
3882 /* only restart when the pads were not yet activated, else we were
3883 * streaming over UDP */
3884 restart = src->need_activate;
3885 GST_OBJECT_UNLOCK (src);
3887 /* no need to restart, we're done */
3891 /* we can try only TCP now */
3892 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
3894 /* close and cleanup our state */
3895 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
3898 /* see if we have TCP left to try. Also don't try TCP when we were configured
3900 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
3903 /* We post a warning message now to inform the user
3904 * that nothing happened. It's most likely a firewall thing. */
3905 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3906 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3907 "firewall is blocking it. Retrying using a TCP connection.",
3908 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3910 /* open new connection using tcp */
3911 if (gst_rtspsrc_open (src, async) < 0)
3914 /* start playback */
3915 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
3924 src->cur_protocols = 0;
3925 /* no transport possible, post an error and stop */
3926 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3927 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3928 "firewall is blocking it. No other protocols to try.",
3929 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3930 return GST_FLOW_ERROR;
3934 GST_DEBUG_OBJECT (src, "open failed");
3939 GST_DEBUG_OBJECT (src, "play failed");
3945 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
3949 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
3952 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
3955 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
3958 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
3966 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
3970 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
3973 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
3976 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
3979 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
3987 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
3991 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
3994 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
3997 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4000 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4008 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4012 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4015 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4018 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4021 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4029 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4031 if (ret == GST_RTSP_OK)
4032 gst_rtspsrc_loop_complete_cmd (src, cmd);
4033 else if (ret == GST_RTSP_EINTR)
4034 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4036 gst_rtspsrc_loop_error_cmd (src, cmd);
4040 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
4044 /* start new request */
4045 gst_rtspsrc_loop_start_cmd (src, cmd);
4047 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4049 GST_OBJECT_LOCK (src);
4050 old = src->loop_cmd;
4051 if (old != CMD_WAIT) {
4052 src->loop_cmd = CMD_WAIT;
4053 GST_OBJECT_UNLOCK (src);
4054 /* cancel previous request */
4055 gst_rtspsrc_loop_cancel_cmd (src, old);
4056 GST_OBJECT_LOCK (src);
4058 src->loop_cmd = cmd;
4059 /* interrupt if allowed */
4061 GST_DEBUG_OBJECT (src, "start connection flush");
4062 gst_rtspsrc_connection_flush (src, TRUE);
4065 gst_task_start (src->task);
4066 GST_OBJECT_UNLOCK (src);
4070 gst_rtspsrc_loop (GstRTSPSrc * src)
4074 if (!src->conninfo.connection || !src->conninfo.connected)
4077 if (src->interleaved)
4078 ret = gst_rtspsrc_loop_interleaved (src);
4080 ret = gst_rtspsrc_loop_udp (src);
4082 if (ret != GST_FLOW_OK)
4090 GST_WARNING_OBJECT (src, "we are not connected");
4091 ret = GST_FLOW_WRONG_STATE;
4096 const gchar *reason = gst_flow_get_name (ret);
4098 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4099 src->running = FALSE;
4100 if (ret == GST_FLOW_EOS) {
4101 /* perform EOS logic */
4102 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4103 gst_element_post_message (GST_ELEMENT_CAST (src),
4104 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4105 src->segment.format, src->segment.position));
4107 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4109 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4110 /* for fatal errors we post an error message, post the error before the
4111 * EOS so the app knows about the error first. */
4112 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4113 ("Internal data flow error."),
4114 ("streaming task paused, reason %s (%d)", reason, ret));
4115 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4121 #ifndef GST_DISABLE_GST_DEBUG
4122 static const gchar *
4123 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4127 while (method != 0) {
4144 static const gchar *
4145 gst_rtspsrc_skip_lws (const gchar * s)
4147 while (g_ascii_isspace (*s))
4152 static const gchar *
4153 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4155 while (s > start && g_ascii_isspace (*(s - 1)))
4160 static const gchar *
4161 gst_rtspsrc_skip_commas (const gchar * s)
4163 /* The grammar allows for multiple commas */
4164 while (g_ascii_isspace (*s) || *s == ',')
4169 static const gchar *
4170 gst_rtspsrc_skip_item (const gchar * s)
4172 gboolean quoted = FALSE;
4173 const gchar *start = s;
4175 /* A list item ends at the last non-whitespace character
4176 * before a comma which is not inside a quoted-string. Or at
4177 * the end of the string.
4183 if (*s == '\\' && *(s + 1))
4192 return gst_rtspsrc_unskip_lws (s, start);
4196 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4200 src = quoted_string + 1;
4201 dst = quoted_string;
4202 while (*src && *src != '"') {
4203 if (*src == '\\' && *(src + 1))
4210 /* Extract the authentication tokens that the server provided for each method
4211 * into an array of structures and give those to the connection object.
4214 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4215 const gchar * header, gboolean * stale)
4217 GSList *list = NULL, *iter;
4219 gchar *item, *eq, *name_end, *value;
4221 g_return_if_fail (stale != NULL);
4223 gst_rtsp_connection_clear_auth_params (conn);
4226 /* Parse a header whose content is described by RFC2616 as
4227 * "#something", where "something" does not itself contain commas,
4228 * except as part of quoted-strings, into a list of allocated strings.
4230 header = gst_rtspsrc_skip_commas (header);
4232 end = gst_rtspsrc_skip_item (header);
4233 list = g_slist_prepend (list, g_strndup (header, end - header));
4234 header = gst_rtspsrc_skip_commas (end);
4239 list = g_slist_reverse (list);
4240 for (iter = list; iter; iter = iter->next) {
4243 eq = strchr (item, '=');
4245 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4246 if (name_end == item) {
4247 /* That's no good... */
4254 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4256 gst_rtsp_decode_quoted_string (value);
4260 if ((strcmp (item, "stale") == 0) && (strcmp (value, "TRUE") == 0))
4262 gst_rtsp_connection_set_auth_param (conn, item, value);
4266 g_slist_free (list);
4269 /* Parse a WWW-Authenticate Response header and determine the
4270 * available authentication methods
4272 * This code should also cope with the fact that each WWW-Authenticate
4273 * header can contain multiple challenge methods + tokens
4275 * At the moment, for Basic auth, we just do a minimal check and don't
4276 * even parse out the realm */
4278 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4279 GstRTSPConnection * conn, gboolean * stale)
4283 g_return_if_fail (hdr != NULL);
4284 g_return_if_fail (methods != NULL);
4285 g_return_if_fail (stale != NULL);
4287 /* Skip whitespace at the start of the string */
4288 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4290 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4291 *methods |= GST_RTSP_AUTH_BASIC;
4292 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4293 *methods |= GST_RTSP_AUTH_DIGEST;
4294 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4299 * gst_rtspsrc_setup_auth:
4300 * @src: the rtsp source
4302 * Configure a username and password and auth method on the
4303 * connection object based on a response we received from the
4306 * Currently, this requires that a username and password were supplied
4307 * in the uri. In the future, they may be requested on demand by sending
4308 * a message up the bus.
4310 * Returns: TRUE if authentication information could be set up correctly.
4313 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4317 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4318 GstRTSPAuthMethod method;
4319 GstRTSPResult auth_result;
4321 GstRTSPConnection *conn;
4323 gboolean stale = FALSE;
4325 conn = src->conninfo.connection;
4327 /* Identify the available auth methods and see if any are supported */
4328 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4329 &hdr, 0) == GST_RTSP_OK) {
4330 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4333 if (avail_methods == GST_RTSP_AUTH_NONE)
4334 goto no_auth_available;
4336 /* For digest auth, if the response indicates that the session
4337 * data are stale, we just update them in the connection object and
4338 * return TRUE to retry the request */
4340 src->tried_url_auth = FALSE;
4342 url = gst_rtsp_connection_get_url (conn);
4344 /* Do we have username and password available? */
4345 if (url != NULL && !src->tried_url_auth && url->user != NULL
4346 && url->passwd != NULL) {
4349 src->tried_url_auth = TRUE;
4350 GST_DEBUG_OBJECT (src,
4351 "Attempting authentication using credentials from the URL");
4353 user = src->user_id;
4354 pass = src->user_pw;
4355 GST_DEBUG_OBJECT (src,
4356 "Attempting authentication using credentials from the properties");
4359 /* FIXME: If the url didn't contain username and password or we tried them
4360 * already, request a username and passwd from the application via some kind
4361 * of credentials request message */
4363 /* If we don't have a username and passwd at this point, bail out. */
4364 if (user == NULL || pass == NULL)
4367 /* Try to configure for each available authentication method, strongest to
4369 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
4370 /* Check if this method is available on the server */
4371 if ((method & avail_methods) == 0)
4374 /* Pass the credentials to the connection to try on the next request */
4375 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
4376 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
4377 * ignore it and end up retrying later */
4378 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
4379 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
4380 gst_rtsp_auth_method_to_string (method));
4385 if (method == GST_RTSP_AUTH_NONE)
4386 goto no_auth_available;
4392 /* Output an error indicating that we couldn't connect because there were
4393 * no supported authentication protocols */
4394 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4395 ("No supported authentication protocol was found"));
4400 /* We don't fire an error message, we just return FALSE and let the
4401 * normal NOT_AUTHORIZED error be propagated */
4406 static GstRTSPResult
4407 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4408 GstRTSPMessage * request, GstRTSPMessage * response,
4409 GstRTSPStatusCode * code)
4412 GstRTSPStatusCode thecode;
4413 gchar *content_base = NULL;
4417 if (!src->short_header)
4418 gst_rtsp_ext_list_before_send (src->extensions, request);
4420 GST_DEBUG_OBJECT (src, "sending message");
4423 gst_rtsp_message_dump (request);
4425 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
4429 gst_rtsp_connection_reset_timeout (conn);
4432 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
4437 gst_rtsp_message_dump (response);
4439 switch (response->type) {
4440 case GST_RTSP_MESSAGE_REQUEST:
4441 res = gst_rtspsrc_handle_request (src, conn, response);
4442 if (res == GST_RTSP_EEOF)
4445 goto handle_request_failed;
4447 case GST_RTSP_MESSAGE_RESPONSE:
4448 /* ok, a response is good */
4449 GST_DEBUG_OBJECT (src, "received response message");
4451 case GST_RTSP_MESSAGE_DATA:
4452 /* get next response */
4453 GST_DEBUG_OBJECT (src, "ignoring data response message");
4456 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4461 thecode = response->type_data.response.code;
4463 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
4465 /* if the caller wanted the result code, we store it. */
4469 /* If the request didn't succeed, bail out before doing any more */
4470 if (thecode != GST_RTSP_STS_OK)
4473 /* store new content base if any */
4474 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
4477 g_free (src->content_base);
4478 src->content_base = g_strdup (content_base);
4480 gst_rtsp_ext_list_after_send (src->extensions, request, response);
4487 gchar *str = gst_rtsp_strresult (res);
4489 if (res != GST_RTSP_EINTR) {
4490 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4491 ("Could not send message. (%s)", str));
4493 GST_WARNING_OBJECT (src, "send interrupted");
4502 GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
4505 /* if reconnect succeeds, try again */
4507 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
4511 /* only try once after reconnect, then fallthrough and error out */
4514 gchar *str = gst_rtsp_strresult (res);
4516 if (res != GST_RTSP_EINTR) {
4517 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4518 ("Could not receive message. (%s)", str));
4520 GST_WARNING_OBJECT (src, "receive interrupted");
4528 handle_request_failed:
4530 /* ERROR was posted */
4531 gst_rtsp_message_unset (response);
4536 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4537 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4538 ("The server closed the connection."));
4539 gst_rtsp_message_unset (response);
4546 * @src: the rtsp source
4547 * @conn: the connection to send on
4548 * @request: must point to a valid request
4549 * @response: must point to an empty #GstRTSPMessage
4550 * @code: an optional code result
4552 * send @request and retrieve the response in @response. optionally @code can be
4553 * non-NULL in which case it will contain the status code of the response.
4555 * If This function returns #GST_RTSP_OK, @response will contain a valid response
4556 * message that should be cleaned with gst_rtsp_message_unset() after usage.
4558 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
4559 * @response message) if the response code was not 200 (OK).
4561 * If the attempt results in an authentication failure, then this will attempt
4562 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
4565 * Returns: #GST_RTSP_OK if the processing was successful.
4567 static GstRTSPResult
4568 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4569 GstRTSPMessage * request, GstRTSPMessage * response,
4570 GstRTSPStatusCode * code)
4572 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
4573 GstRTSPResult res = GST_RTSP_ERROR;
4576 GstRTSPMethod method = GST_RTSP_INVALID;
4582 /* make sure we don't loop forever */
4586 /* save method so we can disable it when the server complains */
4587 method = request->type_data.request.method;
4590 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
4594 case GST_RTSP_STS_UNAUTHORIZED:
4595 if (gst_rtspsrc_setup_auth (src, response)) {
4596 /* Try the request/response again after configuring the auth info
4604 } while (retry == TRUE);
4606 /* If the user requested the code, let them handle errors, otherwise
4607 * post an error below */
4610 else if (int_code != GST_RTSP_STS_OK)
4611 goto error_response;
4618 GST_DEBUG_OBJECT (src, "got error %d", res);
4623 res = GST_RTSP_ERROR;
4625 switch (response->type_data.response.code) {
4626 case GST_RTSP_STS_NOT_FOUND:
4627 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
4628 response->type_data.response.reason));
4630 case GST_RTSP_STS_MOVED_PERMANENTLY:
4631 case GST_RTSP_STS_MOVE_TEMPORARILY:
4633 gchar *new_location;
4634 GstRTSPLowerTrans transports;
4636 GST_DEBUG_OBJECT (src, "got redirection");
4637 /* if we don't have a Location Header, we must error */
4638 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
4639 &new_location, 0) < 0)
4642 /* When we receive a redirect result, we go back to the INIT state after
4643 * parsing the new URI. The caller should do the needed steps to issue
4644 * a new setup when it detects this state change. */
4645 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
4647 /* save current transports */
4648 if (src->conninfo.url)
4649 transports = src->conninfo.url->transports;
4651 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
4653 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
4655 /* set old transports */
4656 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
4657 src->conninfo.url->transports = transports;
4659 src->need_redirect = TRUE;
4660 src->state = GST_RTSP_STATE_INIT;
4664 case GST_RTSP_STS_NOT_ACCEPTABLE:
4665 case GST_RTSP_STS_NOT_IMPLEMENTED:
4666 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
4667 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
4668 gst_rtsp_method_as_text (method));
4669 src->methods &= ~method;
4673 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4674 ("Got error response: %d (%s).", response->type_data.response.code,
4675 response->type_data.response.reason));
4678 /* if we return ERROR we should unset the response ourselves */
4679 if (res == GST_RTSP_ERROR)
4680 gst_rtsp_message_unset (response);
4686 static GstRTSPResult
4687 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
4688 GstRTSPMessage * response, GstRTSPSrc * src)
4690 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
4695 /* parse the response and collect all the supported methods. We need this
4696 * information so that we don't try to send an unsupported request to the
4700 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
4702 GstRTSPHeaderField field;
4708 /* reset supported methods */
4711 /* Try Allow Header first */
4712 field = GST_RTSP_HDR_ALLOW;
4715 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4716 if (indx == 0 && !respoptions) {
4717 /* if no Allow header was found then try the Public header... */
4718 field = GST_RTSP_HDR_PUBLIC;
4719 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4724 /* If we get here, the server gave a list of supported methods, parse
4725 * them here. The string is like:
4727 * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
4729 options = g_strsplit (respoptions, ",", 0);
4731 for (i = 0; options[i]; i++) {
4735 stripped = g_strstrip (options[i]);
4736 method = gst_rtsp_find_method (stripped);
4738 /* keep bitfield of supported methods */
4739 if (method != GST_RTSP_INVALID)
4740 src->methods |= method;
4742 g_strfreev (options);
4747 if (src->methods == 0) {
4748 /* neither Allow nor Public are required, assume the server supports
4749 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
4751 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
4752 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
4754 /* always assume PLAY, FIXME, extensions should be able to override
4756 src->methods |= GST_RTSP_PLAY;
4757 /* also assume it will support Range */
4758 src->seekable = TRUE;
4760 /* we need describe and setup */
4761 if (!(src->methods & GST_RTSP_DESCRIBE))
4763 if (!(src->methods & GST_RTSP_SETUP))
4771 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4772 ("Server does not support DESCRIBE."));
4777 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4778 ("Server does not support SETUP."));
4783 /* masks to be kept in sync with the hardcoded protocol order of preference
4785 static guint protocol_masks[] = {
4786 GST_RTSP_LOWER_TRANS_UDP,
4787 GST_RTSP_LOWER_TRANS_UDP_MCAST,
4788 GST_RTSP_LOWER_TRANS_TCP,
4792 static GstRTSPResult
4793 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
4794 GstRTSPLowerTrans protocols, gchar ** transports)
4798 gboolean add_udp_str;
4803 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
4808 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
4810 /* extension listed transports, use those */
4811 if (*transports != NULL)
4814 /* it's the default */
4815 add_udp_str = FALSE;
4817 /* the default RTSP transports */
4818 result = g_string_new ("");
4819 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
4820 GST_DEBUG_OBJECT (src, "adding UDP unicast");
4822 g_string_append (result, "RTP/AVP");
4824 g_string_append (result, "/UDP");
4825 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
4826 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
4827 GST_DEBUG_OBJECT (src, "adding UDP multicast");
4829 /* we don't have to allocate any UDP ports yet, if the selected transport
4830 * turns out to be multicast we can create them and join the multicast
4831 * group indicated in the transport reply */
4832 if (result->len > 0)
4833 g_string_append (result, ",");
4834 g_string_append (result, "RTP/AVP");
4836 g_string_append (result, "/UDP");
4837 g_string_append (result, ";multicast");
4838 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
4839 GST_DEBUG_OBJECT (src, "adding TCP");
4841 if (result->len > 0)
4842 g_string_append (result, ",");
4843 g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
4845 *transports = g_string_free (result, FALSE);
4847 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
4858 static GstRTSPResult
4859 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
4860 gint orig_rtpport, gint orig_rtcpport)
4863 gint nr_udp, nr_int;
4865 gint rtpport = 0, rtcpport = 0;
4868 src = stream->parent;
4870 /* find number of placeholders first */
4871 if (strstr (*transports, "%%i2"))
4873 else if (strstr (*transports, "%%i1"))
4878 if (strstr (*transports, "%%u2"))
4880 else if (strstr (*transports, "%%u1"))
4885 if (nr_udp == 0 && nr_int == 0)
4889 if (!orig_rtpport || !orig_rtcpport) {
4890 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
4893 rtpport = orig_rtpport;
4894 rtcpport = orig_rtcpport;
4898 str = g_string_new ("");
4900 while ((next = strstr (p, "%%"))) {
4901 g_string_append_len (str, p, next - p);
4902 if (next[2] == 'u') {
4904 g_string_append_printf (str, "%d", rtpport);
4905 else if (next[3] == '2')
4906 g_string_append_printf (str, "%d", rtcpport);
4908 if (next[2] == 'i') {
4910 g_string_append_printf (str, "%d", src->free_channel);
4911 else if (next[3] == '2')
4912 g_string_append_printf (str, "%d", src->free_channel + 1);
4917 /* append final part */
4918 g_string_append (str, p);
4920 g_free (*transports);
4921 *transports = g_string_free (str, FALSE);
4929 return GST_RTSP_ERROR;
4934 gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
4936 gboolean res = FALSE;
4940 const gchar *enc = NULL;
4942 s = gst_caps_get_structure (stream->caps, 0);
4943 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
4944 res = (strstr (enc, "-REAL") != NULL);
4950 /* Perform the SETUP request for all the streams.
4952 * We ask the server for a specific transport, which initially includes all the
4953 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
4954 * two local UDP ports that we send to the server.
4956 * Once the server replied with a transport, we configure the other streams
4957 * with the same transport.
4959 * This function will also configure the stream for the selected transport,
4960 * which basically means creating the pipeline.
4962 static GstRTSPResult
4963 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
4966 GstRTSPResult res = GST_RTSP_ERROR;
4967 GstRTSPMessage request = { 0 };
4968 GstRTSPMessage response = { 0 };
4969 GstRTSPStream *stream = NULL;
4970 GstRTSPLowerTrans protocols;
4971 GstRTSPStatusCode code;
4972 gboolean unsupported_real = FALSE;
4973 gint rtpport, rtcpport;
4977 if (src->conninfo.connection) {
4978 url = gst_rtsp_connection_get_url (src->conninfo.connection);
4979 /* we initially allow all configured lower transports. based on the URL
4980 * transports and the replies from the server we narrow them down. */
4981 protocols = url->transports & src->cur_protocols;
4984 protocols = src->cur_protocols;
4990 /* reset some state */
4991 src->free_channel = 0;
4992 src->interleaved = FALSE;
4993 src->need_activate = FALSE;
4994 /* keep track of next port number, 0 is random */
4995 src->next_port_num = src->client_port_range.min;
4996 rtpport = rtcpport = 0;
4998 if (G_UNLIKELY (src->streams == NULL))
5001 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5002 GstRTSPConnection *conn;
5007 stream = (GstRTSPStream *) walk->data;
5009 /* see if we need to configure this stream */
5010 if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
5011 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5013 stream->disabled = TRUE;
5017 /* merge/overwrite global caps */
5022 s = gst_caps_get_structure (stream->caps, 0);
5024 num = gst_structure_n_fields (src->props);
5025 for (j = 0; j < num; j++) {
5029 name = gst_structure_nth_field_name (src->props, j);
5030 val = gst_structure_get_value (src->props, name);
5031 gst_structure_set_value (s, name, val);
5033 GST_DEBUG_OBJECT (src, "copied %s", name);
5037 /* skip setup if we have no URL for it */
5038 if (stream->conninfo.location == NULL) {
5039 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5043 if (src->conninfo.connection == NULL) {
5044 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5045 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5048 conn = stream->conninfo.connection;
5050 conn = src->conninfo.connection;
5052 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5053 stream->conninfo.location);
5055 /* if we have a multicast connection, only suggest multicast from now on */
5056 if (stream->is_multicast)
5057 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5060 /* first selectable protocol */
5061 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5063 if (!protocol_masks[mask])
5067 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5068 protocol_masks[mask]);
5069 /* create a string with first transport in line */
5071 res = gst_rtspsrc_create_transports_string (src,
5072 protocols & protocol_masks[mask], &transports);
5073 if (res < 0 || transports == NULL)
5074 goto setup_transport_failed;
5076 if (strlen (transports) == 0) {
5077 g_free (transports);
5078 GST_DEBUG_OBJECT (src, "no transports found");
5083 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5085 /* replace placeholders with real values, this function will optionally
5086 * allocate UDP ports and other info needed to execute the setup request */
5087 res = gst_rtspsrc_prepare_transports (stream, &transports,
5088 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5090 g_free (transports);
5091 goto setup_transport_failed;
5094 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5096 /* create SETUP request */
5098 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5099 stream->conninfo.location);
5101 g_free (transports);
5102 goto create_request_failed;
5105 /* select transport, copy is made when adding to header so we can free it. */
5106 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5107 g_free (transports);
5109 /* if the user wants a non default RTP packet size we add the blocksize
5111 if (src->rtp_blocksize > 0) {
5112 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5113 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5118 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5121 /* handle the code ourselves */
5122 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5126 case GST_RTSP_STS_OK:
5128 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5129 gst_rtsp_message_unset (&request);
5130 gst_rtsp_message_unset (&response);
5131 /* cleanup of leftover transport */
5132 gst_rtspsrc_stream_free_udp (stream);
5133 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5134 * we might be in this case */
5135 if (stream->container && rtpport && rtcpport && !retry) {
5136 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5141 /* this transport did not go down well, but we may have others to try
5142 * that we did not send yet, try those and only give up then
5143 * but not without checking for lost cause/extension so we can
5144 * post a nicer/more useful error message later */
5145 if (!unsupported_real)
5146 unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
5147 /* select next available protocol, give up on this stream if none */
5149 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5151 if (!protocol_masks[mask] || unsupported_real)
5156 /* cleanup of leftover transport and move to the next stream */
5157 gst_rtspsrc_stream_free_udp (stream);
5158 goto response_error;
5161 /* parse response transport */
5163 gchar *resptrans = NULL;
5164 GstRTSPTransport transport = { 0 };
5166 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5169 gst_rtspsrc_stream_free_udp (stream);
5173 /* parse transport, go to next stream on parse error */
5174 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5175 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5179 /* update allowed transports for other streams. once the transport of
5180 * one stream has been determined, we make sure that all other streams
5181 * are configured in the same way */
5182 switch (transport.lower_transport) {
5183 case GST_RTSP_LOWER_TRANS_TCP:
5184 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5185 protocols = GST_RTSP_LOWER_TRANS_TCP;
5186 src->interleaved = TRUE;
5187 /* update free channels */
5189 MAX (transport.interleaved.min, src->free_channel);
5191 MAX (transport.interleaved.max, src->free_channel);
5192 src->free_channel++;
5194 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5195 /* only allow multicast for other streams */
5196 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5197 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5199 case GST_RTSP_LOWER_TRANS_UDP:
5200 /* only allow unicast for other streams */
5201 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5202 protocols = GST_RTSP_LOWER_TRANS_UDP;
5205 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5206 transport.lower_transport);
5210 if (!stream->container || (!src->interleaved && !retry)) {
5211 /* now configure the stream with the selected transport */
5212 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5213 GST_DEBUG_OBJECT (src,
5214 "could not configure stream %p transport, skipping stream",
5217 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5218 /* retain the first allocated UDP port pair */
5219 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5220 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5223 /* we need to activate at least one streams when we detect activity */
5224 src->need_activate = TRUE;
5226 /* clean up our transport struct */
5227 gst_rtsp_transport_init (&transport);
5228 /* clean up used RTSP messages */
5229 gst_rtsp_message_unset (&request);
5230 gst_rtsp_message_unset (&response);
5234 /* store the transport protocol that was configured */
5235 src->cur_protocols = protocols;
5237 gst_rtsp_ext_list_stream_select (src->extensions, url);
5239 /* if there is nothing to activate, error out */
5240 if (!src->need_activate)
5241 goto nothing_to_activate;
5248 /* no transport possible, post an error and stop */
5249 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5250 ("Could not connect to server, no protocols left"));
5251 return GST_RTSP_ERROR;
5255 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5256 ("SDP contains no streams"));
5257 return GST_RTSP_ERROR;
5259 create_request_failed:
5261 gchar *str = gst_rtsp_strresult (res);
5263 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5264 ("Could not create request. (%s)", str));
5268 setup_transport_failed:
5270 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5271 ("Could not setup transport."));
5272 res = GST_RTSP_ERROR;
5277 const gchar *str = gst_rtsp_status_as_text (code);
5279 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5280 ("Error (%d): %s", code, GST_STR_NULL (str)));
5281 res = GST_RTSP_ERROR;
5286 gchar *str = gst_rtsp_strresult (res);
5288 if (res != GST_RTSP_EINTR) {
5289 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5290 ("Could not send message. (%s)", str));
5292 GST_WARNING_OBJECT (src, "send interrupted");
5299 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5300 ("Server did not select transport."));
5301 res = GST_RTSP_ERROR;
5304 nothing_to_activate:
5306 /* none of the available error codes is really right .. */
5307 if (unsupported_real) {
5308 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5309 (_("No supported stream was found. You might need to install a "
5310 "GStreamer RTSP extension plugin for Real media streams.")),
5313 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5314 (_("No supported stream was found. You might need to allow "
5315 "more transport protocols or may otherwise be missing "
5316 "the right GStreamer RTSP extension plugin.")), (NULL));
5318 return GST_RTSP_ERROR;
5322 gst_rtsp_message_unset (&request);
5323 gst_rtsp_message_unset (&response);
5329 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
5330 GstSegment * segment)
5333 GstRTSPTimeRange *therange;
5336 gst_rtsp_range_free (src->range);
5338 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
5339 GST_DEBUG_OBJECT (src, "parsed range %s", range);
5340 src->range = therange;
5342 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
5344 gst_segment_init (segment, GST_FORMAT_TIME);
5348 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
5349 therange->min.type, therange->min.seconds, therange->max.type,
5350 therange->max.seconds);
5352 if (therange->min.type == GST_RTSP_TIME_NOW)
5354 else if (therange->min.type == GST_RTSP_TIME_END)
5357 seconds = therange->min.seconds * GST_SECOND;
5359 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
5360 GST_TIME_ARGS (seconds));
5362 /* we need to start playback without clipping from the position reported by
5364 segment->start = seconds;
5365 segment->position = seconds;
5367 if (therange->max.type == GST_RTSP_TIME_NOW)
5369 else if (therange->max.type == GST_RTSP_TIME_END)
5372 seconds = therange->max.seconds * GST_SECOND;
5374 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
5375 GST_TIME_ARGS (seconds));
5377 /* live (WMS) server might send overflowed large max as its idea of infinity,
5378 * compensate to prevent problems later on */
5379 if (seconds != -1 && seconds < 0) {
5381 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
5384 /* live (WMS) might send min == max, which is not worth recording */
5385 if (segment->duration == -1 && seconds == segment->start)
5388 /* don't change duration with unknown value, we might have a valid value
5389 * there that we want to keep. */
5391 segment->duration = seconds;
5396 /* must be called with the RTSP state lock */
5397 static GstRTSPResult
5398 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
5404 /* prepare global stream caps properties */
5406 gst_structure_remove_all_fields (src->props);
5408 src->props = gst_structure_new_empty ("RTSPProperties");
5411 gst_sdp_message_dump (sdp);
5413 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
5415 gst_segment_init (&src->segment, GST_FORMAT_TIME);
5417 /* parse range for duration reporting. */
5422 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
5426 /* keep track of the range and configure it in the segment */
5427 if (gst_rtspsrc_parse_range (src, range, &src->segment))
5431 /* try to find a global control attribute. Note that a '*' means that we should
5432 * do aggregate control with the current url (so we don't do anything and
5433 * leave the current connection as is) */
5435 const gchar *control;
5438 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
5439 if (control == NULL)
5442 /* only take fully qualified urls */
5443 if (g_str_has_prefix (control, "rtsp://"))
5447 g_free (src->conninfo.location);
5448 src->conninfo.location = g_strdup (control);
5449 /* make a connection for this, if there was a connection already, nothing
5451 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
5452 GST_ERROR_OBJECT (src, "could not connect");
5455 /* we need to keep the control url separate from the connection url because
5456 * the rules for constructing the media control url need it */
5457 g_free (src->control);
5458 src->control = g_strdup (control);
5461 /* create streams */
5462 n_streams = gst_sdp_message_medias_len (sdp);
5463 for (i = 0; i < n_streams; i++) {
5464 gst_rtspsrc_create_stream (src, sdp, i);
5467 src->state = GST_RTSP_STATE_INIT;
5470 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
5473 /* reset our state */
5474 src->need_range = TRUE;
5477 src->state = GST_RTSP_STATE_READY;
5484 GST_ERROR_OBJECT (src, "setup failed");
5489 static GstRTSPResult
5490 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
5494 GstRTSPMessage request = { 0 };
5495 GstRTSPMessage response = { 0 };
5498 gchar *respcont = NULL;
5501 src->need_redirect = FALSE;
5503 /* can't continue without a valid url */
5504 if (G_UNLIKELY (src->conninfo.url == NULL)) {
5505 res = GST_RTSP_EINVAL;
5508 src->tried_url_auth = FALSE;
5510 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
5511 goto connect_failed;
5513 /* create OPTIONS */
5514 GST_DEBUG_OBJECT (src, "create options...");
5516 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
5517 src->conninfo.url_str);
5519 goto create_request_failed;
5522 GST_DEBUG_OBJECT (src, "send options...");
5525 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
5528 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5533 if (!gst_rtspsrc_parse_methods (src, &response))
5536 /* create DESCRIBE */
5537 GST_DEBUG_OBJECT (src, "create describe...");
5539 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
5540 src->conninfo.url_str);
5542 goto create_request_failed;
5544 /* we only accept SDP for now */
5545 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
5549 GST_DEBUG_OBJECT (src, "send describe...");
5552 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
5555 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5559 /* we only perform redirect for the describe, currently */
5560 if (src->need_redirect) {
5561 /* close connection, we don't have to send a TEARDOWN yet, ignore the
5563 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5565 gst_rtsp_message_unset (&request);
5566 gst_rtsp_message_unset (&response);
5572 /* it could be that the DESCRIBE method was not implemented */
5573 if (!src->methods & GST_RTSP_DESCRIBE)
5576 /* check if reply is SDP */
5577 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
5579 /* could not be set but since the request returned OK, we assume it
5580 * was SDP, else check it. */
5582 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
5583 goto wrong_content_type;
5586 /* get message body and parse as SDP */
5587 gst_rtsp_message_get_body (&response, &data, &size);
5588 if (data == NULL || size == 0)
5591 GST_DEBUG_OBJECT (src, "parse SDP...");
5592 gst_sdp_message_new (sdp);
5593 gst_sdp_message_parse_buffer (data, size, *sdp);
5595 /* clean up any messages */
5596 gst_rtsp_message_unset (&request);
5597 gst_rtsp_message_unset (&response);
5604 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
5605 ("No valid RTSP URL was provided"));
5610 gchar *str = gst_rtsp_strresult (res);
5612 if (res != GST_RTSP_EINTR) {
5613 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5614 ("Failed to connect. (%s)", str));
5616 GST_WARNING_OBJECT (src, "connect interrupted");
5621 create_request_failed:
5623 gchar *str = gst_rtsp_strresult (res);
5625 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5626 ("Could not create request. (%s)", str));
5632 /* Don't post a message - the rtsp_send method will have
5633 * taken care of it because we passed NULL for the response code */
5638 /* error was posted */
5639 res = GST_RTSP_ERROR;
5644 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5645 ("Server does not support SDP, got %s.", respcont));
5646 res = GST_RTSP_ERROR;
5651 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5652 ("Server can not provide an SDP."));
5653 res = GST_RTSP_ERROR;
5658 if (src->conninfo.connection) {
5659 GST_DEBUG_OBJECT (src, "free connection");
5660 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5662 gst_rtsp_message_unset (&request);
5663 gst_rtsp_message_unset (&response);
5668 static GstRTSPResult
5669 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
5674 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
5676 if (src->sdp == NULL) {
5677 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
5681 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
5686 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
5693 GST_WARNING_OBJECT (src, "can't get sdp");
5694 src->open_error = TRUE;
5699 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
5700 src->open_error = TRUE;
5705 static GstRTSPResult
5706 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
5708 GstRTSPMessage request = { 0 };
5709 GstRTSPMessage response = { 0 };
5710 GstRTSPResult res = GST_RTSP_OK;
5714 GST_DEBUG_OBJECT (src, "TEARDOWN...");
5716 if (src->state < GST_RTSP_STATE_READY) {
5717 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
5724 /* construct a control url */
5726 control = src->control;
5728 control = src->conninfo.url_str;
5730 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
5733 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5734 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5736 GstRTSPConnInfo *info;
5738 /* try aggregate control first but do non-aggregate control otherwise */
5740 setup_url = control;
5741 else if ((setup_url = stream->conninfo.location) == NULL)
5744 if (src->conninfo.connection) {
5745 info = &src->conninfo;
5746 } else if (stream->conninfo.connection) {
5747 info = &stream->conninfo;
5751 if (!info->connected)
5756 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
5758 goto create_request_failed;
5761 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
5764 gst_rtspsrc_send (src, info->connection, &request, &response,
5768 /* FIXME, parse result? */
5769 gst_rtsp_message_unset (&request);
5770 gst_rtsp_message_unset (&response);
5773 /* early exit when we did aggregate control */
5779 /* close connections */
5780 GST_DEBUG_OBJECT (src, "closing connection...");
5781 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5782 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5783 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5784 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
5788 gst_rtspsrc_cleanup (src);
5790 src->state = GST_RTSP_STATE_INVALID;
5793 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
5798 create_request_failed:
5800 gchar *str = gst_rtsp_strresult (res);
5802 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5803 ("Could not create request. (%s)", str));
5809 gchar *str = gst_rtsp_strresult (res);
5811 gst_rtsp_message_unset (&request);
5812 if (res != GST_RTSP_EINTR) {
5813 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5814 ("Could not send message. (%s)", str));
5816 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
5823 GST_DEBUG_OBJECT (src,
5824 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
5829 /* RTP-Info is of the format:
5831 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
5833 * rtptime corresponds to the timestamp for the NPT time given in the header
5834 * seqbase corresponds to the next sequence number we received. This number
5835 * indicates the first seqnum after the seek and should be used to discard
5836 * packets that are from before the seek.
5839 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
5844 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
5846 infos = g_strsplit (rtpinfo, ",", 0);
5847 for (i = 0; infos[i]; i++) {
5849 GstRTSPStream *stream;
5853 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
5855 /* init values, types of seqbase and timebase are bigger than needed so we
5856 * can store -1 as uninitialized values */
5861 /* parse url, find stream for url.
5862 * parse seq and rtptime. The seq number should be configured in the rtp
5863 * depayloader or session manager to detect gaps. Same for the rtptime, it
5864 * should be used to create an initial time newsegment. */
5865 fields = g_strsplit (infos[i], ";", 0);
5866 for (j = 0; fields[j]; j++) {
5867 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
5868 /* remove leading whitespace */
5869 fields[j] = g_strchug (fields[j]);
5870 if (g_str_has_prefix (fields[j], "url=")) {
5871 /* get the url and the stream */
5873 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
5874 } else if (g_str_has_prefix (fields[j], "seq=")) {
5875 seqbase = atoi (fields[j] + 4);
5876 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
5877 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
5880 g_strfreev (fields);
5881 /* now we need to store the values for the caps of the stream */
5882 if (stream != NULL) {
5883 GST_DEBUG_OBJECT (src,
5884 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
5885 stream, seqbase, timebase);
5887 /* we have a stream, configure detected params */
5888 stream->seqbase = seqbase;
5889 stream->timebase = timebase;
5898 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
5903 interval = strtoul (rtcp, NULL, 10);
5904 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
5909 interval *= GST_MSECOND;
5911 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5912 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5914 /* already (optionally) retrieved this when configuring manager */
5915 if (stream->session) {
5916 GObject *rtpsession = stream->session;
5918 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
5920 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
5924 /* now it happens that (Xenon) server sending this may also provide bogus
5925 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
5926 * and just use RTP-Info to sync */
5928 GObjectClass *klass;
5930 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
5931 if (g_object_class_find_property (klass, "rtcp-sync")) {
5932 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
5933 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
5939 gst_rtspsrc_get_float (const gchar * dstr)
5941 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5943 /* canonicalise floating point string so we can handle float strings
5944 * in the form "24.930" or "24,930" irrespective of the current locale */
5945 g_strlcpy (s, dstr, sizeof (s));
5946 g_strdelimit (s, ",", '.');
5947 return g_ascii_strtod (s, NULL);
5951 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
5953 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5955 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
5956 g_strlcpy (val_str, "now", sizeof (val_str));
5958 if (segment->position == 0) {
5959 g_strlcpy (val_str, "0", sizeof (val_str));
5961 g_ascii_dtostr (val_str, sizeof (val_str),
5962 ((gdouble) segment->position) / GST_SECOND);
5965 return g_strdup_printf ("npt=%s-", val_str);
5969 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
5971 stream->timebase = -1;
5972 stream->seqbase = -1;
5976 stream->caps = gst_caps_make_writable (stream->caps);
5977 s = gst_caps_get_structure (stream->caps, 0);
5978 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
5982 static GstRTSPResult
5983 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
5985 GstRTSPResult res = GST_RTSP_OK;
5987 if (src->state < GST_RTSP_STATE_READY) {
5988 res = GST_RTSP_ERROR;
5989 if (src->open_error) {
5990 GST_DEBUG_OBJECT (src, "the stream was in error");
5994 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
5996 if ((res = gst_rtspsrc_open (src, async)) < 0) {
5997 GST_DEBUG_OBJECT (src, "failed to open stream");
6006 static GstRTSPResult
6007 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6009 GstRTSPMessage request = { 0 };
6010 GstRTSPMessage response = { 0 };
6011 GstRTSPResult res = GST_RTSP_OK;
6017 GST_DEBUG_OBJECT (src, "PLAY...");
6019 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6022 if (!(src->methods & GST_RTSP_PLAY))
6025 if (src->state == GST_RTSP_STATE_PLAYING)
6028 if (!src->conninfo.connection || !src->conninfo.connected)
6031 /* send some dummy packets before we activate the receive in the
6033 gst_rtspsrc_send_dummy_packets (src);
6035 /* activate receive elements;
6036 * only in async case, since receive elements may not have been affected
6037 * by overall state change (e.g. not around yet),
6038 * do not mess with state in sync case (e.g. seeking) */
6040 gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
6042 /* construct a control url */
6044 control = src->control;
6046 control = src->conninfo.url_str;
6048 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6049 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6051 GstRTSPConnection *conn;
6053 /* try aggregate control first but do non-aggregate control otherwise */
6055 setup_url = control;
6056 else if ((setup_url = stream->conninfo.location) == NULL)
6059 if (src->conninfo.connection) {
6060 conn = src->conninfo.connection;
6061 } else if (stream->conninfo.connection) {
6062 conn = stream->conninfo.connection;
6068 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6070 goto create_request_failed;
6072 if (src->need_range) {
6073 hval = gen_range_header (src, segment);
6075 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
6079 if (segment->rate != 1.0) {
6080 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6082 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6084 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6086 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6090 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6092 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6095 /* seek may have silently failed as it is not supported */
6096 if (!(src->methods & GST_RTSP_PLAY)) {
6097 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6098 /* obviously it is supported as we made it here */
6099 src->methods |= GST_RTSP_PLAY;
6100 src->seekable = FALSE;
6101 /* but there is nothing to parse in the response,
6102 * so convey we have no idea and not to expect anything particular */
6103 clear_rtp_base (src, stream);
6107 /* need to do for all streams */
6108 for (run = src->streams; run; run = g_list_next (run))
6109 clear_rtp_base (src, (GstRTSPStream *) run->data);
6111 /* NOTE the above also disables npt based eos detection */
6112 /* and below forces position to 0,
6113 * which is visible feedback we lost the plot */
6114 segment->start = segment->position = src->last_pos;
6117 gst_rtsp_message_unset (&request);
6119 /* parse RTP npt field. This is the current position in the stream (Normal
6120 * Play Time) and should be put in the NEWSEGMENT position field. */
6121 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6123 gst_rtspsrc_parse_range (src, hval, segment);
6125 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6126 segment->rate = 1.0;
6128 /* parse Speed header. This is the intended playback rate of the stream
6129 * and should be put in the NEWSEGMENT rate field. */
6130 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6131 0) == GST_RTSP_OK) {
6132 segment->rate = gst_rtspsrc_get_float (hval);
6133 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6134 &hval, 0) == GST_RTSP_OK) {
6135 segment->rate = gst_rtspsrc_get_float (hval);
6138 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6139 * for the RTP packets. If this is not present, we assume all starts from 0...
6140 * This is info for the RTP session manager that we pass to it in caps. */
6142 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6143 &hval, hval_idx++) == GST_RTSP_OK)
6144 gst_rtspsrc_parse_rtpinfo (src, hval);
6146 /* some servers indicate RTCP parameters in PLAY response,
6147 * rather than properly in SDP */
6148 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6149 &hval, 0) == GST_RTSP_OK)
6150 gst_rtspsrc_handle_rtcp_interval (src, hval);
6152 gst_rtsp_message_unset (&response);
6154 /* early exit when we did aggregate control */
6158 /* set again when needed */
6159 src->need_range = FALSE;
6161 /* configure the caps of the streams after we parsed all headers. */
6162 gst_rtspsrc_configure_caps (src, segment);
6164 src->running = TRUE;
6165 src->base_time = -1;
6166 src->state = GST_RTSP_STATE_PLAYING;
6169 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6170 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6171 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6172 stream->discont = TRUE;
6177 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6184 GST_DEBUG_OBJECT (src, "failed to open stream");
6189 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6194 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6197 create_request_failed:
6199 gchar *str = gst_rtsp_strresult (res);
6201 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6202 ("Could not create request. (%s)", str));
6208 gchar *str = gst_rtsp_strresult (res);
6210 gst_rtsp_message_unset (&request);
6211 if (res != GST_RTSP_EINTR) {
6212 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6213 ("Could not send message. (%s)", str));
6215 GST_WARNING_OBJECT (src, "PLAY interrupted");
6222 static GstRTSPResult
6223 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
6225 GstRTSPResult res = GST_RTSP_OK;
6226 GstRTSPMessage request = { 0 };
6227 GstRTSPMessage response = { 0 };
6231 GST_DEBUG_OBJECT (src, "PAUSE...");
6233 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6236 if (!(src->methods & GST_RTSP_PAUSE))
6239 if (src->state == GST_RTSP_STATE_READY)
6242 if (!src->conninfo.connection || !src->conninfo.connected)
6245 /* construct a control url */
6247 control = src->control;
6249 control = src->conninfo.url_str;
6251 /* loop over the streams. We might exit the loop early when we could do an
6252 * aggregate control */
6253 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6254 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6255 GstRTSPConnection *conn;
6258 /* try aggregate control first but do non-aggregate control otherwise */
6260 setup_url = control;
6261 else if ((setup_url = stream->conninfo.location) == NULL)
6264 if (src->conninfo.connection) {
6265 conn = src->conninfo.connection;
6266 } else if (stream->conninfo.connection) {
6267 conn = stream->conninfo.connection;
6273 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
6274 ("Sending PAUSE request"));
6277 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
6279 goto create_request_failed;
6281 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6284 gst_rtsp_message_unset (&request);
6285 gst_rtsp_message_unset (&response);
6287 /* exit early when we did agregate control */
6293 src->state = GST_RTSP_STATE_READY;
6297 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
6304 GST_DEBUG_OBJECT (src, "failed to open stream");
6309 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
6314 GST_DEBUG_OBJECT (src, "we were already PAUSED");
6317 create_request_failed:
6319 gchar *str = gst_rtsp_strresult (res);
6321 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6322 ("Could not create request. (%s)", str));
6328 gchar *str = gst_rtsp_strresult (res);
6330 gst_rtsp_message_unset (&request);
6331 if (res != GST_RTSP_EINTR) {
6332 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6333 ("Could not send message. (%s)", str));
6335 GST_WARNING_OBJECT (src, "PAUSE interrupted");
6343 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
6345 GstRTSPSrc *rtspsrc;
6347 rtspsrc = GST_RTSPSRC (bin);
6349 switch (GST_MESSAGE_TYPE (message)) {
6350 case GST_MESSAGE_EOS:
6351 gst_message_unref (message);
6353 case GST_MESSAGE_ELEMENT:
6355 const GstStructure *s = gst_message_get_structure (message);
6357 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
6358 gboolean ignore_timeout;
6360 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
6362 GST_OBJECT_LOCK (rtspsrc);
6363 ignore_timeout = rtspsrc->ignore_timeout;
6364 rtspsrc->ignore_timeout = TRUE;
6365 GST_OBJECT_UNLOCK (rtspsrc);
6367 /* we only act on the first udp timeout message, others are irrelevant
6368 * and can be ignored. */
6369 if (!ignore_timeout)
6370 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
6372 gst_message_unref (message);
6375 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6378 case GST_MESSAGE_ERROR:
6381 GstRTSPStream *stream;
6384 udpsrc = GST_MESSAGE_SRC (message);
6386 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
6387 GST_ELEMENT_NAME (udpsrc));
6389 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
6393 /* we ignore the RTCP udpsrc */
6394 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
6397 /* if we get error messages from the udp sources, that's not a problem as
6398 * long as not all of them error out. We also don't really know what the
6399 * problem is, the message does not give enough detail... */
6400 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
6401 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
6402 if (ret != GST_FLOW_OK)
6406 gst_message_unref (message);
6410 /* fatal but not our message, forward */
6411 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6416 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6422 /* the thread where everything happens */
6424 gst_rtspsrc_thread (GstRTSPSrc * src)
6428 gboolean running = FALSE;
6430 GST_OBJECT_LOCK (src);
6431 cmd = src->loop_cmd;
6432 src->loop_cmd = CMD_WAIT;
6433 GST_DEBUG_OBJECT (src, "got command %d", cmd);
6435 /* we got the message command, so ensure communication is possible again */
6436 gst_rtspsrc_connection_flush (src, FALSE);
6438 /* we allow these to be interrupted */
6439 if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
6440 src->waiting = TRUE;
6441 GST_OBJECT_UNLOCK (src);
6445 ret = gst_rtspsrc_open (src, TRUE);
6448 ret = gst_rtspsrc_play (src, &src->segment, TRUE);
6449 if (ret == GST_RTSP_OK)
6453 ret = gst_rtspsrc_pause (src, TRUE, TRUE);
6454 if (ret == GST_RTSP_OK)
6458 ret = gst_rtspsrc_close (src, TRUE, FALSE);
6461 running = gst_rtspsrc_loop (src);
6464 ret = gst_rtspsrc_reconnect (src, FALSE);
6465 if (ret == GST_RTSP_OK)
6472 GST_OBJECT_LOCK (src);
6473 /* and go back to sleep */
6474 if (src->loop_cmd == CMD_WAIT) {
6476 src->loop_cmd = CMD_LOOP;
6478 gst_task_pause (src->task);
6481 src->waiting = FALSE;
6482 GST_OBJECT_UNLOCK (src);
6486 gst_rtspsrc_start (GstRTSPSrc * src)
6488 GST_DEBUG_OBJECT (src, "starting");
6490 GST_OBJECT_LOCK (src);
6492 src->loop_cmd = CMD_WAIT;
6494 if (src->task == NULL) {
6495 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src);
6496 if (src->task == NULL)
6499 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
6501 GST_OBJECT_UNLOCK (src);
6508 GST_ERROR_OBJECT (src, "failed to create task");
6514 gst_rtspsrc_stop (GstRTSPSrc * src)
6518 GST_DEBUG_OBJECT (src, "stopping");
6520 /* also cancels pending task */
6521 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
6523 GST_OBJECT_LOCK (src);
6524 if ((task = src->task)) {
6526 GST_OBJECT_UNLOCK (src);
6528 gst_task_stop (task);
6530 /* make sure it is not running */
6531 GST_RTSP_STREAM_LOCK (src);
6532 GST_RTSP_STREAM_UNLOCK (src);
6534 /* now wait for the task to finish */
6535 gst_task_join (task);
6537 /* and free the task */
6538 gst_object_unref (GST_OBJECT (task));
6540 GST_OBJECT_LOCK (src);
6542 GST_OBJECT_UNLOCK (src);
6544 /* ensure synchronously all is closed and clean */
6545 gst_rtspsrc_close (src, FALSE, TRUE);
6550 static GstStateChangeReturn
6551 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
6553 GstRTSPSrc *rtspsrc;
6554 GstStateChangeReturn ret;
6556 rtspsrc = GST_RTSPSRC (element);
6558 switch (transition) {
6559 case GST_STATE_CHANGE_NULL_TO_READY:
6560 if (!gst_rtspsrc_start (rtspsrc))
6563 case GST_STATE_CHANGE_READY_TO_PAUSED:
6564 /* init some state */
6565 rtspsrc->cur_protocols = rtspsrc->protocols;
6566 /* first attempt, don't ignore timeouts */
6567 rtspsrc->ignore_timeout = FALSE;
6568 rtspsrc->open_error = FALSE;
6569 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN);
6571 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6572 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6573 /* unblock the tcp tasks and make the loop waiting */
6574 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT);
6576 case GST_STATE_CHANGE_PAUSED_TO_READY:
6582 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
6583 if (ret == GST_STATE_CHANGE_FAILURE)
6586 switch (transition) {
6587 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6588 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY);
6590 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6591 /* send pause request and keep the idle task around */
6592 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE);
6593 ret = GST_STATE_CHANGE_NO_PREROLL;
6595 case GST_STATE_CHANGE_READY_TO_PAUSED:
6596 ret = GST_STATE_CHANGE_NO_PREROLL;
6598 case GST_STATE_CHANGE_PAUSED_TO_READY:
6599 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE);
6601 case GST_STATE_CHANGE_READY_TO_NULL:
6602 gst_rtspsrc_stop (rtspsrc);
6613 GST_DEBUG_OBJECT (rtspsrc, "start failed");
6614 return GST_STATE_CHANGE_FAILURE;
6619 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
6622 GstRTSPSrc *rtspsrc;
6624 rtspsrc = GST_RTSPSRC (element);
6626 if (GST_EVENT_IS_DOWNSTREAM (event)) {
6627 res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
6629 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
6636 /*** GSTURIHANDLER INTERFACE *************************************************/
6639 gst_rtspsrc_uri_get_type (GType type)
6644 static const gchar *const *
6645 gst_rtspsrc_uri_get_protocols (GType type)
6647 static const gchar *protocols[] =
6648 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
6654 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
6656 GstRTSPSrc *src = GST_RTSPSRC (handler);
6658 /* FIXME: make thread-safe */
6659 return g_strdup (src->conninfo.location);
6663 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
6668 GstRTSPUrl *newurl = NULL;
6669 GstSDPMessage *sdp = NULL;
6671 src = GST_RTSPSRC (handler);
6673 /* same URI, we're fine */
6674 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
6677 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
6678 if ((res = gst_sdp_message_new (&sdp) < 0))
6681 GST_DEBUG_OBJECT (src, "parsing SDP message");
6682 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
6686 GST_DEBUG_OBJECT (src, "parsing URI");
6687 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
6691 /* if worked, free previous and store new url object along with the original
6693 GST_DEBUG_OBJECT (src, "configuring URI");
6694 g_free (src->conninfo.location);
6695 src->conninfo.location = g_strdup (uri);
6696 gst_rtsp_url_free (src->conninfo.url);
6697 src->conninfo.url = newurl;
6698 g_free (src->conninfo.url_str);
6700 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
6702 src->conninfo.url_str = NULL;
6705 gst_sdp_message_free (src->sdp);
6707 src->from_sdp = sdp != NULL;
6709 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
6710 GST_DEBUG_OBJECT (src, "request uri is: %s",
6711 GST_STR_NULL (src->conninfo.url_str));
6718 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
6723 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
6724 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6725 "Could not create SDP");
6730 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
6731 GST_STR_NULL (uri));
6732 gst_sdp_message_free (sdp);
6733 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6739 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
6740 GST_STR_NULL (uri), res);
6741 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6742 "Invalid RTSP URI");
6748 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
6750 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
6752 iface->get_type = gst_rtspsrc_uri_get_type;
6753 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
6754 iface->get_uri = gst_rtspsrc_uri_get_uri;
6755 iface->set_uri = gst_rtspsrc_uri_set_uri;