2 * Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
23 * @short description: element with Uri interface to get RTP data from
26 * RTP (RFC 3550) is a protocol to stream media over the network while
27 * retaining the timing information and providing enough information to
28 * reconstruct the correct timing domain by the receiver.
30 * The RTP data port should be even, while the RTCP port should be
31 * odd. The URI that is entered defines the data port, the RTCP port will
32 * be allocated to the next port.
34 * This element hooks up the correct sockets to support both RTP as the
35 * accompanying RTCP layer.
37 * This Bin handles taking in of data from the network and provides the
40 * This element also implements the URI scheme `rtp://` allowing to render
41 * RTP streams in GStreamer based media players. The RTP URI handler also
42 * allows setting properties through the URI query.
48 #include <gst/net/net.h>
49 #include <gst/rtp/gstrtppayloads.h>
51 #include "gstrtpsrc.h"
52 #include "gstrtp-utils.h"
54 GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug);
55 #define GST_CAT_DEFAULT gst_rtp_src_debug
57 #define DEFAULT_PROP_TTL 64
58 #define DEFAULT_PROP_TTL_MC 1
59 #define DEFAULT_PROP_ENCODING_NAME NULL
60 #define DEFAULT_PROP_LATENCY 200
62 #define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
77 static void gst_rtp_src_uri_handler_init (gpointer g_iface,
80 #define gst_rtp_src_parent_class parent_class
81 G_DEFINE_TYPE_WITH_CODE (GstRtpSrc, gst_rtp_src, GST_TYPE_BIN,
82 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_src_uri_handler_init);
83 GST_DEBUG_CATEGORY_INIT (gst_rtp_src_debug, "rtpsrc", 0, "RTP Source"));
85 #define GST_RTP_SRC_GET_LOCK(obj) (&((GstRtpSrc*)(obj))->lock)
86 #define GST_RTP_SRC_LOCK(obj) (g_mutex_lock (GST_RTP_SRC_GET_LOCK(obj)))
87 #define GST_RTP_SRC_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SRC_GET_LOCK(obj)))
89 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
92 GST_STATIC_CAPS ("application/x-rtp"));
94 static GstStateChangeReturn
95 gst_rtp_src_change_state (GstElement * element, GstStateChange transition);
98 * gst_rtp_src_rtpbin_erquest_pt_map_cb:
99 * @self: The current #GstRtpSrc object
101 * #GstRtpBin callback to map a pt on RTP caps.
103 * Returns: (transfer none): the guess on the RTP caps based on the PT
107 gst_rtp_src_rtpbin_request_pt_map_cb (GstElement * rtpbin, guint session_id,
108 guint pt, gpointer data)
110 GstRtpSrc *self = GST_RTP_SRC (data);
111 const GstRTPPayloadInfo *p = NULL;
113 GST_DEBUG_OBJECT (self,
114 "Requesting caps for session-id 0x%x and pt %u.", session_id, pt);
116 /* the encoding-name has more relevant information */
117 if (self->encoding_name != NULL) {
118 /* Unfortunately, the media needs to be passed in the function. Since
119 * it is not known, try for video if video not found. */
120 p = gst_rtp_payload_info_for_name ("video", self->encoding_name);
122 p = gst_rtp_payload_info_for_name ("audio", self->encoding_name);
126 /* Static payload types, this is a simple lookup */
127 if (!GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
128 p = gst_rtp_payload_info_for_pt (pt);
132 GstCaps *ret = gst_caps_new_simple ("application/x-rtp",
133 "encoding-name", G_TYPE_STRING, p->encoding_name,
134 "clock-rate", G_TYPE_INT, p->clock_rate,
135 "media", G_TYPE_STRING, p->media, NULL);
137 GST_DEBUG_OBJECT (self, "Decided on caps %" GST_PTR_FORMAT, ret);
142 GST_DEBUG_OBJECT (self, "Could not determine caps based on pt and"
143 " the encoding-name was not set.");
148 gst_rtp_src_set_property (GObject * object, guint prop_id,
149 const GValue * value, GParamSpec * pspec)
151 GstRtpSrc *self = GST_RTP_SRC (object);
158 GST_RTP_SRC_LOCK (object);
159 uri = gst_uri_from_string (g_value_get_string (value));
164 gst_uri_unref (self->uri);
166 if (gst_uri_get_port (self->uri) % 2)
167 GST_WARNING_OBJECT (self,
168 "Port %u is not even, this is not standard (see RFC 3550).",
169 gst_uri_get_port (self->uri));
170 gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
171 GST_RTP_SRC_UNLOCK (object);
175 self->ttl = g_value_get_int (value);
178 self->ttl_mc = g_value_get_int (value);
180 case PROP_ENCODING_NAME:
181 g_free (self->encoding_name);
182 self->encoding_name = g_value_dup_string (value);
184 caps = gst_rtp_src_rtpbin_request_pt_map_cb (NULL, 0, 96, self);
185 g_object_set (G_OBJECT (self->rtp_src), "caps", caps, NULL);
186 gst_caps_unref (caps);
190 self->latency = g_value_get_uint (value);
193 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
199 gst_rtp_src_get_property (GObject * object, guint prop_id,
200 GValue * value, GParamSpec * pspec)
202 GstRtpSrc *self = GST_RTP_SRC (object);
206 GST_RTP_SRC_LOCK (object);
208 g_value_take_string (value, gst_uri_to_string (self->uri));
210 g_value_set_string (value, NULL);
211 GST_RTP_SRC_UNLOCK (object);
214 g_value_set_int (value, self->ttl);
217 g_value_set_int (value, self->ttl_mc);
219 case PROP_ENCODING_NAME:
220 g_value_set_string (value, self->encoding_name);
223 g_value_set_uint (value, self->latency);
226 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
232 gst_rtp_src_finalize (GObject * gobject)
234 GstRtpSrc *self = GST_RTP_SRC (gobject);
237 gst_uri_unref (self->uri);
238 g_free (self->encoding_name);
240 g_mutex_clear (&self->lock);
241 G_OBJECT_CLASS (parent_class)->finalize (gobject);
245 gst_rtp_src_class_init (GstRtpSrcClass * klass)
247 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
248 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
250 gobject_class->set_property = gst_rtp_src_set_property;
251 gobject_class->get_property = gst_rtp_src_get_property;
252 gobject_class->finalize = gst_rtp_src_finalize;
253 gstelement_class->change_state = gst_rtp_src_change_state;
258 * uri to an RTP from. All GStreamer parameters can be
259 * encoded in the URI, this URI format is RFC compliant.
261 g_object_class_install_property (gobject_class, PROP_URI,
262 g_param_spec_string ("uri", "URI",
263 "URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
264 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
269 * Set the unicast TTL parameter. In RTP this of importance for RTCP.
271 g_object_class_install_property (gobject_class, PROP_TTL,
272 g_param_spec_int ("ttl", "Unicast TTL",
273 "Used for setting the unicast TTL parameter",
274 0, 255, DEFAULT_PROP_TTL,
275 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
280 * Set the multicast TTL parameter. In RTP this of importance for RTCP.
282 g_object_class_install_property (gobject_class, PROP_TTL_MC,
283 g_param_spec_int ("ttl-mc", "Multicast TTL",
284 "Used for setting the multicast TTL parameter", 0, 255,
285 DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
288 * GstRtpSrc:encoding-name:
290 * Set the encoding name of the stream to use. This is a short-hand for
291 * the full caps and maps typically to the encoding-name in the RTP caps.
293 g_object_class_install_property (gobject_class, PROP_ENCODING_NAME,
294 g_param_spec_string ("encoding-name", "Caps encoding name",
295 "Encoding name use to determine caps parameters",
296 DEFAULT_PROP_ENCODING_NAME,
297 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
302 * Set the size of the latency buffer in the
303 * GstRtpBin/GstRtpJitterBuffer to compensate for network jitter.
305 g_object_class_install_property (gobject_class, PROP_LATENCY,
306 g_param_spec_uint ("latency", "Buffer latency in ms",
307 "Default amount of ms to buffer in the jitterbuffers", 0,
308 G_MAXUINT, DEFAULT_PROP_LATENCY,
309 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
311 gst_element_class_add_pad_template (gstelement_class,
312 gst_static_pad_template_get (&src_template));
314 gst_element_class_set_static_metadata (gstelement_class,
315 "RTP Source element",
317 "Simple RTP src", "Marc Leeman <marc.leeman@gmail.com>");
321 gst_rtp_src_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
324 GstRtpSrc *self = GST_RTP_SRC (data);
325 GstCaps *caps = gst_pad_query_caps (pad, NULL);
329 /* Expose RTP data pad only */
330 GST_INFO_OBJECT (self,
331 "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
332 GST_PTR_FORMAT ".", element, pad, caps);
335 if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
336 /* Sink pad, do not expose */
337 gst_caps_unref (caps);
341 if (G_LIKELY (caps)) {
342 GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
344 if (gst_caps_can_intersect (caps, ref_caps)) {
345 /* SRC RTCP caps, do not expose */
346 gst_caps_unref (ref_caps);
347 gst_caps_unref (caps);
351 gst_caps_unref (ref_caps);
353 GST_ERROR_OBJECT (self, "Pad with no caps detected.");
354 gst_caps_unref (caps);
358 gst_caps_unref (caps);
360 GST_RTP_SRC_LOCK (self);
361 g_snprintf (name, 48, "src_%u", GST_ELEMENT (self)->numpads);
362 upad = gst_ghost_pad_new (name, pad);
364 gst_pad_set_active (upad, TRUE);
365 gst_element_add_pad (GST_ELEMENT (self), upad);
367 GST_RTP_SRC_UNLOCK (self);
371 gst_rtp_src_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
374 GstRtpSrc *self = GST_RTP_SRC (data);
375 GST_INFO_OBJECT (self,
376 "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
381 gst_rtp_src_rtpbin_on_ssrc_collision_cb (GstElement * rtpbin, guint session_id,
382 guint ssrc, gpointer data)
384 GstRtpSrc *self = GST_RTP_SRC (data);
386 GST_INFO_OBJECT (self,
387 "Dectected an SSRC collision: session-id 0x%x, ssrc 0x%x.", session_id,
392 gst_rtp_src_rtpbin_on_new_ssrc_cb (GstElement * rtpbin, guint session_id,
393 guint ssrc, gpointer data)
395 GstRtpSrc *self = GST_RTP_SRC (data);
397 GST_INFO_OBJECT (self, "Dectected a new SSRC: session-id 0x%x, ssrc 0x%x.",
401 static GstPadProbeReturn
402 gst_rtp_src_on_recv_rtcp (GstPad * pad, GstPadProbeInfo * info,
405 GstRtpSrc *self = GST_RTP_SRC (user_data);
407 GstNetAddressMeta *meta;
409 if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
410 GstBufferList *buffer_list = info->data;
411 buffer = gst_buffer_list_get (buffer_list, 0);
416 meta = gst_buffer_get_net_address_meta (buffer);
418 GST_OBJECT_LOCK (self);
419 g_clear_object (&self->rtcp_send_addr);
420 self->rtcp_send_addr = g_object_ref (meta->addr);
421 GST_OBJECT_UNLOCK (self);
423 return GST_PAD_PROBE_OK;
427 gst_rtp_src_attach_net_address_meta (GstRtpSrc * self, GstBuffer * buffer)
429 GST_OBJECT_LOCK (self);
430 if (self->rtcp_send_addr)
431 gst_buffer_add_net_address_meta (buffer, self->rtcp_send_addr);
432 GST_OBJECT_UNLOCK (self);
435 static GstPadProbeReturn
436 gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info,
439 GstRtpSrc *self = GST_RTP_SRC (user_data);
441 if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
442 GstBufferList *buffer_list = info->data;
446 info->data = buffer_list = gst_buffer_list_make_writable (buffer_list);
447 for (i = 0; i < gst_buffer_list_length (buffer_list); i++) {
448 buffer = gst_buffer_list_get (buffer_list, i);
449 gst_rtp_src_attach_net_address_meta (self, buffer);
452 GstBuffer *buffer = info->data;
453 info->data = buffer = gst_buffer_make_writable (buffer);
454 gst_rtp_src_attach_net_address_meta (self, buffer);
457 return GST_PAD_PROBE_OK;
461 gst_rtp_src_setup_elements (GstRtpSrc * self)
471 /* Construct the RTP receiver pipeline.
473 * udpsrc -> [recv_rtp_sink_%u] -------- [recv_rtp_src_%u_%u_%u]
475 * udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> udpsink
477 * This pipeline is fixed for now, note that optionally an FEC stream could
481 /* Should not be NULL */
482 g_return_val_if_fail (self->uri != NULL, FALSE);
484 self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
485 if (self->rtpbin == NULL) {
486 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
487 ("%s", "rtpbin element is not available"));
491 self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
492 if (self->rtp_src == NULL) {
493 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
494 ("%s", "rtp_src element is not available"));
498 self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
499 if (self->rtcp_src == NULL) {
500 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
501 ("%s", "rtcp_src element is not available"));
505 self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
506 if (self->rtcp_sink == NULL) {
507 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
508 ("%s", "rtcp_sink element is not available"));
512 /* Add rtpbin callbacks to monitor the operation of rtpbin */
513 g_signal_connect (self->rtpbin, "pad-added",
514 G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
515 g_signal_connect (self->rtpbin, "pad-removed",
516 G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
517 g_signal_connect (self->rtpbin, "request-pt-map",
518 G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
519 g_signal_connect (self->rtpbin, "on-new-ssrc",
520 G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
521 g_signal_connect (self->rtpbin, "on-ssrc-collision",
522 G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
524 g_object_set (self->rtpbin, "latency", self->latency, NULL);
526 /* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
527 * not all at the same moment */
528 gst_bin_add (GST_BIN (self), self->rtpbin);
529 gst_bin_add (GST_BIN (self), self->rtp_src);
531 g_object_set (self->rtp_src,
532 "address", gst_uri_get_host (self->uri),
533 "port", gst_uri_get_port (self->uri), NULL);
535 gst_bin_add (GST_BIN (self), self->rtcp_sink);
537 /* no need to set address if unicast */
538 caps = gst_caps_new_empty_simple ("application/x-rtcp");
539 g_object_set (self->rtcp_src,
540 "port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
541 gst_caps_unref (caps);
543 addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
544 if (g_inet_address_get_is_multicast (addr)) {
545 g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
548 g_object_unref (addr);
550 g_object_set (self->rtcp_sink,
551 "host", gst_uri_get_host (self->uri),
552 "port", gst_uri_get_port (self->uri) + 1,
553 "ttl", self->ttl, "ttl-mc", self->ttl_mc,
554 /* Set false since we're reusing a socket */
555 "auto-multicast", FALSE, NULL);
557 gst_bin_add (GST_BIN (self), self->rtcp_src);
559 /* share the socket created by the source */
560 g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket,
561 "address", &address, "port", &rtcp_port, NULL);
563 addr = g_inet_address_new_from_string (address);
566 if (g_inet_address_get_is_multicast (addr)) {
567 /* mc-ttl is not supported by dynudpsink */
568 g_socket_set_multicast_ttl (socket, self->ttl_mc);
569 /* In multicast, send RTCP to the multicast group */
570 self->rtcp_send_addr = g_inet_socket_address_new (addr, rtcp_port);
572 /* In unicast, send RTCP to the detected sender address */
573 pad = gst_element_get_static_pad (self->rtcp_src, "src");
574 self->rtcp_recv_probe = gst_pad_add_probe (pad,
575 GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
576 gst_rtp_src_on_recv_rtcp, self, NULL);
577 gst_object_unref (pad);
579 g_object_unref (addr);
581 pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
582 self->rtcp_send_probe = gst_pad_add_probe (pad,
583 GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
584 gst_rtp_src_on_send_rtcp, self, NULL);
585 gst_object_unref (pad);
587 g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
589 /* pads are all named */
590 g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
591 gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
593 g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
594 gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
596 gst_element_sync_state_with_parent (self->rtpbin);
597 gst_element_sync_state_with_parent (self->rtp_src);
598 gst_element_sync_state_with_parent (self->rtcp_sink);
600 g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
601 gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
603 gst_element_sync_state_with_parent (self->rtcp_src);
609 gst_rtp_src_stop (GstRtpSrc * self)
613 if (self->rtcp_recv_probe) {
614 pad = gst_element_get_static_pad (self->rtcp_src, "src");
615 gst_pad_remove_probe (pad, self->rtcp_recv_probe);
616 self->rtcp_recv_probe = 0;
617 gst_object_unref (pad);
620 pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
621 gst_pad_remove_probe (pad, self->rtcp_send_probe);
622 self->rtcp_send_probe = 0;
623 gst_object_unref (pad);
626 static GstStateChangeReturn
627 gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
629 GstRtpSrc *self = GST_RTP_SRC (element);
630 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
632 GST_DEBUG_OBJECT (self, "Changing state: %s => %s",
633 gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
634 gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
636 switch (transition) {
637 case GST_STATE_CHANGE_NULL_TO_READY:
638 if (gst_rtp_src_setup_elements (self) == FALSE)
639 return GST_STATE_CHANGE_FAILURE;
645 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
646 if (ret == GST_STATE_CHANGE_FAILURE)
649 switch (transition) {
650 case GST_STATE_CHANGE_READY_TO_PAUSED:
651 ret = GST_STATE_CHANGE_NO_PREROLL;
653 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
654 ret = GST_STATE_CHANGE_NO_PREROLL;
656 case GST_STATE_CHANGE_READY_TO_NULL:
657 gst_rtp_src_stop (self);
667 gst_rtp_src_init (GstRtpSrc * self)
670 self->rtp_src = NULL;
671 self->rtcp_src = NULL;
672 self->rtcp_sink = NULL;
674 self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
675 self->ttl = DEFAULT_PROP_TTL;
676 self->ttl_mc = DEFAULT_PROP_TTL_MC;
677 self->encoding_name = DEFAULT_PROP_ENCODING_NAME;
678 self->latency = DEFAULT_PROP_LATENCY;
680 GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SOURCE);
681 gst_bin_set_suppressed_flags (GST_BIN (self),
682 GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
684 g_mutex_init (&self->lock);
688 gst_rtp_src_uri_get_type (GType type)
693 static const gchar *const *
694 gst_rtp_src_uri_get_protocols (GType type)
696 static const gchar *protocols[] = { (char *) "rtp", NULL };
702 gst_rtp_src_uri_get_uri (GstURIHandler * handler)
704 GstRtpSrc *self = (GstRtpSrc *) handler;
706 return gst_uri_to_string (self->uri);
710 gst_rtp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
713 GstRtpSrc *self = (GstRtpSrc *) handler;
715 g_object_set (G_OBJECT (self), "uri", uri, NULL);
721 gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
723 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
725 iface->get_type = gst_rtp_src_uri_get_type;
726 iface->get_protocols = gst_rtp_src_uri_get_protocols;
727 iface->get_uri = gst_rtp_src_uri_get_uri;
728 iface->set_uri = gst_rtp_src_uri_set_uri;
731 /* ex: set tabstop=2 shiftwidth=2 expandtab: */