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 stream RTP data to
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 streaming RTP payloaded data on the network.
39 * This element also implements the URI scheme `rtp://` allowing to send
40 * data on the network by bins that allow use the URI to determine the sink.
41 * The RTP URI handler also allows setting properties through the URI query.
49 #include "gstrtpsink.h"
50 #include "gstrtp-utils.h"
52 GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
53 #define GST_CAT_DEFAULT gst_rtp_sink_debug
55 #define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
56 #define DEFAULT_PROP_TTL 64
57 #define DEFAULT_PROP_TTL_MC 1
70 static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
73 #define gst_rtp_sink_parent_class parent_class
74 G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
75 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
76 GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
78 #define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
79 #define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
80 #define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
82 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
85 GST_STATIC_CAPS ("application/x-rtp"));
87 static GstStateChangeReturn
88 gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
91 gst_rtp_sink_set_property (GObject * object, guint prop_id,
92 const GValue * value, GParamSpec * pspec)
94 GstRtpSink *self = GST_RTP_SINK (object);
100 GST_RTP_SINK_LOCK (object);
101 uri = gst_uri_from_string (g_value_get_string (value));
106 gst_uri_unref (self->uri);
108 /* RTP data ports should be even according to RFC 3550, while the
109 * RTCP is sent on odd ports. Just warn if there is a mismatch. */
110 if (gst_uri_get_port (self->uri) % 2)
111 GST_WARNING_OBJECT (self,
112 "Port %u is not even, this is not standard (see RFC 3550).",
113 gst_uri_get_port (self->uri));
115 gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
116 GST_RTP_SINK_UNLOCK (object);
120 self->ttl = g_value_get_int (value);
123 self->ttl_mc = g_value_get_int (value);
126 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
132 gst_rtp_sink_get_property (GObject * object, guint prop_id,
133 GValue * value, GParamSpec * pspec)
135 GstRtpSink *self = GST_RTP_SINK (object);
139 GST_RTP_SINK_LOCK (object);
141 g_value_take_string (value, gst_uri_to_string (self->uri));
143 g_value_set_string (value, NULL);
144 GST_RTP_SINK_UNLOCK (object);
147 g_value_set_int (value, self->ttl);
150 g_value_set_int (value, self->ttl_mc);
153 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
159 gst_rtp_sink_finalize (GObject * gobject)
161 GstRtpSink *self = GST_RTP_SINK (gobject);
164 gst_uri_unref (self->uri);
166 g_mutex_clear (&self->lock);
167 G_OBJECT_CLASS (parent_class)->finalize (gobject);
171 gst_rtp_sink_setup_elements (GstRtpSink * self)
179 /* Should not be NULL */
180 g_return_val_if_fail (self->uri != NULL, FALSE);
182 /* if not already configured */
183 if (self->funnel_rtp == NULL) {
184 self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
185 if (self->funnel_rtp == NULL) {
186 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
187 ("%s", "funnel_rtp element is not available"));
191 self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
192 if (self->funnel_rtcp == NULL) {
193 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
194 ("%s", "funnel_rtcp element is not available"));
198 self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
199 if (self->rtp_sink == NULL) {
200 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
201 ("%s", "rtp_sink element is not available"));
205 self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
206 if (self->rtcp_src == NULL) {
207 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
208 ("%s", "rtcp_src element is not available"));
212 self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
213 if (self->rtcp_sink == NULL) {
214 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
215 ("%s", "rtcp_sink element is not available"));
219 gst_bin_add (GST_BIN (self), self->funnel_rtp);
220 gst_bin_add (GST_BIN (self), self->funnel_rtcp);
222 /* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
223 * not all at the same moment */
224 g_object_set (self->rtp_sink,
225 "host", gst_uri_get_host (self->uri),
226 "port", gst_uri_get_port (self->uri),
227 "ttl", self->ttl, "ttl-mc", self->ttl_mc, NULL);
229 gst_bin_add (GST_BIN (self), self->rtp_sink);
231 g_object_set (self->rtcp_sink,
232 "host", gst_uri_get_host (self->uri),
233 "port", gst_uri_get_port (self->uri) + 1,
234 "ttl", self->ttl, "ttl-mc", self->ttl_mc,
235 /* Set false since we're reusing a socket */
236 "auto-multicast", FALSE, NULL);
238 gst_bin_add (GST_BIN (self), self->rtcp_sink);
240 /* no need to set address if unicast */
241 caps = gst_caps_new_empty_simple ("application/x-rtcp");
242 g_object_set (self->rtcp_src,
243 "port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
244 gst_caps_unref (caps);
246 addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
247 if (g_inet_address_get_is_multicast (addr)) {
248 g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
251 g_object_unref (addr);
253 gst_bin_add (GST_BIN (self), self->rtcp_src);
255 gst_element_link (self->funnel_rtp, self->rtp_sink);
256 gst_element_link (self->funnel_rtcp, self->rtcp_sink);
258 gst_element_sync_state_with_parent (self->funnel_rtp);
259 gst_element_sync_state_with_parent (self->funnel_rtcp);
260 gst_element_sync_state_with_parent (self->rtp_sink);
261 gst_element_sync_state_with_parent (self->rtcp_src);
263 g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
264 g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
266 gst_element_sync_state_with_parent (self->rtcp_sink);
270 /* pads are all named */
271 g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
272 gst_element_link_pads (self->rtpbin, name, self->funnel_rtp, "sink_%u");
274 g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
275 gst_element_link_pads (self->rtpbin, name, self->funnel_rtcp, "sink_%u");
277 g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
278 gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
284 gst_rtp_sink_request_new_pad (GstElement * element,
285 GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
287 GstRtpSink *self = GST_RTP_SINK (element);
290 if (self->rtpbin == NULL) {
291 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
292 ("%s", "rtpbin element is not available"));
296 if (gst_rtp_sink_setup_elements (self) == FALSE)
299 GST_RTP_SINK_LOCK (self);
301 pad = gst_element_get_request_pad (self->rtpbin, "send_rtp_sink_%u");
302 g_return_val_if_fail (pad != NULL, NULL);
304 GST_RTP_SINK_UNLOCK (self);
310 gst_rtp_sink_release_pad (GstElement * element, GstPad * pad)
312 GstRtpSink *self = GST_RTP_SINK (element);
313 GstPad *rpad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
315 GST_RTP_SINK_LOCK (self);
316 gst_element_release_request_pad (self->rtpbin, rpad);
317 gst_object_unref (rpad);
319 gst_pad_set_active (pad, FALSE);
320 gst_element_remove_pad (GST_ELEMENT (self), pad);
322 GST_RTP_SINK_UNLOCK (self);
326 gst_rtp_sink_class_init (GstRtpSinkClass * klass)
328 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
329 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
331 gobject_class->set_property = gst_rtp_sink_set_property;
332 gobject_class->get_property = gst_rtp_sink_get_property;
333 gobject_class->finalize = gst_rtp_sink_finalize;
334 gstelement_class->change_state = gst_rtp_sink_change_state;
336 gstelement_class->request_new_pad =
337 GST_DEBUG_FUNCPTR (gst_rtp_sink_request_new_pad);
338 gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_sink_release_pad);
343 * uri to stream RTP to. All GStreamer parameters can be
344 * encoded in the URI, this URI format is RFC compliant.
346 g_object_class_install_property (gobject_class, PROP_URI,
347 g_param_spec_string ("uri", "URI",
348 "URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
349 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
354 * Set the unicast TTL parameter.
356 g_object_class_install_property (gobject_class, PROP_TTL,
357 g_param_spec_int ("ttl", "Unicast TTL",
358 "Used for setting the unicast TTL parameter",
359 0, 255, DEFAULT_PROP_TTL,
360 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
365 * Set the multicast TTL parameter.
367 g_object_class_install_property (gobject_class, PROP_TTL_MC,
368 g_param_spec_int ("ttl-mc", "Multicast TTL",
369 "Used for setting the multicast TTL parameter", 0, 255,
370 DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
372 gst_element_class_add_pad_template (gstelement_class,
373 gst_static_pad_template_get (&sink_template));
375 gst_element_class_set_static_metadata (gstelement_class,
378 "Simple RTP sink", "Marc Leeman <marc.leeman@gmail.com>");
382 gst_rtp_sink_rtpbin_element_added_cb (GstBin * element,
383 GstElement * new_element, gpointer data)
385 GstRtpSink *self = GST_RTP_SINK (data);
386 GST_INFO_OBJECT (self,
387 "Element %" GST_PTR_FORMAT " added element %" GST_PTR_FORMAT ".", element,
392 gst_rtp_sink_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
395 GstRtpSink *self = GST_RTP_SINK (data);
396 GstCaps *caps = gst_pad_query_caps (pad, NULL);
399 /* Expose RTP data pad only */
400 GST_INFO_OBJECT (self,
401 "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
402 GST_PTR_FORMAT ".", element, pad, caps);
405 if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
406 /* Src pad, do not expose */
407 gst_caps_unref (caps);
411 if (G_LIKELY (caps)) {
412 GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
414 if (gst_caps_can_intersect (caps, ref_caps)) {
415 /* SRC RTCP caps, do not expose */
416 gst_caps_unref (ref_caps);
417 gst_caps_unref (caps);
421 gst_caps_unref (ref_caps);
423 GST_ERROR_OBJECT (self, "Pad with no caps detected.");
424 gst_caps_unref (caps);
428 gst_caps_unref (caps);
430 upad = gst_element_get_compatible_pad (self->funnel_rtp, pad, NULL);
432 GST_ERROR_OBJECT (self, "No compatible pad found to link pad.");
433 gst_caps_unref (caps);
437 GST_INFO_OBJECT (self, "Linking with pad %" GST_PTR_FORMAT ".", upad);
438 gst_pad_link (pad, upad);
439 gst_object_unref (upad);
443 gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
446 GstRtpSink *self = GST_RTP_SINK (data);
447 GST_INFO_OBJECT (self,
448 "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
453 gst_rtp_sink_setup_rtpbin (GstRtpSink * self)
455 self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
456 if (self->rtpbin == NULL) {
457 GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
458 ("%s", "rtpbin element is not available"));
462 /* Add rtpbin callbacks to monitor the operation of rtpbin */
463 g_signal_connect (self->rtpbin, "element-added",
464 G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
465 g_signal_connect (self->rtpbin, "pad-added",
466 G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
467 g_signal_connect (self->rtpbin, "pad-removed",
468 G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
470 gst_bin_add (GST_BIN (self), self->rtpbin);
472 gst_element_sync_state_with_parent (self->rtpbin);
477 static GstStateChangeReturn
478 gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
480 GstRtpSink *self = GST_RTP_SINK (element);
481 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
483 GST_DEBUG_OBJECT (self, "changing state: %s => %s",
484 gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
485 gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
487 switch (transition) {
488 case GST_STATE_CHANGE_NULL_TO_READY:
490 case GST_STATE_CHANGE_READY_TO_PAUSED:
496 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
497 if (ret == GST_STATE_CHANGE_FAILURE)
500 switch (transition) {
501 case GST_STATE_CHANGE_READY_TO_PAUSED:
503 case GST_STATE_CHANGE_PAUSED_TO_READY:
514 gst_rtp_sink_init (GstRtpSink * self)
517 self->funnel_rtp = NULL;
518 self->funnel_rtcp = NULL;
519 self->rtp_sink = NULL;
520 self->rtcp_src = NULL;
521 self->rtcp_sink = NULL;
523 self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
524 self->ttl = DEFAULT_PROP_TTL;
525 self->ttl_mc = DEFAULT_PROP_TTL_MC;
527 if (gst_rtp_sink_setup_rtpbin (self) == FALSE)
530 GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SINK);
531 gst_bin_set_suppressed_flags (GST_BIN (self),
532 GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
534 g_mutex_init (&self->lock);
538 gst_rtp_sink_uri_get_type (GType type)
543 static const gchar *const *
544 gst_rtp_sink_uri_get_protocols (GType type)
546 static const gchar *protocols[] = { (char *) "rtp", NULL };
552 gst_rtp_sink_uri_get_uri (GstURIHandler * handler)
554 GstRtpSink *self = (GstRtpSink *) handler;
556 return gst_uri_to_string (self->uri);
560 gst_rtp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
563 GstRtpSink *self = (GstRtpSink *) handler;
565 g_object_set (G_OBJECT (self), "uri", uri, NULL);
571 gst_rtp_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
573 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
575 iface->get_type = gst_rtp_sink_uri_get_type;
576 iface->get_protocols = gst_rtp_sink_uri_get_protocols;
577 iface->get_uri = gst_rtp_sink_uri_get_uri;
578 iface->set_uri = gst_rtp_sink_uri_set_uri;
581 /* ex: set tabstop=2 shiftwidth=2 expandtab: */