gstreamer/gst/rtp/gstrtpsink.c
Matthew Waters 67e4684932 build: fix werror build with newer gcc
In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstbin.h:27,
                 from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:35,
                 from ../gst/rtp/gstrtpsink.h:23,
                 from ../gst/rtp/gstrtpsink.c:49:
In function ‘gst_rtp_sink_start’,
    inlined from ‘gst_rtp_sink_change_state’ at ../gst/rtp/gstrtpsink.c:509:11:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstelement.h:422:18: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
  422 |   gchar *__txt = _gst_element_error_printf text;                        \
../gst/rtp/gstrtpsink.c:476:3: note: in expansion of macro ‘GST_ELEMENT_ERROR’
  476 |   GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
      |   ^~~~~~~~~~~~~~~~~
../gst/rtp/gstrtpsink.c: In function ‘gst_rtp_sink_change_state’:
../gst/rtp/gstrtpsink.c:477:37: note: format string is defined here
  477 |       ("Could not resolve hostname '%s'", remote_addr),
      |                                     ^~

In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstbin.h:27,
                 from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:35,
                 from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/rtp/gstrtpdefs.h:27,
                 from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/rtp/rtp.h:25,
                 from ../gst/rist/gstristsink.c:72:
In function ‘gst_rist_sink_setup_rtcp_socket’,
    inlined from ‘gst_rist_sink_start’ at ../gst/rist/gstristsink.c:658:10,
    inlined from ‘gst_rist_sink_change_state’ at ../gst/rist/gstristsink.c:801:13:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstelement.h:422:18: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
  422 |   gchar *__txt = _gst_element_error_printf text;                        \
../gst/rist/gstristsink.c:595:3: note: in expansion of macro ‘GST_ELEMENT_ERROR’
  595 |   GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
      |   ^~~~~~~~~~~~~~~~~
../gst/rist/gstristsink.c: In function ‘gst_rist_sink_change_state’:
../gst/rist/gstristsink.c:596:37: note: format string is defined here
  596 |       ("Could not resolve hostname '%s'", remote_addr),
      |                                     ^~
2019-09-24 10:29:44 +10:00

673 lines
19 KiB
C

/* GStreamer
* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION: gstrtsinkp
* @title: GstRtpSink
* @short description: element with Uri interface to stream RTP data to
* the network.
*
* RTP (RFC 3550) is a protocol to stream media over the network while
* retaining the timing information and providing enough information to
* reconstruct the correct timing domain by the receiver.
*
* The RTP data port should be even, while the RTCP port should be
* odd. The URI that is entered defines the data port, the RTCP port will
* be allocated to the next port.
*
* This element hooks up the correct sockets to support both RTP as the
* accompanying RTCP layer.
*
* This Bin handles streaming RTP payloaded data on the network.
*
* This element also implements the URI scheme `rtp://` allowing to send
* data on the network by bins that allow use the URI to determine the sink.
* The RTP URI handler also allows setting properties through the URI query.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gio/gio.h>
#include "gstrtpsink.h"
#include "gstrtp-utils.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
#define GST_CAT_DEFAULT gst_rtp_sink_debug
#define DEFAULT_PROP_TTL 64
#define DEFAULT_PROP_TTL_MC 1
#define DEFAULT_PROP_ADDRESS "0.0.0.0"
#define DEFAULT_PROP_PORT 5004
#define DEFAULT_PROP_URI "rtp://"DEFAULT_PROP_ADDRESS":"G_STRINGIFY(DEFAULT_PROP_PORT)
enum
{
PROP_0,
PROP_URI,
PROP_ADDRESS,
PROP_PORT,
PROP_TTL,
PROP_TTL_MC,
PROP_LAST
};
static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
gpointer iface_data);
#define gst_rtp_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
#define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
#define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
#define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStateChangeReturn
gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
static void
gst_rtp_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpSink *self = GST_RTP_SINK (object);
switch (prop_id) {
case PROP_URI:{
GstUri *uri = NULL;
GST_RTP_SINK_LOCK (object);
uri = gst_uri_from_string (g_value_get_string (value));
if (uri == NULL)
break;
if (self->uri)
gst_uri_unref (self->uri);
self->uri = uri;
gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
g_object_set (self, "address", gst_uri_get_host (self->uri), NULL);
g_object_set (self, "port", gst_uri_get_port (self->uri), NULL);
GST_RTP_SINK_UNLOCK (object);
break;
}
case PROP_ADDRESS:
gst_uri_set_host (self->uri, g_value_get_string (value));
g_object_set_property (G_OBJECT (self->rtp_sink), "host", value);
g_object_set_property (G_OBJECT (self->rtcp_sink), "host", value);
break;
case PROP_PORT:{
guint port = g_value_get_uint (value);
/* According to RFC 3550, 11, RTCP receiver port should be even
* number and RTCP port should be the RTP port + 1 */
if (port & 0x1)
GST_WARNING_OBJECT (self,
"Port %u is odd, this is not standard (see RFC 3550).", port);
gst_uri_set_port (self->uri, port);
g_object_set (self->rtp_sink, "port", port, NULL);
g_object_set (self->rtcp_sink, "port", port + 1, NULL);
break;
}
case PROP_TTL:
self->ttl = g_value_get_int (value);
g_object_set (self->rtp_sink, "ttl", self->ttl, NULL);
g_object_set (self->rtcp_sink, "ttl", self->ttl, NULL);
break;
case PROP_TTL_MC:
self->ttl_mc = g_value_get_int (value);
g_object_set (self->rtp_sink, "ttl-mc", self->ttl_mc, NULL);
g_object_set (self->rtcp_sink, "ttl-mc", self->ttl_mc, NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpSink *self = GST_RTP_SINK (object);
switch (prop_id) {
case PROP_URI:
GST_RTP_SINK_LOCK (object);
if (self->uri)
g_value_take_string (value, gst_uri_to_string (self->uri));
else
g_value_set_string (value, NULL);
GST_RTP_SINK_UNLOCK (object);
break;
case PROP_ADDRESS:
g_value_set_string (value, gst_uri_get_host (self->uri));
break;
case PROP_PORT:
g_value_set_uint (value, gst_uri_get_port (self->uri));
break;
case PROP_TTL:
g_value_set_int (value, self->ttl);
break;
case PROP_TTL_MC:
g_value_set_int (value, self->ttl_mc);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_sink_finalize (GObject * gobject)
{
GstRtpSink *self = GST_RTP_SINK (gobject);
if (self->uri)
gst_uri_unref (self->uri);
g_mutex_clear (&self->lock);
G_OBJECT_CLASS (parent_class)->finalize (gobject);
}
static gboolean
gst_rtp_sink_setup_elements (GstRtpSink * self)
{
/*GstPad *pad; */
gchar name[48];
/* pads are all named */
g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->funnel_rtp, "sink_%u");
g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->funnel_rtcp, "sink_%u");
g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
return TRUE;
}
static GstPad *
gst_rtp_sink_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRtpSink *self = GST_RTP_SINK (element);
GstPad *pad = NULL;
if (self->rtpbin == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtpbin element is not available"));
return NULL;
}
if (gst_rtp_sink_setup_elements (self) == FALSE)
return NULL;
GST_RTP_SINK_LOCK (self);
pad = gst_element_get_request_pad (self->rtpbin, "send_rtp_sink_%u");
g_return_val_if_fail (pad != NULL, NULL);
GST_RTP_SINK_UNLOCK (self);
return pad;
}
static void
gst_rtp_sink_release_pad (GstElement * element, GstPad * pad)
{
GstRtpSink *self = GST_RTP_SINK (element);
GstPad *rpad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
GST_RTP_SINK_LOCK (self);
gst_element_release_request_pad (self->rtpbin, rpad);
gst_object_unref (rpad);
gst_pad_set_active (pad, FALSE);
gst_element_remove_pad (GST_ELEMENT (self), pad);
GST_RTP_SINK_UNLOCK (self);
}
static void
gst_rtp_sink_class_init (GstRtpSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->set_property = gst_rtp_sink_set_property;
gobject_class->get_property = gst_rtp_sink_get_property;
gobject_class->finalize = gst_rtp_sink_finalize;
gstelement_class->change_state = gst_rtp_sink_change_state;
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_sink_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_sink_release_pad);
/**
* GstRtpSink:uri:
*
* uri to stream RTP to. All GStreamer parameters can be
* encoded in the URI, this URI format is RFC compliant.
*/
g_object_class_install_property (gobject_class, PROP_URI,
g_param_spec_string ("uri", "URI",
"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:address:
*
* Address to receive packets from (can be IPv4 or IPv6).
*/
g_object_class_install_property (gobject_class, PROP_ADDRESS,
g_param_spec_string ("address", "Address",
"Address to send packets to (can be IPv4 or IPv6).",
DEFAULT_PROP_ADDRESS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:port:
*
* The port to listen to RTP packets, the RTCP port is this value
* +1. This port must be an even number.
*/
g_object_class_install_property (gobject_class, PROP_PORT,
g_param_spec_uint ("port", "Port", "The port RTP packets will be sent, "
"the RTCP port is this value + 1. This port must be an even number.",
2, 65534, DEFAULT_PROP_PORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT));
/**
* GstRtpSink:ttl:
*
* Set the unicast TTL parameter.
*/
g_object_class_install_property (gobject_class, PROP_TTL,
g_param_spec_int ("ttl", "Unicast TTL",
"Used for setting the unicast TTL parameter",
0, 255, DEFAULT_PROP_TTL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:ttl-mc:
*
* Set the multicast TTL parameter.
*/
g_object_class_install_property (gobject_class, PROP_TTL_MC,
g_param_spec_int ("ttl-mc", "Multicast TTL",
"Used for setting the multicast TTL parameter", 0, 255,
DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP Sink element",
"Generic/Bin/Sink",
"Simple RTP sink", "Marc Leeman <marc.leeman@gmail.com>");
}
static void
gst_rtp_sink_rtpbin_element_added_cb (GstBin * element,
GstElement * new_element, gpointer data)
{
GstRtpSink *self = GST_RTP_SINK (data);
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " added element %" GST_PTR_FORMAT ".", element,
new_element);
}
static void
gst_rtp_sink_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
gpointer data)
{
GstRtpSink *self = GST_RTP_SINK (data);
GstCaps *caps = gst_pad_query_caps (pad, NULL);
GstPad *upad;
/* Expose RTP data pad only */
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
GST_PTR_FORMAT ".", element, pad, caps);
/* Sanity checks */
if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
/* Src pad, do not expose */
gst_caps_unref (caps);
return;
}
if (G_LIKELY (caps)) {
GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
if (gst_caps_can_intersect (caps, ref_caps)) {
/* SRC RTCP caps, do not expose */
gst_caps_unref (ref_caps);
gst_caps_unref (caps);
return;
}
gst_caps_unref (ref_caps);
} else {
GST_ERROR_OBJECT (self, "Pad with no caps detected.");
gst_caps_unref (caps);
return;
}
gst_caps_unref (caps);
upad = gst_element_get_compatible_pad (self->funnel_rtp, pad, NULL);
if (upad == NULL) {
GST_ERROR_OBJECT (self, "No compatible pad found to link pad.");
gst_caps_unref (caps);
return;
}
GST_INFO_OBJECT (self, "Linking with pad %" GST_PTR_FORMAT ".", upad);
gst_pad_link (pad, upad);
gst_object_unref (upad);
}
static void
gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
gpointer data)
{
GstRtpSink *self = GST_RTP_SINK (data);
GST_INFO_OBJECT (self,
"Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
pad);
}
static gboolean
gst_rtp_sink_start (GstRtpSink * self)
{
GSocket *socket = NULL;
GInetAddress *iaddr = NULL;
gchar *remote_addr = NULL;
GError *error = NULL;
/* Should not be NULL */
g_return_val_if_fail (self->uri != NULL, FALSE);
iaddr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (!iaddr) {
GList *results;
GResolver *resolver = NULL;
resolver = g_resolver_get_default ();
results =
g_resolver_lookup_by_name (resolver, gst_uri_get_host (self->uri), NULL,
&error);
if (!results) {
g_object_unref (resolver);
goto dns_resolve_failed;
}
iaddr = G_INET_ADDRESS (g_object_ref (results->data));
g_resolver_free_addresses (results);
g_object_unref (resolver);
}
remote_addr = g_inet_address_to_string (iaddr);
if (g_inet_address_get_is_multicast (iaddr)) {
g_object_set (self->rtcp_src, "address", remote_addr, "port",
gst_uri_get_port (self->uri) + 1, NULL);
} else {
const gchar *any_addr;
if (g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6)
any_addr = "::";
else
any_addr = "0.0.0.0";
g_object_set (self->rtcp_src, "address", any_addr, "port", 0, NULL);
}
g_object_unref (iaddr);
gst_element_set_locked_state (self->rtcp_src, FALSE);
gst_element_sync_state_with_parent (self->rtcp_src);
/* share the socket created by the sink */
g_object_get (self->rtcp_src, "used-socket", &socket, NULL);
g_object_set (self->rtcp_sink, "socket", socket, "auto-multicast", FALSE,
"close-socket", FALSE, NULL);
g_object_unref (socket);
gst_element_set_locked_state (self->rtcp_sink, FALSE);
gst_element_sync_state_with_parent (self->rtcp_sink);
return TRUE;
dns_resolve_failed:
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("Could not resolve hostname '%s'", GST_STR_NULL (remote_addr)),
("DNS resolver reported: %s", error->message));
g_free (remote_addr);
g_error_free (error);
return FALSE;
}
static GstStateChangeReturn
gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
{
GstRtpSink *self = GST_RTP_SINK (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GST_DEBUG_OBJECT (self, "changing state: %s => %s",
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (gst_rtp_sink_start (self) == FALSE)
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
default:
break;
}
return ret;
}
static void
gst_rtp_sink_init (GstRtpSink * self)
{
const gchar *missing_plugin = NULL;
GstCaps *caps;
self->rtpbin = NULL;
self->funnel_rtp = NULL;
self->funnel_rtcp = NULL;
self->rtp_sink = NULL;
self->rtcp_src = NULL;
self->rtcp_sink = NULL;
self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
self->ttl = DEFAULT_PROP_TTL;
self->ttl_mc = DEFAULT_PROP_TTL_MC;
g_mutex_init (&self->lock);
/* Construct the RTP sender pipeline.
*
* *-> [send_rtp_sink_%u] -------- [send_rtp_src_%u] -> udpsink
* | rtpbin |
* udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> * udpsink
*/
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
missing_plugin = "rtpmanager";
goto missing_plugin;
}
gst_bin_add (GST_BIN (self), self->rtpbin);
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "element-added",
G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SINK);
gst_bin_set_suppressed_flags (GST_BIN (self),
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtp == NULL) {
missing_plugin = "funnel";
goto missing_plugin;
}
self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtcp == NULL) {
missing_plugin = "funnel";
goto missing_plugin;
}
self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtp_sink == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtcp_sink == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
gst_bin_add (GST_BIN (self), self->funnel_rtp);
gst_bin_add (GST_BIN (self), self->funnel_rtcp);
gst_bin_add (GST_BIN (self), self->rtp_sink);
gst_bin_add (GST_BIN (self), self->rtcp_src);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
gst_element_set_locked_state (self->rtcp_src, TRUE);
gst_element_set_locked_state (self->rtcp_sink, TRUE);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src, "caps", caps, NULL);
gst_caps_unref (caps);
gst_element_link (self->funnel_rtp, self->rtp_sink);
gst_element_link (self->funnel_rtcp, self->rtcp_sink);
if (missing_plugin == NULL)
return;
missing_plugin:
{
GST_ERROR_OBJECT (self, "'%s' plugin is missing.", missing_plugin);
/* Just make our element valid, so we fail cleanly */
gst_element_add_pad (GST_ELEMENT (self),
gst_pad_new_from_static_template (&sink_template, "sink_%u"));
}
}
static GstURIType
gst_rtp_sink_uri_get_type (GType type)
{
return GST_URI_SINK;
}
static const gchar *const *
gst_rtp_sink_uri_get_protocols (GType type)
{
static const gchar *protocols[] = { (char *) "rtp", NULL };
return protocols;
}
static gchar *
gst_rtp_sink_uri_get_uri (GstURIHandler * handler)
{
GstRtpSink *self = (GstRtpSink *) handler;
return gst_uri_to_string (self->uri);
}
static gboolean
gst_rtp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRtpSink *self = (GstRtpSink *) handler;
g_object_set (G_OBJECT (self), "uri", uri, NULL);
return TRUE;
}
static void
gst_rtp_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtp_sink_uri_get_type;
iface->get_protocols = gst_rtp_sink_uri_get_protocols;
iface->get_uri = gst_rtp_sink_uri_get_uri;
iface->set_uri = gst_rtp_sink_uri_set_uri;
}
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