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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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The properties on the udpsink/udpsrc elements need to be set before there is any state change. If not, in a network without default gateway, udpsink tries to bind an a NULL interface and fails. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1398>
726 lines
22 KiB
C
726 lines
22 KiB
C
/* GStreamer
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* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION: gstrtsinkp
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* @title: GstRtpSink
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* @short description: element with Uri interface to stream RTP data to
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* the network.
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*
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* RTP (RFC 3550) is a protocol to stream media over the network while
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* retaining the timing information and providing enough information to
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* reconstruct the correct timing domain by the receiver.
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*
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* The RTP data port should be even, while the RTCP port should be
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* odd. The URI that is entered defines the data port, the RTCP port will
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* be allocated to the next port.
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*
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* This element hooks up the correct sockets to support both RTP as the
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* accompanying RTCP layer.
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*
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* This Bin handles streaming RTP payloaded data on the network.
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*
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* This element also implements the URI scheme `rtp://` allowing to send
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* data on the network by bins that allow use the URI to determine the sink.
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* The RTP URI handler also allows setting properties through the URI query.
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <gio/gio.h>
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#include "gstrtpsink.h"
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#include "gstrtp-utils.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
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#define GST_CAT_DEFAULT gst_rtp_sink_debug
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#define DEFAULT_PROP_TTL 64
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#define DEFAULT_PROP_TTL_MC 1
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#define DEFAULT_PROP_ADDRESS "0.0.0.0"
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#define DEFAULT_PROP_PORT 5004
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#define DEFAULT_PROP_URI "rtp://"DEFAULT_PROP_ADDRESS":"G_STRINGIFY(DEFAULT_PROP_PORT)
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#define DEFAULT_PROP_MULTICAST_IFACE NULL
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enum
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{
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PROP_0,
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PROP_URI,
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PROP_ADDRESS,
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PROP_PORT,
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PROP_TTL,
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PROP_TTL_MC,
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PROP_MULTICAST_IFACE,
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PROP_LAST
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};
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static void gst_rtp_sink_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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#define gst_rtp_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRtpSink, gst_rtp_sink, GST_TYPE_BIN,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_sink_uri_handler_init);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_sink_debug, "rtpsink", 0, "RTP Sink"));
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GST_ELEMENT_REGISTER_DEFINE (rtpsink, "rtpsink", GST_RANK_PRIMARY + 1,
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GST_TYPE_RTP_SINK);
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#define GST_RTP_SINK_GET_LOCK(obj) (&((GstRtpSink*)(obj))->lock)
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#define GST_RTP_SINK_LOCK(obj) (g_mutex_lock (GST_RTP_SINK_GET_LOCK(obj)))
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#define GST_RTP_SINK_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SINK_GET_LOCK(obj)))
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStateChangeReturn
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gst_rtp_sink_change_state (GstElement * element, GstStateChange transition);
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static void
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gst_rtp_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpSink *self = GST_RTP_SINK (object);
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switch (prop_id) {
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case PROP_URI:{
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GstUri *uri = NULL;
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GST_RTP_SINK_LOCK (object);
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uri = gst_uri_from_string (g_value_get_string (value));
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if (uri == NULL)
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break;
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if (self->uri)
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gst_uri_unref (self->uri);
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self->uri = uri;
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gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
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g_object_set (self, "address", gst_uri_get_host (self->uri), NULL);
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g_object_set (self, "port", gst_uri_get_port (self->uri), NULL);
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GST_RTP_SINK_UNLOCK (object);
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break;
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}
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case PROP_ADDRESS:
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gst_uri_set_host (self->uri, g_value_get_string (value));
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g_object_set_property (G_OBJECT (self->rtp_sink), "host", value);
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g_object_set_property (G_OBJECT (self->rtcp_sink), "host", value);
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break;
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case PROP_PORT:{
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guint port = g_value_get_uint (value);
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/* According to RFC 3550, 11, RTCP receiver port should be even
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* number and RTCP port should be the RTP port + 1 */
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if (port & 0x1)
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GST_WARNING_OBJECT (self,
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"Port %u is odd, this is not standard (see RFC 3550).", port);
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gst_uri_set_port (self->uri, port);
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g_object_set (self->rtp_sink, "port", port, NULL);
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g_object_set (self->rtcp_sink, "port", port + 1, NULL);
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break;
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}
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case PROP_TTL:
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self->ttl = g_value_get_int (value);
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g_object_set (self->rtp_sink, "ttl", self->ttl, NULL);
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g_object_set (self->rtcp_sink, "ttl", self->ttl, NULL);
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break;
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case PROP_TTL_MC:
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self->ttl_mc = g_value_get_int (value);
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g_object_set (self->rtp_sink, "ttl-mc", self->ttl_mc, NULL);
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g_object_set (self->rtcp_sink, "ttl-mc", self->ttl_mc, NULL);
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break;
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case PROP_MULTICAST_IFACE:
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g_free (self->multi_iface);
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if (g_value_get_string (value) == NULL)
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self->multi_iface = g_strdup (DEFAULT_PROP_MULTICAST_IFACE);
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else
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self->multi_iface = g_value_dup_string (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpSink *self = GST_RTP_SINK (object);
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switch (prop_id) {
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case PROP_URI:
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GST_RTP_SINK_LOCK (object);
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if (self->uri)
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g_value_take_string (value, gst_uri_to_string (self->uri));
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else
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g_value_set_string (value, NULL);
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GST_RTP_SINK_UNLOCK (object);
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break;
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case PROP_ADDRESS:
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g_value_set_string (value, gst_uri_get_host (self->uri));
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break;
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case PROP_PORT:
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g_value_set_uint (value, gst_uri_get_port (self->uri));
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break;
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case PROP_TTL:
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g_value_set_int (value, self->ttl);
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break;
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case PROP_TTL_MC:
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g_value_set_int (value, self->ttl_mc);
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break;
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case PROP_MULTICAST_IFACE:
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g_value_set_string (value, self->multi_iface);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_sink_finalize (GObject * gobject)
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{
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GstRtpSink *self = GST_RTP_SINK (gobject);
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if (self->uri)
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gst_uri_unref (self->uri);
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g_free (self->multi_iface);
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g_mutex_clear (&self->lock);
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G_OBJECT_CLASS (parent_class)->finalize (gobject);
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}
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static gboolean
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gst_rtp_sink_setup_elements (GstRtpSink * self)
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{
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/*GstPad *pad; */
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gchar name[48];
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/* pads are all named */
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g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
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gst_element_link_pads (self->rtpbin, name, self->funnel_rtp, "sink_%u");
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g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
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gst_element_link_pads (self->rtpbin, name, self->funnel_rtcp, "sink_%u");
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g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
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gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
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return TRUE;
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}
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static GstPad *
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gst_rtp_sink_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
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{
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GstRtpSink *self = GST_RTP_SINK (element);
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GstPad *rpad, *pad = NULL;
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if (self->rtpbin == NULL) {
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GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
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("%s", "rtpbin element is not available"));
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return NULL;
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}
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if (gst_rtp_sink_setup_elements (self) == FALSE)
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return NULL;
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GST_RTP_SINK_LOCK (self);
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rpad = gst_element_request_pad_simple (self->rtpbin, "send_rtp_sink_%u");
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if (rpad) {
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pad = gst_ghost_pad_new (GST_PAD_NAME (rpad), rpad);
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gst_element_add_pad (element, pad);
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gst_clear_object (&rpad);
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}
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GST_RTP_SINK_UNLOCK (self);
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g_return_val_if_fail (pad != NULL, NULL);
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return pad;
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}
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static void
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gst_rtp_sink_release_pad (GstElement * element, GstPad * pad)
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{
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GstRtpSink *self = GST_RTP_SINK (element);
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GstPad *rpad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
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GST_RTP_SINK_LOCK (self);
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gst_element_release_request_pad (self->rtpbin, rpad);
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gst_object_unref (rpad);
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gst_pad_set_active (pad, FALSE);
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gst_element_remove_pad (GST_ELEMENT (self), pad);
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GST_RTP_SINK_UNLOCK (self);
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}
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static void
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gst_rtp_sink_class_init (GstRtpSinkClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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gobject_class->set_property = gst_rtp_sink_set_property;
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gobject_class->get_property = gst_rtp_sink_get_property;
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gobject_class->finalize = gst_rtp_sink_finalize;
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gstelement_class->change_state = gst_rtp_sink_change_state;
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_sink_request_new_pad);
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gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_sink_release_pad);
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/**
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* GstRtpSink:uri:
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*
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* uri to stream RTP to. All GStreamer parameters can be
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* encoded in the URI, this URI format is RFC compliant.
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*/
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g_object_class_install_property (gobject_class, PROP_URI,
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g_param_spec_string ("uri", "URI",
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"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSink:address:
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*
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* Address to receive packets from (can be IPv4 or IPv6).
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*/
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g_object_class_install_property (gobject_class, PROP_ADDRESS,
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g_param_spec_string ("address", "Address",
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"Address to send packets to (can be IPv4 or IPv6).",
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DEFAULT_PROP_ADDRESS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSink:port:
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*
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* The port to listen to RTP packets, the RTCP port is this value
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* +1. This port must be an even number.
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*/
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g_object_class_install_property (gobject_class, PROP_PORT,
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g_param_spec_uint ("port", "Port", "The port RTP packets will be sent, "
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"the RTCP port is this value + 1. This port must be an even number.",
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2, 65534, DEFAULT_PROP_PORT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT));
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/**
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* GstRtpSink:ttl:
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*
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* Set the unicast TTL parameter.
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*/
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g_object_class_install_property (gobject_class, PROP_TTL,
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g_param_spec_int ("ttl", "Unicast TTL",
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"Used for setting the unicast TTL parameter",
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0, 255, DEFAULT_PROP_TTL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSink:ttl-mc:
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*
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* Set the multicast TTL parameter.
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*/
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g_object_class_install_property (gobject_class, PROP_TTL_MC,
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g_param_spec_int ("ttl-mc", "Multicast TTL",
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"Used for setting the multicast TTL parameter", 0, 255,
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DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/* GstRtpSink:multicast-iface:
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*
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* The networkinterface on which to join the multicast group
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*/
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g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
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g_param_spec_string ("multicast-iface", "Multicast Interface",
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"The network interface on which to join the multicast group."
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"This allows multiple interfaces separated by comma. (\"eth0,eth1\")",
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DEFAULT_PROP_MULTICAST_IFACE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Sink element",
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"Generic/Bin/Sink",
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"Simple RTP sink", "Marc Leeman <marc.leeman@gmail.com>");
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}
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static void
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gst_rtp_sink_rtpbin_element_added_cb (GstBin * element,
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GstElement * new_element, gpointer data)
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{
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GstRtpSink *self = GST_RTP_SINK (data);
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GST_INFO_OBJECT (self,
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"Element %" GST_PTR_FORMAT " added element %" GST_PTR_FORMAT ".", element,
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new_element);
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}
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static void
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gst_rtp_sink_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
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gpointer data)
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{
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GstRtpSink *self = GST_RTP_SINK (data);
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GstCaps *caps = gst_pad_query_caps (pad, NULL);
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GstPad *upad;
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/* Expose RTP data pad only */
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GST_INFO_OBJECT (self,
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"Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
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GST_PTR_FORMAT ".", element, pad, caps);
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/* Sanity checks */
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if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
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/* Src pad, do not expose */
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gst_caps_unref (caps);
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return;
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}
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if (G_LIKELY (caps)) {
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GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
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if (gst_caps_can_intersect (caps, ref_caps)) {
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/* SRC RTCP caps, do not expose */
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gst_caps_unref (ref_caps);
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gst_caps_unref (caps);
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return;
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}
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gst_caps_unref (ref_caps);
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} else {
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GST_ERROR_OBJECT (self, "Pad with no caps detected.");
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gst_caps_unref (caps);
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return;
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}
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gst_caps_unref (caps);
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upad = gst_element_get_compatible_pad (self->funnel_rtp, pad, NULL);
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if (upad == NULL) {
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GST_ERROR_OBJECT (self, "No compatible pad found to link pad.");
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gst_caps_unref (caps);
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return;
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}
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GST_INFO_OBJECT (self, "Linking with pad %" GST_PTR_FORMAT ".", upad);
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gst_pad_link (pad, upad);
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gst_object_unref (upad);
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}
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static void
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gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
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gpointer data)
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{
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GstRtpSink *self = GST_RTP_SINK (data);
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GST_INFO_OBJECT (self,
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"Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
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pad);
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}
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static gboolean
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gst_rtp_sink_reuse_socket (GstRtpSink * self)
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{
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GSocket *socket = NULL;
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gst_element_set_locked_state (self->rtcp_src, FALSE);
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gst_element_sync_state_with_parent (self->rtcp_src);
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/* share the socket created by the sink */
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|
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);
|
|
|
|
g_object_set (self->rtcp_sink, "sync", FALSE, "async", FALSE, NULL);
|
|
gst_element_set_locked_state (self->rtcp_sink, FALSE);
|
|
gst_element_sync_state_with_parent (self->rtcp_sink);
|
|
|
|
return TRUE;
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_sink_start (GstRtpSink * self)
|
|
{
|
|
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);
|
|
|
|
/* set multicast-iface on the udpsrc and udpsink elements */
|
|
g_object_set (self->rtcp_src, "multicast-iface", self->multi_iface, NULL);
|
|
g_object_set (self->rtcp_sink, "multicast-iface", self->multi_iface, NULL);
|
|
g_object_set (self->rtp_sink, "multicast-iface", self->multi_iface, 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_free (remote_addr);
|
|
g_object_unref (iaddr);
|
|
|
|
return TRUE;
|
|
|
|
dns_resolve_failed:
|
|
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
|
|
("Could not resolve hostname '%s'", gst_uri_get_host (self->uri)),
|
|
("DNS resolver reported: %s", error->message));
|
|
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:
|
|
/* Set the properties to the child elements to avoid binding to
|
|
* a NULL interface on a network without a default gateway */
|
|
if (gst_rtp_sink_start (self) == FALSE)
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
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:
|
|
/* re-use the sockets after they have been initialised */
|
|
if (gst_rtp_sink_reuse_socket (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;
|
|
self->multi_iface = g_strdup (DEFAULT_PROP_MULTICAST_IFACE);
|
|
|
|
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", "rtp_send_rtpbin0");
|
|
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_object (self->rtpbin, "element-added",
|
|
G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self, 0);
|
|
g_signal_connect_object (self->rtpbin, "pad-added",
|
|
G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self, 0);
|
|
g_signal_connect_object (self->rtpbin, "pad-removed",
|
|
G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self, 0);
|
|
|
|
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", "rtp_rtp_funnel0");
|
|
if (self->funnel_rtp == NULL) {
|
|
missing_plugin = "funnel";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
self->funnel_rtcp = gst_element_factory_make ("funnel", "rtp_rtcp_funnel0");
|
|
if (self->funnel_rtcp == NULL) {
|
|
missing_plugin = "funnel";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
self->rtp_sink = gst_element_factory_make ("udpsink", "rtp_rtp_udpsink0");
|
|
if (self->rtp_sink == NULL) {
|
|
missing_plugin = "udp";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
self->rtcp_src = gst_element_factory_make ("udpsrc", "rtp_rtcp_udpsrc0");
|
|
if (self->rtcp_src == NULL) {
|
|
missing_plugin = "udp";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
self->rtcp_sink = gst_element_factory_make ("udpsink", "rtp_rtcp_udpsink0");
|
|
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;
|
|
}
|
|
|
|
/* ex: set tabstop=2 shiftwidth=2 expandtab: */
|