mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1842ffc906
NiceAgent and it's associated thread is alive for as long as GstWebRTCICE is alive so make sure any signal handlers connected to NiceAgent do not access data that is deleted earlier. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2073>
417 lines
12 KiB
C
417 lines
12 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "nicetransport.h"
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#include "icestream.h"
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#include <gio/gnetworking.h>
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#define GST_CAT_DEFAULT gst_webrtc_nice_transport_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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enum
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{
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SIGNAL_0,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_STREAM,
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PROP_SEND_BUFFER_SIZE,
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PROP_RECEIVE_BUFFER_SIZE
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};
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//static guint gst_webrtc_nice_transport_signals[LAST_SIGNAL] = { 0 };
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struct _GstWebRTCNiceTransportPrivate
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{
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gboolean running;
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gint send_buffer_size;
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gint receive_buffer_size;
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gulong on_new_selected_pair_id;
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gulong on_component_state_changed_id;
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};
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#define gst_webrtc_nice_transport_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCNiceTransport, gst_webrtc_nice_transport,
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GST_TYPE_WEBRTC_ICE_TRANSPORT, G_ADD_PRIVATE (GstWebRTCNiceTransport)
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_nice_transport_debug,
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"webrtcnicetransport", 0, "webrtcnicetransport");
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);
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static NiceComponentType
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_gst_component_to_nice (GstWebRTCICEComponent component)
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{
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switch (component) {
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case GST_WEBRTC_ICE_COMPONENT_RTP:
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return NICE_COMPONENT_TYPE_RTP;
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case GST_WEBRTC_ICE_COMPONENT_RTCP:
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return NICE_COMPONENT_TYPE_RTCP;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static GstWebRTCICEComponent
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_nice_component_to_gst (NiceComponentType component)
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{
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switch (component) {
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case NICE_COMPONENT_TYPE_RTP:
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return GST_WEBRTC_ICE_COMPONENT_RTP;
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case NICE_COMPONENT_TYPE_RTCP:
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return GST_WEBRTC_ICE_COMPONENT_RTCP;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static GstWebRTCICEConnectionState
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_nice_component_state_to_gst (NiceComponentState state)
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{
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switch (state) {
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case NICE_COMPONENT_STATE_DISCONNECTED:
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return GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED;
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case NICE_COMPONENT_STATE_GATHERING:
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return GST_WEBRTC_ICE_CONNECTION_STATE_NEW;
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case NICE_COMPONENT_STATE_CONNECTING:
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return GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING;
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case NICE_COMPONENT_STATE_CONNECTED:
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return GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED;
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case NICE_COMPONENT_STATE_READY:
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return GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED;
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case NICE_COMPONENT_STATE_FAILED:
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return GST_WEBRTC_ICE_CONNECTION_STATE_FAILED;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static void
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gst_webrtc_nice_transport_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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switch (prop_id) {
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case PROP_STREAM:
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if (nice->stream)
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gst_object_unref (nice->stream);
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nice->stream = g_value_dup_object (value);
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break;
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case PROP_SEND_BUFFER_SIZE:
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nice->priv->send_buffer_size = g_value_get_int (value);
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gst_webrtc_nice_transport_update_buffer_size (nice);
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break;
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case PROP_RECEIVE_BUFFER_SIZE:
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nice->priv->receive_buffer_size = g_value_get_int (value);
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gst_webrtc_nice_transport_update_buffer_size (nice);
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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_webrtc_nice_transport_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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switch (prop_id) {
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case PROP_STREAM:
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g_value_set_object (value, nice->stream);
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break;
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case PROP_SEND_BUFFER_SIZE:
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g_value_set_int (value, nice->priv->send_buffer_size);
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break;
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case PROP_RECEIVE_BUFFER_SIZE:
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g_value_set_int (value, nice->priv->receive_buffer_size);
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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_webrtc_nice_transport_finalize (GObject * object)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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NiceAgent *agent;
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GstWebRTCICE *webrtc_ice = g_weak_ref_get (&nice->stream->ice_weak);
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if (webrtc_ice) {
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g_object_get (webrtc_ice, "agent", &agent, NULL);
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if (nice->priv->on_component_state_changed_id != 0) {
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g_signal_handler_disconnect (agent,
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nice->priv->on_component_state_changed_id);
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}
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if (nice->priv->on_new_selected_pair_id != 0) {
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g_signal_handler_disconnect (agent, nice->priv->on_new_selected_pair_id);
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}
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g_object_unref (agent);
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gst_object_unref (webrtc_ice);
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}
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gst_object_unref (nice->stream);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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void
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gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice)
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{
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NiceAgent *agent = NULL;
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GPtrArray *sockets;
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guint i;
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GstWebRTCICE *webrtc_ice = g_weak_ref_get (&nice->stream->ice_weak);
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g_assert (webrtc_ice != NULL);
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g_object_get (webrtc_ice, "agent", &agent, NULL);
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g_assert (agent != NULL);
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sockets = nice_agent_get_sockets (agent, nice->stream->stream_id, 1);
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if (sockets == NULL) {
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g_object_unref (agent);
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gst_object_unref (webrtc_ice);
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return;
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}
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for (i = 0; i < sockets->len; i++) {
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GSocket *gsocket = g_ptr_array_index (sockets, i);
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#ifdef SO_SNDBUF
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if (nice->priv->send_buffer_size != 0) {
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GError *gerror = NULL;
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if (!g_socket_set_option (gsocket, SOL_SOCKET, SO_SNDBUF,
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nice->priv->send_buffer_size, &gerror))
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GST_WARNING_OBJECT (nice, "Could not set send buffer size : %s",
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gerror->message);
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g_clear_error (&gerror);
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}
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#endif
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#ifdef SO_RCVBUF
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if (nice->priv->receive_buffer_size != 0) {
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GError *gerror = NULL;
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if (!g_socket_set_option (gsocket, SOL_SOCKET, SO_RCVBUF,
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nice->priv->receive_buffer_size, &gerror))
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GST_WARNING_OBJECT (nice, "Could not set send receive size : %s",
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gerror->message);
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g_clear_error (&gerror);
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}
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#endif
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}
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g_ptr_array_unref (sockets);
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g_object_unref (agent);
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gst_object_unref (webrtc_ice);
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}
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static void
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_on_new_selected_pair (NiceAgent * agent, guint stream_id,
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NiceComponentType component, NiceCandidate * lcandidate,
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NiceCandidate * rcandidate, GWeakRef * nice_weak)
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{
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GstWebRTCNiceTransport *nice = g_weak_ref_get (nice_weak);
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GstWebRTCICETransport *ice;
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GstWebRTCICEComponent comp = _nice_component_to_gst (component);
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guint our_stream_id;
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if (!nice)
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return;
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ice = GST_WEBRTC_ICE_TRANSPORT (nice);
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g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
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if (stream_id != our_stream_id)
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goto cleanup;
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if (comp != ice->component)
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goto cleanup;
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gst_webrtc_ice_transport_selected_pair_change (ice);
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cleanup:
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gst_object_unref (nice);
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}
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static void
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_on_component_state_changed (NiceAgent * agent, guint stream_id,
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NiceComponentType component, NiceComponentState state, GWeakRef * nice_weak)
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{
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GstWebRTCNiceTransport *nice = g_weak_ref_get (nice_weak);
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GstWebRTCICETransport *ice;
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GstWebRTCICEComponent comp = _nice_component_to_gst (component);
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guint our_stream_id;
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if (!nice)
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return;
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ice = GST_WEBRTC_ICE_TRANSPORT (nice);
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g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
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if (stream_id != our_stream_id)
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goto cleanup;
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if (comp != ice->component)
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goto cleanup;
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GST_DEBUG_OBJECT (ice, "%u %u %s", stream_id, component,
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nice_component_state_to_string (state));
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gst_webrtc_ice_transport_connection_state_change (ice,
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_nice_component_state_to_gst (state));
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cleanup:
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gst_object_unref (nice);
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}
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static GWeakRef *
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weak_new (GstWebRTCNiceTransport * nice)
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{
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GWeakRef *weak = g_new0 (GWeakRef, 1);
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g_weak_ref_init (weak, nice);
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return weak;
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}
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static void
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weak_free (GWeakRef * weak)
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{
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g_weak_ref_clear (weak);
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g_free (weak);
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}
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static void
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gst_webrtc_nice_transport_constructed (GObject * object)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (object);
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NiceComponentType component = _gst_component_to_nice (ice->component);
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gboolean controlling_mode;
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guint our_stream_id;
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NiceAgent *agent;
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GstWebRTCICE *webrtc_ice = g_weak_ref_get (&nice->stream->ice_weak);
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g_assert (webrtc_ice != NULL);
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g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
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g_object_get (webrtc_ice, "agent", &agent, NULL);
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g_object_get (agent, "controlling-mode", &controlling_mode, NULL);
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ice->role =
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controlling_mode ? GST_WEBRTC_ICE_ROLE_CONTROLLING :
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GST_WEBRTC_ICE_ROLE_CONTROLLED;
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nice->priv->on_component_state_changed_id = g_signal_connect_data (agent,
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"component-state-changed", G_CALLBACK (_on_component_state_changed),
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weak_new (nice), (GClosureNotify) weak_free, (GConnectFlags) 0);
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nice->priv->on_new_selected_pair_id = g_signal_connect_data (agent,
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"new-selected-pair-full", G_CALLBACK (_on_new_selected_pair),
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weak_new (nice), (GClosureNotify) weak_free, (GConnectFlags) 0);
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ice->src = gst_element_factory_make ("nicesrc", NULL);
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if (ice->src) {
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g_object_set (ice->src, "agent", agent, "stream", our_stream_id,
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"component", component, NULL);
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}
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ice->sink = gst_element_factory_make ("nicesink", NULL);
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if (ice->sink) {
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g_object_set (ice->sink, "agent", agent, "stream", our_stream_id,
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"component", component, "async", FALSE, "enable-last-sample", FALSE,
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"sync", FALSE, NULL);
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}
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g_object_unref (agent);
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gst_object_unref (webrtc_ice);
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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gst_webrtc_nice_transport_class_init (GstWebRTCNiceTransportClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->constructed = gst_webrtc_nice_transport_constructed;
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gobject_class->get_property = gst_webrtc_nice_transport_get_property;
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gobject_class->set_property = gst_webrtc_nice_transport_set_property;
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gobject_class->finalize = gst_webrtc_nice_transport_finalize;
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g_object_class_install_property (gobject_class,
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PROP_STREAM,
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g_param_spec_object ("stream",
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"WebRTC ICE Stream", "ICE stream associated with this transport",
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GST_TYPE_WEBRTC_ICE_STREAM,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCNiceTransport:send-buffer-size:
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*
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* Size of the kernel send buffer in bytes, 0=default
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*
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* Since: 1.20
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_SEND_BUFFER_SIZE, g_param_spec_int ("send-buffer-size",
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"Send Buffer Size",
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"Size of the kernel send buffer in bytes, 0=default", 0, G_MAXINT, 0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCNiceTransport:receive-buffer-size:
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*
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* Size of the kernel receive buffer in bytes, 0=default
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*
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* Since: 1.20
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_RECEIVE_BUFFER_SIZE, g_param_spec_int ("receive-buffer-size",
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"Receive Buffer Size",
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"Size of the kernel receive buffer in bytes, 0=default", 0, G_MAXINT,
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0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_nice_transport_init (GstWebRTCNiceTransport * nice)
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{
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nice->priv = gst_webrtc_nice_transport_get_instance_private (nice);
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}
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GstWebRTCNiceTransport *
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gst_webrtc_nice_transport_new (GstWebRTCICEStream * stream,
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GstWebRTCICEComponent component)
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{
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return g_object_new (GST_TYPE_WEBRTC_NICE_TRANSPORT, "stream", stream,
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"component", component, NULL);
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}
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