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And mark string parameters as const. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
212 lines
6.3 KiB
C
212 lines
6.3 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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/**
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* SECTION:gstwebrtc-icetransport
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* @short_description: RTCIceTransport object
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* @title: GstWebRTCICETransport
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCDTLSTransport
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* @symbols:
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* - GstWebRTCICETransport
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*
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* See the [specification](https://www.w3.org/TR/webrtc/#rtcicetransport)
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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 "icetransport.h"
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#include "webrtc-enumtypes.h"
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#include "webrtc-priv.h"
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#define GST_CAT_DEFAULT gst_webrtc_ice_transport_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define gst_webrtc_ice_transport_parent_class parent_class
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/* We would inherit from GstBin however when combined with the dtls transport,
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* this causes loops in the graph. */
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G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICETransport,
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gst_webrtc_ice_transport, GST_TYPE_OBJECT,
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_transport_debug,
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"webrtcicetransport", 0, "webrtcicetransport"););
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enum
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{
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SIGNAL_0,
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ON_SELECTED_CANDIDATE_PAIR_CHANGE_SIGNAL,
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ON_NEW_CANDIDATE_SIGNAL,
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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_COMPONENT,
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PROP_STATE,
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PROP_GATHERING_STATE,
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};
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static guint gst_webrtc_ice_transport_signals[LAST_SIGNAL] = { 0 };
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void
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gst_webrtc_ice_transport_connection_state_change (GstWebRTCICETransport * ice,
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GstWebRTCICEConnectionState new_state)
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{
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GST_OBJECT_LOCK (ice);
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ice->state = new_state;
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GST_OBJECT_UNLOCK (ice);
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g_object_notify (G_OBJECT (ice), "state");
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}
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void
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gst_webrtc_ice_transport_gathering_state_change (GstWebRTCICETransport * ice,
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GstWebRTCICEGatheringState new_state)
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{
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GST_OBJECT_LOCK (ice);
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ice->gathering_state = new_state;
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GST_OBJECT_UNLOCK (ice);
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g_object_notify (G_OBJECT (ice), "gathering-state");
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}
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void
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gst_webrtc_ice_transport_selected_pair_change (GstWebRTCICETransport * ice)
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{
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g_signal_emit (ice,
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gst_webrtc_ice_transport_signals
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[ON_SELECTED_CANDIDATE_PAIR_CHANGE_SIGNAL], 0);
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}
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void
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gst_webrtc_ice_transport_new_candidate (GstWebRTCICETransport * ice,
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guint stream_id, GstWebRTCICEComponent component, const gchar * attr)
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{
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g_signal_emit (ice, gst_webrtc_ice_transport_signals[ON_NEW_CANDIDATE_SIGNAL],
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stream_id, component, attr);
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}
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static void
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gst_webrtc_ice_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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GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
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switch (prop_id) {
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case PROP_COMPONENT:
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webrtc->component = g_value_get_enum (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_webrtc_ice_transport_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
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switch (prop_id) {
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case PROP_COMPONENT:
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g_value_set_enum (value, webrtc->component);
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break;
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case PROP_STATE:
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g_value_set_enum (value, webrtc->state);
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break;
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case PROP_GATHERING_STATE:
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g_value_set_enum (value, webrtc->gathering_state);
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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_ice_transport_finalize (GObject * object)
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{
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// GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_webrtc_ice_transport_constructed (GObject * object)
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{
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// GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
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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_ice_transport_class_init (GstWebRTCICETransportClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->constructed = gst_webrtc_ice_transport_constructed;
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gobject_class->get_property = gst_webrtc_ice_transport_get_property;
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gobject_class->set_property = gst_webrtc_ice_transport_set_property;
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gobject_class->finalize = gst_webrtc_ice_transport_finalize;
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g_object_class_install_property (gobject_class,
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PROP_COMPONENT,
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g_param_spec_enum ("component",
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"ICE component", "The ICE component of this transport",
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GST_TYPE_WEBRTC_ICE_COMPONENT, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_STATE,
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g_param_spec_enum ("state",
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"ICE connection state", "The ICE connection state of this transport",
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GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, 0,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_GATHERING_STATE,
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g_param_spec_enum ("gathering-state",
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"ICE gathering state", "The ICE gathering state of this transport",
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GST_TYPE_WEBRTC_ICE_GATHERING_STATE, 0,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTC::on-selected_candidate-pair-change:
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* @object: the #GstWebRTCICETransport
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*/
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gst_webrtc_ice_transport_signals[ON_SELECTED_CANDIDATE_PAIR_CHANGE_SIGNAL] =
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g_signal_new ("on-selected-candidate-pair-change",
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G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
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G_TYPE_NONE, 0);
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/**
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* GstWebRTC::on-new-candidate:
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* @object: the #GstWebRTCICETransport
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*/
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gst_webrtc_ice_transport_signals[ON_NEW_CANDIDATE_SIGNAL] =
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g_signal_new ("on-new-candidate",
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G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
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G_TYPE_NONE, 1, G_TYPE_STRING);
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}
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static void
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gst_webrtc_ice_transport_init (GstWebRTCICETransport * webrtc)
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{
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}
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