gstreamer/gst-libs/gst/webrtc/icetransport.c
Matthew Waters 1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523
2018-02-02 15:02:21 +11:00

204 lines
6.2 KiB
C

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.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:gstwebrtc-icetransport
* @short_description: RTCIceTransport object
* @title: GstWebRTCICETransport
* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCDTLSTransport
*
* <ulink url="https://www.w3.org/TR/webrtc/#rtcicetransport">https://www.w3.org/TR/webrtc/#rtcicetransport</ulink>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "icetransport.h"
#include "webrtc-enumtypes.h"
#define GST_CAT_DEFAULT gst_webrtc_ice_transport_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_ice_transport_parent_class parent_class
/* We would inherit from GstBin however when combined with the dtls transport,
* this causes loops in the graph. */
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICETransport,
gst_webrtc_ice_transport, GST_TYPE_OBJECT,
GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_transport_debug,
"webrtcicetransport", 0, "webrtcicetransport"););
enum
{
SIGNAL_0,
ON_SELECTED_CANDIDATE_PAIR_CHANGE_SIGNAL,
ON_NEW_CANDIDATE_SIGNAL,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_COMPONENT,
PROP_STATE,
PROP_GATHERING_STATE,
};
static guint gst_webrtc_ice_transport_signals[LAST_SIGNAL] = { 0 };
void
gst_webrtc_ice_transport_connection_state_change (GstWebRTCICETransport * ice,
GstWebRTCICEConnectionState new_state)
{
ice->state = new_state;
g_object_notify (G_OBJECT (ice), "state");
}
void
gst_webrtc_ice_transport_gathering_state_change (GstWebRTCICETransport * ice,
GstWebRTCICEGatheringState new_state)
{
ice->gathering_state = new_state;
g_object_notify (G_OBJECT (ice), "gathering-state");
}
void
gst_webrtc_ice_transport_selected_pair_change (GstWebRTCICETransport * ice)
{
g_signal_emit (ice,
gst_webrtc_ice_transport_signals
[ON_SELECTED_CANDIDATE_PAIR_CHANGE_SIGNAL], 0);
}
void
gst_webrtc_ice_transport_new_candidate (GstWebRTCICETransport * ice,
guint stream_id, GstWebRTCICEComponent component, gchar * attr)
{
g_signal_emit (ice, gst_webrtc_ice_transport_signals[ON_NEW_CANDIDATE_SIGNAL],
stream_id, component, attr);
}
static void
gst_webrtc_ice_transport_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
switch (prop_id) {
case PROP_COMPONENT:
webrtc->component = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_ice_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
switch (prop_id) {
case PROP_COMPONENT:
g_value_set_enum (value, webrtc->component);
break;
case PROP_STATE:
g_value_set_enum (value, webrtc->state);
break;
case PROP_GATHERING_STATE:
g_value_set_enum (value, webrtc->gathering_state);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_ice_transport_finalize (GObject * object)
{
// GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_ice_transport_constructed (GObject * object)
{
// GstWebRTCICETransport *webrtc = GST_WEBRTC_ICE_TRANSPORT (object);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
gst_webrtc_ice_transport_class_init (GstWebRTCICETransportClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = gst_webrtc_ice_transport_constructed;
gobject_class->get_property = gst_webrtc_ice_transport_get_property;
gobject_class->set_property = gst_webrtc_ice_transport_set_property;
gobject_class->finalize = gst_webrtc_ice_transport_finalize;
g_object_class_install_property (gobject_class,
PROP_COMPONENT,
g_param_spec_enum ("component",
"ICE component", "The ICE component of this transport",
GST_TYPE_WEBRTC_ICE_COMPONENT, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_STATE,
g_param_spec_enum ("state",
"ICE connection state", "The ICE connection state of this transport",
GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, 0,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_GATHERING_STATE,
g_param_spec_enum ("gathering-state",
"ICE gathering state", "The ICE gathering state of this transport",
GST_TYPE_WEBRTC_ICE_GATHERING_STATE, 0,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTC::on-selected_candidate-pair-change:
* @object: the #GstWebRTCICETransport
*/
gst_webrtc_ice_transport_signals[ON_SELECTED_CANDIDATE_PAIR_CHANGE_SIGNAL] =
g_signal_new ("on-selected-candidate-pair-change",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 0);
/**
* GstWebRTC::on-new-candidate:
* @object: the #GstWebRTCICETransport
*/
gst_webrtc_ice_transport_signals[ON_NEW_CANDIDATE_SIGNAL] =
g_signal_new ("on-new-candidate",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_STRING);
}
static void
gst_webrtc_ice_transport_init (GstWebRTCICETransport * webrtc)
{
}