gstreamer/ext/webrtc/nicetransport.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

268 lines
7.9 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "nicetransport.h"
#include "icestream.h"
#define GST_CAT_DEFAULT gst_webrtc_nice_transport_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_nice_transport_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWebRTCNiceTransport, gst_webrtc_nice_transport,
GST_TYPE_WEBRTC_ICE_TRANSPORT,
GST_DEBUG_CATEGORY_INIT (gst_webrtc_nice_transport_debug,
"webrtcnicetransport", 0, "webrtcnicetransport");
);
enum
{
SIGNAL_0,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_STREAM,
};
//static guint gst_webrtc_nice_transport_signals[LAST_SIGNAL] = { 0 };
struct _GstWebRTCNiceTransportPrivate
{
gboolean running;
};
static NiceComponentType
_gst_component_to_nice (GstWebRTCICEComponent component)
{
switch (component) {
case GST_WEBRTC_ICE_COMPONENT_RTP:
return NICE_COMPONENT_TYPE_RTP;
case GST_WEBRTC_ICE_COMPONENT_RTCP:
return NICE_COMPONENT_TYPE_RTCP;
default:
g_assert_not_reached ();
return 0;
}
}
static GstWebRTCICEComponent
_nice_component_to_gst (NiceComponentType component)
{
switch (component) {
case NICE_COMPONENT_TYPE_RTP:
return GST_WEBRTC_ICE_COMPONENT_RTP;
case NICE_COMPONENT_TYPE_RTCP:
return GST_WEBRTC_ICE_COMPONENT_RTCP;
default:
g_assert_not_reached ();
return 0;
}
}
static GstWebRTCICEConnectionState
_nice_component_state_to_gst (NiceComponentState state)
{
switch (state) {
case NICE_COMPONENT_STATE_DISCONNECTED:
return GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED;
case NICE_COMPONENT_STATE_GATHERING:
return GST_WEBRTC_ICE_CONNECTION_STATE_NEW;
case NICE_COMPONENT_STATE_CONNECTING:
return GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING;
case NICE_COMPONENT_STATE_CONNECTED:
return GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED;
case NICE_COMPONENT_STATE_READY:
return GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED;
case NICE_COMPONENT_STATE_FAILED:
return GST_WEBRTC_ICE_CONNECTION_STATE_FAILED;
default:
g_assert_not_reached ();
return 0;
}
}
static void
gst_webrtc_nice_transport_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
switch (prop_id) {
case PROP_STREAM:
if (nice->stream)
gst_object_unref (nice->stream);
nice->stream = g_value_dup_object (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_nice_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
switch (prop_id) {
case PROP_STREAM:
g_value_set_object (value, nice->stream);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_nice_transport_finalize (GObject * object)
{
GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
gst_object_unref (nice->stream);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
_on_new_selected_pair (NiceAgent * agent, guint stream_id,
NiceComponentType component, NiceCandidate * lcandidate,
NiceCandidate * rcandidate, GstWebRTCNiceTransport * nice)
{
GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (nice);
GstWebRTCICEComponent comp = _nice_component_to_gst (component);
guint our_stream_id;
g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
if (stream_id != our_stream_id)
return;
if (comp != ice->component)
return;
gst_webrtc_ice_transport_selected_pair_change (ice);
}
static void
_on_component_state_changed (NiceAgent * agent, guint stream_id,
NiceComponentType component, NiceComponentState state,
GstWebRTCNiceTransport * nice)
{
GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (nice);
GstWebRTCICEComponent comp = _nice_component_to_gst (component);
guint our_stream_id;
g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
if (stream_id != our_stream_id)
return;
if (comp != ice->component)
return;
GST_DEBUG_OBJECT (ice, "%u %u %s", stream_id, component,
nice_component_state_to_string (state));
gst_webrtc_ice_transport_connection_state_change (ice,
_nice_component_state_to_gst (state));
}
static void
gst_webrtc_nice_transport_constructed (GObject * object)
{
GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (object);
NiceComponentType component = _gst_component_to_nice (ice->component);
gboolean controlling_mode;
guint our_stream_id;
NiceAgent *agent;
g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
g_object_get (nice->stream->ice, "agent", &agent, NULL);
g_object_get (agent, "controlling-mode", &controlling_mode, NULL);
ice->role =
controlling_mode ? GST_WEBRTC_ICE_ROLE_CONTROLLING :
GST_WEBRTC_ICE_ROLE_CONTROLLED;
g_signal_connect (agent, "component-state-changed",
G_CALLBACK (_on_component_state_changed), nice);
g_signal_connect (agent, "new-selected-pair-full",
G_CALLBACK (_on_new_selected_pair), nice);
ice->src = gst_element_factory_make ("nicesrc", NULL);
if (ice->src) {
g_object_set (ice->src, "agent", agent, "stream", our_stream_id,
"component", component, NULL);
}
ice->sink = gst_element_factory_make ("nicesink", NULL);
if (ice->sink) {
g_object_set (ice->sink, "agent", agent, "stream", our_stream_id,
"component", component, "async", FALSE, "enable-last-sample", FALSE,
NULL);
if (ice->component == GST_WEBRTC_ICE_COMPONENT_RTCP)
g_object_set (ice->sink, "sync", FALSE, NULL);
}
g_object_unref (agent);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
gst_webrtc_nice_transport_class_init (GstWebRTCNiceTransportClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
g_type_class_add_private (klass, sizeof (GstWebRTCNiceTransportPrivate));
gobject_class->constructed = gst_webrtc_nice_transport_constructed;
gobject_class->get_property = gst_webrtc_nice_transport_get_property;
gobject_class->set_property = gst_webrtc_nice_transport_set_property;
gobject_class->finalize = gst_webrtc_nice_transport_finalize;
g_object_class_install_property (gobject_class,
PROP_STREAM,
g_param_spec_object ("stream",
"WebRTC ICE Stream", "ICE stream associated with this transport",
GST_TYPE_WEBRTC_ICE_STREAM,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_nice_transport_init (GstWebRTCNiceTransport * nice)
{
nice->priv =
G_TYPE_INSTANCE_GET_PRIVATE ((nice), GST_TYPE_WEBRTC_NICE_TRANSPORT,
GstWebRTCNiceTransportPrivate);
}
GstWebRTCNiceTransport *
gst_webrtc_nice_transport_new (GstWebRTCICEStream * stream,
GstWebRTCICEComponent component)
{
return g_object_new (GST_TYPE_WEBRTC_NICE_TRANSPORT, "stream", stream,
"component", component, NULL);
}