gstreamer/ext/webrtc/transportstream.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

252 lines
7.5 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 "transportstream.h"
#include "transportsendbin.h"
#include "transportreceivebin.h"
#include "gstwebrtcice.h"
#include "gstwebrtcbin.h"
#include "utils.h"
#define transport_stream_parent_class parent_class
G_DEFINE_TYPE (TransportStream, transport_stream, GST_TYPE_OBJECT);
enum
{
PROP_0,
PROP_WEBRTC,
PROP_SESSION_ID,
PROP_RTCP_MUX,
PROP_DTLS_CLIENT,
};
static void
transport_stream_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
TransportStream *stream = TRANSPORT_STREAM (object);
switch (prop_id) {
case PROP_WEBRTC:
gst_object_set_parent (GST_OBJECT (stream), g_value_get_object (value));
break;
}
GST_OBJECT_LOCK (stream);
switch (prop_id) {
case PROP_WEBRTC:
break;
case PROP_SESSION_ID:
stream->session_id = g_value_get_uint (value);
break;
case PROP_RTCP_MUX:
stream->rtcp_mux = g_value_get_boolean (value);
break;
case PROP_DTLS_CLIENT:
stream->dtls_client = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (stream);
}
static void
transport_stream_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
TransportStream *stream = TRANSPORT_STREAM (object);
GST_OBJECT_LOCK (stream);
switch (prop_id) {
case PROP_SESSION_ID:
g_value_set_uint (value, stream->session_id);
break;
case PROP_RTCP_MUX:
g_value_set_boolean (value, stream->rtcp_mux);
break;
case PROP_DTLS_CLIENT:
g_value_set_boolean (value, stream->dtls_client);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (stream);
}
static void
transport_stream_dispose (GObject * object)
{
TransportStream *stream = TRANSPORT_STREAM (object);
if (stream->send_bin)
gst_object_unref (stream->send_bin);
stream->send_bin = NULL;
if (stream->receive_bin)
gst_object_unref (stream->receive_bin);
stream->receive_bin = NULL;
if (stream->transport)
gst_object_unref (stream->transport);
stream->transport = NULL;
if (stream->rtcp_transport)
gst_object_unref (stream->rtcp_transport);
stream->rtcp_transport = NULL;
GST_OBJECT_PARENT (object) = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
transport_stream_finalize (GObject * object)
{
TransportStream *stream = TRANSPORT_STREAM (object);
g_array_free (stream->ptmap, TRUE);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
transport_stream_constructed (GObject * object)
{
TransportStream *stream = TRANSPORT_STREAM (object);
GstWebRTCBin *webrtc;
GstWebRTCICETransport *ice_trans;
stream->transport = gst_webrtc_dtls_transport_new (stream->session_id, FALSE);
stream->rtcp_transport =
gst_webrtc_dtls_transport_new (stream->session_id, TRUE);
webrtc = GST_WEBRTC_BIN (gst_object_get_parent (GST_OBJECT (object)));
g_object_bind_property (stream->transport, "client", stream, "dtls-client",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (stream->rtcp_transport, "client", stream,
"dtls-client", G_BINDING_BIDIRECTIONAL);
g_object_bind_property (stream->transport, "certificate",
stream->rtcp_transport, "certificate", G_BINDING_BIDIRECTIONAL);
/* Need to go full Java and have a transport manager?
* Or make the caller set the ICE transport up? */
stream->stream = _find_ice_stream_for_session (webrtc, stream->session_id);
if (stream->stream == NULL) {
stream->stream = gst_webrtc_ice_add_stream (webrtc->priv->ice,
stream->session_id);
_add_ice_stream_item (webrtc, stream->session_id, stream->stream);
}
ice_trans =
gst_webrtc_ice_find_transport (webrtc->priv->ice, stream->stream,
GST_WEBRTC_ICE_COMPONENT_RTP);
gst_webrtc_dtls_transport_set_transport (stream->transport, ice_trans);
gst_object_unref (ice_trans);
ice_trans =
gst_webrtc_ice_find_transport (webrtc->priv->ice, stream->stream,
GST_WEBRTC_ICE_COMPONENT_RTCP);
gst_webrtc_dtls_transport_set_transport (stream->rtcp_transport, ice_trans);
gst_object_unref (ice_trans);
stream->send_bin = g_object_new (transport_send_bin_get_type (), "stream",
stream, NULL);
gst_object_ref_sink (stream->send_bin);
stream->receive_bin = g_object_new (transport_receive_bin_get_type (),
"stream", stream, NULL);
gst_object_ref_sink (stream->receive_bin);
gst_object_unref (webrtc);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
transport_stream_class_init (TransportStreamClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = transport_stream_constructed;
gobject_class->get_property = transport_stream_get_property;
gobject_class->set_property = transport_stream_set_property;
gobject_class->dispose = transport_stream_dispose;
gobject_class->finalize = transport_stream_finalize;
/* some acrobatics are required to set the parent before _constructed()
* has been called */
g_object_class_install_property (gobject_class,
PROP_WEBRTC,
g_param_spec_object ("webrtc", "Parent webrtcbin",
"Parent webrtcbin",
GST_TYPE_WEBRTC_BIN,
G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_SESSION_ID,
g_param_spec_uint ("session-id", "Session ID",
"Session ID used for this transport",
0, G_MAXUINT, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RTCP_MUX,
g_param_spec_boolean ("rtcp-mux", "RTCP Mux",
"Whether RTCP packets are muxed with RTP packets",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DTLS_CLIENT,
g_param_spec_boolean ("dtls-client", "DTLS client",
"Whether we take the client role in DTLS negotiation",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
clear_ptmap_item (PtMapItem * item)
{
if (item->caps)
gst_caps_unref (item->caps);
}
static void
transport_stream_init (TransportStream * stream)
{
stream->ptmap = g_array_new (FALSE, TRUE, sizeof (PtMapItem));
g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
}
TransportStream *
transport_stream_new (GstWebRTCBin * webrtc, guint session_id)
{
TransportStream *stream;
stream = g_object_new (transport_stream_get_type (), "webrtc", webrtc,
"session-id", session_id, NULL);
return stream;
}