gstreamer/gst-libs/gst/webrtc/rtptransceiver.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

186 lines
5.5 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.
*/
/**
* SECTION:gstwebrtc-transceiver
* @short_description: RTCRtpTransceiver object
* @title: GstWebRTCRTPTransceiver
* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver
*
* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface</ulink>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "rtptransceiver.h"
#define GST_CAT_DEFAULT gst_webrtc_rtp_transceiver_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_rtp_transceiver_parent_class parent_class
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCRTPTransceiver,
gst_webrtc_rtp_transceiver, GST_TYPE_OBJECT,
GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_transceiver_debug,
"webrtctransceiver", 0, "webrtctransceiver");
);
enum
{
SIGNAL_0,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_MID,
PROP_SENDER,
PROP_RECEIVER,
PROP_STOPPED, // FIXME
PROP_DIRECTION, // FIXME
PROP_MLINE,
};
//static guint gst_webrtc_rtp_transceiver_signals[LAST_SIGNAL] = { 0 };
static void
gst_webrtc_rtp_transceiver_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
switch (prop_id) {
case PROP_SENDER:
webrtc->sender = g_value_dup_object (value);
break;
case PROP_RECEIVER:
webrtc->receiver = g_value_dup_object (value);
break;
case PROP_MLINE:
webrtc->mline = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_rtp_transceiver_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
switch (prop_id) {
case PROP_SENDER:
g_value_set_object (value, webrtc->sender);
break;
case PROP_RECEIVER:
g_value_set_object (value, webrtc->receiver);
break;
case PROP_MLINE:
g_value_set_uint (value, webrtc->mline);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_rtp_transceiver_constructed (GObject * object)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
gst_object_set_parent (GST_OBJECT (webrtc->sender), GST_OBJECT (webrtc));
gst_object_set_parent (GST_OBJECT (webrtc->receiver), GST_OBJECT (webrtc));
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
gst_webrtc_rtp_transceiver_dispose (GObject * object)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
if (webrtc->sender) {
GST_OBJECT_PARENT (webrtc->sender) = NULL;
gst_object_unref (webrtc->sender);
}
webrtc->sender = NULL;
if (webrtc->receiver) {
GST_OBJECT_PARENT (webrtc->receiver) = NULL;
gst_object_unref (webrtc->receiver);
}
webrtc->receiver = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_webrtc_rtp_transceiver_finalize (GObject * object)
{
GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
g_free (webrtc->mid);
if (webrtc->codec_preferences)
gst_caps_unref (webrtc->codec_preferences);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_rtp_transceiver_class_init (GstWebRTCRTPTransceiverClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_rtp_transceiver_get_property;
gobject_class->set_property = gst_webrtc_rtp_transceiver_set_property;
gobject_class->constructed = gst_webrtc_rtp_transceiver_constructed;
gobject_class->dispose = gst_webrtc_rtp_transceiver_dispose;
gobject_class->finalize = gst_webrtc_rtp_transceiver_finalize;
g_object_class_install_property (gobject_class,
PROP_SENDER,
g_param_spec_object ("sender", "Sender",
"The RTP sender for this transceiver",
GST_TYPE_WEBRTC_RTP_SENDER,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RECEIVER,
g_param_spec_object ("receiver", "Receiver",
"The RTP receiver for this transceiver",
GST_TYPE_WEBRTC_RTP_RECEIVER,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MLINE,
g_param_spec_uint ("mlineindex", "Media Line Index",
"Index in the SDP of the Media",
0, G_MAXUINT, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_rtp_transceiver_init (GstWebRTCRTPTransceiver * webrtc)
{
}