mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 02:31:03 +00:00
28b0be4036
It was changed from a function to a property in the latest WebRTC spec.
235 lines
6.8 KiB
C
235 lines
6.8 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 "gstwebrtcbin.h"
|
|
#include "utils.h"
|
|
#include "webrtctransceiver.h"
|
|
|
|
#define GST_CAT_DEFAULT webrtc_transceiver_debug
|
|
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
|
|
|
|
#define webrtc_transceiver_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
|
|
GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
|
|
GST_DEBUG_CATEGORY_INIT (webrtc_transceiver_debug,
|
|
"webrtctransceiver", 0, "webrtctransceiver"););
|
|
|
|
#define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
|
|
#define DEFAULT_DO_NACK FALSE
|
|
#define DEFAULT_FEC_PERCENTAGE 100
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_WEBRTC,
|
|
PROP_FEC_TYPE,
|
|
PROP_FEC_PERCENTAGE,
|
|
PROP_DO_NACK,
|
|
};
|
|
|
|
void
|
|
webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
|
|
TransportStream * stream)
|
|
{
|
|
GstWebRTCRTPTransceiver *rtp_trans;
|
|
|
|
g_return_if_fail (WEBRTC_IS_TRANSCEIVER (trans));
|
|
|
|
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
|
|
|
|
gst_object_replace ((GstObject **) & trans->stream, (GstObject *) stream);
|
|
|
|
if (rtp_trans->sender)
|
|
gst_object_replace ((GstObject **) & rtp_trans->sender->transport,
|
|
(GstObject *) stream->transport);
|
|
if (rtp_trans->receiver)
|
|
gst_object_replace ((GstObject **) & rtp_trans->receiver->transport,
|
|
(GstObject *) stream->transport);
|
|
|
|
if (rtp_trans->sender)
|
|
gst_object_replace ((GstObject **) & rtp_trans->sender->rtcp_transport,
|
|
(GstObject *) stream->rtcp_transport);
|
|
if (rtp_trans->receiver)
|
|
gst_object_replace ((GstObject **) & rtp_trans->receiver->rtcp_transport,
|
|
(GstObject *) stream->rtcp_transport);
|
|
}
|
|
|
|
GstWebRTCDTLSTransport *
|
|
webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans)
|
|
{
|
|
g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
|
|
|
|
if (trans->sender) {
|
|
return trans->sender->transport;
|
|
} else if (trans->receiver) {
|
|
return trans->receiver->transport;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
GstWebRTCDTLSTransport *
|
|
webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans)
|
|
{
|
|
g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
|
|
|
|
if (trans->sender) {
|
|
return trans->sender->rtcp_transport;
|
|
} else if (trans->receiver) {
|
|
return trans->receiver->rtcp_transport;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
webrtc_transceiver_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_WEBRTC:
|
|
gst_object_set_parent (GST_OBJECT (trans), g_value_get_object (value));
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (trans);
|
|
switch (prop_id) {
|
|
case PROP_WEBRTC:
|
|
break;
|
|
case PROP_FEC_TYPE:
|
|
trans->fec_type = g_value_get_enum (value);
|
|
break;
|
|
case PROP_DO_NACK:
|
|
trans->do_nack = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_FEC_PERCENTAGE:
|
|
trans->fec_percentage = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
GST_OBJECT_UNLOCK (trans);
|
|
}
|
|
|
|
static void
|
|
webrtc_transceiver_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
|
|
|
|
GST_OBJECT_LOCK (trans);
|
|
switch (prop_id) {
|
|
case PROP_FEC_TYPE:
|
|
g_value_set_enum (value, trans->fec_type);
|
|
break;
|
|
case PROP_DO_NACK:
|
|
g_value_set_boolean (value, trans->do_nack);
|
|
break;
|
|
case PROP_FEC_PERCENTAGE:
|
|
g_value_set_uint (value, trans->fec_percentage);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
GST_OBJECT_UNLOCK (trans);
|
|
}
|
|
|
|
static void
|
|
webrtc_transceiver_finalize (GObject * object)
|
|
{
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
|
|
|
|
if (trans->stream)
|
|
gst_object_unref (trans->stream);
|
|
trans->stream = NULL;
|
|
|
|
if (trans->local_rtx_ssrc_map)
|
|
gst_structure_free (trans->local_rtx_ssrc_map);
|
|
trans->local_rtx_ssrc_map = NULL;
|
|
|
|
gst_caps_replace (&trans->last_configured_caps, NULL);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
webrtc_transceiver_class_init (WebRTCTransceiverClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = (GObjectClass *) klass;
|
|
|
|
gobject_class->get_property = webrtc_transceiver_get_property;
|
|
gobject_class->set_property = webrtc_transceiver_set_property;
|
|
gobject_class->finalize = webrtc_transceiver_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_FEC_TYPE,
|
|
g_param_spec_enum ("fec-type", "FEC type",
|
|
"The type of Forward Error Correction to use",
|
|
GST_TYPE_WEBRTC_FEC_TYPE,
|
|
DEFAULT_FEC_TYPE,
|
|
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_DO_NACK,
|
|
g_param_spec_boolean ("do-nack", "Do nack",
|
|
"Whether to send negative acknowledgements for feedback",
|
|
DEFAULT_DO_NACK,
|
|
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_FEC_PERCENTAGE,
|
|
g_param_spec_uint ("fec-percentage", "FEC percentage",
|
|
"The amount of Forward Error Correction to apply",
|
|
0, 100, DEFAULT_FEC_PERCENTAGE,
|
|
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
}
|
|
|
|
static void
|
|
webrtc_transceiver_init (WebRTCTransceiver * trans)
|
|
{
|
|
}
|
|
|
|
WebRTCTransceiver *
|
|
webrtc_transceiver_new (GstWebRTCBin * webrtc, GstWebRTCRTPSender * sender,
|
|
GstWebRTCRTPReceiver * receiver)
|
|
{
|
|
WebRTCTransceiver *trans;
|
|
|
|
trans = g_object_new (webrtc_transceiver_get_type (), "sender", sender,
|
|
"receiver", receiver, "webrtc", webrtc, NULL);
|
|
|
|
return trans;
|
|
}
|