gstreamer/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c
Matthew Waters b7d0ddd1a4 webrtc: support renegotiating adding/removing RTX
We need to always add the RTX/RED/ULPFEC elements as rtpbin will only
call us once to request aux/fec senders/receivers.

We also need to regenerate the media section of the SDP instead of
blindly copying from the previous offer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1762>
2022-03-04 19:21:59 +11:00

223 lines
6.4 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);
g_object_notify (G_OBJECT (rtp_trans->sender), "transport");
}
if (rtp_trans->receiver) {
gst_object_replace ((GstObject **) & rtp_trans->receiver->transport,
(GstObject *) stream->transport);
g_object_notify (G_OBJECT (rtp_trans->receiver), "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;
}
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);
gst_clear_object (&trans->stream);
gst_clear_object (&trans->ulpfecdec);
gst_clear_object (&trans->ulpfecenc);
gst_clear_object (&trans->redenc);
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);
gst_event_replace (&trans->ssrc_event, 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;
}