mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 17:20:36 +00:00
303 lines
9.3 KiB
C
303 lines
9.3 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
|
|
*
|
|
* <https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include "config.h"
|
|
#endif
|
|
|
|
#include "rtptransceiver.h"
|
|
|
|
#include "webrtc-priv.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,
|
|
"webrtcrtptransceiver", 0, "webrtcrtptransceiver");
|
|
);
|
|
|
|
enum
|
|
{
|
|
SIGNAL_0,
|
|
LAST_SIGNAL,
|
|
};
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_SENDER,
|
|
PROP_RECEIVER,
|
|
PROP_DIRECTION,
|
|
PROP_MLINE,
|
|
PROP_MID,
|
|
PROP_CURRENT_DIRECTION,
|
|
PROP_KIND,
|
|
PROP_CODEC_PREFERENCES,
|
|
PROP_STOPPED, // FIXME
|
|
};
|
|
|
|
//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;
|
|
case PROP_DIRECTION:
|
|
GST_OBJECT_LOCK (webrtc);
|
|
webrtc->direction = g_value_get_enum (value);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
break;
|
|
case PROP_CODEC_PREFERENCES:
|
|
GST_OBJECT_LOCK (webrtc);
|
|
gst_caps_replace (&webrtc->codec_preferences, g_value_get_boxed (value));
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
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_MID:
|
|
g_value_set_string (value, webrtc->mid);
|
|
break;
|
|
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;
|
|
case PROP_DIRECTION:
|
|
GST_OBJECT_LOCK (webrtc);
|
|
g_value_set_enum (value, webrtc->direction);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
break;
|
|
case PROP_CURRENT_DIRECTION:
|
|
g_value_set_enum (value, webrtc->current_direction);
|
|
break;
|
|
case PROP_KIND:
|
|
g_value_set_enum (value, webrtc->kind);
|
|
break;
|
|
case PROP_CODEC_PREFERENCES:
|
|
GST_OBJECT_LOCK (webrtc);
|
|
gst_value_set_caps (value, webrtc->codec_preferences);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
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));
|
|
|
|
/**
|
|
* GstWebRTCRTPTransceiver:direction:
|
|
*
|
|
* Direction of the transceiver.
|
|
*
|
|
* Since: 1.18
|
|
**/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_DIRECTION,
|
|
g_param_spec_enum ("direction", "Direction",
|
|
"Transceiver direction",
|
|
GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstWebRTCRTPTransceiver:mid:
|
|
*
|
|
* The media ID of the m-line associated with this transceiver. This
|
|
* association is established, when possible, whenever either a
|
|
* local or remote description is applied. This field is null if
|
|
* neither a local or remote description has been applied, or if its
|
|
* associated m-line is rejected by either a remote offer or any
|
|
* answer.
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_MID,
|
|
g_param_spec_string ("mid", "Media ID",
|
|
"The media ID of the m-line associated with this transceiver. This "
|
|
" association is established, when possible, whenever either a local"
|
|
" or remote description is applied. This field is null if neither a"
|
|
" local or remote description has been applied, or if its associated"
|
|
" m-line is rejected by either a remote offer or any answer.",
|
|
NULL, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstWebRTCRTPTransceiver:current-direction:
|
|
*
|
|
* The transceiver's current directionality, or none if the
|
|
* transceiver is stopped or has never participated in an exchange
|
|
* of offers and answers. To change the transceiver's
|
|
* directionality, set the value of the direction property.
|
|
*
|
|
* Since: 1.20
|
|
**/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_DIRECTION,
|
|
g_param_spec_enum ("current-direction", "Current Direction",
|
|
"Transceiver current direction",
|
|
GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstWebRTCRTPTransceiver:kind:
|
|
*
|
|
* The kind of media this transceiver transports
|
|
*
|
|
* Since: 1.20
|
|
**/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_KIND,
|
|
g_param_spec_enum ("kind", "Media Kind",
|
|
"Kind of media this transceiver transports",
|
|
GST_TYPE_WEBRTC_KIND, GST_WEBRTC_KIND_UNKNOWN,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstWebRTCRTPTransceiver:codec-preferences:
|
|
*
|
|
* Caps representing the codec preferences.
|
|
*
|
|
* Since: 1.20
|
|
**/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_CODEC_PREFERENCES,
|
|
g_param_spec_boxed ("codec-preferences", "Codec Preferences",
|
|
"Caps representing the codec preferences.",
|
|
GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_rtp_transceiver_init (GstWebRTCRTPTransceiver * webrtc)
|
|
{
|
|
webrtc->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE;
|
|
}
|