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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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As those symbols are documented in a 'fwd' header smart indexing in hotdoc wasn't working. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2891>
305 lines
9.4 KiB
C
305 lines
9.4 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstwebrtc-transceiver
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* @short_description: RTCRtpTransceiver object
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* @title: GstWebRTCRTPTransceiver
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver
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* @symbols:
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* - GstWebRTCRTPTransceiver
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*
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* <https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface>
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "rtptransceiver.h"
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#include "webrtc-priv.h"
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#define GST_CAT_DEFAULT gst_webrtc_rtp_transceiver_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define gst_webrtc_rtp_transceiver_parent_class parent_class
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G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCRTPTransceiver,
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gst_webrtc_rtp_transceiver, GST_TYPE_OBJECT,
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_rtp_transceiver_debug,
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"webrtcrtptransceiver", 0, "webrtcrtptransceiver");
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);
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enum
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{
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SIGNAL_0,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_SENDER,
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PROP_RECEIVER,
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PROP_DIRECTION,
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PROP_MLINE,
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PROP_MID,
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PROP_CURRENT_DIRECTION,
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PROP_KIND,
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PROP_CODEC_PREFERENCES,
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PROP_STOPPED, // FIXME
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};
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//static guint gst_webrtc_rtp_transceiver_signals[LAST_SIGNAL] = { 0 };
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static void
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gst_webrtc_rtp_transceiver_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
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switch (prop_id) {
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case PROP_SENDER:
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webrtc->sender = g_value_dup_object (value);
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break;
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case PROP_RECEIVER:
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webrtc->receiver = g_value_dup_object (value);
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break;
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case PROP_MLINE:
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webrtc->mline = g_value_get_uint (value);
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break;
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case PROP_DIRECTION:
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GST_OBJECT_LOCK (webrtc);
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webrtc->direction = g_value_get_enum (value);
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GST_OBJECT_UNLOCK (webrtc);
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break;
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case PROP_CODEC_PREFERENCES:
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GST_OBJECT_LOCK (webrtc);
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gst_caps_replace (&webrtc->codec_preferences, g_value_get_boxed (value));
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GST_OBJECT_UNLOCK (webrtc);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_rtp_transceiver_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
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switch (prop_id) {
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case PROP_MID:
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g_value_set_string (value, webrtc->mid);
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break;
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case PROP_SENDER:
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g_value_set_object (value, webrtc->sender);
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break;
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case PROP_RECEIVER:
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g_value_set_object (value, webrtc->receiver);
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break;
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case PROP_MLINE:
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g_value_set_uint (value, webrtc->mline);
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break;
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case PROP_DIRECTION:
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GST_OBJECT_LOCK (webrtc);
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g_value_set_enum (value, webrtc->direction);
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GST_OBJECT_UNLOCK (webrtc);
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break;
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case PROP_CURRENT_DIRECTION:
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g_value_set_enum (value, webrtc->current_direction);
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break;
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case PROP_KIND:
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g_value_set_enum (value, webrtc->kind);
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break;
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case PROP_CODEC_PREFERENCES:
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GST_OBJECT_LOCK (webrtc);
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gst_value_set_caps (value, webrtc->codec_preferences);
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GST_OBJECT_UNLOCK (webrtc);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_rtp_transceiver_constructed (GObject * object)
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{
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GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
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gst_object_set_parent (GST_OBJECT (webrtc->sender), GST_OBJECT (webrtc));
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gst_object_set_parent (GST_OBJECT (webrtc->receiver), GST_OBJECT (webrtc));
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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gst_webrtc_rtp_transceiver_dispose (GObject * object)
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{
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GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
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if (webrtc->sender) {
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GST_OBJECT_PARENT (webrtc->sender) = NULL;
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gst_object_unref (webrtc->sender);
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}
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webrtc->sender = NULL;
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if (webrtc->receiver) {
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GST_OBJECT_PARENT (webrtc->receiver) = NULL;
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gst_object_unref (webrtc->receiver);
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}
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webrtc->receiver = NULL;
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_webrtc_rtp_transceiver_finalize (GObject * object)
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{
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GstWebRTCRTPTransceiver *webrtc = GST_WEBRTC_RTP_TRANSCEIVER (object);
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g_free (webrtc->mid);
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if (webrtc->codec_preferences)
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gst_caps_unref (webrtc->codec_preferences);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_webrtc_rtp_transceiver_class_init (GstWebRTCRTPTransceiverClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->get_property = gst_webrtc_rtp_transceiver_get_property;
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gobject_class->set_property = gst_webrtc_rtp_transceiver_set_property;
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gobject_class->constructed = gst_webrtc_rtp_transceiver_constructed;
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gobject_class->dispose = gst_webrtc_rtp_transceiver_dispose;
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gobject_class->finalize = gst_webrtc_rtp_transceiver_finalize;
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g_object_class_install_property (gobject_class,
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PROP_SENDER,
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g_param_spec_object ("sender", "Sender",
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"The RTP sender for this transceiver",
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GST_TYPE_WEBRTC_RTP_SENDER,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_RECEIVER,
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g_param_spec_object ("receiver", "Receiver",
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"The RTP receiver for this transceiver",
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GST_TYPE_WEBRTC_RTP_RECEIVER,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_MLINE,
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g_param_spec_uint ("mlineindex", "Media Line Index",
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"Index in the SDP of the Media",
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0, G_MAXUINT, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCRTPTransceiver:direction:
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*
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* Direction of the transceiver.
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*
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* Since: 1.18
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**/
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g_object_class_install_property (gobject_class,
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PROP_DIRECTION,
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g_param_spec_enum ("direction", "Direction",
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"Transceiver direction",
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GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCRTPTransceiver:mid:
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*
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* The media ID of the m-line associated with this transceiver. This
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* association is established, when possible, whenever either a
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* local or remote description is applied. This field is null if
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* neither a local or remote description has been applied, or if its
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* associated m-line is rejected by either a remote offer or any
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* answer.
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class,
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PROP_MID,
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g_param_spec_string ("mid", "Media ID",
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"The media ID of the m-line associated with this transceiver. This "
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" association is established, when possible, whenever either a local"
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" or remote description is applied. This field is null if neither a"
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" local or remote description has been applied, or if its associated"
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" m-line is rejected by either a remote offer or any answer.",
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NULL, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCRTPTransceiver:current-direction:
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*
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* The transceiver's current directionality, or none if the
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* transceiver is stopped or has never participated in an exchange
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* of offers and answers. To change the transceiver's
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* directionality, set the value of the direction property.
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*
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* Since: 1.20
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**/
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g_object_class_install_property (gobject_class,
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PROP_DIRECTION,
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g_param_spec_enum ("current-direction", "Current Direction",
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"Transceiver current direction",
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GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCRTPTransceiver:kind:
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*
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* The kind of media this transceiver transports
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*
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* Since: 1.20
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**/
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g_object_class_install_property (gobject_class,
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PROP_KIND,
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g_param_spec_enum ("kind", "Media Kind",
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"Kind of media this transceiver transports",
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GST_TYPE_WEBRTC_KIND, GST_WEBRTC_KIND_UNKNOWN,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCRTPTransceiver:codec-preferences:
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*
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* Caps representing the codec preferences.
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*
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* Since: 1.20
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**/
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g_object_class_install_property (gobject_class,
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PROP_CODEC_PREFERENCES,
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g_param_spec_boxed ("codec-preferences", "Codec Preferences",
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"Caps representing the codec preferences.",
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GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_rtp_transceiver_init (GstWebRTCRTPTransceiver * webrtc)
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{
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webrtc->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE;
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}
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