diff options
Diffstat (limited to 'gst-libs/gst/webrtc/rtptransceiver.c')
| -rw-r--r-- | gst-libs/gst/webrtc/rtptransceiver.c | 186 |
1 files changed, 186 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/rtptransceiver.c b/gst-libs/gst/webrtc/rtptransceiver.c new file mode 100644 index 000000000..d0d9628d0 --- /dev/null +++ b/gst-libs/gst/webrtc/rtptransceiver.c @@ -0,0 +1,186 @@ +/* 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) +{ +} |