| /* 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_retrieved_caps, NULL); | |
| gst_caps_replace (&trans->last_send_configured_caps, NULL); | |
| g_free (trans->pending_mid); | |
| gst_event_replace (&trans->tos_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; | |
| } |