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235 lines
6.8 KiB
C
235 lines
6.8 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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "gstwebrtcbin.h"
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#include "utils.h"
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#include "webrtctransceiver.h"
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#define GST_CAT_DEFAULT webrtc_transceiver_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define webrtc_transceiver_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
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GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
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GST_DEBUG_CATEGORY_INIT (webrtc_transceiver_debug,
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"webrtctransceiver", 0, "webrtctransceiver"););
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#define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
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#define DEFAULT_DO_NACK FALSE
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#define DEFAULT_FEC_PERCENTAGE 100
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enum
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{
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PROP_0,
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PROP_WEBRTC,
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PROP_FEC_TYPE,
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PROP_FEC_PERCENTAGE,
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PROP_DO_NACK,
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};
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void
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webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
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TransportStream * stream)
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{
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GstWebRTCRTPTransceiver *rtp_trans;
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g_return_if_fail (WEBRTC_IS_TRANSCEIVER (trans));
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rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
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gst_object_replace ((GstObject **) & trans->stream, (GstObject *) stream);
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if (rtp_trans->sender)
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gst_object_replace ((GstObject **) & rtp_trans->sender->transport,
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(GstObject *) stream->transport);
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if (rtp_trans->receiver)
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gst_object_replace ((GstObject **) & rtp_trans->receiver->transport,
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(GstObject *) stream->transport);
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if (rtp_trans->sender)
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gst_object_replace ((GstObject **) & rtp_trans->sender->rtcp_transport,
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(GstObject *) stream->rtcp_transport);
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if (rtp_trans->receiver)
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gst_object_replace ((GstObject **) & rtp_trans->receiver->rtcp_transport,
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(GstObject *) stream->rtcp_transport);
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}
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GstWebRTCDTLSTransport *
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webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans)
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{
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g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
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if (trans->sender) {
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return trans->sender->transport;
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} else if (trans->receiver) {
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return trans->receiver->transport;
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}
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return NULL;
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}
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GstWebRTCDTLSTransport *
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webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans)
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{
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g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
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if (trans->sender) {
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return trans->sender->rtcp_transport;
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} else if (trans->receiver) {
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return trans->receiver->rtcp_transport;
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}
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return NULL;
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}
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static void
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webrtc_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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WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
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switch (prop_id) {
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case PROP_WEBRTC:
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gst_object_set_parent (GST_OBJECT (trans), g_value_get_object (value));
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break;
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}
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GST_OBJECT_LOCK (trans);
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switch (prop_id) {
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case PROP_WEBRTC:
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break;
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case PROP_FEC_TYPE:
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trans->fec_type = g_value_get_enum (value);
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break;
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case PROP_DO_NACK:
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trans->do_nack = g_value_get_boolean (value);
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break;
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case PROP_FEC_PERCENTAGE:
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trans->fec_percentage = g_value_get_uint (value);
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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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GST_OBJECT_UNLOCK (trans);
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}
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static void
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webrtc_transceiver_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
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GST_OBJECT_LOCK (trans);
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switch (prop_id) {
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case PROP_FEC_TYPE:
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g_value_set_enum (value, trans->fec_type);
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break;
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case PROP_DO_NACK:
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g_value_set_boolean (value, trans->do_nack);
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break;
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case PROP_FEC_PERCENTAGE:
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g_value_set_uint (value, trans->fec_percentage);
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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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GST_OBJECT_UNLOCK (trans);
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}
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static void
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webrtc_transceiver_finalize (GObject * object)
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{
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WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
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if (trans->stream)
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gst_object_unref (trans->stream);
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trans->stream = NULL;
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if (trans->local_rtx_ssrc_map)
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gst_structure_free (trans->local_rtx_ssrc_map);
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trans->local_rtx_ssrc_map = NULL;
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gst_caps_replace (&trans->last_configured_caps, NULL);
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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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webrtc_transceiver_class_init (WebRTCTransceiverClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->get_property = webrtc_transceiver_get_property;
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gobject_class->set_property = webrtc_transceiver_set_property;
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gobject_class->finalize = webrtc_transceiver_finalize;
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/* some acrobatics are required to set the parent before _constructed()
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* has been called */
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g_object_class_install_property (gobject_class,
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PROP_WEBRTC,
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g_param_spec_object ("webrtc", "Parent webrtcbin",
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"Parent webrtcbin",
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GST_TYPE_WEBRTC_BIN,
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G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_FEC_TYPE,
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g_param_spec_enum ("fec-type", "FEC type",
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"The type of Forward Error Correction to use",
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GST_TYPE_WEBRTC_FEC_TYPE,
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DEFAULT_FEC_TYPE,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_DO_NACK,
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g_param_spec_boolean ("do-nack", "Do nack",
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"Whether to send negative acknowledgements for feedback",
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DEFAULT_DO_NACK,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_FEC_PERCENTAGE,
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g_param_spec_uint ("fec-percentage", "FEC percentage",
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"The amount of Forward Error Correction to apply",
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0, 100, DEFAULT_FEC_PERCENTAGE,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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webrtc_transceiver_init (WebRTCTransceiver * trans)
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{
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}
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WebRTCTransceiver *
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webrtc_transceiver_new (GstWebRTCBin * webrtc, GstWebRTCRTPSender * sender,
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GstWebRTCRTPReceiver * receiver)
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{
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WebRTCTransceiver *trans;
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trans = g_object_new (webrtc_transceiver_get_type (), "sender", sender,
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"receiver", receiver, "webrtc", webrtc, NULL);
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return trans;
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}
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