subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.c

   1 /* GStreamer
   2  * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
   3  *
   4  * This library is free software; you can redistribute it and/or
   5  * modify it under the terms of the GNU Library General Public
   6  * License as published by the Free Software Foundation; either
   7  * version 2 of the License, or (at your option) any later version.
   8  *
   9  * This library is distributed in the hope that it will be useful,
  10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  12  * Library General Public License for more details.
  13  *
  14  * You should have received a copy of the GNU Library General Public
  15  * License along with this library; if not, write to the
  16  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
  17  * Boston, MA 02110-1301, USA.
  18  */
  19
  20 #ifdef HAVE_CONFIG_H
  21 # include "config.h"
  22 #endif
  23
  24 #include "gstwebrtcbin.h"
  25 #include "utils.h"
  26 #include "webrtctransceiver.h"
  27
  28 #define GST_CAT_DEFAULT webrtc_transceiver_debug
  29 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
  30
  31 #define webrtc_transceiver_parent_class parent_class
  32 G_DEFINE_TYPE_WITH_CODE (WebRTCTransceiver, webrtc_transceiver,
  33     GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
  34     GST_DEBUG_CATEGORY_INIT (webrtc_transceiver_debug,
  35         "webrtctransceiver", 0, "webrtctransceiver"););
  36
  37 #define DEFAULT_FEC_TYPE GST_WEBRTC_FEC_TYPE_NONE
  38 #define DEFAULT_DO_NACK FALSE
  39 #define DEFAULT_FEC_PERCENTAGE 100
  40
  41 enum
  42 {
  43   PROP_0,
  44   PROP_WEBRTC,
  45   PROP_FEC_TYPE,
  46   PROP_FEC_PERCENTAGE,
  47   PROP_DO_NACK,
  48 };
  49
  50 void
  51 webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
  52     TransportStream * stream)
  53 {
  54   GstWebRTCRTPTransceiver *rtp_trans;
  55
  56   g_return_if_fail (WEBRTC_IS_TRANSCEIVER (trans));
  57
  58   rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
  59
  60   gst_object_replace ((GstObject **) & trans->stream, (GstObject *) stream);
  61
  62   if (rtp_trans->sender) {
  63     gst_object_replace ((GstObject **) & rtp_trans->sender->transport,
  64         (GstObject *) stream->transport);
  65     g_object_notify (G_OBJECT (rtp_trans->sender), "transport");
  66   }
  67
  68   if (rtp_trans->receiver) {
  69     gst_object_replace ((GstObject **) & rtp_trans->receiver->transport,
  70         (GstObject *) stream->transport);
  71     g_object_notify (G_OBJECT (rtp_trans->receiver), "transport");
  72   }
  73 }
  74
  75 GstWebRTCDTLSTransport *
  76 webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans)
  77 {
  78   g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
  79
  80   if (trans->sender) {
  81     return trans->sender->transport;
  82   } else if (trans->receiver) {
  83     return trans->receiver->transport;
  84   }
  85
  86   return NULL;
  87 }
  88
  89 static void
  90 webrtc_transceiver_set_property (GObject * object, guint prop_id,
  91     const GValue * value, GParamSpec * pspec)
  92 {
  93   WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
  94
  95   switch (prop_id) {
  96     case PROP_WEBRTC:
  97       gst_object_set_parent (GST_OBJECT (trans), g_value_get_object (value));
  98       break;
  99   }
 100
 101   GST_OBJECT_LOCK (trans);
 102   switch (prop_id) {
 103     case PROP_WEBRTC:
 104       break;
 105     case PROP_FEC_TYPE:
 106       trans->fec_type = g_value_get_enum (value);
 107       break;
 108     case PROP_DO_NACK:
 109       trans->do_nack = g_value_get_boolean (value);
 110       break;
 111     case PROP_FEC_PERCENTAGE:
 112       trans->fec_percentage = g_value_get_uint (value);
 113       break;
 114     default:
 115       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
 116       break;
 117   }
 118   GST_OBJECT_UNLOCK (trans);
 119 }
 120
 121 static void
 122 webrtc_transceiver_get_property (GObject * object, guint prop_id,
 123     GValue * value, GParamSpec * pspec)
 124 {
 125   WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
 126
 127   GST_OBJECT_LOCK (trans);
 128   switch (prop_id) {
 129     case PROP_FEC_TYPE:
 130       g_value_set_enum (value, trans->fec_type);
 131       break;
 132     case PROP_DO_NACK:
 133       g_value_set_boolean (value, trans->do_nack);
 134       break;
 135     case PROP_FEC_PERCENTAGE:
 136       g_value_set_uint (value, trans->fec_percentage);
 137       break;
 138     default:
 139       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
 140       break;
 141   }
 142   GST_OBJECT_UNLOCK (trans);
 143 }
 144
 145 static void
 146 webrtc_transceiver_finalize (GObject * object)
 147 {
 148   WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (object);
 149
 150   if (trans->stream)
 151     gst_object_unref (trans->stream);
 152   trans->stream = NULL;
 153
 154   if (trans->local_rtx_ssrc_map)
 155     gst_structure_free (trans->local_rtx_ssrc_map);
 156   trans->local_rtx_ssrc_map = NULL;
 157
 158   gst_caps_replace (&trans->last_configured_caps, NULL);
 159
 160   if (trans->ssrc_event)
 161     gst_event_unref (trans->ssrc_event);
 162   trans->ssrc_event = NULL;
 163
 164   G_OBJECT_CLASS (parent_class)->finalize (object);
 165 }
 166
 167 static void
 168 webrtc_transceiver_class_init (WebRTCTransceiverClass * klass)
 169 {
 170   GObjectClass *gobject_class = (GObjectClass *) klass;
 171
 172   gobject_class->get_property = webrtc_transceiver_get_property;
 173   gobject_class->set_property = webrtc_transceiver_set_property;
 174   gobject_class->finalize = webrtc_transceiver_finalize;
 175
 176   /* some acrobatics are required to set the parent before _constructed()
 177    * has been called */
 178   g_object_class_install_property (gobject_class,
 179       PROP_WEBRTC,
 180       g_param_spec_object ("webrtc", "Parent webrtcbin",
 181           "Parent webrtcbin",
 182           GST_TYPE_WEBRTC_BIN,
 183           G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
 184
 185   g_object_class_install_property (gobject_class,
 186       PROP_FEC_TYPE,
 187       g_param_spec_enum ("fec-type", "FEC type",
 188           "The type of Forward Error Correction to use",
 189           GST_TYPE_WEBRTC_FEC_TYPE,
 190           DEFAULT_FEC_TYPE,
 191           G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 192
 193   g_object_class_install_property (gobject_class,
 194       PROP_DO_NACK,
 195       g_param_spec_boolean ("do-nack", "Do nack",
 196           "Whether to send negative acknowledgements for feedback",
 197           DEFAULT_DO_NACK,
 198           G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 199
 200   g_object_class_install_property (gobject_class,
 201       PROP_FEC_PERCENTAGE,
 202       g_param_spec_uint ("fec-percentage", "FEC percentage",
 203           "The amount of Forward Error Correction to apply",
 204           0, 100, DEFAULT_FEC_PERCENTAGE,
 205           G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 206 }
 207
 208 static void
 209 webrtc_transceiver_init (WebRTCTransceiver * trans)
 210 {
 211 }
 212
 213 WebRTCTransceiver *
 214 webrtc_transceiver_new (GstWebRTCBin * webrtc, GstWebRTCRTPSender * sender,
 215     GstWebRTCRTPReceiver * receiver)
 216 {
 217   WebRTCTransceiver *trans;
 218
 219   trans = g_object_new (webrtc_transceiver_get_type (), "sender", sender,
 220       "receiver", receiver, "webrtc", webrtc, NULL);
 221
 222   return trans;
 223 }