webrtcbin: Remove remnant of non-rtcp-mux mode

There was some code left that wasn't used anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1930>

ext/webrtc/gstwebrtcbin.c

@@ -949,7 +949,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
     GstWebRTCDTLSTransport *dtls_transport;
     GstWebRTCICETransport *transport;
     GstWebRTCICEGatheringState ice_state;
-    gboolean rtcp_mux = FALSE;
 
     if (rtp_trans->stopped || stream == NULL) {
       GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated",
@@ -963,8 +962,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
       GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
     }
 
-    g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
-
     dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
     if (dtls_transport == NULL) {
       GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans);
@@ -1023,12 +1020,9 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
   for (i = 0; i < webrtc->priv->transceivers->len; i++) {
     GstWebRTCRTPTransceiver *rtp_trans =
         g_ptr_array_index (webrtc->priv->transceivers, i);
-    WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
-    TransportStream *stream = trans->stream;
     GstWebRTCDTLSTransport *transport;
     GstWebRTCICEConnectionState ice_state;
     GstWebRTCDTLSTransportState dtls_state;
-    gboolean rtcp_mux = FALSE;
 
     if (rtp_trans->stopped) {
       GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
@@ -1039,7 +1033,6 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
       continue;
     }
 
-    g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
     transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
 
     /* get transport state */
@@ -2137,8 +2130,6 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
 
     webrtc->priv->data_channel_transport = stream;
 
-    g_object_set (stream, "rtcp-mux", TRUE, NULL);
-
     if (!(sctp_transport = webrtc->priv->sctp_transport)) {
       sctp_transport = gst_webrtc_sctp_transport_new ();
       sctp_transport->transport =
@@ -4028,7 +4019,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
   GstWebRTCRTPTransceiverDirection new_dir;
   const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
   GstWebRTCDTLSSetup new_setup;
-  gboolean new_rtcp_mux, new_rtcp_rsize;
+  gboolean new_rtcp_rsize;
   ReceiveState receive_state = RECEIVE_STATE_UNSET;
   int i;
 
@@ -4098,8 +4089,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
     }
 
     if (!bundled || bundle_idx == media_idx) {
-      new_rtcp_mux = _media_has_attribute_key (local_media, "rtcp-mux")
-          && _media_has_attribute_key (remote_media, "rtcp-mux");
       new_rtcp_rsize = _media_has_attribute_key (local_media, "rtcp-rsize")
           && _media_has_attribute_key (remote_media, "rtcp-rsize");
 
@@ -4112,8 +4101,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
           g_object_unref (session);
         }
       }
-
-      g_object_set (stream, "rtcp-mux", new_rtcp_mux, NULL);
     }
   }
 
@@ -4319,7 +4306,7 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
 
   /* data channel is always rtcp-muxed to avoid generating ICE candidates
    * for RTCP */
-  g_object_set (stream, "rtcp-mux", TRUE, "dtls-client",
+  g_object_set (stream, "dtls-client",
       new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL);
 
   local_port = _get_sctp_port_from_media (local_media);

ext/webrtc/gstwebrtcice.c

@@ -138,35 +138,6 @@ _stop_thread (GstWebRTCICE * ice)
   g_thread_unref (ice->priv->thread);
 }
 
-#if 0
-static NiceComponentType
-_webrtc_component_to_nice (GstWebRTCICEComponent comp)
-{
-  switch (comp) {
-    case GST_WEBRTC_ICE_COMPONENT_RTP:
-      return NICE_COMPONENT_TYPE_RTP;
-    case GST_WEBRTC_ICE_COMPONENT_RTCP:
-      return NICE_COMPONENT_TYPE_RTCP;
-    default:
-      g_assert_not_reached ();
-      return 0;
-  }
-}
-
-static GstWebRTCICEComponent
-_nice_component_to_webrtc (NiceComponentType comp)
-{
-  switch (comp) {
-    case NICE_COMPONENT_TYPE_RTP:
-      return GST_WEBRTC_ICE_COMPONENT_RTP;
-    case NICE_COMPONENT_TYPE_RTCP:
-      return GST_WEBRTC_ICE_COMPONENT_RTCP;
-    default:
-      g_assert_not_reached ();
-      return 0;
-  }
-}
-#endif
 struct NiceStreamItem
 {
   guint session_id;
@@ -380,16 +351,6 @@ _add_turn_server (GstWebRTCICE * ice, struct NiceStreamItem *item,
       g_free (uri);
       break;
     }
-    ret = nice_agent_set_relay_info (ice->priv->nice_agent,
-        item->nice_stream_id, NICE_COMPONENT_TYPE_RTCP,
-        gst_uri_get_host (turn_server), gst_uri_get_port (turn_server),
-        user, pass, relays[i]);
-    if (!ret) {
-      gchar *uri = gst_uri_to_string (turn_server);
-      GST_ERROR_OBJECT (ice, "Failed to set TURN server '%s'", uri);
-      g_free (uri);
-      break;
-    }
   }
   g_free (user);
   g_free (pass);

ext/webrtc/transportsendbin.c

@@ -69,7 +69,6 @@ enum
 {
   PROP_0,
   PROP_STREAM,
-  PROP_RTCP_MUX,
 };
 
 #define TSB_GET_LOCK(tsb) (&tsb->lock)

ext/webrtc/transportsendbin.h

@@ -59,7 +59,6 @@ struct _TransportSendBin
 
   /*
   struct pad_block          *rtp_block;
-  struct pad_block          *rtcp_mux_block;
   struct pad_block          *rtp_nice_block;
 
   struct pad_block          *rtcp_block;

ext/webrtc/transportstream.c

@@ -36,7 +36,6 @@ enum
   PROP_0,
   PROP_WEBRTC,
   PROP_SESSION_ID,
-  PROP_RTCP_MUX,
   PROP_DTLS_CLIENT,
 };
 
@@ -124,9 +123,6 @@ transport_stream_set_property (GObject * object, guint prop_id,
     case PROP_SESSION_ID:
       stream->session_id = g_value_get_uint (value);
       break;
-    case PROP_RTCP_MUX:
-      stream->rtcp_mux = g_value_get_boolean (value);
-      break;
     case PROP_DTLS_CLIENT:
       stream->dtls_client = g_value_get_boolean (value);
       break;
@@ -148,9 +144,6 @@ transport_stream_get_property (GObject * object, guint prop_id,
     case PROP_SESSION_ID:
       g_value_set_uint (value, stream->session_id);
       break;
-    case PROP_RTCP_MUX:
-      g_value_set_boolean (value, stream->rtcp_mux);
-      break;
     case PROP_DTLS_CLIENT:
       g_value_set_boolean (value, stream->dtls_client);
       break;
@@ -270,12 +263,6 @@ transport_stream_class_init (TransportStreamClass * klass)
           0, G_MAXUINT, 0,
           G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
 
-  g_object_class_install_property (gobject_class,
-      PROP_RTCP_MUX,
-      g_param_spec_boolean ("rtcp-mux", "RTCP Mux",
-          "Whether RTCP packets are muxed with RTP packets",
-          FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
   g_object_class_install_property (gobject_class,
       PROP_DTLS_CLIENT,
       g_param_spec_boolean ("dtls-client", "DTLS client",

ext/webrtc/transportstream.h

@@ -52,9 +52,6 @@ struct _TransportStream
   GstObject                 parent;
 
   guint                     session_id;             /* session_id */
-  gboolean                  rtcp;
-  gboolean                  rtcp_mux;
-  gboolean                  rtcp_rsize;
   gboolean                  dtls_client;
   gboolean                  active;                 /* TRUE if any mline in the bundle/transport is active */
   TransportSendBin         *send_bin;               /* bin containing all the sending transport elements */

ext/webrtc/webrtctransceiver.h

@@ -63,7 +63,6 @@ void                      webrtc_transceiver_set_transport  (WebRTCTransceiver *
                                                              TransportStream * stream);
 
 GstWebRTCDTLSTransport *  webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans);
-GstWebRTCDTLSTransport *  webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans);
 
 G_END_DECLS