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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-24 01:00:37 +00:00
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>
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parent
3c3e89304e
commit
df8d29e9c3
7 changed files with 2 additions and 73 deletions
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@ -949,7 +949,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
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GstWebRTCDTLSTransport *dtls_transport;
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GstWebRTCICETransport *transport;
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GstWebRTCICEGatheringState ice_state;
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gboolean rtcp_mux = FALSE;
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if (rtp_trans->stopped || stream == NULL) {
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GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated",
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@ -963,8 +962,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
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GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
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}
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g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
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dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
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if (dtls_transport == NULL) {
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GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans);
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@ -1023,12 +1020,9 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
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for (i = 0; i < webrtc->priv->transceivers->len; i++) {
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GstWebRTCRTPTransceiver *rtp_trans =
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g_ptr_array_index (webrtc->priv->transceivers, i);
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WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
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TransportStream *stream = trans->stream;
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GstWebRTCDTLSTransport *transport;
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GstWebRTCICEConnectionState ice_state;
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GstWebRTCDTLSTransportState dtls_state;
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gboolean rtcp_mux = FALSE;
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if (rtp_trans->stopped) {
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GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
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@ -1039,7 +1033,6 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
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continue;
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}
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g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
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transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
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/* get transport state */
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@ -2137,8 +2130,6 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
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webrtc->priv->data_channel_transport = stream;
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g_object_set (stream, "rtcp-mux", TRUE, NULL);
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if (!(sctp_transport = webrtc->priv->sctp_transport)) {
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sctp_transport = gst_webrtc_sctp_transport_new ();
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sctp_transport->transport =
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@ -4028,7 +4019,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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GstWebRTCRTPTransceiverDirection new_dir;
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const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
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GstWebRTCDTLSSetup new_setup;
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gboolean new_rtcp_mux, new_rtcp_rsize;
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gboolean new_rtcp_rsize;
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ReceiveState receive_state = RECEIVE_STATE_UNSET;
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int i;
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@ -4098,8 +4089,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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}
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if (!bundled || bundle_idx == media_idx) {
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new_rtcp_mux = _media_has_attribute_key (local_media, "rtcp-mux")
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&& _media_has_attribute_key (remote_media, "rtcp-mux");
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new_rtcp_rsize = _media_has_attribute_key (local_media, "rtcp-rsize")
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&& _media_has_attribute_key (remote_media, "rtcp-rsize");
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@ -4112,8 +4101,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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g_object_unref (session);
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}
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}
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g_object_set (stream, "rtcp-mux", new_rtcp_mux, NULL);
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}
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}
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@ -4319,7 +4306,7 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
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/* data channel is always rtcp-muxed to avoid generating ICE candidates
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* for RTCP */
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g_object_set (stream, "rtcp-mux", TRUE, "dtls-client",
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g_object_set (stream, "dtls-client",
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new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL);
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local_port = _get_sctp_port_from_media (local_media);
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@ -138,35 +138,6 @@ _stop_thread (GstWebRTCICE * ice)
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g_thread_unref (ice->priv->thread);
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}
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#if 0
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static NiceComponentType
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_webrtc_component_to_nice (GstWebRTCICEComponent comp)
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{
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switch (comp) {
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case GST_WEBRTC_ICE_COMPONENT_RTP:
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return NICE_COMPONENT_TYPE_RTP;
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case GST_WEBRTC_ICE_COMPONENT_RTCP:
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return NICE_COMPONENT_TYPE_RTCP;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static GstWebRTCICEComponent
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_nice_component_to_webrtc (NiceComponentType comp)
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{
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switch (comp) {
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case NICE_COMPONENT_TYPE_RTP:
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return GST_WEBRTC_ICE_COMPONENT_RTP;
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case NICE_COMPONENT_TYPE_RTCP:
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return GST_WEBRTC_ICE_COMPONENT_RTCP;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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#endif
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struct NiceStreamItem
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{
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guint session_id;
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@ -380,16 +351,6 @@ _add_turn_server (GstWebRTCICE * ice, struct NiceStreamItem *item,
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g_free (uri);
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break;
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}
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ret = nice_agent_set_relay_info (ice->priv->nice_agent,
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item->nice_stream_id, NICE_COMPONENT_TYPE_RTCP,
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gst_uri_get_host (turn_server), gst_uri_get_port (turn_server),
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user, pass, relays[i]);
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if (!ret) {
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gchar *uri = gst_uri_to_string (turn_server);
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GST_ERROR_OBJECT (ice, "Failed to set TURN server '%s'", uri);
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g_free (uri);
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break;
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}
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}
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g_free (user);
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g_free (pass);
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@ -69,7 +69,6 @@ enum
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{
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PROP_0,
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PROP_STREAM,
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PROP_RTCP_MUX,
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};
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#define TSB_GET_LOCK(tsb) (&tsb->lock)
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@ -59,7 +59,6 @@ struct _TransportSendBin
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/*
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struct pad_block *rtp_block;
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struct pad_block *rtcp_mux_block;
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struct pad_block *rtp_nice_block;
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struct pad_block *rtcp_block;
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@ -36,7 +36,6 @@ enum
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PROP_0,
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PROP_WEBRTC,
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PROP_SESSION_ID,
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PROP_RTCP_MUX,
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PROP_DTLS_CLIENT,
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};
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@ -124,9 +123,6 @@ transport_stream_set_property (GObject * object, guint prop_id,
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case PROP_SESSION_ID:
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stream->session_id = g_value_get_uint (value);
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break;
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case PROP_RTCP_MUX:
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stream->rtcp_mux = g_value_get_boolean (value);
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break;
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case PROP_DTLS_CLIENT:
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stream->dtls_client = g_value_get_boolean (value);
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break;
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@ -148,9 +144,6 @@ transport_stream_get_property (GObject * object, guint prop_id,
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case PROP_SESSION_ID:
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g_value_set_uint (value, stream->session_id);
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break;
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case PROP_RTCP_MUX:
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g_value_set_boolean (value, stream->rtcp_mux);
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break;
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case PROP_DTLS_CLIENT:
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g_value_set_boolean (value, stream->dtls_client);
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break;
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@ -270,12 +263,6 @@ transport_stream_class_init (TransportStreamClass * klass)
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0, G_MAXUINT, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_RTCP_MUX,
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g_param_spec_boolean ("rtcp-mux", "RTCP Mux",
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"Whether RTCP packets are muxed with RTP packets",
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FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_DTLS_CLIENT,
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g_param_spec_boolean ("dtls-client", "DTLS client",
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@ -52,9 +52,6 @@ struct _TransportStream
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GstObject parent;
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guint session_id; /* session_id */
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gboolean rtcp;
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gboolean rtcp_mux;
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gboolean rtcp_rsize;
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gboolean dtls_client;
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gboolean active; /* TRUE if any mline in the bundle/transport is active */
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TransportSendBin *send_bin; /* bin containing all the sending transport elements */
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@ -63,7 +63,6 @@ void webrtc_transceiver_set_transport (WebRTCTransceiver *
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TransportStream * stream);
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GstWebRTCDTLSTransport * webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans);
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GstWebRTCDTLSTransport * webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans);
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G_END_DECLS
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