[vala/0.42] gstreamer: Update from 1.15+ git master

[Rico Tzschichholz <ricotz src gnome org>]

commit 2b656a8f1b06457c5f2227f65ae3ae254b1903b1
Author: Rico Tzschichholz <ricotz ubuntu com>
Date:   Sun Sep 23 12:30:17 2018 +0200

    gstreamer: Update from 1.15+ git master

 vapi/gstreamer-webrtc-1.0.vapi | 22 ++++++++++++++++++++++
 1 file changed, 22 insertions(+)
---
diff --git a/vapi/gstreamer-webrtc-1.0.vapi b/vapi/gstreamer-webrtc-1.0.vapi
index 511381e8c..bda7c17de 100644
--- a/vapi/gstreamer-webrtc-1.0.vapi
+++ b/vapi/gstreamer-webrtc-1.0.vapi
@@ -118,6 +118,14 @@ namespace Gst {
                CONNECTING,
                CONNECTED
        }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DATA_CHANNEL_STATE_", type_id 
= "gst_webrtc_data_channel_state_get_type ()")]
+       public enum WebRTCDataChannelState {
+               NEW,
+               CONNECTING,
+               OPEN,
+               CLOSING,
+               CLOSED
+       }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_FEC_TYPE_", type_id = 
"gst_webrtc_fec_type_get_type ()")]
        public enum WebRTCFECType {
                NONE,
@@ -158,6 +166,13 @@ namespace Gst {
                FAILED,
                CLOSED
        }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PRIORITY_TYPE_", type_id = 
"gst_webrtc_priority_type_get_type ()")]
+       public enum WebRTCPriorityType {
+               VERY_LOW,
+               LOW,
+               MEDIUM,
+               HIGH
+       }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_", 
type_id = "gst_webrtc_rtp_transceiver_direction_get_type ()")]
        public enum WebRTCRTPTransceiverDirection {
                NONE,
@@ -166,6 +181,13 @@ namespace Gst {
                RECVONLY,
                SENDRECV
        }
+       [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SCTP_TRANSPORT_STATE_", 
type_id = "gst_webrtc_sctp_transport_state_get_type ()")]
+       public enum WebRTCSCTPTransportState {
+               NEW,
+               CONNECTING,
+               CONNECTED,
+               CLOSED
+       }
        [CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SDP_TYPE_", type_id = 
"gst_webrtc_sdp_type_get_type ()")]
        public enum WebRTCSDPType {
                OFFER,