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-rw-r--r--sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs46
-rw-r--r--sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs31
-rw-r--r--sources/generated/Gst.WebRTC/WebRTCRTPSender.cs31
3 files changed, 7 insertions, 101 deletions
diff --git a/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs b/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs
index e8fdfd2..300296c 100644
--- a/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs
+++ b/sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs
@@ -14,21 +14,19 @@ namespace Gst.WebRTC {
public WebRTCDTLSTransport (IntPtr raw) : base(raw) {}
[DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
- static extern IntPtr gst_webrtc_dtls_transport_new(uint session_id, bool rtcp);
+ static extern IntPtr gst_webrtc_dtls_transport_new(uint session_id);
- public WebRTCDTLSTransport (uint session_id, bool rtcp) : base (IntPtr.Zero)
+ public WebRTCDTLSTransport (uint session_id) : base (IntPtr.Zero)
{
if (GetType () != typeof (WebRTCDTLSTransport)) {
var vals = new List<GLib.Value> ();
var names = new List<string> ();
names.Add ("session_id");
vals.Add (new GLib.Value (session_id));
- names.Add ("rtcp");
- vals.Add (new GLib.Value (rtcp));
CreateNativeObject (names.ToArray (), vals.ToArray ());
return;
}
- Raw = gst_webrtc_dtls_transport_new(session_id, rtcp);
+ Raw = gst_webrtc_dtls_transport_new(session_id);
}
[GLib.Property ("certificate")]
@@ -71,16 +69,6 @@ namespace Gst.WebRTC {
}
}
- [GLib.Property ("rtcp")]
- public bool Rtcp {
- get {
- GLib.Value val = GetProperty ("rtcp");
- bool ret = (bool) val;
- val.Dispose ();
- return ret;
- }
- }
-
[GLib.Property ("session-id")]
public uint SessionId {
get {
@@ -135,15 +123,6 @@ namespace Gst.WebRTC {
}
}
- public bool IsRtcp {
- get {
- unsafe {
- bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("is_rtcp"));
- return (*raw_ptr);
- }
- }
- }
-
public bool ClientField {
get {
unsafe {
@@ -239,22 +218,14 @@ namespace Gst.WebRTC {
, -1
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCDTLSTransportState))) // state
, "transport"
- , "is_rtcp"
- , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_stateAlign), "state")
- , 0
- ),
- new GLib.AbiField("is_rtcp"
- , -1
- , (uint) Marshal.SizeOf(typeof(bool)) // is_rtcp
- , "state"
, "client"
- , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_is_rtcpAlign), "is_rtcp")
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_stateAlign), "state")
, 0
),
new GLib.AbiField("client"
, -1
, (uint) Marshal.SizeOf(typeof(bool)) // client
- , "is_rtcp"
+ , "state"
, "session_id"
, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_clientAlign), "client")
, 0
@@ -305,13 +276,6 @@ namespace Gst.WebRTC {
}
[StructLayout(LayoutKind.Sequential)]
- public struct GstWebRTCDTLSTransport_is_rtcpAlign
- {
- sbyte f1;
- private bool is_rtcp;
- }
-
- [StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCDTLSTransport_clientAlign
{
sbyte f1;
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs b/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs
index f5c4ca8..0417077 100644
--- a/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs
@@ -25,9 +25,6 @@ namespace Gst.WebRTC {
Raw = gst_webrtc_rtp_receiver_new();
}
- [DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
- static extern void gst_webrtc_rtp_receiver_set_transport(IntPtr raw, IntPtr transport);
-
public Gst.WebRTC.WebRTCDTLSTransport Transport {
get {
unsafe {
@@ -35,24 +32,6 @@ namespace Gst.WebRTC {
return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
}
}
- set {
- gst_webrtc_rtp_receiver_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
- }
- }
-
- [DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
- static extern void gst_webrtc_rtp_receiver_set_rtcp_transport(IntPtr raw, IntPtr transport);
-
- public Gst.WebRTC.WebRTCDTLSTransport RtcpTransport {
- get {
- unsafe {
- IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("rtcp_transport"));
- return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
- }
- }
- set {
- gst_webrtc_rtp_receiver_set_rtcp_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
- }
}
@@ -106,14 +85,6 @@ namespace Gst.WebRTC {
, Gst.Object.abi_info.Fields
, (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
, null
- , "rtcp_transport"
- , (uint) Marshal.SizeOf(typeof(IntPtr))
- , 0
- ),
- new GLib.AbiField("rtcp_transport"
- , -1
- , (uint) Marshal.SizeOf(typeof(IntPtr)) // rtcp_transport
- , "transport"
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
@@ -121,7 +92,7 @@ namespace Gst.WebRTC {
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
- , "rtcp_transport"
+ , "transport"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs b/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
index 3a89c19..5669142 100644
--- a/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
@@ -41,9 +41,6 @@ namespace Gst.WebRTC {
}
}
- [DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
- static extern void gst_webrtc_rtp_sender_set_transport(IntPtr raw, IntPtr transport);
-
public Gst.WebRTC.WebRTCDTLSTransport Transport {
get {
unsafe {
@@ -51,24 +48,6 @@ namespace Gst.WebRTC {
return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
}
}
- set {
- gst_webrtc_rtp_sender_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
- }
- }
-
- [DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
- static extern void gst_webrtc_rtp_sender_set_rtcp_transport(IntPtr raw, IntPtr transport);
-
- public Gst.WebRTC.WebRTCDTLSTransport RtcpTransport {
- get {
- unsafe {
- IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("rtcp_transport"));
- return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCDTLSTransport;
- }
- }
- set {
- gst_webrtc_rtp_sender_set_rtcp_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
- }
}
public Gst.WebRTC.WebRTCPriorityType PriorityField {
@@ -131,14 +110,6 @@ namespace Gst.WebRTC {
, Gst.Object.abi_info.Fields
, (uint) Marshal.SizeOf(typeof(IntPtr)) // transport
, null
- , "rtcp_transport"
- , (uint) Marshal.SizeOf(typeof(IntPtr))
- , 0
- ),
- new GLib.AbiField("rtcp_transport"
- , -1
- , (uint) Marshal.SizeOf(typeof(IntPtr)) // rtcp_transport
- , "transport"
, "send_encodings"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
@@ -146,7 +117,7 @@ namespace Gst.WebRTC {
new GLib.AbiField("send_encodings"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // send_encodings
- , "rtcp_transport"
+ , "transport"
, "priority"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
generated by cgit v1.2.3 (git 2.39.1) at 2024-10-16 13:28:22 +0000