gstreamer/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
Aaron Boxer 38a0731461 build: on Windows, use MSVC format for gst*, glib and gobject dlls
Generated files were generated using 'ninja -C build update-code'
except for libgstfft, which had to be updated manually
(see issue #25).

Note: with these changes, building on MS Windows will require
the msvc compiler - mingw will no longer work.
2019-11-18 14:19:39 -05:00

148 lines
4 KiB
C#

// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Collections;
using System.Collections.Generic;
using System.Runtime.InteropServices;
#region Autogenerated code
public partial class WebRTCRTPSender : Gst.Object {
public WebRTCRTPSender (IntPtr raw) : base(raw) {}
[DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_rtp_sender_new();
public WebRTCRTPSender () : base (IntPtr.Zero)
{
if (GetType () != typeof (WebRTCRTPSender)) {
CreateNativeObject (new string [0], new GLib.Value[0]);
return;
}
Raw = gst_webrtc_rtp_sender_new();
}
[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 {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
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);
}
}
// Internal representation of the wrapped structure ABI.
static GLib.AbiStruct _class_abi = null;
static public new GLib.AbiStruct class_abi {
get {
if (_class_abi == null)
_class_abi = new GLib.AbiStruct (new List<GLib.AbiField>{
new GLib.AbiField("_padding"
, Gst.Object.class_abi.Fields
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, null
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _class_abi;
}
}
// End of the ABI representation.
[DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_rtp_sender_get_type();
public static new GLib.GType GType {
get {
IntPtr raw_ret = gst_webrtc_rtp_sender_get_type();
GLib.GType ret = new GLib.GType(raw_ret);
return ret;
}
}
static WebRTCRTPSender ()
{
GtkSharp.GstreamerSharp.ObjectManager.Initialize ();
}
// Internal representation of the wrapped structure ABI.
static GLib.AbiStruct _abi_info = null;
static public new GLib.AbiStruct abi_info {
get {
if (_abi_info == null)
_abi_info = new GLib.AbiStruct (new List<GLib.AbiField>{
new GLib.AbiField("transport"
, 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
),
new GLib.AbiField("send_encodings"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // send_encodings
, "rtcp_transport"
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, "send_encodings"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _abi_info;
}
}
// End of the ABI representation.
#endregion
}
}