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Generate bindings for the new GstWebRTC library

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This commit is contained in:
Thibault Saunier 2018-03-19 15:49:25 -03:00
parent 2a9149734f
commit 6bada6f67d
32 changed files with 3589 additions and 6 deletions
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1003
girs/GstWebRTC-1.0.gir Normal file
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File diff suppressed because it is too large Load diff

8
meson.build
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@ -79,7 +79,9 @@ gst_deps_defs = [
['gstreamer-rtsp', ['gst-plugins-base', 'rtsp_dep'], 'gst_rtsp'],
['gstreamer-sdp', ['gst-plugins-base', 'sdp_dep'], 'gstsdp'],
['gstreamer-tag', ['gst-plugins-base', 'tag_dep'], 'gsttag'],
['gstreamer-video', ['gst-plugins-base', 'video_dep'], 'gstvideo'],]
['gstreamer-video', ['gst-plugins-base', 'video_dep'], 'gstvideo'],
['gstreamer-webrtc', ['gst-plugins-bad', 'gstwebrtc_dep'], 'gstwebrtc'],
]
foreach dep: gst_deps_defs
gst_deps += [dependency(dep.get(0) + '-' + apiversion, version: gst_required_version,
@ -165,7 +167,7 @@ if bindinator.get_variable('found')
run_target('bindinate_gstreamer',
command: [bindinate,
'--name=gstreamer', '--regenerate=true',
'--merge-with=GstApp-1.0,GstAudio-1.0,GstBase-1.0,GstController-1.0,GstNet-1.0,GstPbutils-1.0,GstRtp-1.0,GstRtsp-1.0,GstSdp-1.0,GstTag-1.0,GstVideo-1.0',
'--merge-with=GstApp-1.0,GstAudio-1.0,GstBase-1.0,GstController-1.0,GstNet-1.0,GstPbutils-1.0,GstRtp-1.0,GstRtsp-1.0,GstSdp-1.0,GstTag-1.0,GstVideo-1.0,GstWebRTC-1.0',
'--gir=Gst-1.0',
'--copy-girs=@0@'.format(join_paths(meson.current_source_dir(), 'girs'))],
depends: []
@ -183,4 +185,4 @@ if bindinator.get_variable('found')
run_target('update-all', command: [find_program('update_sources.py'), 'bindinate'])
else
warning('Bindinator not usable as some required dependencies are not avalaible.')
endif
endif

3
sources/custom/Application.cs
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@ -32,7 +32,8 @@ namespace Gst {
GLib.GType.Register (FractionRange.GType, typeof(FractionRange));
GLib.GType.Register (DateTime.GType, typeof(DateTime));
GLib.GType.Register (Gst.Array.GType, typeof(Gst.Array));
GLib.GType.Register(Promise.GType, typeof(Promise));
GLib.GType.Register(Gst.WebRTC.WebRTCSessionDescription.GType, typeof(Gst.WebRTC.WebRTCSessionDescription));
}
[DllImport("libgstreamer-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]

16
sources/generated/Gst.WebRTC/Constants.cs Normal file
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@ -0,0 +1,16 @@
// 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 Constants {
#endregion
}
}

25
sources/generated/Gst.WebRTC/Global.cs Normal file
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@ -0,0 +1,25 @@
// 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 Global {
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_sdp_type_to_string(int type);
public static string WebrtcSdpTypeToString(Gst.WebRTC.WebRTCSDPType type) {
IntPtr raw_ret = gst_webrtc_sdp_type_to_string((int) type);
string ret = GLib.Marshaller.Utf8PtrToString (raw_ret);
return ret;
}
#endregion
}
}

18
sources/generated/Gst.WebRTC/OnNewCandidateHandler.cs Normal file
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@ -0,0 +1,18 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
public delegate void OnNewCandidateHandler(object o, OnNewCandidateArgs args);
public class OnNewCandidateArgs : GLib.SignalArgs {
public string Object{
get {
return (string) Args [0];
}
}
}
}

30
sources/generated/Gst.WebRTC/WebRTCDTLSSetup.cs Normal file
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@ -0,0 +1,30 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCDTLSSetupGType))]
public enum WebRTCDTLSSetup {
None = 0,
Actpass = 1,
Active = 2,
Passive = 3,
}
internal class WebRTCDTLSSetupGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_dtls_setup_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_dtls_setup_get_type ());
}
}
}
#endregion
}

333
sources/generated/Gst.WebRTC/WebRTCDTLSTransport.cs Normal file
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@ -0,0 +1,333 @@
// 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 WebRTCDTLSTransport : Gst.Object {
public WebRTCDTLSTransport (IntPtr raw) : base(raw) {}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_dtls_transport_new(uint session_id, bool rtcp);
public WebRTCDTLSTransport (uint session_id, bool rtcp) : 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);
}
[GLib.Property ("certificate")]
public string Certificate {
get {
GLib.Value val = GetProperty ("certificate");
string ret = (string) val;
val.Dispose ();
return ret;
}
set {
GLib.Value val = new GLib.Value(value);
SetProperty("certificate", val);
val.Dispose ();
}
}
[GLib.Property ("client")]
public bool Client {
get {
GLib.Value val = GetProperty ("client");
bool ret = (bool) val;
val.Dispose ();
return ret;
}
set {
GLib.Value val = new GLib.Value(value);
SetProperty("client", val);
val.Dispose ();
}
}
[GLib.Property ("remote-certificate")]
public string RemoteCertificate {
get {
GLib.Value val = GetProperty ("remote-certificate");
string ret = (string) val;
val.Dispose ();
return ret;
}
}
[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 {
GLib.Value val = GetProperty ("session-id");
uint ret = (uint) val;
val.Dispose ();
return ret;
}
}
[GLib.Property ("state")]
public Gst.WebRTC.WebRTCDTLSTransportState State {
get {
GLib.Value val = GetProperty ("state");
Gst.WebRTC.WebRTCDTLSTransportState ret = (Gst.WebRTC.WebRTCDTLSTransportState) (Enum) val;
val.Dispose ();
return ret;
}
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern void gst_webrtc_dtls_transport_set_transport(IntPtr raw, IntPtr ice);
[GLib.Property ("transport")]
public Gst.WebRTC.WebRTCICETransport Transport {
get {
GLib.Value val = GetProperty ("transport");
Gst.WebRTC.WebRTCICETransport ret = (Gst.WebRTC.WebRTCICETransport) val;
val.Dispose ();
return ret;
}
set {
gst_webrtc_dtls_transport_set_transport(Handle, value == null ? IntPtr.Zero : value.Handle);
}
}
public Gst.WebRTC.WebRTCICETransport TransportField {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCICETransport;
}
}
}
public Gst.WebRTC.WebRTCDTLSTransportState StateField {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("state"));
return (Gst.WebRTC.WebRTCDTLSTransportState) (*raw_ptr);
}
}
}
public bool IsRtcp {
get {
unsafe {
bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("is_rtcp"));
return (*raw_ptr);
}
}
}
public bool ClientField {
get {
unsafe {
bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("client"));
return (*raw_ptr);
}
}
}
public uint SessionIdField {
get {
unsafe {
uint* raw_ptr = (uint*)(((byte*)Handle) + abi_info.GetFieldOffset("session_id"));
return (*raw_ptr);
}
}
}
public Gst.Element Dtlssrtpenc {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("dtlssrtpenc"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
}
}
}
public Gst.Element Dtlssrtpdec {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("dtlssrtpdec"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
}
}
}
// 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("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_dtls_transport_get_type();
public static new GLib.GType GType {
get {
IntPtr raw_ret = gst_webrtc_dtls_transport_get_type();
GLib.GType ret = new GLib.GType(raw_ret);
return ret;
}
}
static WebRTCDTLSTransport ()
{
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
, "state"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("state"
, -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")
, 0
),
new GLib.AbiField("client"
, -1
, (uint) Marshal.SizeOf(typeof(bool)) // client
, "is_rtcp"
, "session_id"
, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_clientAlign), "client")
, 0
),
new GLib.AbiField("session_id"
, -1
, (uint) Marshal.SizeOf(typeof(uint)) // session_id
, "client"
, "dtlssrtpenc"
, (long) Marshal.OffsetOf(typeof(GstWebRTCDTLSTransport_session_idAlign), "session_id")
, 0
),
new GLib.AbiField("dtlssrtpenc"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // dtlssrtpenc
, "session_id"
, "dtlssrtpdec"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("dtlssrtpdec"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // dtlssrtpdec
, "dtlssrtpenc"
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, "dtlssrtpdec"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _abi_info;
}
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCDTLSTransport_stateAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCDTLSTransportState state;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCDTLSTransport_is_rtcpAlign
{
sbyte f1;
private bool is_rtcp;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCDTLSTransport_clientAlign
{
sbyte f1;
private bool client;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCDTLSTransport_session_idAlign
{
sbyte f1;
private uint session_id;
}
// End of the ABI representation.
#endregion
}
}

31
sources/generated/Gst.WebRTC/WebRTCDTLSTransportState.cs Normal file
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@ -0,0 +1,31 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCDTLSTransportStateGType))]
public enum WebRTCDTLSTransportState {
New = 0,
Closed = 1,
Failed = 2,
Connecting = 3,
Connected = 4,
}
internal class WebRTCDTLSTransportStateGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_dtls_transport_state_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_dtls_transport_state_get_type ());
}
}
}
#endregion
}

28
sources/generated/Gst.WebRTC/WebRTCICEComponent.cs Normal file
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@ -0,0 +1,28 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCICEComponentGType))]
public enum WebRTCICEComponent {
Rtp = 0,
Rtcp = 1,
}
internal class WebRTCICEComponentGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_ice_component_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_ice_component_get_type ());
}
}
}
#endregion
}

33
sources/generated/Gst.WebRTC/WebRTCICEConnectionState.cs Normal file
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@ -0,0 +1,33 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCICEConnectionStateGType))]
public enum WebRTCICEConnectionState {
New = 0,
Checking = 1,
Connected = 2,
Completed = 3,
Failed = 4,
Disconnected = 5,
Closed = 6,
}
internal class WebRTCICEConnectionStateGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_ice_connection_state_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_ice_connection_state_get_type ());
}
}
}
#endregion
}

29
sources/generated/Gst.WebRTC/WebRTCICEGatheringState.cs Normal file
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@ -0,0 +1,29 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCICEGatheringStateGType))]
public enum WebRTCICEGatheringState {
New = 0,
Gathering = 1,
Complete = 2,
}
internal class WebRTCICEGatheringStateGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_ice_gathering_state_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_ice_gathering_state_get_type ());
}
}
}
#endregion
}

28
sources/generated/Gst.WebRTC/WebRTCICERole.cs Normal file
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@ -0,0 +1,28 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCICERoleGType))]
public enum WebRTCICERole {
Controlled = 0,
Controlling = 1,
}
internal class WebRTCICERoleGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_ice_role_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_ice_role_get_type ());
}
}
}
#endregion
}

463
sources/generated/Gst.WebRTC/WebRTCICETransport.cs Normal file
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@ -0,0 +1,463 @@
// 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 WebRTCICETransport : Gst.Object {
protected WebRTCICETransport (IntPtr raw) : base(raw) {}
protected WebRTCICETransport() : base(IntPtr.Zero)
{
CreateNativeObject (new string [0], new GLib.Value [0]);
}
[GLib.Property ("component")]
public Gst.WebRTC.WebRTCICEComponent Component {
get {
GLib.Value val = GetProperty ("component");
Gst.WebRTC.WebRTCICEComponent ret = (Gst.WebRTC.WebRTCICEComponent) (Enum) val;
val.Dispose ();
return ret;
}
}
[GLib.Property ("gathering-state")]
public Gst.WebRTC.WebRTCICEGatheringState GatheringState {
get {
GLib.Value val = GetProperty ("gathering-state");
Gst.WebRTC.WebRTCICEGatheringState ret = (Gst.WebRTC.WebRTCICEGatheringState) (Enum) val;
val.Dispose ();
return ret;
}
}
[GLib.Property ("state")]
public Gst.WebRTC.WebRTCICEConnectionState State {
get {
GLib.Value val = GetProperty ("state");
Gst.WebRTC.WebRTCICEConnectionState ret = (Gst.WebRTC.WebRTCICEConnectionState) (Enum) val;
val.Dispose ();
return ret;
}
}
public Gst.WebRTC.WebRTCICERole Role {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("role"));
return (Gst.WebRTC.WebRTCICERole) (*raw_ptr);
}
}
}
public Gst.WebRTC.WebRTCICEComponent ComponentField {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("component"));
return (Gst.WebRTC.WebRTCICEComponent) (*raw_ptr);
}
}
}
public Gst.WebRTC.WebRTCICEConnectionState StateField {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("state"));
return (Gst.WebRTC.WebRTCICEConnectionState) (*raw_ptr);
}
}
}
public Gst.WebRTC.WebRTCICEGatheringState GatheringStateField {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("gathering_state"));
return (Gst.WebRTC.WebRTCICEGatheringState) (*raw_ptr);
}
}
}
public Gst.Element Src {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("src"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
}
}
}
public Gst.Element Sink {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("sink"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.Element;
}
}
}
[GLib.Signal("on-selected-candidate-pair-change")]
public event System.EventHandler OnSelectedCandidatePairChange {
add {
this.AddSignalHandler ("on-selected-candidate-pair-change", value);
}
remove {
this.RemoveSignalHandler ("on-selected-candidate-pair-change", value);
}
}
[GLib.Signal("on-new-candidate")]
public event Gst.WebRTC.OnNewCandidateHandler OnNewCandidate {
add {
this.AddSignalHandler ("on-new-candidate", value, typeof (Gst.WebRTC.OnNewCandidateArgs));
}
remove {
this.RemoveSignalHandler ("on-new-candidate", value);
}
}
static OnNewCandidateNativeDelegate OnNewCandidate_cb_delegate;
static OnNewCandidateNativeDelegate OnNewCandidateVMCallback {
get {
if (OnNewCandidate_cb_delegate == null)
OnNewCandidate_cb_delegate = new OnNewCandidateNativeDelegate (OnNewCandidate_cb);
return OnNewCandidate_cb_delegate;
}
}
static void OverrideOnNewCandidate (GLib.GType gtype)
{
OverrideOnNewCandidate (gtype, OnNewCandidateVMCallback);
}
static void OverrideOnNewCandidate (GLib.GType gtype, OnNewCandidateNativeDelegate callback)
{
OverrideVirtualMethod (gtype, "on-new-candidate", callback);
}
[UnmanagedFunctionPointer (CallingConvention.Cdecl)]
delegate void OnNewCandidateNativeDelegate (IntPtr inst, IntPtr _object);
static void OnNewCandidate_cb (IntPtr inst, IntPtr _object)
{
try {
WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
__obj.OnOnNewCandidate (GLib.Marshaller.Utf8PtrToString (_object));
} catch (Exception e) {
GLib.ExceptionManager.RaiseUnhandledException (e, false);
}
}
[GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideOnNewCandidate")]
protected virtual void OnOnNewCandidate (string _object)
{
InternalOnNewCandidate (_object);
}
private void InternalOnNewCandidate (string _object)
{
GLib.Value ret = GLib.Value.Empty;
GLib.ValueArray inst_and_params = new GLib.ValueArray (2);
GLib.Value[] vals = new GLib.Value [2];
vals [0] = new GLib.Value (this);
inst_and_params.Append (vals [0]);
vals [1] = new GLib.Value (_object);
inst_and_params.Append (vals [1]);
g_signal_chain_from_overridden (inst_and_params.ArrayPtr, ref ret);
foreach (GLib.Value v in vals)
v.Dispose ();
}
static OnSelectedCandidatePairChangeNativeDelegate OnSelectedCandidatePairChange_cb_delegate;
static OnSelectedCandidatePairChangeNativeDelegate OnSelectedCandidatePairChangeVMCallback {
get {
if (OnSelectedCandidatePairChange_cb_delegate == null)
OnSelectedCandidatePairChange_cb_delegate = new OnSelectedCandidatePairChangeNativeDelegate (OnSelectedCandidatePairChange_cb);
return OnSelectedCandidatePairChange_cb_delegate;
}
}
static void OverrideOnSelectedCandidatePairChange (GLib.GType gtype)
{
OverrideOnSelectedCandidatePairChange (gtype, OnSelectedCandidatePairChangeVMCallback);
}
static void OverrideOnSelectedCandidatePairChange (GLib.GType gtype, OnSelectedCandidatePairChangeNativeDelegate callback)
{
OverrideVirtualMethod (gtype, "on-selected-candidate-pair-change", callback);
}
[UnmanagedFunctionPointer (CallingConvention.Cdecl)]
delegate void OnSelectedCandidatePairChangeNativeDelegate (IntPtr inst);
static void OnSelectedCandidatePairChange_cb (IntPtr inst)
{
try {
WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
__obj.OnOnSelectedCandidatePairChange ();
} catch (Exception e) {
GLib.ExceptionManager.RaiseUnhandledException (e, false);
}
}
[GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideOnSelectedCandidatePairChange")]
protected virtual void OnOnSelectedCandidatePairChange ()
{
InternalOnSelectedCandidatePairChange ();
}
private void InternalOnSelectedCandidatePairChange ()
{
GLib.Value ret = GLib.Value.Empty;
GLib.ValueArray inst_and_params = new GLib.ValueArray (1);
GLib.Value[] vals = new GLib.Value [1];
vals [0] = new GLib.Value (this);
inst_and_params.Append (vals [0]);
g_signal_chain_from_overridden (inst_and_params.ArrayPtr, ref ret);
foreach (GLib.Value v in vals)
v.Dispose ();
}
static GatherCandidatesNativeDelegate GatherCandidates_cb_delegate;
static GatherCandidatesNativeDelegate GatherCandidatesVMCallback {
get {
if (GatherCandidates_cb_delegate == null)
GatherCandidates_cb_delegate = new GatherCandidatesNativeDelegate (GatherCandidates_cb);
return GatherCandidates_cb_delegate;
}
}
static void OverrideGatherCandidates (GLib.GType gtype)
{
OverrideGatherCandidates (gtype, GatherCandidatesVMCallback);
}
static void OverrideGatherCandidates (GLib.GType gtype, GatherCandidatesNativeDelegate callback)
{
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((long) gtype.GetClassPtr()) + (long) class_abi.GetFieldOffset("gather_candidates"));
*raw_ptr = Marshal.GetFunctionPointerForDelegate((Delegate) callback);
}
}
[UnmanagedFunctionPointer (CallingConvention.Cdecl)]
delegate bool GatherCandidatesNativeDelegate (IntPtr inst);
static bool GatherCandidates_cb (IntPtr inst)
{
try {
WebRTCICETransport __obj = GLib.Object.GetObject (inst, false) as WebRTCICETransport;
bool __result;
__result = __obj.OnGatherCandidates ();
return __result;
} catch (Exception e) {
GLib.ExceptionManager.RaiseUnhandledException (e, true);
// NOTREACHED: above call does not return.
throw e;
}
}
[GLib.DefaultSignalHandler(Type=typeof(Gst.WebRTC.WebRTCICETransport), ConnectionMethod="OverrideGatherCandidates")]
protected virtual bool OnGatherCandidates ()
{
return InternalGatherCandidates ();
}
private bool InternalGatherCandidates ()
{
GatherCandidatesNativeDelegate unmanaged = null;
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((long) this.LookupGType().GetThresholdType().GetClassPtr()) + (long) class_abi.GetFieldOffset("gather_candidates"));
unmanaged = (GatherCandidatesNativeDelegate) Marshal.GetDelegateForFunctionPointer(*raw_ptr, typeof(GatherCandidatesNativeDelegate));
}
if (unmanaged == null) return false;
bool __result = unmanaged (this.Handle);
return __result;
}
// 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("gather_candidates"
, Gst.Object.class_abi.Fields
, (uint) Marshal.SizeOf(typeof(IntPtr)) // gather_candidates
, null
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, "gather_candidates"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _class_abi;
}
}
// End of the ABI representation.
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_ice_transport_get_type();
public static new GLib.GType GType {
get {
IntPtr raw_ret = gst_webrtc_ice_transport_get_type();
GLib.GType ret = new GLib.GType(raw_ret);
return ret;
}
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern void gst_webrtc_ice_transport_connection_state_change(IntPtr raw, int new_state);
public void ConnectionStateChange(Gst.WebRTC.WebRTCICEConnectionState new_state) {
gst_webrtc_ice_transport_connection_state_change(Handle, (int) new_state);
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern void gst_webrtc_ice_transport_gathering_state_change(IntPtr raw, int new_state);
public void GatheringStateChange(Gst.WebRTC.WebRTCICEGatheringState new_state) {
gst_webrtc_ice_transport_gathering_state_change(Handle, (int) new_state);
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern void gst_webrtc_ice_transport_new_candidate(IntPtr raw, uint stream_id, int component, IntPtr attr);
public void NewCandidate(uint stream_id, Gst.WebRTC.WebRTCICEComponent component, string attr) {
IntPtr native_attr = GLib.Marshaller.StringToPtrGStrdup (attr);
gst_webrtc_ice_transport_new_candidate(Handle, stream_id, (int) component, native_attr);
GLib.Marshaller.Free (native_attr);
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern void gst_webrtc_ice_transport_selected_pair_change(IntPtr raw);
public void SelectedPairChange() {
gst_webrtc_ice_transport_selected_pair_change(Handle);
}
static WebRTCICETransport ()
{
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("role"
, Gst.Object.abi_info.Fields
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICERole))) // role
, null
, "component"
, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_roleAlign), "role")
, 0
),
new GLib.AbiField("component"
, -1
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEComponent))) // component
, "role"
, "state"
, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_componentAlign), "component")
, 0
),
new GLib.AbiField("state"
, -1
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEConnectionState))) // state
, "component"
, "gathering_state"
, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_stateAlign), "state")
, 0
),
new GLib.AbiField("gathering_state"
, -1
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCICEGatheringState))) // gathering_state
, "state"
, "src"
, (long) Marshal.OffsetOf(typeof(GstWebRTCICETransport_gathering_stateAlign), "gathering_state")
, 0
),
new GLib.AbiField("src"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // src
, "gathering_state"
, "sink"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("sink"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // sink
, "src"
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, "sink"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _abi_info;
}
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCICETransport_roleAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCICERole role;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCICETransport_componentAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCICEComponent component;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCICETransport_stateAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCICEConnectionState state;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCICETransport_gathering_stateAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCICEGatheringState gathering_state;
}
// End of the ABI representation.
#endregion
}
}

32
sources/generated/Gst.WebRTC/WebRTCPeerConnectionState.cs Normal file
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// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCPeerConnectionStateGType))]
public enum WebRTCPeerConnectionState {
New = 0,
Connecting = 1,
Connected = 2,
Disconnected = 3,
Failed = 4,
Closed = 5,
}
internal class WebRTCPeerConnectionStateGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_peer_connection_state_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_peer_connection_state_get_type ());
}
}
}
#endregion
}

140
sources/generated/Gst.WebRTC/WebRTCRTPReceiver.cs Normal file
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// 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 WebRTCRTPReceiver : Gst.Object {
public WebRTCRTPReceiver (IntPtr raw) : base(raw) {}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_rtp_receiver_new();
public WebRTCRTPReceiver () : base (IntPtr.Zero)
{
if (GetType () != typeof (WebRTCRTPReceiver)) {
CreateNativeObject (new string [0], new GLib.Value[0]);
return;
}
Raw = gst_webrtc_rtp_receiver_new();
}
[DllImport("libgstwebrtc-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 {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("transport"));
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("libgstwebrtc-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);
}
}
// 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("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_rtp_receiver_get_type();
public static new GLib.GType GType {
get {
IntPtr raw_ret = gst_webrtc_rtp_receiver_get_type();
GLib.GType ret = new GLib.GType(raw_ret);
return ret;
}
}
static WebRTCRTPReceiver ()
{
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"
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, "rtcp_transport"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _abi_info;
}
}
// End of the ABI representation.
#endregion
}
}

148
sources/generated/Gst.WebRTC/WebRTCRTPSender.cs Normal file
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// 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("libgstwebrtc-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("libgstwebrtc-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("libgstwebrtc-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("libgstwebrtc-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
}
}

281
sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs Normal file
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// 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 WebRTCRTPTransceiver : Gst.Object {
protected WebRTCRTPTransceiver (IntPtr raw) : base(raw) {}
protected WebRTCRTPTransceiver() : base(IntPtr.Zero)
{
CreateNativeObject (new string [0], new GLib.Value [0]);
}
[GLib.Property ("mlineindex")]
public uint Mlineindex {
get {
GLib.Value val = GetProperty ("mlineindex");
uint ret = (uint) val;
val.Dispose ();
return ret;
}
}
[GLib.Property ("receiver")]
public Gst.WebRTC.WebRTCRTPReceiver Receiver {
get {
GLib.Value val = GetProperty ("receiver");
Gst.WebRTC.WebRTCRTPReceiver ret = (Gst.WebRTC.WebRTCRTPReceiver) val;
val.Dispose ();
return ret;
}
}
[GLib.Property ("sender")]
public Gst.WebRTC.WebRTCRTPSender Sender {
get {
GLib.Value val = GetProperty ("sender");
Gst.WebRTC.WebRTCRTPSender ret = (Gst.WebRTC.WebRTCRTPSender) val;
val.Dispose ();
return ret;
}
}
public uint Mline {
get {
unsafe {
uint* raw_ptr = (uint*)(((byte*)Handle) + abi_info.GetFieldOffset("mline"));
return (*raw_ptr);
}
}
}
public string Mid {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("mid"));
return GLib.Marshaller.Utf8PtrToString ((*raw_ptr));
}
}
}
public bool Stopped {
get {
unsafe {
bool* raw_ptr = (bool*)(((byte*)Handle) + abi_info.GetFieldOffset("stopped"));
return (*raw_ptr);
}
}
}
public Gst.WebRTC.WebRTCRTPSender SenderField {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("sender"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCRTPSender;
}
}
}
public Gst.WebRTC.WebRTCRTPReceiver ReceiverField {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("receiver"));
return GLib.Object.GetObject((*raw_ptr)) as Gst.WebRTC.WebRTCRTPReceiver;
}
}
}
public Gst.WebRTC.WebRTCRTPTransceiverDirection Direction {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("direction"));
return (Gst.WebRTC.WebRTCRTPTransceiverDirection) (*raw_ptr);
}
}
}
public Gst.WebRTC.WebRTCRTPTransceiverDirection CurrentDirection {
get {
unsafe {
int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("current_direction"));
return (Gst.WebRTC.WebRTCRTPTransceiverDirection) (*raw_ptr);
}
}
}
public Gst.Caps CodecPreferences {
get {
unsafe {
IntPtr* raw_ptr = (IntPtr*)(((byte*)Handle) + abi_info.GetFieldOffset("codec_preferences"));
return (*raw_ptr) == IntPtr.Zero ? null : (Gst.Caps) GLib.Opaque.GetOpaque ((*raw_ptr), typeof (Gst.Caps), false);
}
}
}
// 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("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_rtp_transceiver_get_type();
public static new GLib.GType GType {
get {
IntPtr raw_ret = gst_webrtc_rtp_transceiver_get_type();
GLib.GType ret = new GLib.GType(raw_ret);
return ret;
}
}
static WebRTCRTPTransceiver ()
{
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("mline"
, Gst.Object.abi_info.Fields
, (uint) Marshal.SizeOf(typeof(uint)) // mline
, null
, "mid"
, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_mlineAlign), "mline")
, 0
),
new GLib.AbiField("mid"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // mid
, "mline"
, "stopped"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("stopped"
, -1
, (uint) Marshal.SizeOf(typeof(bool)) // stopped
, "mid"
, "sender"
, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_stoppedAlign), "stopped")
, 0
),
new GLib.AbiField("sender"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // sender
, "stopped"
, "receiver"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("receiver"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // receiver
, "sender"
, "direction"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("direction"
, -1
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCRTPTransceiverDirection))) // direction
, "receiver"
, "current_direction"
, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_directionAlign), "direction")
, 0
),
new GLib.AbiField("current_direction"
, -1
, (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCRTPTransceiverDirection))) // current_direction
, "direction"
, "codec_preferences"
, (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_current_directionAlign), "current_direction")
, 0
),
new GLib.AbiField("codec_preferences"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // codec_preferences
, "current_direction"
, "_padding"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
, "codec_preferences"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
});
return _abi_info;
}
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCRTPTransceiver_mlineAlign
{
sbyte f1;
private uint mline;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCRTPTransceiver_stoppedAlign
{
sbyte f1;
private bool stopped;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCRTPTransceiver_directionAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCRTPTransceiverDirection direction;
}
[StructLayout(LayoutKind.Sequential)]
public struct GstWebRTCRTPTransceiver_current_directionAlign
{
sbyte f1;
private Gst.WebRTC.WebRTCRTPTransceiverDirection current_direction;
}
// End of the ABI representation.
#endregion
}
}

31
sources/generated/Gst.WebRTC/WebRTCRTPTransceiverDirection.cs Normal file
View file

@ -0,0 +1,31 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCRTPTransceiverDirectionGType))]
public enum WebRTCRTPTransceiverDirection {
None = 0,
Inactive = 1,
Sendonly = 2,
Recvonly = 3,
Sendrecv = 4,
}
internal class WebRTCRTPTransceiverDirectionGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_rtp_transceiver_direction_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_rtp_transceiver_direction_get_type ());
}
}
}
#endregion
}

30
sources/generated/Gst.WebRTC/WebRTCSDPType.cs Normal file
View file

@ -0,0 +1,30 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCSDPTypeGType))]
public enum WebRTCSDPType {
Offer = 1,
Pranswer = 2,
Answer = 3,
Rollback = 4,
}
internal class WebRTCSDPTypeGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_sdp_type_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_sdp_type_get_type ());
}
}
}
#endregion
}

83
sources/generated/Gst.WebRTC/WebRTCSessionDescription.cs Normal file
View file

@ -0,0 +1,83 @@
// 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
[StructLayout(LayoutKind.Sequential)]
public partial struct WebRTCSessionDescription : IEquatable<WebRTCSessionDescription> {
public Gst.WebRTC.WebRTCSDPType Type;
private IntPtr _sdp;
public Gst.Sdp.SDPMessage Sdp {
get {
return _sdp == IntPtr.Zero ? null : (Gst.Sdp.SDPMessage) GLib.Opaque.GetOpaque (_sdp, typeof (Gst.Sdp.SDPMessage), false);
}
set {
_sdp = value == null ? IntPtr.Zero : value.Handle;
}
}
public static Gst.WebRTC.WebRTCSessionDescription Zero = new Gst.WebRTC.WebRTCSessionDescription ();
public static Gst.WebRTC.WebRTCSessionDescription New(IntPtr raw) {
if (raw == IntPtr.Zero)
return Gst.WebRTC.WebRTCSessionDescription.Zero;
return (Gst.WebRTC.WebRTCSessionDescription) Marshal.PtrToStructure (raw, typeof (Gst.WebRTC.WebRTCSessionDescription));
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_session_description_new(int type, IntPtr sdp);
public static WebRTCSessionDescription New(Gst.WebRTC.WebRTCSDPType type, Gst.Sdp.SDPMessage sdp)
{
WebRTCSessionDescription result = WebRTCSessionDescription.New (gst_webrtc_session_description_new((int) type, sdp == null ? IntPtr.Zero : sdp.Handle));
return result;
}
[DllImport("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_session_description_get_type();
public static GLib.GType GType {
get {
IntPtr raw_ret = gst_webrtc_session_description_get_type();
GLib.GType ret = new GLib.GType(raw_ret);
return ret;
}
}
public bool Equals (WebRTCSessionDescription other)
{
return true && Type.Equals (other.Type) && Sdp.Equals (other.Sdp);
}
public override bool Equals (object other)
{
return other is WebRTCSessionDescription && Equals ((WebRTCSessionDescription) other);
}
public override int GetHashCode ()
{
return this.GetType ().FullName.GetHashCode () ^ Type.GetHashCode () ^ Sdp.GetHashCode ();
}
public static explicit operator GLib.Value (Gst.WebRTC.WebRTCSessionDescription boxed)
{
GLib.Value val = GLib.Value.Empty;
val.Init (Gst.WebRTC.WebRTCSessionDescription.GType);
val.Val = boxed;
return val;
}
public static explicit operator Gst.WebRTC.WebRTCSessionDescription (GLib.Value val)
{
return (Gst.WebRTC.WebRTCSessionDescription) val.Val;
}
#endregion
}
}

32
sources/generated/Gst.WebRTC/WebRTCSignalingState.cs Normal file
View file

@ -0,0 +1,32 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCSignalingStateGType))]
public enum WebRTCSignalingState {
Stable = 0,
Closed = 1,
HaveLocalOffer = 2,
HaveRemoteOffer = 3,
HaveLocalPranswer = 4,
HaveRemotePranswer = 5,
}
internal class WebRTCSignalingStateGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_signaling_state_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_signaling_state_get_type ());
}
}
}
#endregion
}

40
sources/generated/Gst.WebRTC/WebRTCStatsType.cs Normal file
View file

@ -0,0 +1,40 @@
// This file was generated by the Gtk# code generator.
// Any changes made will be lost if regenerated.
namespace Gst.WebRTC {
using System;
using System.Runtime.InteropServices;
#region Autogenerated code
[GLib.GType (typeof (Gst.WebRTC.WebRTCStatsTypeGType))]
public enum WebRTCStatsType {
Codec = 1,
InboundRtp = 2,
OutboundRtp = 3,
RemoteInboundRtp = 4,
RemoteOutboundRtp = 5,
Csrc = 6,
PeerConnection = 7,
DataChannel = 8,
Stream = 9,
Transport = 10,
CandidatePair = 11,
LocalCandidate = 12,
RemoteCandidate = 13,
Certificate = 14,
}
internal class WebRTCStatsTypeGType {
[DllImport ("libgstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern IntPtr gst_webrtc_stats_type_get_type ();
public static GLib.GType GType {
get {
return new GLib.GType (gst_webrtc_stats_type_get_type ());
}
}
}
#endregion
}

5
sources/generated/GtkSharp/ObjectManager.cs
View file

@ -69,6 +69,11 @@ namespace GtkSharp.GstreamerSharp {
GLib.GType.Register (Gst.Video.VideoEncoder.GType, typeof (Gst.Video.VideoEncoder));
GLib.GType.Register (Gst.Video.VideoFilter.GType, typeof (Gst.Video.VideoFilter));
GLib.GType.Register (Gst.Video.VideoSink.GType, typeof (Gst.Video.VideoSink));
GLib.GType.Register (Gst.WebRTC.WebRTCDTLSTransport.GType, typeof (Gst.WebRTC.WebRTCDTLSTransport));
GLib.GType.Register (Gst.WebRTC.WebRTCICETransport.GType, typeof (Gst.WebRTC.WebRTCICETransport));
GLib.GType.Register (Gst.WebRTC.WebRTCRTPReceiver.GType, typeof (Gst.WebRTC.WebRTCRTPReceiver));
GLib.GType.Register (Gst.WebRTC.WebRTCRTPSender.GType, typeof (Gst.WebRTC.WebRTCRTPSender));
GLib.GType.Register (Gst.WebRTC.WebRTCRTPTransceiver.GType, typeof (Gst.WebRTC.WebRTCRTPTransceiver));
}
}
}

48
sources/generated/gstreamer-sharp-abi.c
View file

@ -21,6 +21,7 @@
#include <gst/video/video.h>
#include <gst/video/gstvideoaffinetransformationmeta.h>
#include <gst/net/gstnetcontrolmessagemeta.h>
#include <gst/webrtc/webrtc.h>
int main (int argc, char *argv[]) {
g_print("\"sizeof(GstAllocatorClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstAllocatorClass));
@ -944,5 +945,52 @@ int main (int argc, char *argv[]) {
g_print("\"GstVideoInfo.fps_d\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, fps_d));
g_print("\"GstVideoInfo.offset\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, offset));
g_print("\"GstVideoInfo.stride\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstVideoInfo, stride));
g_print("\"sizeof(GstWebRTCDTLSTransportClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCDTLSTransportClass));
g_print("\"GstWebRTCDTLSTransportClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransportClass, _padding));
g_print("\"sizeof(GstWebRTCDTLSTransport)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCDTLSTransport));
g_print("\"GstWebRTCDTLSTransport.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, transport));
g_print("\"GstWebRTCDTLSTransport.state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, state));
g_print("\"GstWebRTCDTLSTransport.is_rtcp\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, is_rtcp));
g_print("\"GstWebRTCDTLSTransport.client\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, client));
g_print("\"GstWebRTCDTLSTransport.session_id\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, session_id));
g_print("\"GstWebRTCDTLSTransport.dtlssrtpenc\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, dtlssrtpenc));
g_print("\"GstWebRTCDTLSTransport.dtlssrtpdec\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, dtlssrtpdec));
g_print("\"GstWebRTCDTLSTransport._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCDTLSTransport, _padding));
g_print("\"sizeof(GstWebRTCICETransportClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCICETransportClass));
g_print("\"GstWebRTCICETransportClass.gather_candidates\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransportClass, gather_candidates));
g_print("\"GstWebRTCICETransportClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransportClass, _padding));
g_print("\"sizeof(GstWebRTCICETransport)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCICETransport));
g_print("\"GstWebRTCICETransport.role\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, role));
g_print("\"GstWebRTCICETransport.component\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, component));
g_print("\"GstWebRTCICETransport.state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, state));
g_print("\"GstWebRTCICETransport.gathering_state\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, gathering_state));
g_print("\"GstWebRTCICETransport.src\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, src));
g_print("\"GstWebRTCICETransport.sink\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, sink));
g_print("\"GstWebRTCICETransport._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCICETransport, _padding));
g_print("\"sizeof(GstWebRTCRTPReceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPReceiverClass));
g_print("\"GstWebRTCRTPReceiverClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiverClass, _padding));
g_print("\"sizeof(GstWebRTCRTPReceiver)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPReceiver));
g_print("\"GstWebRTCRTPReceiver.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, transport));
g_print("\"GstWebRTCRTPReceiver.rtcp_transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, rtcp_transport));
g_print("\"GstWebRTCRTPReceiver._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPReceiver, _padding));
g_print("\"sizeof(GstWebRTCRTPSenderClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPSenderClass));
g_print("\"GstWebRTCRTPSenderClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSenderClass, _padding));
g_print("\"sizeof(GstWebRTCRTPSender)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPSender));
g_print("\"GstWebRTCRTPSender.transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, transport));
g_print("\"GstWebRTCRTPSender.rtcp_transport\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, rtcp_transport));
g_print("\"GstWebRTCRTPSender.send_encodings\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, send_encodings));
g_print("\"GstWebRTCRTPSender._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, _padding));
g_print("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiverClass));
g_print("\"GstWebRTCRTPTransceiverClass._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiverClass, _padding));
g_print("\"sizeof(GstWebRTCRTPTransceiver)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiver));
g_print("\"GstWebRTCRTPTransceiver.mline\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, mline));
g_print("\"GstWebRTCRTPTransceiver.mid\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, mid));
g_print("\"GstWebRTCRTPTransceiver.stopped\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, stopped));
g_print("\"GstWebRTCRTPTransceiver.sender\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, sender));
g_print("\"GstWebRTCRTPTransceiver.receiver\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, receiver));
g_print("\"GstWebRTCRTPTransceiver.direction\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, direction));
g_print("\"GstWebRTCRTPTransceiver.current_direction\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, current_direction));
g_print("\"GstWebRTCRTPTransceiver.codec_preferences\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, codec_preferences));
g_print("\"GstWebRTCRTPTransceiver._padding\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, _padding));
return 0;
}

47
sources/generated/gstreamer-sharp-abi.cs
View file

@ -939,6 +939,53 @@ namespace AbiTester {
Console.WriteLine("\"GstVideoInfo.fps_d\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("fps_d") + "\"");
Console.WriteLine("\"GstVideoInfo.offset\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("offset") + "\"");
Console.WriteLine("\"GstVideoInfo.stride\": \"" + Gst.Video.VideoInfo.abi_info.GetFieldOffset("stride") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCDTLSTransportClass)\": \"" + Gst.WebRTC.WebRTCDTLSTransport.class_abi.Size + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransportClass._padding\": \"" + Gst.WebRTC.WebRTCDTLSTransport.class_abi.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCDTLSTransport)\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.Size + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.transport\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("transport") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.state\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("state") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.is_rtcp\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("is_rtcp") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.client\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("client") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.session_id\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("session_id") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.dtlssrtpenc\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("dtlssrtpenc") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport.dtlssrtpdec\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("dtlssrtpdec") + "\"");
Console.WriteLine("\"GstWebRTCDTLSTransport._padding\": \"" + Gst.WebRTC.WebRTCDTLSTransport.abi_info.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCICETransportClass)\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.Size + "\"");
Console.WriteLine("\"GstWebRTCICETransportClass.gather_candidates\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.GetFieldOffset("gather_candidates") + "\"");
Console.WriteLine("\"GstWebRTCICETransportClass._padding\": \"" + Gst.WebRTC.WebRTCICETransport.class_abi.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCICETransport)\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.Size + "\"");
Console.WriteLine("\"GstWebRTCICETransport.role\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("role") + "\"");
Console.WriteLine("\"GstWebRTCICETransport.component\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("component") + "\"");
Console.WriteLine("\"GstWebRTCICETransport.state\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("state") + "\"");
Console.WriteLine("\"GstWebRTCICETransport.gathering_state\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("gathering_state") + "\"");
Console.WriteLine("\"GstWebRTCICETransport.src\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("src") + "\"");
Console.WriteLine("\"GstWebRTCICETransport.sink\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("sink") + "\"");
Console.WriteLine("\"GstWebRTCICETransport._padding\": \"" + Gst.WebRTC.WebRTCICETransport.abi_info.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPReceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPReceiver.class_abi.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPReceiverClass._padding\": \"" + Gst.WebRTC.WebRTCRTPReceiver.class_abi.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPReceiver)\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPReceiver.transport\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("transport") + "\"");
Console.WriteLine("\"GstWebRTCRTPReceiver.rtcp_transport\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("rtcp_transport") + "\"");
Console.WriteLine("\"GstWebRTCRTPReceiver._padding\": \"" + Gst.WebRTC.WebRTCRTPReceiver.abi_info.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPSenderClass)\": \"" + Gst.WebRTC.WebRTCRTPSender.class_abi.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPSenderClass._padding\": \"" + Gst.WebRTC.WebRTCRTPSender.class_abi.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPSender)\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPSender.transport\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("transport") + "\"");
Console.WriteLine("\"GstWebRTCRTPSender.rtcp_transport\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("rtcp_transport") + "\"");
Console.WriteLine("\"GstWebRTCRTPSender.send_encodings\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("send_encodings") + "\"");
Console.WriteLine("\"GstWebRTCRTPSender._padding\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiverClass._padding\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.GetFieldOffset("_padding") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiver)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.mline\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mline") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.mid\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mid") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.stopped\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("stopped") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.sender\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("sender") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.receiver\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("receiver") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.direction\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("direction") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.current_direction\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("current_direction") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.codec_preferences\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("codec_preferences") + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver._padding\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("_padding") + "\"");
}
}
}

301
sources/generated/gstreamer-sharp-api.xml
View file

@ -28814,4 +28814,305 @@
<constant value="16" ctype="gint" gtype="gint" name="VIDEO_TILE_Y_TILES_SHIFT" />
</object>
</namespace>
<namespace name="Gst.WebRTC" library="libgstwebrtc-1.0-0.dll">
<enum name="WebRTCDTLSSetup" cname="GstWebRTCDTLSSetup" type="enum" gtype="gst_webrtc_dtls_setup_get_type">
<member cname="GST_WEBRTC_DTLS_SETUP_NONE" name="None" value="0" />
<member cname="GST_WEBRTC_DTLS_SETUP_ACTPASS" name="Actpass" value="1" />
<member cname="GST_WEBRTC_DTLS_SETUP_ACTIVE" name="Active" value="2" />
<member cname="GST_WEBRTC_DTLS_SETUP_PASSIVE" name="Passive" value="3" />
</enum>
<enum name="WebRTCDTLSTransportState" cname="GstWebRTCDTLSTransportState" type="enum" gtype="gst_webrtc_dtls_transport_state_get_type">
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" name="New" value="0" />
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" name="Closed" value="1" />
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" name="Failed" value="2" />
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" name="Connecting" value="3" />
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" name="Connected" value="4" />
</enum>
<enum name="WebRTCICEComponent" cname="GstWebRTCICEComponent" type="enum" gtype="gst_webrtc_ice_component_get_type">
<member cname="GST_WEBRTC_ICE_COMPONENT_RTP" name="Rtp" value="0" />
<member cname="GST_WEBRTC_ICE_COMPONENT_RTCP" name="Rtcp" value="1" />
</enum>
<enum name="WebRTCICEConnectionState" cname="GstWebRTCICEConnectionState" type="enum" gtype="gst_webrtc_ice_connection_state_get_type">
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" name="New" value="0" />
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" name="Checking" value="1" />
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" name="Connected" value="2" />
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" name="Completed" value="3" />
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" name="Failed" value="4" />
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="5" />
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" name="Closed" value="6" />
</enum>
<enum name="WebRTCICEGatheringState" cname="GstWebRTCICEGatheringState" type="enum" gtype="gst_webrtc_ice_gathering_state_get_type">
<member cname="GST_WEBRTC_ICE_GATHERING_STATE_NEW" name="New" value="0" />
<member cname="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" name="Gathering" value="1" />
<member cname="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" name="Complete" value="2" />
</enum>
<enum name="WebRTCICERole" cname="GstWebRTCICERole" type="enum" gtype="gst_webrtc_ice_role_get_type">
<member cname="GST_WEBRTC_ICE_ROLE_CONTROLLED" name="Controlled" value="0" />
<member cname="GST_WEBRTC_ICE_ROLE_CONTROLLING" name="Controlling" value="1" />
</enum>
<enum name="WebRTCPeerConnectionState" cname="GstWebRTCPeerConnectionState" type="enum" gtype="gst_webrtc_peer_connection_state_get_type">
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" name="New" value="0" />
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" name="Connecting" value="1" />
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" name="Connected" value="2" />
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="3" />
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" name="Failed" value="4" />
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" name="Closed" value="5" />
</enum>
<enum name="WebRTCRTPTransceiverDirection" cname="GstWebRTCRTPTransceiverDirection" type="enum" gtype="gst_webrtc_rtp_transceiver_direction_get_type">
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" name="None" value="0" />
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" name="Inactive" value="1" />
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" name="Sendonly" value="2" />
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" name="Recvonly" value="3" />
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" name="Sendrecv" value="4" />
</enum>
<enum name="WebRTCSDPType" cname="GstWebRTCSDPType" type="enum" gtype="gst_webrtc_sdp_type_get_type">
<member cname="GST_WEBRTC_SDP_TYPE_OFFER" name="Offer" value="1" />
<member cname="GST_WEBRTC_SDP_TYPE_PRANSWER" name="Pranswer" value="2" />
<member cname="GST_WEBRTC_SDP_TYPE_ANSWER" name="Answer" value="3" />
<member cname="GST_WEBRTC_SDP_TYPE_ROLLBACK" name="Rollback" value="4" />
</enum>
<enum name="WebRTCSignalingState" cname="GstWebRTCSignalingState" type="enum" gtype="gst_webrtc_signaling_state_get_type">
<member cname="GST_WEBRTC_SIGNALING_STATE_STABLE" name="Stable" value="0" />
<member cname="GST_WEBRTC_SIGNALING_STATE_CLOSED" name="Closed" value="1" />
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" name="HaveLocalOffer" value="2" />
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" name="HaveRemoteOffer" value="3" />
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" name="HaveLocalPranswer" value="4" />
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" name="HaveRemotePranswer" value="5" />
</enum>
<enum name="WebRTCStatsType" cname="GstWebRTCStatsType" type="enum" gtype="gst_webrtc_stats_type_get_type">
<member cname="GST_WEBRTC_STATS_CODEC" name="Codec" value="1" />
<member cname="GST_WEBRTC_STATS_INBOUND_RTP" name="InboundRtp" value="2" />
<member cname="GST_WEBRTC_STATS_OUTBOUND_RTP" name="OutboundRtp" value="3" />
<member cname="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" name="RemoteInboundRtp" value="4" />
<member cname="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" name="RemoteOutboundRtp" value="5" />
<member cname="GST_WEBRTC_STATS_CSRC" name="Csrc" value="6" />
<member cname="GST_WEBRTC_STATS_PEER_CONNECTION" name="PeerConnection" value="7" />
<member cname="GST_WEBRTC_STATS_DATA_CHANNEL" name="DataChannel" value="8" />
<member cname="GST_WEBRTC_STATS_STREAM" name="Stream" value="9" />
<member cname="GST_WEBRTC_STATS_TRANSPORT" name="Transport" value="10" />
<member cname="GST_WEBRTC_STATS_CANDIDATE_PAIR" name="CandidatePair" value="11" />
<member cname="GST_WEBRTC_STATS_LOCAL_CANDIDATE" name="LocalCandidate" value="12" />
<member cname="GST_WEBRTC_STATS_REMOTE_CANDIDATE" name="RemoteCandidate" value="13" />
<member cname="GST_WEBRTC_STATS_CERTIFICATE" name="Certificate" value="14" />
</enum>
<object name="WebRTCDTLSTransport" cname="GstWebRTCDTLSTransport" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCDTLSTransportClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</class_struct>
<method name="GetType" cname="gst_webrtc_dtls_transport_get_type" shared="true">
<return-type type="GType" />
</method>
<constructor cname="gst_webrtc_dtls_transport_new">
<parameters>
<parameter name="session_id" type="guint" />
<parameter name="rtcp" type="gboolean" />
</parameters>
</constructor>
<method name="SetTransport" cname="gst_webrtc_dtls_transport_set_transport">
<return-type type="void" />
<parameters>
<parameter name="ice" type="GstWebRTCICETransport*" />
</parameters>
</method>
<property name="Certificate" cname="certificate" type="gchar*" readable="true" writeable="true" construct="false" construct-only="false" />
<property name="Client" cname="client" type="gboolean" readable="true" writeable="true" construct="false" construct-only="false" />
<property name="RemoteCertificate" cname="remote-certificate" type="gchar*" readable="true" writeable="false" construct="false" construct-only="false" />
<property name="Rtcp" cname="rtcp" type="gboolean" readable="true" writeable="true" construct="false" construct-only="true" />
<property name="SessionId" cname="session-id" type="guint" readable="true" writeable="true" construct="false" construct-only="true" />
<property name="State" cname="state" type="GstWebRTCDTLSTransportState" readable="true" writeable="false" construct="false" construct-only="false" />
<property name="Transport" cname="transport" type="GstWebRTCICETransport*" readable="true" writeable="false" construct="false" construct-only="false" />
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
<field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="TransportField" type="GstWebRTCICETransport*" />
<field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCDTLSTransportState" />
<field cname="is_rtcp" access="public" writeable="false" readable="true" is_callback="false" name="IsRtcp" type="gboolean" />
<field cname="client" access="public" writeable="false" readable="true" is_callback="false" name="ClientField" type="gboolean" />
<field cname="session_id" access="public" writeable="false" readable="true" is_callback="false" name="SessionIdField" type="guint" />
<field cname="dtlssrtpenc" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpenc" type="GstElement*" />
<field cname="dtlssrtpdec" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpdec" type="GstElement*" />
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</object>
<object name="WebRTCICETransport" cname="GstWebRTCICETransport" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCICETransportClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
<warning>missing glib:type-name</warning>
</field>
<method vm="gather_candidates" />
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</class_struct>
<method name="GetType" cname="gst_webrtc_ice_transport_get_type" shared="true">
<return-type type="GType" />
</method>
<virtual_method name="GatherCandidates" cname="gather_candidates">
<return-type type="gboolean" />
<parameters />
</virtual_method>
<method name="ConnectionStateChange" cname="gst_webrtc_ice_transport_connection_state_change">
<return-type type="void" />
<parameters>
<parameter name="new_state" type="GstWebRTCICEConnectionState" />
</parameters>
</method>
<method name="GatheringStateChange" cname="gst_webrtc_ice_transport_gathering_state_change">
<return-type type="void" />
<parameters>
<parameter name="new_state" type="GstWebRTCICEGatheringState" />
</parameters>
</method>
<method name="NewCandidate" cname="gst_webrtc_ice_transport_new_candidate">
<return-type type="void" />
<parameters>
<parameter name="stream_id" type="guint" />
<parameter name="component" type="GstWebRTCICEComponent" />
<parameter name="attr" type="const-gchar*" />
</parameters>
</method>
<method name="SelectedPairChange" cname="gst_webrtc_ice_transport_selected_pair_change">
<return-type type="void" />
<parameters />
</method>
<property name="Component" cname="component" type="GstWebRTCICEComponent" readable="true" writeable="true" construct="false" construct-only="true" />
<property name="GatheringState" cname="gathering-state" type="GstWebRTCICEGatheringState" readable="true" writeable="false" construct="false" construct-only="false" />
<property name="State" cname="state" type="GstWebRTCICEConnectionState" readable="true" writeable="false" construct="false" construct-only="false" />
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
<field cname="role" access="public" writeable="false" readable="true" is_callback="false" name="Role" type="GstWebRTCICERole" />
<field cname="component" access="public" writeable="false" readable="true" is_callback="false" name="ComponentField" type="GstWebRTCICEComponent" />
<field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCICEConnectionState" />
<field cname="gathering_state" access="public" writeable="false" readable="true" is_callback="false" name="GatheringStateField" type="GstWebRTCICEGatheringState" />
<field cname="src" access="public" writeable="false" readable="true" is_callback="false" name="Src" type="GstElement*" />
<field cname="sink" access="public" writeable="false" readable="true" is_callback="false" name="Sink" type="GstElement*" />
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
<signal name="OnNewCandidate" cname="on-new-candidate" when="last">
<return-type type="void" />
<parameters>
<parameter name="_object" type="const-gchar*" />
</parameters>
</signal>
<signal name="OnSelectedCandidatePairChange" cname="on-selected-candidate-pair-change" when="last">
<return-type type="void" />
<parameters />
</signal>
</object>
<object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCRTPReceiverClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</class_struct>
<method name="GetType" cname="gst_webrtc_rtp_receiver_get_type" shared="true">
<return-type type="GType" />
</method>
<constructor cname="gst_webrtc_rtp_receiver_new" disable_void_ctor="" />
<method name="SetRtcpTransport" cname="gst_webrtc_rtp_receiver_set_rtcp_transport">
<return-type type="void" />
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*" />
</parameters>
</method>
<method name="SetTransport" cname="gst_webrtc_rtp_receiver_set_transport">
<return-type type="void" />
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*" />
</parameters>
</method>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
<field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*" />
<field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*" />
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</object>
<object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCRTPSenderClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</class_struct>
<method name="GetType" cname="gst_webrtc_rtp_sender_get_type" shared="true">
<return-type type="GType" />
</method>
<constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor="" />
<method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
<return-type type="void" />
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*" />
</parameters>
</method>
<method name="SetTransport" cname="gst_webrtc_rtp_sender_set_transport">
<return-type type="void" />
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*" />
</parameters>
</method>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
<field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*" />
<field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*" />
<field cname="send_encodings" access="public" writeable="false" readable="true" is_callback="false" name="SendEncodings" type="GArray*" array="true" null_term_array="true" />
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</object>
<object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCRTPTransceiverClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</class_struct>
<method name="GetType" cname="gst_webrtc_rtp_transceiver_get_type" shared="true">
<return-type type="GType" />
</method>
<property name="Mlineindex" cname="mlineindex" type="guint" readable="true" writeable="true" construct="false" construct-only="true" />
<property name="Receiver" cname="receiver" type="GstWebRTCRTPReceiver*" readable="true" writeable="true" construct="false" construct-only="true" />
<property name="Sender" cname="sender" type="GstWebRTCRTPSender*" readable="true" writeable="true" construct="false" construct-only="true" />
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*" hidden="true" />
<field cname="mline" access="public" writeable="false" readable="true" is_callback="false" name="Mline" type="guint" />
<field cname="mid" access="public" writeable="false" readable="true" is_callback="false" name="Mid" type="gchar*" />
<field cname="stopped" access="public" writeable="false" readable="true" is_callback="false" name="Stopped" type="gboolean" />
<field cname="sender" access="public" writeable="false" readable="true" is_callback="false" name="SenderField" type="GstWebRTCRTPSender*" />
<field cname="receiver" access="public" writeable="false" readable="true" is_callback="false" name="ReceiverField" type="GstWebRTCRTPReceiver*" />
<field cname="direction" access="public" writeable="false" readable="true" is_callback="false" name="Direction" type="GstWebRTCRTPTransceiverDirection" />
<field cname="current_direction" access="public" writeable="false" readable="true" is_callback="false" name="CurrentDirection" type="GstWebRTCRTPTransceiverDirection" />
<field cname="codec_preferences" access="public" writeable="false" readable="true" is_callback="false" name="CodecPreferences" type="GstCaps*">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" />
</object>
<boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
<method name="GetType" cname="gst_webrtc_session_description_get_type" shared="true">
<return-type type="GType" />
</method>
<field cname="type" access="public" writeable="true" readable="true" is_callback="false" name="Type" type="GstWebRTCSDPType" />
<field cname="sdp" access="public" writeable="true" readable="true" is_callback="false" name="Sdp" type="GstSDPMessage*">
<warning>missing glib:type-name</warning>
</field>
<constructor cname="gst_webrtc_session_description_new">
<parameters>
<parameter name="type" type="GstWebRTCSDPType" />
<parameter name="sdp" type="GstSDPMessage*">
<warning>missing glib:type-name</warning>
</parameter>
</parameters>
</constructor>
<method name="Copy" cname="gst_webrtc_session_description_copy">
<return-type type="GstWebRTCSessionDescription*" owned="true">
<warning>missing glib:type-name</warning>
</return-type>
<parameters />
</method>
<method name="Free" cname="gst_webrtc_session_description_free">
<return-type type="void" />
<parameters />
</method>
</boxed>
<object name="Global" cname="GstWebRTCGlobal" opaque="true">
<method name="WebrtcSdpTypeToString" cname="gst_webrtc_sdp_type_to_string" shared="true">
<return-type type="const-gchar*" />
<parameters>
<parameter name="type" type="GstWebRTCSDPType" />
</parameters>
</method>
</object>
<object name="Constants" cname="GstWebRTCConstants" opaque="true" />
</namespace>
</api>

20
sources/generated/meson.build
View file

@ -722,6 +722,26 @@ generated_sources = [
'Gst.Rtsp/Gst.RtspSharp.RTSPConnectionAcceptCertificateFuncNative.cs',
'Gst.Audio/AudioStreamAlign.cs',
'Gst.Video/VideoOverlayProperties.cs',
'Gst.WebRTC/WebRTCPeerConnectionState.cs',
'Gst.WebRTC/WebRTCSessionDescription.cs',
'Gst.WebRTC/WebRTCICEGatheringState.cs',
'Gst.WebRTC/WebRTCRTPTransceiverDirection.cs',
'Gst.WebRTC/WebRTCRTPTransceiver.cs',
'Gst.WebRTC/OnNewCandidateHandler.cs',
'Gst.WebRTC/WebRTCICERole.cs',
'Gst.WebRTC/Global.cs',
'Gst.WebRTC/WebRTCICEComponent.cs',
'Gst.WebRTC/WebRTCICEConnectionState.cs',
'Gst.WebRTC/WebRTCDTLSTransport.cs',
'Gst.WebRTC/WebRTCICETransport.cs',
'Gst.WebRTC/WebRTCSDPType.cs',
'Gst.WebRTC/WebRTCRTPSender.cs',
'Gst.WebRTC/WebRTCSignalingState.cs',
'Gst.WebRTC/WebRTCDTLSTransportState.cs',
'Gst.WebRTC/WebRTCDTLSSetup.cs',
'Gst.WebRTC/WebRTCRTPReceiver.cs',
'Gst.WebRTC/WebRTCStatsType.cs',
'Gst.WebRTC/Constants.cs',
]
run_target('update_gstreamer_code',

301
sources/gstreamer-sharp-api.raw
View file

@ -29184,4 +29184,305 @@
<constant value="16" ctype="gint" gtype="gint" name="VIDEO_TILE_Y_TILES_SHIFT"/>
</object>
</namespace>
<namespace name="GstWebRTC" library="gstwebrtc-1.0">
<enum name="WebRTCDTLSSetup" cname="GstWebRTCDTLSSetup" type="enum" gtype="gst_webrtc_dtls_setup_get_type">
<member cname="GST_WEBRTC_DTLS_SETUP_NONE" name="None" value="0"/>
<member cname="GST_WEBRTC_DTLS_SETUP_ACTPASS" name="Actpass" value="1"/>
<member cname="GST_WEBRTC_DTLS_SETUP_ACTIVE" name="Active" value="2"/>
<member cname="GST_WEBRTC_DTLS_SETUP_PASSIVE" name="Passive" value="3"/>
</enum>
<enum name="WebRTCDTLSTransportState" cname="GstWebRTCDTLSTransportState" type="enum" gtype="gst_webrtc_dtls_transport_state_get_type">
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" name="New" value="0"/>
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" name="Closed" value="1"/>
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" name="Failed" value="2"/>
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" name="Connecting" value="3"/>
<member cname="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" name="Connected" value="4"/>
</enum>
<enum name="WebRTCICEComponent" cname="GstWebRTCICEComponent" type="enum" gtype="gst_webrtc_ice_component_get_type">
<member cname="GST_WEBRTC_ICE_COMPONENT_RTP" name="Rtp" value="0"/>
<member cname="GST_WEBRTC_ICE_COMPONENT_RTCP" name="Rtcp" value="1"/>
</enum>
<enum name="WebRTCICEConnectionState" cname="GstWebRTCICEConnectionState" type="enum" gtype="gst_webrtc_ice_connection_state_get_type">
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" name="New" value="0"/>
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" name="Checking" value="1"/>
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" name="Connected" value="2"/>
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" name="Completed" value="3"/>
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" name="Failed" value="4"/>
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="5"/>
<member cname="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" name="Closed" value="6"/>
</enum>
<enum name="WebRTCICEGatheringState" cname="GstWebRTCICEGatheringState" type="enum" gtype="gst_webrtc_ice_gathering_state_get_type">
<member cname="GST_WEBRTC_ICE_GATHERING_STATE_NEW" name="New" value="0"/>
<member cname="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" name="Gathering" value="1"/>
<member cname="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" name="Complete" value="2"/>
</enum>
<enum name="WebRTCICERole" cname="GstWebRTCICERole" type="enum" gtype="gst_webrtc_ice_role_get_type">
<member cname="GST_WEBRTC_ICE_ROLE_CONTROLLED" name="Controlled" value="0"/>
<member cname="GST_WEBRTC_ICE_ROLE_CONTROLLING" name="Controlling" value="1"/>
</enum>
<enum name="WebRTCPeerConnectionState" cname="GstWebRTCPeerConnectionState" type="enum" gtype="gst_webrtc_peer_connection_state_get_type">
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" name="New" value="0"/>
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" name="Connecting" value="1"/>
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" name="Connected" value="2"/>
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" name="Disconnected" value="3"/>
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" name="Failed" value="4"/>
<member cname="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" name="Closed" value="5"/>
</enum>
<enum name="WebRTCRTPTransceiverDirection" cname="GstWebRTCRTPTransceiverDirection" type="enum" gtype="gst_webrtc_rtp_transceiver_direction_get_type">
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" name="None" value="0"/>
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" name="Inactive" value="1"/>
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" name="Sendonly" value="2"/>
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" name="Recvonly" value="3"/>
<member cname="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" name="Sendrecv" value="4"/>
</enum>
<enum name="WebRTCSDPType" cname="GstWebRTCSDPType" type="enum" gtype="gst_webrtc_sdp_type_get_type">
<member cname="GST_WEBRTC_SDP_TYPE_OFFER" name="Offer" value="1"/>
<member cname="GST_WEBRTC_SDP_TYPE_PRANSWER" name="Pranswer" value="2"/>
<member cname="GST_WEBRTC_SDP_TYPE_ANSWER" name="Answer" value="3"/>
<member cname="GST_WEBRTC_SDP_TYPE_ROLLBACK" name="Rollback" value="4"/>
</enum>
<enum name="WebRTCSignalingState" cname="GstWebRTCSignalingState" type="enum" gtype="gst_webrtc_signaling_state_get_type">
<member cname="GST_WEBRTC_SIGNALING_STATE_STABLE" name="Stable" value="0"/>
<member cname="GST_WEBRTC_SIGNALING_STATE_CLOSED" name="Closed" value="1"/>
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" name="HaveLocalOffer" value="2"/>
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" name="HaveRemoteOffer" value="3"/>
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" name="HaveLocalPranswer" value="4"/>
<member cname="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" name="HaveRemotePranswer" value="5"/>
</enum>
<enum name="WebRTCStatsType" cname="GstWebRTCStatsType" type="enum" gtype="gst_webrtc_stats_type_get_type">
<member cname="GST_WEBRTC_STATS_CODEC" name="Codec" value="1"/>
<member cname="GST_WEBRTC_STATS_INBOUND_RTP" name="InboundRtp" value="2"/>
<member cname="GST_WEBRTC_STATS_OUTBOUND_RTP" name="OutboundRtp" value="3"/>
<member cname="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" name="RemoteInboundRtp" value="4"/>
<member cname="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" name="RemoteOutboundRtp" value="5"/>
<member cname="GST_WEBRTC_STATS_CSRC" name="Csrc" value="6"/>
<member cname="GST_WEBRTC_STATS_PEER_CONNECTION" name="PeerConnection" value="7"/>
<member cname="GST_WEBRTC_STATS_DATA_CHANNEL" name="DataChannel" value="8"/>
<member cname="GST_WEBRTC_STATS_STREAM" name="Stream" value="9"/>
<member cname="GST_WEBRTC_STATS_TRANSPORT" name="Transport" value="10"/>
<member cname="GST_WEBRTC_STATS_CANDIDATE_PAIR" name="CandidatePair" value="11"/>
<member cname="GST_WEBRTC_STATS_LOCAL_CANDIDATE" name="LocalCandidate" value="12"/>
<member cname="GST_WEBRTC_STATS_REMOTE_CANDIDATE" name="RemoteCandidate" value="13"/>
<member cname="GST_WEBRTC_STATS_CERTIFICATE" name="Certificate" value="14"/>
</enum>
<object name="WebRTCDTLSTransport" cname="GstWebRTCDTLSTransport" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCDTLSTransportClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</class_struct>
<method name="GetType" cname="gst_webrtc_dtls_transport_get_type" shared="true">
<return-type type="GType"/>
</method>
<constructor cname="gst_webrtc_dtls_transport_new">
<parameters>
<parameter name="session_id" type="guint"/>
<parameter name="rtcp" type="gboolean"/>
</parameters>
</constructor>
<method name="SetTransport" cname="gst_webrtc_dtls_transport_set_transport">
<return-type type="void"/>
<parameters>
<parameter name="ice" type="GstWebRTCICETransport*"/>
</parameters>
</method>
<property name="Certificate" cname="certificate" type="gchar*" readable="true" writeable="true" construct="false" construct-only="false"/>
<property name="Client" cname="client" type="gboolean" readable="true" writeable="true" construct="false" construct-only="false"/>
<property name="RemoteCertificate" cname="remote-certificate" type="gchar*" readable="true" writeable="false" construct="false" construct-only="false"/>
<property name="Rtcp" cname="rtcp" type="gboolean" readable="true" writeable="true" construct="false" construct-only="true"/>
<property name="SessionId" cname="session-id" type="guint" readable="true" writeable="true" construct="false" construct-only="true"/>
<property name="State" cname="state" type="GstWebRTCDTLSTransportState" readable="true" writeable="false" construct="false" construct-only="false"/>
<property name="Transport" cname="transport" type="GstWebRTCICETransport*" readable="true" writeable="false" construct="false" construct-only="false"/>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
<field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="TransportField" type="GstWebRTCICETransport*"/>
<field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCDTLSTransportState"/>
<field cname="is_rtcp" access="public" writeable="false" readable="true" is_callback="false" name="IsRtcp" type="gboolean"/>
<field cname="client" access="public" writeable="false" readable="true" is_callback="false" name="ClientField" type="gboolean"/>
<field cname="session_id" access="public" writeable="false" readable="true" is_callback="false" name="SessionIdField" type="guint"/>
<field cname="dtlssrtpenc" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpenc" type="GstElement*"/>
<field cname="dtlssrtpdec" access="public" writeable="false" readable="true" is_callback="false" name="Dtlssrtpdec" type="GstElement*"/>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</object>
<object name="WebRTCICETransport" cname="GstWebRTCICETransport" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCICETransportClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstBinClass">
<warning>missing glib:type-name</warning>
</field>
<method vm="gather_candidates"/>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</class_struct>
<method name="GetType" cname="gst_webrtc_ice_transport_get_type" shared="true">
<return-type type="GType"/>
</method>
<virtual_method name="GatherCandidates" cname="gather_candidates">
<return-type type="gboolean"/>
<parameters/>
</virtual_method>
<method name="ConnectionStateChange" cname="gst_webrtc_ice_transport_connection_state_change">
<return-type type="void"/>
<parameters>
<parameter name="new_state" type="GstWebRTCICEConnectionState"/>
</parameters>
</method>
<method name="GatheringStateChange" cname="gst_webrtc_ice_transport_gathering_state_change">
<return-type type="void"/>
<parameters>
<parameter name="new_state" type="GstWebRTCICEGatheringState"/>
</parameters>
</method>
<method name="NewCandidate" cname="gst_webrtc_ice_transport_new_candidate">
<return-type type="void"/>
<parameters>
<parameter name="stream_id" type="guint"/>
<parameter name="component" type="GstWebRTCICEComponent"/>
<parameter name="attr" type="const-gchar*"/>
</parameters>
</method>
<method name="SelectedPairChange" cname="gst_webrtc_ice_transport_selected_pair_change">
<return-type type="void"/>
<parameters/>
</method>
<property name="Component" cname="component" type="GstWebRTCICEComponent" readable="true" writeable="true" construct="false" construct-only="true"/>
<property name="GatheringState" cname="gathering-state" type="GstWebRTCICEGatheringState" readable="true" writeable="false" construct="false" construct-only="false"/>
<property name="State" cname="state" type="GstWebRTCICEConnectionState" readable="true" writeable="false" construct="false" construct-only="false"/>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
<field cname="role" access="public" writeable="false" readable="true" is_callback="false" name="Role" type="GstWebRTCICERole"/>
<field cname="component" access="public" writeable="false" readable="true" is_callback="false" name="ComponentField" type="GstWebRTCICEComponent"/>
<field cname="state" access="public" writeable="false" readable="true" is_callback="false" name="StateField" type="GstWebRTCICEConnectionState"/>
<field cname="gathering_state" access="public" writeable="false" readable="true" is_callback="false" name="GatheringStateField" type="GstWebRTCICEGatheringState"/>
<field cname="src" access="public" writeable="false" readable="true" is_callback="false" name="Src" type="GstElement*"/>
<field cname="sink" access="public" writeable="false" readable="true" is_callback="false" name="Sink" type="GstElement*"/>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
<signal name="OnNewCandidate" cname="on-new-candidate" when="last">
<return-type type="void"/>
<parameters>
<parameter name="_object" type="const-gchar*"/>
</parameters>
</signal>
<signal name="OnSelectedCandidatePairChange" cname="on-selected-candidate-pair-change" when="last">
<return-type type="void"/>
<parameters/>
</signal>
</object>
<object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCRTPReceiverClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</class_struct>
<method name="GetType" cname="gst_webrtc_rtp_receiver_get_type" shared="true">
<return-type type="GType"/>
</method>
<constructor cname="gst_webrtc_rtp_receiver_new" disable_void_ctor=""/>
<method name="SetRtcpTransport" cname="gst_webrtc_rtp_receiver_set_rtcp_transport">
<return-type type="void"/>
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*"/>
</parameters>
</method>
<method name="SetTransport" cname="gst_webrtc_rtp_receiver_set_transport">
<return-type type="void"/>
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*"/>
</parameters>
</method>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
<field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*"/>
<field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*"/>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</object>
<object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCRTPSenderClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</class_struct>
<method name="GetType" cname="gst_webrtc_rtp_sender_get_type" shared="true">
<return-type type="GType"/>
</method>
<constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor=""/>
<method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
<return-type type="void"/>
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*"/>
</parameters>
</method>
<method name="SetTransport" cname="gst_webrtc_rtp_sender_set_transport">
<return-type type="void"/>
<parameters>
<parameter name="transport" type="GstWebRTCDTLSTransport*"/>
</parameters>
</method>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
<field cname="transport" access="public" writeable="false" readable="true" is_callback="false" name="Transport" type="GstWebRTCDTLSTransport*"/>
<field cname="rtcp_transport" access="public" writeable="false" readable="true" is_callback="false" name="RtcpTransport" type="GstWebRTCDTLSTransport*"/>
<field cname="send_encodings" access="public" writeable="false" readable="true" is_callback="false" name="SendEncodings" type="GArray*" array="true" null_term_array="true"/>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</object>
<object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
<class_struct cname="GstWebRTCRTPTransceiverClass">
<field cname="parent_class" access="public" writeable="false" readable="true" is_callback="false" name="ParentClass" type="GstObjectClass">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</class_struct>
<method name="GetType" cname="gst_webrtc_rtp_transceiver_get_type" shared="true">
<return-type type="GType"/>
</method>
<property name="Mlineindex" cname="mlineindex" type="guint" readable="true" writeable="true" construct="false" construct-only="true"/>
<property name="Receiver" cname="receiver" type="GstWebRTCRTPReceiver*" readable="true" writeable="true" construct="false" construct-only="true"/>
<property name="Sender" cname="sender" type="GstWebRTCRTPSender*" readable="true" writeable="true" construct="false" construct-only="true"/>
<field cname="parent" access="public" writeable="false" readable="true" is_callback="false" name="Parent" type="GstObject*"/>
<field cname="mline" access="public" writeable="false" readable="true" is_callback="false" name="Mline" type="guint"/>
<field cname="mid" access="public" writeable="false" readable="true" is_callback="false" name="Mid" type="gchar*"/>
<field cname="stopped" access="public" writeable="false" readable="true" is_callback="false" name="Stopped" type="gboolean"/>
<field cname="sender" access="public" writeable="false" readable="true" is_callback="false" name="SenderField" type="GstWebRTCRTPSender*"/>
<field cname="receiver" access="public" writeable="false" readable="true" is_callback="false" name="ReceiverField" type="GstWebRTCRTPReceiver*"/>
<field cname="direction" access="public" writeable="false" readable="true" is_callback="false" name="Direction" type="GstWebRTCRTPTransceiverDirection"/>
<field cname="current_direction" access="public" writeable="false" readable="true" is_callback="false" name="CurrentDirection" type="GstWebRTCRTPTransceiverDirection"/>
<field cname="codec_preferences" access="public" writeable="false" readable="true" is_callback="false" name="CodecPreferences" type="GstCaps*">
<warning>missing glib:type-name</warning>
</field>
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4"/>
</object>
<boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
<method name="GetType" cname="gst_webrtc_session_description_get_type" shared="true">
<return-type type="GType"/>
</method>
<field cname="type" access="public" writeable="true" readable="true" is_callback="false" name="Type" type="GstWebRTCSDPType"/>
<field cname="sdp" access="public" writeable="true" readable="true" is_callback="false" name="Sdp" type="GstSDPMessage*">
<warning>missing glib:type-name</warning>
</field>
<constructor cname="gst_webrtc_session_description_new">
<parameters>
<parameter name="type" type="GstWebRTCSDPType"/>
<parameter name="sdp" type="GstSDPMessage*">
<warning>missing glib:type-name</warning>
</parameter>
</parameters>
</constructor>
<method name="Copy" cname="gst_webrtc_session_description_copy">
<return-type type="GstWebRTCSessionDescription*" owned="true">
<warning>missing glib:type-name</warning>
</return-type>
<parameters/>
</method>
<method name="Free" cname="gst_webrtc_session_description_free">
<return-type type="void"/>
<parameters/>
</method>
</boxed>
<object name="Global" cname="GstWebRTCGlobal" opaque="true">
<method name="WebrtcSdpTypeToString" cname="gst_webrtc_sdp_type_to_string" shared="true">
<return-type type="const-gchar*"/>
<parameters>
<parameter name="type" type="GstWebRTCSDPType"/>
</parameters>
</method>
</object>
<object name="Constants" cname="GstWebRTCConstants" opaque="true"/>
</namespace>
</api>

2
sources/gstreamer-sharp.dll.config
View file

@ -12,6 +12,7 @@
<dllmap dll="libgstrtp-1.0-0.dll" target="libgstrtp-1.0.so.0" os="linux"/>
<dllmap dll="libgstrtsp-1.0-0.dll" target="libgstrtsp-1.0.so.0" os="linux"/>
<dllmap dll="libgstsdp-1.0-0.dll" target="libgstsdp-1.0.so.0" os="linux"/>
<dllmap dll="libgstwebrtc-1.0-0.dll" target="libgstwebrtc-1.0.so.0" os="linux"/>
<dllmap dll="libgstcontroller-1.0-0.dll" target="libgstcontroller-1.0.so.0" os="linux"/>
<dllmap dll="libglib-2.0-0.dll" target="libglib-2.0.so.0" os="linux"/>
<dllmap dll="libgobject-2.0-0.dll" target="libgobject-2.0.so.0" os="linux"/>
@ -29,6 +30,7 @@
<dllmap dll="libgstrtp-1.0-0.dll" target="libgstrtp-1.0.dylib" os="osx"/>
<dllmap dll="libgstrtsp-1.0-0.dll" target="libgstrtsp-1.0.dylib" os="osx"/>
<dllmap dll="libgstsdp-1.0-0.dll" target="libgstsdp-1.0.dylib" os="osx"/>
<dllmap dll="libgstwebrtc-1.0-0.dll" target="libgstwebrtc-1.0.dylib" os="osx"/>
<dllmap dll="libgstcontroller-1.0-0.dll" target="libgstcontroller-1.0.dylib" os="osx"/>
<dllmap dll="libglib-2.0-0.dll" target="libglib-2.0.dylib" os="osx"/>
<dllmap dll="libgobject-2.0-0.dll" target="libgobject-2.0.dylib" os="osx"/>

2
sources/gstreamer-sharp.metadata
View file

@ -243,6 +243,7 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
<attr path="/api/namespace[@name='GstRtp']" name="name">Gst.Rtp</attr>
<attr path="/api/namespace[@name='GstRtsp']" name="name">Gst.Rtsp</attr>
<attr path="/api/namespace[@name='GstSdp']" name="name">Gst.Sdp</attr>
<attr path="/api/namespace[@name='GstWebRTC']" name="name">Gst.WebRTC</attr>
<attr path="/api/namespace" name="library">libgstreamer-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Base']" name="library">libgstbase-1.0-0.dll</attr>
@ -258,6 +259,7 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
<attr path="/api/namespace[@name='Gst.Rtp']" name="library">libgstrtp-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Rtsp']" name="library">libgstrtsp-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.Sdp']" name="library">libgstsdp-1.0-0.dll</attr>
<attr path="/api/namespace[@name='Gst.WebRTC']" name="library">libgstwebrtc-1.0-0.dll</attr>
<!-- DoubleRange and Fraction are in Value.cs -->
<attr path="//struct[@name='DoubleRange' or @name='Fraction' or @name='IntRange' or @name='FractionRange']" name="hidden">true</attr>

4
sources/meson.build
View file

@ -1,7 +1,7 @@
raw_api_fname = join_paths(meson.current_source_dir(), meson.project_name() + '-api.raw')
metadata = files(meson.project_name() + '.metadata')
abi_includes = 'glib.h,gst/gst.h,gst/video/video.h,gst/audio/audio.h,gst/rtsp/rtsp.h,gst/app/app.h,gst/audio/audio.h,gst/base/base.h,gst/controller/controller.h,gst/fft/fft.h,gst/net/net.h,gst/pbutils/gstaudiovisualizer.h,gst/pbutils/pbutils.h,gst/rtp/rtp.h,gst/rtsp/rtsp.h,gst/sdp/sdp.h,gst/tag/tag.h,gst/video/video.h,gst/video/gstvideoaffinetransformationmeta.h,gst/net/gstnetcontrolmessagemeta.h'
abi_includes = 'glib.h,gst/gst.h,gst/video/video.h,gst/audio/audio.h,gst/rtsp/rtsp.h,gst/app/app.h,gst/audio/audio.h,gst/base/base.h,gst/controller/controller.h,gst/fft/fft.h,gst/net/net.h,gst/pbutils/gstaudiovisualizer.h,gst/pbutils/pbutils.h,gst/rtp/rtp.h,gst/rtsp/rtsp.h,gst/sdp/sdp.h,gst/tag/tag.h,gst/video/video.h,gst/video/gstvideoaffinetransformationmeta.h,gst/net/gstnetcontrolmessagemeta.h,gst/webrtc/webrtc.h'
sources = [
'custom/Adapter.cs',
@ -43,7 +43,7 @@ gst_sharp_dep = declare_dependency(dependencies: [glib_sharp_dep, gio_sharp_dep]
if add_languages('c', required: false) and csc.get_id() == 'mono'
c_abi_exe = executable('gst_sharp_c_abi', c_abi,
cs_args: ['-nowarn:169', '-nowarn:108', '-nowarn:114', '-unsafe'],
c_args: ['-DGST_USE_UNSTABLE_API'],
dependencies: [gst_deps])
cs_abi_exe = executable('gst_sharp_cs_abi', cs_abi,