diff --git a/girs/GstWebRTC-1.0.gir b/girs/GstWebRTC-1.0.gir
index e75778b623..dbf87e1848 100644
--- a/girs/GstWebRTC-1.0.gir
+++ b/girs/GstWebRTC-1.0.gir
@@ -1521,6 +1521,39 @@ for more information.
glib:nick="relay">
+
+ https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
+
+ Kind has not yet been set
+
+
+ Kind is audio
+
+
+ Kind is audio
+
+
-
+ An object to track the receiving aspect of the stream
+
+Mostly matches the WebRTC RTCRtpReceiver interface.
+
+ line="68"/>
@@ -1628,7 +1667,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
+ line="73"/>
@@ -1644,7 +1683,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
+ line="70"/>
@@ -1661,9 +1700,15 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
+ The transport for RTP packets
+ The transport for RTCP packets without rtcp-mux
@@ -1675,7 +1720,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
-
+
@@ -1688,20 +1733,53 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
-
+ An object to track the sending aspect of the stream
+
+Mostly matches the WebRTC RTCRtpSender interface.
+
-
+
+
+ Sets the content of the IPv4 Type of Service (ToS), also known as DSCP
+(Differentiated Services Code Point).
+This also sets the Traffic Class field of IPv6.
+
+
+
+
+
+
+ a #GstWebRTCRTPSender
+
+
+
+ The priority of this sender
+
+
+
+
-
+
@@ -1716,7 +1794,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
-
+
@@ -1729,20 +1807,44 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
+
+ The priority from which to set the DSCP field on packets
+
+
+ The transport for RTP packets
+ The transport for RTCP packets without rtcp-mux
+ Unused
+
+ The priority of the stream (Since: 1.20)
+
+
@@ -1752,7 +1854,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
-
+
@@ -1765,13 +1867,17 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
+ Mostly matches the WebRTC RTCRtpTransceiver interface.
+ line="96"/>
+ the mline number this transceiver corresponds to
+ The media ID of the m-line associated with this
+transceiver. This association is established, when possible,
+whenever either a local or remote description is applied. This
+field is NULL if neither a local or remote description has been
+applied, or if its associated m-line is rejected by either a remote
+offer or any answer.
+ Indicates whether or not sending and receiving using the paired
+#GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
+either due to SDP offer/answer
+ The #GstWebRTCRTPSender object responsible sending data to the
+remote peer
+ The #GstWebRTCRTPReceiver object responsible for receiver data from
+the remote peer.
+ The transceiver's desired direction.
+ The transceiver's current direction (read-only)
+ A caps representing the codec preferences (read-only)
+
+ Type of media (Since: 1.20)
+
+
@@ -1838,7 +1983,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
c:type="GstWebRTCRTPTransceiverClass"
glib:is-gtype-struct-for="WebRTCRTPTransceiver">
+ line="96"/>
diff --git a/sources/generated/Gst.WebRTC/WebRTCKind.cs b/sources/generated/Gst.WebRTC/WebRTCKind.cs
new file mode 100644
index 0000000000..e5041a932f
--- /dev/null
+++ b/sources/generated/Gst.WebRTC/WebRTCKind.cs
@@ -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.WebRTCKindGType))]
+ public enum WebRTCKind {
+
+ Unknown = 0,
+ Audio = 1,
+ Video = 2,
+ }
+
+ internal class WebRTCKindGType {
+ [DllImport ("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern IntPtr gst_webrtc_kind_get_type ();
+
+ public static GLib.GType GType {
+ get {
+ return new GLib.GType (gst_webrtc_kind_get_type ());
+ }
+ }
+ }
+#endregion
+}
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs b/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
index d96de34a3f..3a89c19f80 100644
--- a/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPSender.cs
@@ -25,6 +25,22 @@ namespace Gst.WebRTC {
Raw = gst_webrtc_rtp_sender_new();
}
+ [DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
+ static extern void gst_webrtc_rtp_sender_set_priority(IntPtr raw, int priority);
+
+ [GLib.Property ("priority")]
+ public Gst.WebRTC.WebRTCPriorityType Priority {
+ get {
+ GLib.Value val = GetProperty ("priority");
+ Gst.WebRTC.WebRTCPriorityType ret = (Gst.WebRTC.WebRTCPriorityType) (Enum) val;
+ val.Dispose ();
+ return ret;
+ }
+ set {
+ gst_webrtc_rtp_sender_set_priority(Handle, (int) value);
+ }
+ }
+
[DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
static extern void gst_webrtc_rtp_sender_set_transport(IntPtr raw, IntPtr transport);
@@ -55,6 +71,15 @@ namespace Gst.WebRTC {
}
}
+ public Gst.WebRTC.WebRTCPriorityType PriorityField {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("priority"));
+ return (Gst.WebRTC.WebRTCPriorityType) (*raw_ptr);
+ }
+ }
+ }
+
// Internal representation of the wrapped structure ABI.
static GLib.AbiStruct _class_abi = null;
@@ -122,14 +147,22 @@ namespace Gst.WebRTC {
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // send_encodings
, "rtcp_transport"
- , "_padding"
+ , "priority"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
+ new GLib.AbiField("priority"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCPriorityType))) // priority
+ , "send_encodings"
+ , "_padding"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCRTPSender_priorityAlign), "priority")
+ , 0
+ ),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
- , "send_encodings"
+ , "priority"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
@@ -140,6 +173,13 @@ namespace Gst.WebRTC {
}
}
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCRTPSender_priorityAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCPriorityType priority;
+ }
+
// End of the ABI representation.
diff --git a/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs b/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs
index 29779e5f94..0d9478293e 100644
--- a/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs
+++ b/sources/generated/Gst.WebRTC/WebRTCRTPTransceiver.cs
@@ -135,6 +135,15 @@ namespace Gst.WebRTC {
}
}
+ public Gst.WebRTC.WebRTCKind Kind {
+ get {
+ unsafe {
+ int* raw_ptr = (int*)(((byte*)Handle) + abi_info.GetFieldOffset("kind"));
+ return (Gst.WebRTC.WebRTCKind) (*raw_ptr);
+ }
+ }
+ }
+
// Internal representation of the wrapped structure ABI.
static GLib.AbiStruct _class_abi = null;
@@ -242,14 +251,22 @@ namespace Gst.WebRTC {
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) // codec_preferences
, "current_direction"
- , "_padding"
+ , "kind"
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
),
+ new GLib.AbiField("kind"
+ , -1
+ , (uint) Marshal.SizeOf(System.Enum.GetUnderlyingType(typeof(Gst.WebRTC.WebRTCKind))) // kind
+ , "codec_preferences"
+ , "_padding"
+ , (long) Marshal.OffsetOf(typeof(GstWebRTCRTPTransceiver_kindAlign), "kind")
+ , 0
+ ),
new GLib.AbiField("_padding"
, -1
, (uint) Marshal.SizeOf(typeof(IntPtr)) * 4 // _padding
- , "codec_preferences"
+ , "kind"
, null
, (uint) Marshal.SizeOf(typeof(IntPtr))
, 0
@@ -288,6 +305,13 @@ namespace Gst.WebRTC {
private Gst.WebRTC.WebRTCRTPTransceiverDirection current_direction;
}
+ [StructLayout(LayoutKind.Sequential)]
+ public struct GstWebRTCRTPTransceiver_kindAlign
+ {
+ sbyte f1;
+ private Gst.WebRTC.WebRTCKind kind;
+ }
+
// End of the ABI representation.
diff --git a/sources/generated/gstreamer-sharp-abi.c b/sources/generated/gstreamer-sharp-abi.c
index edf0133395..a02758e799 100644
--- a/sources/generated/gstreamer-sharp-abi.c
+++ b/sources/generated/gstreamer-sharp-abi.c
@@ -1021,6 +1021,7 @@ int main (int argc, char *argv[]) {
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.priority\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPSender, priority));
g_print("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) sizeof(GstWebRTCRTPTransceiverClass));
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));
@@ -1031,5 +1032,6 @@ int main (int argc, char *argv[]) {
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.kind\": \"%" G_GUINT64_FORMAT "\"\n", (guint64) G_STRUCT_OFFSET(GstWebRTCRTPTransceiver, kind));
return 0;
}
diff --git a/sources/generated/gstreamer-sharp-abi.cs b/sources/generated/gstreamer-sharp-abi.cs
index f61bc3f751..43e3a3bcc2 100644
--- a/sources/generated/gstreamer-sharp-abi.cs
+++ b/sources/generated/gstreamer-sharp-abi.cs
@@ -1015,6 +1015,7 @@ namespace AbiTester {
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.priority\": \"" + Gst.WebRTC.WebRTCRTPSender.abi_info.GetFieldOffset("priority") + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiverClass)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.class_abi.Size + "\"");
Console.WriteLine("\"sizeof(GstWebRTCRTPTransceiver)\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.Size + "\"");
Console.WriteLine("\"GstWebRTCRTPTransceiver.mline\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("mline") + "\"");
@@ -1025,6 +1026,7 @@ namespace AbiTester {
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.kind\": \"" + Gst.WebRTC.WebRTCRTPTransceiver.abi_info.GetFieldOffset("kind") + "\"");
}
}
}
diff --git a/sources/generated/gstreamer-sharp-api.xml b/sources/generated/gstreamer-sharp-api.xml
index f5d91990b1..31c797de73 100644
--- a/sources/generated/gstreamer-sharp-api.xml
+++ b/sources/generated/gstreamer-sharp-api.xml
@@ -31188,6 +31188,11 @@
+
+
+
+
+
@@ -31497,7 +31502,7 @@
-
+
+
+
+
+
+
@@ -31548,13 +31559,15 @@
+
+
-
-
+
+
+
+
+
+
@@ -31937,13 +31948,15 @@
+
+
-