mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-31 12:38:48 +00:00
Update bindings for new WebRTC symbols
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer-sharp/-/merge_requests/25>
This commit is contained in:
Olivier Crête
parent
2cdb1e714d
commit
9fdd11cda3
9 changed files with 293 additions and 22 deletions
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@ -1521,6 +1521,39 @@ for more information.</doc>
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glib:nick="relay">
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</member>
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</enumeration>
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<enumeration name="WebRTCKind"
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version="1.20"
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glib:type-name="GstWebRTCKind"
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glib:get-type="gst_webrtc_kind_get_type"
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c:type="GstWebRTCKind">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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line="376">https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind</doc>
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<member name="unknown"
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value="0"
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c:identifier="GST_WEBRTC_KIND_UNKNOWN"
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glib:nick="unknown">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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line="378">Kind has not yet been set</doc>
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</member>
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<member name="audio"
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value="1"
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c:identifier="GST_WEBRTC_KIND_AUDIO"
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glib:nick="audio">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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line="379">Kind is audio</doc>
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</member>
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<member name="video"
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value="2"
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c:identifier="GST_WEBRTC_KIND_VIDEO"
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glib:nick="video">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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line="380">Kind is audio</doc>
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</member>
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</enumeration>
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<enumeration name="WebRTCPeerConnectionState"
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glib:type-name="GstWebRTCPeerConnectionState"
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glib:get-type="gst_webrtc_peer_connection_state_get_type"
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@ -1613,14 +1646,20 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<class name="WebRTCRTPReceiver"
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c:symbol-prefix="webrtc_rtp_receiver"
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c:type="GstWebRTCRTPReceiver"
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version="1.16"
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parent="Gst.Object"
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glib:type-name="GstWebRTCRTPReceiver"
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glib:get-type="gst_webrtc_rtp_receiver_get_type"
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glib:type-struct="WebRTCRTPReceiverClass">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="57"/>
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpreceiver.h"
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line="38">An object to track the receiving aspect of the stream
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Mostly matches the WebRTC RTCRtpReceiver interface.</doc>
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="65"/>
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<constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h"
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line="60"/>
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line="68"/>
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<return-value transfer-ownership="none">
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<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
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</return-value>
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@ -1628,7 +1667,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<method name="set_rtcp_transport"
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c:identifier="gst_webrtc_rtp_receiver_set_rtcp_transport">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h"
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line="65"/>
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line="73"/>
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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@ -1644,7 +1683,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<method name="set_transport"
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c:identifier="gst_webrtc_rtp_receiver_set_transport">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h"
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line="62"/>
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line="70"/>
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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@ -1661,9 +1700,15 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="transport">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpreceiver.h"
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line="40">The transport for RTP packets</doc>
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</field>
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<field name="rtcp_transport">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpreceiver.h"
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line="41">The transport for RTCP packets without rtcp-mux</doc>
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</field>
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<field name="_padding">
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@ -1675,7 +1720,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<record name="WebRTCRTPReceiverClass"
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c:type="GstWebRTCRTPReceiverClass"
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glib:is-gtype-struct-for="WebRTCRTPReceiver">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="57"/>
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="65"/>
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<field name="parent_class">
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<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
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</field>
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@ -1688,20 +1733,53 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<class name="WebRTCRTPSender"
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c:symbol-prefix="webrtc_rtp_sender"
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c:type="GstWebRTCRTPSender"
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version="1.16"
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parent="Gst.Object"
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glib:type-name="GstWebRTCRTPSender"
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glib:get-type="gst_webrtc_rtp_sender_get_type"
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glib:type-struct="WebRTCRTPSenderClass">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="59"/>
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.h"
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line="38">An object to track the sending aspect of the stream
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Mostly matches the WebRTC RTCRtpSender interface.</doc>
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="70"/>
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<constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="62"/>
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="73"/>
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<return-value transfer-ownership="none">
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<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
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</return-value>
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</constructor>
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<method name="set_priority"
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c:identifier="gst_webrtc_rtp_sender_set_priority"
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version="1.20">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.c"
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line="85">Sets the content of the IPv4 Type of Service (ToS), also known as DSCP
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(Differentiated Services Code Point).
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This also sets the Traffic Class field of IPv6.</doc>
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="82"/>
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="sender" transfer-ownership="none">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.c"
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line="87">a #GstWebRTCRTPSender</doc>
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<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
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</instance-parameter>
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<parameter name="priority" transfer-ownership="none">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.c"
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line="88">The priority of this sender</doc>
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<type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
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</parameter>
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</parameters>
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</method>
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<method name="set_rtcp_transport"
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c:identifier="gst_webrtc_rtp_sender_set_rtcp_transport">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="68"/>
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="79"/>
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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@ -1716,7 +1794,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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</method>
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<method name="set_transport"
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c:identifier="gst_webrtc_rtp_sender_set_transport">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="65"/>
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="76"/>
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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@ -1729,20 +1807,44 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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</parameter>
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</parameters>
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</method>
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<property name="priority"
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version="1.20"
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writable="1"
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transfer-ownership="none">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.c"
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line="166">The priority from which to set the DSCP field on packets</doc>
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<type name="WebRTCPriorityType"/>
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</property>
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<field name="parent">
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="transport">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.h"
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line="40">The transport for RTP packets</doc>
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</field>
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<field name="rtcp_transport">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.h"
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line="41">The transport for RTCP packets without rtcp-mux</doc>
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</field>
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<field name="send_encodings">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.h"
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line="42">Unused</doc>
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<array name="GLib.Array" c:type="GArray*">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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<field name="priority">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtpsender.h"
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line="43">The priority of the stream (Since: 1.20)</doc>
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<type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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@ -1752,7 +1854,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<record name="WebRTCRTPSenderClass"
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c:type="GstWebRTCRTPSenderClass"
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glib:is-gtype-struct-for="WebRTCRTPSender">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="59"/>
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="70"/>
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<field name="parent_class">
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<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
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</field>
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@ -1765,13 +1867,17 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<class name="WebRTCRTPTransceiver"
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c:symbol-prefix="webrtc_rtp_transceiver"
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c:type="GstWebRTCRTPTransceiver"
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version="1.16"
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parent="Gst.Object"
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abstract="1"
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glib:type-name="GstWebRTCRTPTransceiver"
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glib:get-type="gst_webrtc_rtp_transceiver_get_type"
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glib:type-struct="WebRTCRTPTransceiverClass">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="39">Mostly matches the WebRTC RTCRtpTransceiver interface.</doc>
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="66"/>
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line="96"/>
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<property name="direction"
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version="1.18"
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writable="1"
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@ -1803,31 +1909,70 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="mline">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="41">the mline number this transceiver corresponds to</doc>
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<type name="guint" c:type="guint"/>
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</field>
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<field name="mid">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="42">The media ID of the m-line associated with this
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transceiver. This association is established, when possible,
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whenever either a local or remote description is applied. This
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field is NULL if neither a local or remote description has been
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applied, or if its associated m-line is rejected by either a remote
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offer or any answer.</doc>
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<type name="utf8" c:type="gchar*"/>
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</field>
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<field name="stopped">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="48">Indicates whether or not sending and receiving using the paired
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#GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
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either due to SDP offer/answer</doc>
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<type name="gboolean" c:type="gboolean"/>
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</field>
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<field name="sender">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="51">The #GstWebRTCRTPSender object responsible sending data to the
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remote peer</doc>
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<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
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</field>
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<field name="receiver">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="53">The #GstWebRTCRTPReceiver object responsible for receiver data from
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the remote peer.</doc>
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<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
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</field>
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<field name="direction">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="55">The transceiver's desired direction.</doc>
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<type name="WebRTCRTPTransceiverDirection"
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c:type="GstWebRTCRTPTransceiverDirection"/>
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</field>
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<field name="current_direction">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="56">The transceiver's current direction (read-only)</doc>
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<type name="WebRTCRTPTransceiverDirection"
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c:type="GstWebRTCRTPTransceiverDirection"/>
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</field>
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<field name="codec_preferences">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="57">A caps representing the codec preferences (read-only)</doc>
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<type name="Gst.Caps" c:type="GstCaps*"/>
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</field>
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<field name="kind">
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<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="58">Type of media (Since: 1.20)</doc>
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<type name="WebRTCKind" c:type="GstWebRTCKind"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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@ -1838,7 +1983,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
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c:type="GstWebRTCRTPTransceiverClass"
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glib:is-gtype-struct-for="WebRTCRTPTransceiver">
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="66"/>
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line="96"/>
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<field name="parent_class">
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<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
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</field>
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29
sources/generated/Gst.WebRTC/WebRTCKind.cs
Normal file
29
sources/generated/Gst.WebRTC/WebRTCKind.cs
Normal file
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@ -0,0 +1,29 @@
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// This file was generated by the Gtk# code generator.
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// Any changes made will be lost if regenerated.
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namespace Gst.WebRTC {
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using System;
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using System.Runtime.InteropServices;
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#region Autogenerated code
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[GLib.GType (typeof (Gst.WebRTC.WebRTCKindGType))]
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public enum WebRTCKind {
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Unknown = 0,
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Audio = 1,
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Video = 2,
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}
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internal class WebRTCKindGType {
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[DllImport ("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
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static extern IntPtr gst_webrtc_kind_get_type ();
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public static GLib.GType GType {
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get {
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return new GLib.GType (gst_webrtc_kind_get_type ());
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}
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||||
}
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||||
}
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#endregion
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}
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@ -25,6 +25,22 @@ namespace Gst.WebRTC {
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Raw = gst_webrtc_rtp_sender_new();
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}
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[DllImport("gstwebrtc-1.0-0.dll", CallingConvention = CallingConvention.Cdecl)]
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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.
|
||||
|
||||
|
|
|
|||
|
|
@ -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.
|
||||
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -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") + "\"");
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -31188,6 +31188,11 @@
|
|||
<member cname="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL" name="All" value="0" />
|
||||
<member cname="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY" name="Relay" value="1" />
|
||||
</enum>
|
||||
<enum name="WebRTCKind" cname="GstWebRTCKind" type="enum" gtype="gst_webrtc_kind_get_type" version="1.20">
|
||||
<member cname="GST_WEBRTC_KIND_UNKNOWN" name="Unknown" value="0" />
|
||||
<member cname="GST_WEBRTC_KIND_AUDIO" name="Audio" value="1" />
|
||||
<member cname="GST_WEBRTC_KIND_VIDEO" name="Video" value="2" />
|
||||
</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" />
|
||||
|
|
@ -31497,7 +31502,7 @@
|
|||
<parameters />
|
||||
</signal>
|
||||
</object>
|
||||
<object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
|
||||
<object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject" version="1.16">
|
||||
<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>
|
||||
|
|
@ -31525,7 +31530,7 @@
|
|||
<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" padding="true" />
|
||||
</object>
|
||||
<object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject">
|
||||
<object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject" version="1.16">
|
||||
<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>
|
||||
|
|
@ -31536,6 +31541,12 @@
|
|||
<return-type type="GType" />
|
||||
</method>
|
||||
<constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor="" />
|
||||
<method name="SetPriority" cname="gst_webrtc_rtp_sender_set_priority" version="1.20">
|
||||
<return-type type="void" />
|
||||
<parameters>
|
||||
<parameter name="priority" type="GstWebRTCPriorityType" />
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
|
||||
<return-type type="void" />
|
||||
<parameters>
|
||||
|
|
@ -31548,13 +31559,15 @@
|
|||
<parameter name="transport" type="GstWebRTCDTLSTransport*" />
|
||||
</parameters>
|
||||
</method>
|
||||
<property name="Priority" cname="priority" type="GstWebRTCPriorityType" readable="true" writeable="true" construct="false" construct-only="false" version="1.20" />
|
||||
<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="priority" access="public" writeable="false" readable="true" is_callback="false" name="PriorityField" type="GstWebRTCPriorityType" />
|
||||
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" padding="true" />
|
||||
</object>
|
||||
<object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject">
|
||||
<object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject" version="1.16">
|
||||
<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>
|
||||
|
|
@ -31579,6 +31592,7 @@
|
|||
<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="kind" access="public" writeable="false" readable="true" is_callback="false" name="Kind" type="GstWebRTCKind" />
|
||||
<field cname="_padding" access="public" writeable="false" readable="true" is_callback="false" name="_Padding" type="gpointer" array="true" array_len="4" padding="true" />
|
||||
</object>
|
||||
<boxed name="WebRTCSessionDescription" cname="GstWebRTCSessionDescription" opaque="false" hidden="false">
|
||||
|
|
|
|||
|
|
@ -432,6 +432,7 @@ generated_sources = [
|
|||
'Gst.WebRTC/WebRTCICERole.cs',
|
||||
'Gst.WebRTC/WebRTCICETransport.cs',
|
||||
'Gst.WebRTC/WebRTCICETransportPolicy.cs',
|
||||
'Gst.WebRTC/WebRTCKind.cs',
|
||||
'Gst.WebRTC/WebRTCPeerConnectionState.cs',
|
||||
'Gst.WebRTC/WebRTCPriorityType.cs',
|
||||
'Gst.WebRTC/WebRTCRTPReceiver.cs',
|
||||
|
|
|
|||
|
|
@ -31577,6 +31577,11 @@
|
|||
<member cname="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL" name="All" value="0"/>
|
||||
<member cname="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY" name="Relay" value="1"/>
|
||||
</enum>
|
||||
<enum name="WebRTCKind" cname="GstWebRTCKind" type="enum" gtype="gst_webrtc_kind_get_type" version="1.20">
|
||||
<member cname="GST_WEBRTC_KIND_UNKNOWN" name="Unknown" value="0"/>
|
||||
<member cname="GST_WEBRTC_KIND_AUDIO" name="Audio" value="1"/>
|
||||
<member cname="GST_WEBRTC_KIND_VIDEO" name="Video" value="2"/>
|
||||
</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"/>
|
||||
|
|
@ -31886,7 +31891,7 @@
|
|||
<parameters/>
|
||||
</signal>
|
||||
</object>
|
||||
<object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject">
|
||||
<object name="WebRTCRTPReceiver" cname="GstWebRTCRTPReceiver" opaque="false" hidden="false" parent="GstObject" version="1.16">
|
||||
<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>
|
||||
|
|
@ -31914,7 +31919,7 @@
|
|||
<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">
|
||||
<object name="WebRTCRTPSender" cname="GstWebRTCRTPSender" opaque="false" hidden="false" parent="GstObject" version="1.16">
|
||||
<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>
|
||||
|
|
@ -31925,6 +31930,12 @@
|
|||
<return-type type="GType"/>
|
||||
</method>
|
||||
<constructor cname="gst_webrtc_rtp_sender_new" disable_void_ctor=""/>
|
||||
<method name="SetPriority" cname="gst_webrtc_rtp_sender_set_priority" version="1.20">
|
||||
<return-type type="void"/>
|
||||
<parameters>
|
||||
<parameter name="priority" type="GstWebRTCPriorityType"/>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="SetRtcpTransport" cname="gst_webrtc_rtp_sender_set_rtcp_transport">
|
||||
<return-type type="void"/>
|
||||
<parameters>
|
||||
|
|
@ -31937,13 +31948,15 @@
|
|||
<parameter name="transport" type="GstWebRTCDTLSTransport*"/>
|
||||
</parameters>
|
||||
</method>
|
||||
<property name="Priority" cname="priority" type="GstWebRTCPriorityType" readable="true" writeable="true" construct="false" construct-only="false" version="1.20"/>
|
||||
<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="priority" access="public" writeable="false" readable="true" is_callback="false" name="PriorityField" type="GstWebRTCPriorityType"/>
|
||||
<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">
|
||||
<object name="WebRTCRTPTransceiver" cname="GstWebRTCRTPTransceiver" defaultconstructoraccess="protected" opaque="false" hidden="false" parent="GstObject" version="1.16">
|
||||
<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>
|
||||
|
|
@ -31968,6 +31981,7 @@
|
|||
<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="kind" access="public" writeable="false" readable="true" is_callback="false" name="Kind" type="GstWebRTCKind"/>
|
||||
<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">
|
||||
|
|
|
|||
Loading…
Reference in a new issue