mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-02 12:45:51 +00:00
1743 lines
70 KiB
XML
1743 lines
70 KiB
XML
<?xml version="1.0"?>
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<!-- This file was automatically generated from C sources - DO NOT EDIT!
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To affect the contents of this file, edit the original C definitions,
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and/or use gtk-doc annotations. -->
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<repository version="1.2"
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xmlns="http://www.gtk.org/introspection/core/1.0"
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xmlns:c="http://www.gtk.org/introspection/c/1.0"
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xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
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<include name="Gst" version="1.0"/>
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<include name="GstSdp" version="1.0"/>
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<package name="gstreamer-webrtc-1.0"/>
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<c:include name="gst/webrtc/webrtc.h"/>
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<namespace name="GstWebRTC"
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version="1.0"
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shared-library="libgstwebrtc-1.0.so.0"
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c:identifier-prefixes="Gst"
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c:symbol-prefixes="gst">
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<function-macro name="IS_WEBRTC_DATA_CHANNEL"
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c:identifier="GST_IS_WEBRTC_DATA_CHANNEL"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/datachannel.h" line="34"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_DATA_CHANNEL_CLASS"
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c:identifier="GST_IS_WEBRTC_DATA_CHANNEL_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/datachannel.h" line="36"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_DTLS_TRANSPORT"
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c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
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line="33"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_DTLS_TRANSPORT_CLASS"
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c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
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line="35"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_ICE_TRANSPORT"
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c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/icetransport.h"
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line="32"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_ICE_TRANSPORT_CLASS"
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c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/icetransport.h"
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line="34"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_RECEIVER"
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c:identifier="GST_IS_WEBRTC_RTP_RECEIVER"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="33"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_RECEIVER_CLASS"
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c:identifier="GST_IS_WEBRTC_RTP_RECEIVER_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="35"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_SENDER"
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c:identifier="GST_IS_WEBRTC_RTP_SENDER"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="33"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_SENDER_CLASS"
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c:identifier="GST_IS_WEBRTC_RTP_SENDER_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="35"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_TRANSCEIVER"
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c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="32"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
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c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="34"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DATA_CHANNEL"
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c:identifier="GST_WEBRTC_DATA_CHANNEL"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/datachannel.h" line="33"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DATA_CHANNEL_CLASS"
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c:identifier="GST_WEBRTC_DATA_CHANNEL_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/datachannel.h" line="35"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DATA_CHANNEL_GET_CLASS"
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c:identifier="GST_WEBRTC_DATA_CHANNEL_GET_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/datachannel.h" line="37"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DTLS_TRANSPORT"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
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line="32"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DTLS_TRANSPORT_CLASS"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
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line="34"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DTLS_TRANSPORT_GET_CLASS"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/dtlstransport.h"
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line="36"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_ICE_TRANSPORT"
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c:identifier="GST_WEBRTC_ICE_TRANSPORT"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/icetransport.h"
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line="31"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_ICE_TRANSPORT_CLASS"
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c:identifier="GST_WEBRTC_ICE_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/icetransport.h"
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line="33"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_ICE_TRANSPORT_GET_CLASS"
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c:identifier="GST_WEBRTC_ICE_TRANSPORT_GET_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/icetransport.h"
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line="35"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_RECEIVER"
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c:identifier="GST_WEBRTC_RTP_RECEIVER"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="32"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_RECEIVER_CLASS"
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c:identifier="GST_WEBRTC_RTP_RECEIVER_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="34"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_RECEIVER_GET_CLASS"
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c:identifier="GST_WEBRTC_RTP_RECEIVER_GET_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="36"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_SENDER"
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c:identifier="GST_WEBRTC_RTP_SENDER"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="32"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_SENDER_CLASS"
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c:identifier="GST_WEBRTC_RTP_SENDER_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="34"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_SENDER_GET_CLASS"
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c:identifier="GST_WEBRTC_RTP_SENDER_GET_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="36"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_TRANSCEIVER"
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c:identifier="GST_WEBRTC_RTP_TRANSCEIVER"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="31"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_TRANSCEIVER_CLASS"
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c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="33"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
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c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
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introspectable="0">
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<source-position filename="gst-libs/gst/webrtc/rtptransceiver.h"
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line="35"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<enumeration name="WebRTCBundlePolicy"
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version="1.16"
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glib:type-name="GstWebRTCBundlePolicy"
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glib:get-type="gst_webrtc_bundle_policy_get_type"
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c:type="GstWebRTCBundlePolicy">
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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="340">See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
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for more information.</doc>
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<member name="none"
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value="0"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE"
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glib:nick="none">
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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="342">none</doc>
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</member>
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<member name="balanced"
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value="1"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED"
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glib:nick="balanced">
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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="343">balanced</doc>
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</member>
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<member name="max_compat"
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value="2"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT"
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glib:nick="max-compat">
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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="344">max-compat</doc>
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</member>
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<member name="max_bundle"
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value="3"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE"
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glib:nick="max-bundle">
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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="345">max-bundle</doc>
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</member>
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</enumeration>
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<enumeration name="WebRTCDTLSSetup"
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glib:type-name="GstWebRTCDTLSSetup"
|
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glib:get-type="gst_webrtc_dtls_setup_get_type"
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c:type="GstWebRTCDTLSSetup">
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<member name="none"
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value="0"
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c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
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glib:nick="none">
|
|
<doc xml:space="preserve"
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filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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line="220">none</doc>
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</member>
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<member name="actpass"
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value="1"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
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|
glib:nick="actpass">
|
|
<doc xml:space="preserve"
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|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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line="221">actpass</doc>
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|
</member>
|
|
<member name="active"
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|
value="2"
|
|
c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
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|
glib:nick="active">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
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|
line="222">sendonly</doc>
|
|
</member>
|
|
<member name="passive"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
|
|
glib:nick="passive">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="223">recvonly</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCDTLSTransport"
|
|
c:symbol-prefix="webrtc_dtls_transport"
|
|
c:type="GstWebRTCDTLSTransport"
|
|
parent="Gst.Object"
|
|
glib:type-name="GstWebRTCDTLSTransport"
|
|
glib:get-type="gst_webrtc_dtls_transport_get_type"
|
|
glib:type-struct="WebRTCDTLSTransportClass">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="46"/>
|
|
<property name="certificate" writable="1" transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</property>
|
|
<property name="client" writable="1" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="remote-certificate" transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</property>
|
|
<property name="session-id"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</property>
|
|
<property name="state" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransportState"/>
|
|
</property>
|
|
<property name="transport" transfer-ownership="none">
|
|
<type name="WebRTCICETransport"/>
|
|
</property>
|
|
</class>
|
|
<record name="WebRTCDTLSTransportClass"
|
|
c:type="GstWebRTCDTLSTransportClass"
|
|
disguised="1"
|
|
glib:is-gtype-struct-for="WebRTCDTLSTransport">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="46"/>
|
|
</record>
|
|
<enumeration name="WebRTCDTLSTransportState"
|
|
glib:type-name="GstWebRTCDTLSTransportState"
|
|
glib:get-type="gst_webrtc_dtls_transport_state_get_type"
|
|
c:type="GstWebRTCDTLSTransportState">
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
|
|
glib:nick="new">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="67">new</doc>
|
|
</member>
|
|
<member name="closed"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="68">closed</doc>
|
|
</member>
|
|
<member name="failed"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
|
|
glib:nick="failed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="69">failed</doc>
|
|
</member>
|
|
<member name="connecting"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
|
|
glib:nick="connecting">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="70">connecting</doc>
|
|
</member>
|
|
<member name="connected"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
|
|
glib:nick="connected">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="71">connected</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCDataChannel"
|
|
c:symbol-prefix="webrtc_data_channel"
|
|
c:type="GstWebRTCDataChannel"
|
|
parent="GObject.Object"
|
|
abstract="1"
|
|
glib:type-name="GstWebRTCDataChannel"
|
|
glib:get-type="gst_webrtc_data_channel_get_type"
|
|
glib:type-struct="WebRTCDataChannelClass">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="63"/>
|
|
<method name="close" c:identifier="gst_webrtc_data_channel_close">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="545">Close the @channel.</doc>
|
|
<source-position filename="gst-libs/gst/webrtc/datachannel.h"
|
|
line="46"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="channel" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="547">a #GstWebRTCDataChannel</doc>
|
|
<type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="send_data"
|
|
c:identifier="gst_webrtc_data_channel_send_data">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="507">Send @data as a data message over @channel.</doc>
|
|
<source-position filename="gst-libs/gst/webrtc/datachannel.h"
|
|
line="40"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="channel" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="509">a #GstWebRTCDataChannel</doc>
|
|
<type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
|
|
</instance-parameter>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="510">a #GBytes or %NULL</doc>
|
|
<type name="GLib.Bytes" c:type="GBytes*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="send_string"
|
|
c:identifier="gst_webrtc_data_channel_send_string">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="526">Send @str as a string message over @channel.</doc>
|
|
<source-position filename="gst-libs/gst/webrtc/datachannel.h"
|
|
line="43"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="channel" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="528">a #GstWebRTCDataChannel</doc>
|
|
<type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
|
|
</instance-parameter>
|
|
<parameter name="str"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="529">a string or %NULL</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="buffered-amount" transfer-ownership="none">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<property name="buffered-amount-low-threshold"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<property name="id"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</property>
|
|
<property name="label"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</property>
|
|
<property name="max-packet-lifetime"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</property>
|
|
<property name="max-retransmits"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</property>
|
|
<property name="negotiated"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="ordered"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="priority"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="WebRTCPriorityType"/>
|
|
</property>
|
|
<property name="protocol"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</property>
|
|
<property name="ready-state" transfer-ownership="none">
|
|
<type name="WebRTCDataChannelState"/>
|
|
</property>
|
|
<glib:signal name="close" when="last" action="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="353">Close the data channel</doc>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
</glib:signal>
|
|
<glib:signal name="on-buffered-amount-low" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
</glib:signal>
|
|
<glib:signal name="on-close" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
</glib:signal>
|
|
<glib:signal name="on-error" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="error" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="299">the #GError thrown</doc>
|
|
<type name="GLib.Error"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
<glib:signal name="on-message-data" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="308">a #GBytes of the data received</doc>
|
|
<type name="GLib.Bytes"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
<glib:signal name="on-message-string" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="317">the data received as a string</doc>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
<glib:signal name="on-open" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
</glib:signal>
|
|
<glib:signal name="send-data" when="last" action="1">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="334">a #GBytes with the data</doc>
|
|
<type name="GLib.Bytes"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
<glib:signal name="send-string" when="last" action="1">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="345">the data to send as a string</doc>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
</class>
|
|
<record name="WebRTCDataChannelClass"
|
|
c:type="GstWebRTCDataChannelClass"
|
|
disguised="1"
|
|
glib:is-gtype-struct-for="WebRTCDataChannel">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="63"/>
|
|
</record>
|
|
<enumeration name="WebRTCDataChannelState"
|
|
version="1.16"
|
|
glib:type-name="GstWebRTCDataChannelState"
|
|
glib:get-type="gst_webrtc_data_channel_state_get_type"
|
|
c:type="GstWebRTCDataChannelState">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="319">See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_NEW"
|
|
glib:nick="new">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="321">new</doc>
|
|
</member>
|
|
<member name="connecting"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING"
|
|
glib:nick="connecting">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="322">connection</doc>
|
|
</member>
|
|
<member name="open"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_OPEN"
|
|
glib:nick="open">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="323">open</doc>
|
|
</member>
|
|
<member name="closing"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING"
|
|
glib:nick="closing">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="324">closing</doc>
|
|
</member>
|
|
<member name="closed"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="325">closed</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCFECType"
|
|
version="1.14.1"
|
|
glib:type-name="GstWebRTCFECType"
|
|
glib:get-type="gst_webrtc_fec_type_get_type"
|
|
c:type="GstWebRTCFECType">
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_FEC_TYPE_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="270">none</doc>
|
|
</member>
|
|
<member name="ulp_red"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED"
|
|
glib:nick="ulp-red">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="271">ulpfec + red</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICEComponent"
|
|
glib:type-name="GstWebRTCICEComponent"
|
|
glib:get-type="gst_webrtc_ice_component_get_type"
|
|
c:type="GstWebRTCICEComponent">
|
|
<member name="rtp"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP"
|
|
glib:nick="rtp">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="175">RTP component</doc>
|
|
</member>
|
|
<member name="rtcp"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
|
|
glib:nick="rtcp">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="176">RTCP component</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICEConnectionState"
|
|
glib:type-name="GstWebRTCICEConnectionState"
|
|
glib:get-type="gst_webrtc_ice_connection_state_get_type"
|
|
c:type="GstWebRTCICEConnectionState">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="97">See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
|
|
glib:nick="new">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="99">new</doc>
|
|
</member>
|
|
<member name="checking"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING"
|
|
glib:nick="checking">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="100">checking</doc>
|
|
</member>
|
|
<member name="connected"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED"
|
|
glib:nick="connected">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="101">connected</doc>
|
|
</member>
|
|
<member name="completed"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED"
|
|
glib:nick="completed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="102">completed</doc>
|
|
</member>
|
|
<member name="failed"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED"
|
|
glib:nick="failed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="103">failed</doc>
|
|
</member>
|
|
<member name="disconnected"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED"
|
|
glib:nick="disconnected">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="104">disconnected</doc>
|
|
</member>
|
|
<member name="closed"
|
|
value="6"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="105">closed</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICEGatheringState"
|
|
glib:type-name="GstWebRTCICEGatheringState"
|
|
glib:get-type="gst_webrtc_ice_gathering_state_get_type"
|
|
c:type="GstWebRTCICEGatheringState">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="82">See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW"
|
|
glib:nick="new">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="84">new</doc>
|
|
</member>
|
|
<member name="gathering"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING"
|
|
glib:nick="gathering">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="85">gathering</doc>
|
|
</member>
|
|
<member name="complete"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
|
|
glib:nick="complete">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="86">complete</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICERole"
|
|
glib:type-name="GstWebRTCICERole"
|
|
glib:get-type="gst_webrtc_ice_role_get_type"
|
|
c:type="GstWebRTCICERole">
|
|
<member name="controlled"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
|
|
glib:nick="controlled">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="164">controlled</doc>
|
|
</member>
|
|
<member name="controlling"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
|
|
glib:nick="controlling">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="165">controlling</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCICETransport"
|
|
c:symbol-prefix="webrtc_ice_transport"
|
|
c:type="GstWebRTCICETransport"
|
|
parent="Gst.Object"
|
|
abstract="1"
|
|
glib:type-name="GstWebRTCICETransport"
|
|
glib:get-type="gst_webrtc_ice_transport_get_type"
|
|
glib:type-struct="WebRTCICETransportClass">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="49"/>
|
|
<property name="component"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="WebRTCICEComponent"/>
|
|
</property>
|
|
<property name="gathering-state" transfer-ownership="none">
|
|
<type name="WebRTCICEGatheringState"/>
|
|
</property>
|
|
<property name="state" transfer-ownership="none">
|
|
<type name="WebRTCICEConnectionState"/>
|
|
</property>
|
|
<glib:signal name="on-new-candidate" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="object" transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
<glib:signal name="on-selected-candidate-pair-change" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
</glib:signal>
|
|
</class>
|
|
<record name="WebRTCICETransportClass"
|
|
c:type="GstWebRTCICETransportClass"
|
|
disguised="1"
|
|
glib:is-gtype-struct-for="WebRTCICETransport">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="49"/>
|
|
</record>
|
|
<enumeration name="WebRTCICETransportPolicy"
|
|
version="1.16"
|
|
glib:type-name="GstWebRTCICETransportPolicy"
|
|
glib:get-type="gst_webrtc_ice_transport_policy_get_type"
|
|
c:type="GstWebRTCICETransportPolicy">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="360">See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
|
|
for more information.</doc>
|
|
<member name="all"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL"
|
|
glib:nick="all">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="362">all</doc>
|
|
</member>
|
|
<member name="relay"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY"
|
|
glib:nick="relay">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="363">relay</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCKind"
|
|
version="1.20"
|
|
glib:type-name="GstWebRTCKind"
|
|
glib:get-type="gst_webrtc_kind_get_type"
|
|
c:type="GstWebRTCKind">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="376">https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind</doc>
|
|
<member name="unknown"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_KIND_UNKNOWN"
|
|
glib:nick="unknown">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="378">Kind has not yet been set</doc>
|
|
</member>
|
|
<member name="audio"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_KIND_AUDIO"
|
|
glib:nick="audio">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="379">Kind is audio</doc>
|
|
</member>
|
|
<member name="video"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_KIND_VIDEO"
|
|
glib:nick="video">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="380">Kind is audio</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCPeerConnectionState"
|
|
glib:type-name="GstWebRTCPeerConnectionState"
|
|
glib:get-type="gst_webrtc_peer_connection_state_get_type"
|
|
c:type="GstWebRTCPeerConnectionState">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="141">See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW"
|
|
glib:nick="new">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="143">new</doc>
|
|
</member>
|
|
<member name="connecting"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING"
|
|
glib:nick="connecting">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="144">connecting</doc>
|
|
</member>
|
|
<member name="connected"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED"
|
|
glib:nick="connected">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="145">connected</doc>
|
|
</member>
|
|
<member name="disconnected"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED"
|
|
glib:nick="disconnected">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="146">disconnected</doc>
|
|
</member>
|
|
<member name="failed"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED"
|
|
glib:nick="failed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="147">failed</doc>
|
|
</member>
|
|
<member name="closed"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="148">closed</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCPriorityType"
|
|
version="1.16"
|
|
glib:type-name="GstWebRTCPriorityType"
|
|
glib:get-type="gst_webrtc_priority_type_get_type"
|
|
c:type="GstWebRTCPriorityType">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="300">See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
|
|
<member name="very_low"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_PRIORITY_TYPE_VERY_LOW"
|
|
glib:nick="very-low">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="302">very-low</doc>
|
|
</member>
|
|
<member name="low"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW"
|
|
glib:nick="low">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="303">low</doc>
|
|
</member>
|
|
<member name="medium"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_PRIORITY_TYPE_MEDIUM"
|
|
glib:nick="medium">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="304">medium</doc>
|
|
</member>
|
|
<member name="high"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_PRIORITY_TYPE_HIGH"
|
|
glib:nick="high">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="305">high</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCRTPReceiver"
|
|
c:symbol-prefix="webrtc_rtp_receiver"
|
|
c:type="GstWebRTCRTPReceiver"
|
|
parent="Gst.Object"
|
|
glib:type-name="GstWebRTCRTPReceiver"
|
|
glib:get-type="gst_webrtc_rtp_receiver_get_type"
|
|
glib:type-struct="WebRTCRTPReceiverClass">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="52"/>
|
|
<property name="transport" version="1.20" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpreceiver.c"
|
|
line="107">The DTLS transport for this receiver</doc>
|
|
<type name="WebRTCDTLSTransport"/>
|
|
</property>
|
|
</class>
|
|
<record name="WebRTCRTPReceiverClass"
|
|
c:type="GstWebRTCRTPReceiverClass"
|
|
disguised="1"
|
|
glib:is-gtype-struct-for="WebRTCRTPReceiver">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="52"/>
|
|
</record>
|
|
<class name="WebRTCRTPSender"
|
|
c:symbol-prefix="webrtc_rtp_sender"
|
|
c:type="GstWebRTCRTPSender"
|
|
parent="Gst.Object"
|
|
glib:type-name="GstWebRTCRTPSender"
|
|
glib:get-type="gst_webrtc_rtp_sender_get_type"
|
|
glib:type-struct="WebRTCRTPSenderClass">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="55"/>
|
|
<method name="set_priority"
|
|
c:identifier="gst_webrtc_rtp_sender_set_priority"
|
|
version="1.20">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpsender.c"
|
|
line="61">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.</doc>
|
|
<source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="39"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sender" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpsender.c"
|
|
line="63">a #GstWebRTCRTPSender</doc>
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</instance-parameter>
|
|
<parameter name="priority" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpsender.c"
|
|
line="64">The priority of this sender</doc>
|
|
<type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="priority"
|
|
version="1.20"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpsender.c"
|
|
line="143">The priority from which to set the DSCP field on packets</doc>
|
|
<type name="WebRTCPriorityType"/>
|
|
</property>
|
|
<property name="transport" version="1.20" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpsender.c"
|
|
line="158">The DTLS transport for this sender</doc>
|
|
<type name="WebRTCDTLSTransport"/>
|
|
</property>
|
|
</class>
|
|
<record name="WebRTCRTPSenderClass"
|
|
c:type="GstWebRTCRTPSenderClass"
|
|
disguised="1"
|
|
glib:is-gtype-struct-for="WebRTCRTPSender">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="55"/>
|
|
</record>
|
|
<class name="WebRTCRTPTransceiver"
|
|
c:symbol-prefix="webrtc_rtp_transceiver"
|
|
c:type="GstWebRTCRTPTransceiver"
|
|
parent="Gst.Object"
|
|
abstract="1"
|
|
glib:type-name="GstWebRTCRTPTransceiver"
|
|
glib:get-type="gst_webrtc_rtp_transceiver_get_type"
|
|
glib:type-struct="WebRTCRTPTransceiverClass">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="60"/>
|
|
<property name="codec-preferences"
|
|
version="1.20"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtptransceiver.c"
|
|
line="285">Caps representing the codec preferences.</doc>
|
|
<type name="Gst.Caps"/>
|
|
</property>
|
|
<property name="current-direction"
|
|
version="1.20"
|
|
transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtptransceiver.c"
|
|
line="253">The transceiver's current directionality, or none if the
|
|
transceiver is stopped or has never participated in an exchange
|
|
of offers and answers. To change the transceiver's
|
|
directionality, set the value of the direction property.</doc>
|
|
<type name="WebRTCRTPTransceiverDirection"/>
|
|
</property>
|
|
<property name="direction"
|
|
version="1.18"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtptransceiver.c"
|
|
line="216">Direction of the transceiver.</doc>
|
|
<type name="WebRTCRTPTransceiverDirection"/>
|
|
</property>
|
|
<property name="kind" version="1.20" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtptransceiver.c"
|
|
line="271">The kind of media this transceiver transports</doc>
|
|
<type name="WebRTCKind"/>
|
|
</property>
|
|
<property name="mid" version="1.20" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtptransceiver.c"
|
|
line="231">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.</doc>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</property>
|
|
<property name="mlineindex"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</property>
|
|
<property name="receiver"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver"/>
|
|
</property>
|
|
<property name="sender"
|
|
writable="1"
|
|
construct-only="1"
|
|
transfer-ownership="none">
|
|
<type name="WebRTCRTPSender"/>
|
|
</property>
|
|
</class>
|
|
<record name="WebRTCRTPTransceiverClass"
|
|
c:type="GstWebRTCRTPTransceiverClass"
|
|
disguised="1"
|
|
glib:is-gtype-struct-for="WebRTCRTPTransceiver">
|
|
<source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="60"/>
|
|
</record>
|
|
<enumeration name="WebRTCRTPTransceiverDirection"
|
|
glib:type-name="GstWebRTCRTPTransceiverDirection"
|
|
glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type"
|
|
c:type="GstWebRTCRTPTransceiverDirection">
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="203">none</doc>
|
|
</member>
|
|
<member name="inactive"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE"
|
|
glib:nick="inactive">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="204">inactive</doc>
|
|
</member>
|
|
<member name="sendonly"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY"
|
|
glib:nick="sendonly">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="205">sendonly</doc>
|
|
</member>
|
|
<member name="recvonly"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY"
|
|
glib:nick="recvonly">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="206">recvonly</doc>
|
|
</member>
|
|
<member name="sendrecv"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
|
|
glib:nick="sendrecv">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="207">sendrecv</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCSCTPTransportState"
|
|
version="1.16"
|
|
glib:type-name="GstWebRTCSCTPTransportState"
|
|
glib:get-type="gst_webrtc_sctp_transport_state_get_type"
|
|
c:type="GstWebRTCSCTPTransportState">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="281">See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW"
|
|
glib:nick="new">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="283">new</doc>
|
|
</member>
|
|
<member name="connecting"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING"
|
|
glib:nick="connecting">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="284">connecting</doc>
|
|
</member>
|
|
<member name="connected"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED"
|
|
glib:nick="connected">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="285">connected</doc>
|
|
</member>
|
|
<member name="closed"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="286">closed</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCSDPType"
|
|
glib:type-name="GstWebRTCSDPType"
|
|
glib:get-type="gst_webrtc_sdp_type_get_type"
|
|
c:type="GstWebRTCSDPType">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="184">See <http://w3c.github.io/webrtc-pc/#rtcsdptype></doc>
|
|
<member name="offer"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_OFFER"
|
|
glib:nick="offer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="186">offer</doc>
|
|
</member>
|
|
<member name="pranswer"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
|
|
glib:nick="pranswer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="187">pranswer</doc>
|
|
</member>
|
|
<member name="answer"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
|
|
glib:nick="answer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="188">answer</doc>
|
|
</member>
|
|
<member name="rollback"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
|
|
glib:nick="rollback">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="189">rollback</doc>
|
|
</member>
|
|
<function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
|
|
<source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="30"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="41">the string representation of @type or "unknown" when @type is not
|
|
recognized.</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="39">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</enumeration>
|
|
<record name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription"
|
|
glib:type-name="GstWebRTCSessionDescription"
|
|
glib:get-type="gst_webrtc_session_description_get_type"
|
|
c:symbol-prefix="webrtc_session_description">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="36">See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class></doc>
|
|
<source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="47"/>
|
|
<field name="type" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="38">the #GstWebRTCSDPType of the description</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</field>
|
|
<field name="sdp" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="39">the #GstSDPMessage of the description</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</field>
|
|
<constructor name="new"
|
|
c:identifier="gst_webrtc_session_description_new">
|
|
<source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="50"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="103">a new #GstWebRTCSessionDescription from @type
|
|
and @sdp</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="100">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
<parameter name="sdp" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="101">a #GstSDPMessage</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="copy" c:identifier="gst_webrtc_session_description_copy">
|
|
<source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="52"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="65">a new copy of @src</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="63">a #GstWebRTCSessionDescription</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="const GstWebRTCSessionDescription*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="free" c:identifier="gst_webrtc_session_description_free">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="83">Free @desc and all associated resources</doc>
|
|
<source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="54"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="desc" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="85">a #GstWebRTCSessionDescription</doc>
|
|
<type name="WebRTCSessionDescription"
|
|
c:type="GstWebRTCSessionDescription*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<enumeration name="WebRTCSignalingState"
|
|
glib:type-name="GstWebRTCSignalingState"
|
|
glib:get-type="gst_webrtc_signaling_state_get_type"
|
|
c:type="GstWebRTCSignalingState">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="120">See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate></doc>
|
|
<member name="stable"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE"
|
|
glib:nick="stable">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="122">stable</doc>
|
|
</member>
|
|
<member name="closed"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="123">closed</doc>
|
|
</member>
|
|
<member name="have_local_offer"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER"
|
|
glib:nick="have-local-offer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="124">have-local-offer</doc>
|
|
</member>
|
|
<member name="have_remote_offer"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER"
|
|
glib:nick="have-remote-offer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="125">have-remote-offer</doc>
|
|
</member>
|
|
<member name="have_local_pranswer"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER"
|
|
glib:nick="have-local-pranswer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="126">have-local-pranswer</doc>
|
|
</member>
|
|
<member name="have_remote_pranswer"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
|
|
glib:nick="have-remote-pranswer">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="127">have-remote-pranswer</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCStatsType"
|
|
glib:type-name="GstWebRTCStatsType"
|
|
glib:get-type="gst_webrtc_stats_type_get_type"
|
|
c:type="GstWebRTCStatsType">
|
|
<member name="codec"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_STATS_CODEC"
|
|
glib:nick="codec">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="235">codec</doc>
|
|
</member>
|
|
<member name="inbound_rtp"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_STATS_INBOUND_RTP"
|
|
glib:nick="inbound-rtp">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="236">inbound-rtp</doc>
|
|
</member>
|
|
<member name="outbound_rtp"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP"
|
|
glib:nick="outbound-rtp">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="237">outbound-rtp</doc>
|
|
</member>
|
|
<member name="remote_inbound_rtp"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
|
|
glib:nick="remote-inbound-rtp">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="238">remote-inbound-rtp</doc>
|
|
</member>
|
|
<member name="remote_outbound_rtp"
|
|
value="5"
|
|
c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
|
|
glib:nick="remote-outbound-rtp">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="239">remote-outbound-rtp</doc>
|
|
</member>
|
|
<member name="csrc"
|
|
value="6"
|
|
c:identifier="GST_WEBRTC_STATS_CSRC"
|
|
glib:nick="csrc">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="240">csrc</doc>
|
|
</member>
|
|
<member name="peer_connection"
|
|
value="7"
|
|
c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION"
|
|
glib:nick="peer-connection">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="241">peer-connectiion</doc>
|
|
</member>
|
|
<member name="data_channel"
|
|
value="8"
|
|
c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
|
|
glib:nick="data-channel">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="242">data-channel</doc>
|
|
</member>
|
|
<member name="stream"
|
|
value="9"
|
|
c:identifier="GST_WEBRTC_STATS_STREAM"
|
|
glib:nick="stream">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="243">stream</doc>
|
|
</member>
|
|
<member name="transport"
|
|
value="10"
|
|
c:identifier="GST_WEBRTC_STATS_TRANSPORT"
|
|
glib:nick="transport">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="244">transport</doc>
|
|
</member>
|
|
<member name="candidate_pair"
|
|
value="11"
|
|
c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR"
|
|
glib:nick="candidate-pair">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="245">candidate-pair</doc>
|
|
</member>
|
|
<member name="local_candidate"
|
|
value="12"
|
|
c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE"
|
|
glib:nick="local-candidate">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="246">local-candidate</doc>
|
|
</member>
|
|
<member name="remote_candidate"
|
|
value="13"
|
|
c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
|
|
glib:nick="remote-candidate">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="247">remote-candidate</doc>
|
|
</member>
|
|
<member name="certificate"
|
|
value="14"
|
|
c:identifier="GST_WEBRTC_STATS_CERTIFICATE"
|
|
glib:nick="certificate">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/webrtc_fwd.h"
|
|
line="248">certificate</doc>
|
|
</member>
|
|
</enumeration>
|
|
<docsection name="gstwebrtc-datachannel">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/datachannel.c"
|
|
line="21"><https://www.w3.org/TR/webrtc/#rtcdatachannel></doc>
|
|
</docsection>
|
|
<docsection name="gstwebrtc-dtlstransport">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/dtlstransport.c"
|
|
line="20"><https://www.w3.org/TR/webrtc/#rtcdtlstransport></doc>
|
|
</docsection>
|
|
<docsection name="gstwebrtc-icetransport">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/icetransport.c"
|
|
line="20"><https://www.w3.org/TR/webrtc/#rtcicetransport></doc>
|
|
</docsection>
|
|
<docsection name="gstwebrtc-receiver">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpreceiver.c"
|
|
line="20"><https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface></doc>
|
|
</docsection>
|
|
<docsection name="gstwebrtc-sender">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtpsender.c"
|
|
line="20"><https://www.w3.org/TR/webrtc/#rtcrtpsender-interface></doc>
|
|
</docsection>
|
|
<docsection name="gstwebrtc-sessiondescription">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="20"><https://www.w3.org/TR/webrtc/#rtcsessiondescription-class></doc>
|
|
</docsection>
|
|
<docsection name="gstwebrtc-transceiver">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtptransceiver.c"
|
|
line="20"><https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface></doc>
|
|
</docsection>
|
|
<function name="webrtc_sdp_type_to_string"
|
|
c:identifier="gst_webrtc_sdp_type_to_string"
|
|
moved-to="WebRTCSDPType.to_string">
|
|
<source-position filename="gst-libs/gst/webrtc/rtcsessiondescription.h"
|
|
line="30"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="41">the string representation of @type or "unknown" when @type is not
|
|
recognized.</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gst-libs/gst/webrtc/rtcsessiondescription.c"
|
|
line="39">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</namespace>
|
|
</repository>
|