diff options
Diffstat (limited to 'girs')
| -rw-r--r-- | girs/GstWebRTC-1.0.gir | 936 |
1 files changed, 156 insertions, 780 deletions
diff --git a/girs/GstWebRTC-1.0.gir b/girs/GstWebRTC-1.0.gir index c43e3dce55..e6712f0d14 100644 --- a/girs/GstWebRTC-1.0.gir +++ b/girs/GstWebRTC-1.0.gir @@ -113,7 +113,7 @@ and/or use gtk-doc annotations. --> c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER" introspectable="0"> <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="34"/> + line="32"/> <parameters> <parameter name="obj"> </parameter> @@ -123,7 +123,7 @@ and/or use gtk-doc annotations. --> c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS" introspectable="0"> <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="36"/> + line="34"/> <parameters> <parameter name="klass"> </parameter> @@ -156,24 +156,6 @@ and/or use gtk-doc annotations. --> </parameter> </parameters> </function-macro> - <function-macro name="WEBRTC_DATA_CHANNEL_LOCK" - c:identifier="GST_WEBRTC_DATA_CHANNEL_LOCK" - introspectable="0"> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="39"/> - <parameters> - <parameter name="channel"> - </parameter> - </parameters> - </function-macro> - <function-macro name="WEBRTC_DATA_CHANNEL_UNLOCK" - c:identifier="GST_WEBRTC_DATA_CHANNEL_UNLOCK" - introspectable="0"> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="40"/> - <parameters> - <parameter name="channel"> - </parameter> - </parameters> - </function-macro> <function-macro name="WEBRTC_DTLS_TRANSPORT" c:identifier="GST_WEBRTC_DTLS_TRANSPORT" introspectable="0"> @@ -292,7 +274,7 @@ and/or use gtk-doc annotations. --> c:identifier="GST_WEBRTC_RTP_TRANSCEIVER" introspectable="0"> <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="33"/> + line="31"/> <parameters> <parameter name="obj"> </parameter> @@ -302,7 +284,7 @@ and/or use gtk-doc annotations. --> c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_CLASS" introspectable="0"> <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="35"/> + line="33"/> <parameters> <parameter name="klass"> </parameter> @@ -312,7 +294,7 @@ and/or use gtk-doc annotations. --> c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS" introspectable="0"> <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="37"/> + line="35"/> <parameters> <parameter name="obj"> </parameter> @@ -325,31 +307,39 @@ and/or use gtk-doc annotations. --> c:type="GstWebRTCBundlePolicy"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/webrtc_fwd.h" - line="340">GST_WEBRTC_BUNDLE_POLICY_NONE: none -GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced -GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat -GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle -See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 + line="340">See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.</doc> <member name="none" value="0" c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE" glib:nick="none"> + <doc xml:space="preserve" + filename="gst-libs/gst/webrtc/webrtc_fwd.h" + line="342">none</doc> </member> <member name="balanced" value="1" c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED" glib:nick="balanced"> + <doc xml:space="preserve" + filename="gst-libs/gst/webrtc/webrtc_fwd.h" + line="343">balanced</doc> </member> <member name="max_compat" value="2" c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT" glib:nick="max-compat"> + <doc xml:space="preserve" + filename="gst-libs/gst/webrtc/webrtc_fwd.h" + line="344">max-compat</doc> </member> <member name="max_bundle" value="3" c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE" glib:nick="max-bundle"> + <doc xml:space="preserve" + filename="gst-libs/gst/webrtc/webrtc_fwd.h" + line="345">max-bundle</doc> </member> </enumeration> <enumeration name="WebRTCDTLSSetup" @@ -396,36 +386,7 @@ for more information.</doc> glib:type-name="GstWebRTCDTLSTransport" glib:get-type="gst_webrtc_dtls_transport_get_type" glib:type-struct="WebRTCDTLSTransportClass"> - <source-position filename="gst-libs/gst/webrtc/dtlstransport.h" - line="61"/> - <constructor name="new" c:identifier="gst_webrtc_dtls_transport_new"> - <source-position filename="gst-libs/gst/webrtc/dtlstransport.h" - line="64"/> - <return-value transfer-ownership="none"> - <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/> - </return-value> - <parameters> - <parameter name="session_id" transfer-ownership="none"> - <type name="guint" c:type="guint"/> - </parameter> - </parameters> - </constructor> - <method name="set_transport" - c:identifier="gst_webrtc_dtls_transport_set_transport"> - <source-position filename="gst-libs/gst/webrtc/dtlstransport.h" - line="67"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <instance-parameter name="transport" transfer-ownership="none"> - <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/> - </instance-parameter> - <parameter name="ice" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </parameter> - </parameters> - </method> + <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> @@ -447,47 +408,12 @@ for more information.</doc> <property name="transport" transfer-ownership="none"> <type name="WebRTCICETransport"/> </property> - <field name="parent"> - <type name="Gst.Object" c:type="GstObject"/> - </field> - <field name="transport"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </field> - <field name="state"> - <type name="WebRTCDTLSTransportState" - c:type="GstWebRTCDTLSTransportState"/> - </field> - <field name="client"> - <type name="gboolean" c:type="gboolean"/> - </field> - <field name="session_id"> - <type name="guint" c:type="guint"/> - </field> - <field name="dtlssrtpenc"> - <type name="Gst.Element" c:type="GstElement*"/> - </field> - <field name="dtlssrtpdec"> - <type name="Gst.Element" c:type="GstElement*"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> </class> <record name="WebRTCDTLSTransportClass" c:type="GstWebRTCDTLSTransportClass" + disguised="1" glib:is-gtype-struct-for="WebRTCDTLSTransport"> - <source-position filename="gst-libs/gst/webrtc/dtlstransport.h" - line="61"/> - <field name="parent_class"> - <type name="Gst.ObjectClass" c:type="GstObjectClass"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="46"/> </record> <enumeration name="WebRTCDTLSTransportState" glib:type-name="GstWebRTCDTLSTransportState" @@ -537,226 +463,18 @@ for more information.</doc> <class name="WebRTCDataChannel" c:symbol-prefix="webrtc_data_channel" c:type="GstWebRTCDataChannel" - version="1.18" 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/datachannel.h" line="82"/> - <virtual-method name="close" invoker="close"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="544">Close the @channel.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="79"/> - <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="546">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </instance-parameter> - </parameters> - </virtual-method> - <virtual-method name="send_data" invoker="send_data"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="506">Send @data as a data message over @channel.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="77"/> - <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="508">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="509">a #GBytes or %NULL</doc> - <type name="GLib.Bytes" c:type="GBytes*"/> - </parameter> - </parameters> - </virtual-method> - <virtual-method name="send_string" invoker="send_string"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="525">Send @str as a string message over @channel.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="78"/> - <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="527">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="528">a string or %NULL</doc> - <type name="utf8" c:type="const gchar*"/> - </parameter> - </parameters> - </virtual-method> + <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="544">Close the @channel.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="109"/> - <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="546">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </instance-parameter> - </parameters> - </method> - <method name="on_buffered_amount_low" - c:identifier="gst_webrtc_data_channel_on_buffered_amount_low"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="490">Signal that the data channel reached a low buffered amount. Should only be used by subclasses.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="100"/> - <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="492">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </instance-parameter> - </parameters> - </method> - <method name="on_close" c:identifier="gst_webrtc_data_channel_on_close"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="405">Signal that the data channel was closed. Should only be used by subclasses.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="88"/> - <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="407">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </instance-parameter> - </parameters> - </method> - <method name="on_error" c:identifier="gst_webrtc_data_channel_on_error"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="434">Signal that the data channel had an error. Should only be used by subclasses.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="91"/> - <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="436">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </instance-parameter> - <parameter name="error" transfer-ownership="full"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="437">a #GError</doc> - <type name="GLib.Error" c:type="GError*"/> - </parameter> - </parameters> - </method> - <method name="on_message_data" - c:identifier="gst_webrtc_data_channel_on_message_data"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="454">Signal that the data channel received a data message. Should only be used by subclasses.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="94"/> - <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="456">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="457">a #GBytes or %NULL</doc> - <type name="GLib.Bytes" c:type="GBytes*"/> - </parameter> - </parameters> - </method> - <method name="on_message_string" - c:identifier="gst_webrtc_data_channel_on_message_string"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="472">Signal that the data channel received a string message. Should only be used by subclasses.</doc> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="97"/> - <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="474">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="475">a string or %NULL</doc> - <type name="utf8" c:type="const gchar*"/> - </parameter> - </parameters> - </method> - <method name="on_open" c:identifier="gst_webrtc_data_channel_on_open"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="371">Signal that the data channel was opened. Should only be used by subclasses.</doc> + line="545">Close the @channel.</doc> <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="85"/> + line="46"/> <return-value transfer-ownership="none"> <type name="none" c:type="void"/> </return-value> @@ -764,7 +482,7 @@ for more information.</doc> <instance-parameter name="channel" transfer-ownership="none"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="373">a #GstWebRTCDataChannel</doc> + line="547">a #GstWebRTCDataChannel</doc> <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> </instance-parameter> </parameters> @@ -773,9 +491,9 @@ for more information.</doc> c:identifier="gst_webrtc_data_channel_send_data"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="506">Send @data as a data message over @channel.</doc> + line="507">Send @data as a data message over @channel.</doc> <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="103"/> + line="40"/> <return-value transfer-ownership="none"> <type name="none" c:type="void"/> </return-value> @@ -783,7 +501,7 @@ for more information.</doc> <instance-parameter name="channel" transfer-ownership="none"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="508">a #GstWebRTCDataChannel</doc> + line="509">a #GstWebRTCDataChannel</doc> <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> </instance-parameter> <parameter name="data" @@ -792,7 +510,7 @@ for more information.</doc> allow-none="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="509">a #GBytes or %NULL</doc> + line="510">a #GBytes or %NULL</doc> <type name="GLib.Bytes" c:type="GBytes*"/> </parameter> </parameters> @@ -801,9 +519,9 @@ for more information.</doc> c:identifier="gst_webrtc_data_channel_send_string"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="525">Send @str as a string message over @channel.</doc> + line="526">Send @str as a string message over @channel.</doc> <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="106"/> + line="43"/> <return-value transfer-ownership="none"> <type name="none" c:type="void"/> </return-value> @@ -811,7 +529,7 @@ for more information.</doc> <instance-parameter name="channel" transfer-ownership="none"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="527">a #GstWebRTCDataChannel</doc> + line="528">a #GstWebRTCDataChannel</doc> <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> </instance-parameter> <parameter name="str" @@ -820,7 +538,7 @@ for more information.</doc> allow-none="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="528">a string or %NULL</doc> + line="529">a string or %NULL</doc> <type name="utf8" c:type="const gchar*"/> </parameter> </parameters> @@ -884,55 +602,10 @@ for more information.</doc> <property name="ready-state" transfer-ownership="none"> <type name="WebRTCDataChannelState"/> </property> - <field name="parent"> - <type name="GObject.Object" c:type="GObject"/> - </field> - <field name="lock"> - <type name="GLib.Mutex" c:type="GMutex"/> - </field> - <field name="label"> - <type name="utf8" c:type="gchar*"/> - </field> - <field name="ordered"> - <type name="gboolean" c:type="gboolean"/> - </field> - <field name="max_packet_lifetime"> - <type name="guint" c:type="guint"/> - </field> - <field name="max_retransmits"> - <type name="guint" c:type="guint"/> - </field> - <field name="protocol"> - <type name="utf8" c:type="gchar*"/> - </field> - <field name="negotiated"> - <type name="gboolean" c:type="gboolean"/> - </field> - <field name="id"> - <type name="gint" c:type="gint"/> - </field> - <field name="priority"> - <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/> - </field> - <field name="ready_state"> - <type name="WebRTCDataChannelState" - c:type="GstWebRTCDataChannelState"/> - </field> - <field name="buffered_amount"> - <type name="guint64" c:type="guint64"/> - </field> - <field name="buffered_amount_low_threshold"> - <type name="guint64" c:type="guint64"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> <glib:signal name="close" when="last" action="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="352">Close the data channel</doc> + line="353">Close the data channel</doc> <return-value transfer-ownership="none"> <type name="none" c:type="void"/> </return-value> @@ -955,7 +628,7 @@ for more information.</doc> <parameter name="error" transfer-ownership="none"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="298">the #GError thrown</doc> + line="299">the #GError thrown</doc> <type name="GLib.Error"/> </parameter> </parameters> @@ -971,7 +644,7 @@ for more information.</doc> allow-none="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="307">a #GBytes of the data received</doc> + line="308">a #GBytes of the data received</doc> <type name="GLib.Bytes"/> </parameter> </parameters> @@ -987,7 +660,7 @@ for more information.</doc> allow-none="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="316">the data received as a string</doc> + line="317">the data received as a string</doc> <type name="utf8" c:type="gchar*"/> </parameter> </parameters> @@ -1008,7 +681,7 @@ for more information.</doc> allow-none="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="333">a #GBytes with the data</doc> + line="334">a #GBytes with the data</doc> <type name="GLib.Bytes"/> </parameter> </parameters> @@ -1024,7 +697,7 @@ for more information.</doc> allow-none="1"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/datachannel.c" - line="344">the data to send as a string</doc> + line="345">the data to send as a string</doc> <type name="utf8" c:type="gchar*"/> </parameter> </parameters> @@ -1032,86 +705,9 @@ for more information.</doc> </class> <record name="WebRTCDataChannelClass" c:type="GstWebRTCDataChannelClass" - glib:is-gtype-struct-for="WebRTCDataChannel" - version="1.18"> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" line="82"/> - <field name="parent_class"> - <type name="GObject.ObjectClass" c:type="GObjectClass"/> - </field> - <field name="send_data"> - <callback name="send_data"> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="77"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <parameter name="channel" transfer-ownership="none"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="508">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </parameter> - <parameter name="data" - transfer-ownership="none" - nullable="1" - allow-none="1"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="509">a #GBytes or %NULL</doc> - <type name="GLib.Bytes" c:type="GBytes*"/> - </parameter> - </parameters> - </callback> - </field> - <field name="send_string"> - <callback name="send_string"> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="78"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <parameter name="channel" transfer-ownership="none"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="527">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </parameter> - <parameter name="str" - transfer-ownership="none" - nullable="1" - allow-none="1"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="528">a string or %NULL</doc> - <type name="utf8" c:type="const gchar*"/> - </parameter> - </parameters> - </callback> - </field> - <field name="close"> - <callback name="close"> - <source-position filename="gst-libs/gst/webrtc/datachannel.h" - line="79"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <parameter name="channel" transfer-ownership="none"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/datachannel.c" - line="546">a #GstWebRTCDataChannel</doc> - <type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/> - </parameter> - </parameters> - </callback> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + 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" @@ -1120,36 +716,46 @@ for more information.</doc> c:type="GstWebRTCDataChannelState"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/webrtc_fwd.h" - line="319">GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new -GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection -GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open -GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing -GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed -See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc> + 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" @@ -1320,89 +926,7 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc> glib:type-name="GstWebRTCICETransport" glib:get-type="gst_webrtc_ice_transport_get_type" glib:type-struct="WebRTCICETransportClass"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="64"/> - <virtual-method name="gather_candidates"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="61"/> - <return-value transfer-ownership="none"> - <type name="gboolean" c:type="gboolean"/> - </return-value> - <parameters> - <instance-parameter name="transport" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </instance-parameter> - </parameters> - </virtual-method> - <method name="connection_state_change" - c:identifier="gst_webrtc_ice_transport_connection_state_change"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="67"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <instance-parameter name="ice" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </instance-parameter> - <parameter name="new_state" transfer-ownership="none"> - <type name="WebRTCICEConnectionState" - c:type="GstWebRTCICEConnectionState"/> - </parameter> - </parameters> - </method> - <method name="gathering_state_change" - c:identifier="gst_webrtc_ice_transport_gathering_state_change"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="70"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <instance-parameter name="ice" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </instance-parameter> - <parameter name="new_state" transfer-ownership="none"> - <type name="WebRTCICEGatheringState" - c:type="GstWebRTCICEGatheringState"/> - </parameter> - </parameters> - </method> - <method name="new_candidate" - c:identifier="gst_webrtc_ice_transport_new_candidate"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="75"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <instance-parameter name="ice" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </instance-parameter> - <parameter name="stream_id" transfer-ownership="none"> - <type name="guint" c:type="guint"/> - </parameter> - <parameter name="component" transfer-ownership="none"> - <type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/> - </parameter> - <parameter name="attr" transfer-ownership="none"> - <type name="utf8" c:type="gchar*"/> - </parameter> - </parameters> - </method> - <method name="selected_pair_change" - c:identifier="gst_webrtc_ice_transport_selected_pair_change"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="73"/> - <return-value transfer-ownership="none"> - <type name="none" c:type="void"/> - </return-value> - <parameters> - <instance-parameter name="ice" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </instance-parameter> - </parameters> - </method> + <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="49"/> <property name="component" writable="1" construct-only="1" @@ -1415,34 +939,6 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc> <property name="state" transfer-ownership="none"> <type name="WebRTCICEConnectionState"/> </property> - <field name="parent"> - <type name="Gst.Object" c:type="GstObject"/> - </field> - <field name="role"> - <type name="WebRTCICERole" c:type="GstWebRTCICERole"/> - </field> - <field name="component"> - <type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/> - </field> - <field name="state"> - <type name="WebRTCICEConnectionState" - c:type="GstWebRTCICEConnectionState"/> - </field> - <field name="gathering_state"> - <type name="WebRTCICEGatheringState" - c:type="GstWebRTCICEGatheringState"/> - </field> - <field name="src"> - <type name="Gst.Element" c:type="GstElement*"/> - </field> - <field name="sink"> - <type name="Gst.Element" c:type="GstElement*"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> <glib:signal name="on-new-candidate" when="last"> <return-value transfer-ownership="none"> <type name="none" c:type="void"/> @@ -1461,31 +957,9 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc> </class> <record name="WebRTCICETransportClass" c:type="GstWebRTCICETransportClass" + disguised="1" glib:is-gtype-struct-for="WebRTCICETransport"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="64"/> - <field name="parent_class"> - <type name="Gst.ObjectClass" c:type="GstObjectClass"/> - </field> - <field name="gather_candidates"> - <callback name="gather_candidates"> - <source-position filename="gst-libs/gst/webrtc/icetransport.h" - line="61"/> - <return-value transfer-ownership="none"> - <type name="gboolean" c:type="gboolean"/> - </return-value> - <parameters> - <parameter name="transport" transfer-ownership="none"> - <type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/> - </parameter> - </parameters> - </callback> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="49"/> </record> <enumeration name="WebRTCICETransportPolicy" version="1.16" @@ -1494,19 +968,23 @@ See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc> c:type="GstWebRTCICETransportPolicy"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/webrtc_fwd.h" - line="360">GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all -GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay -See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 + 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" @@ -1605,110 +1083,78 @@ for more information.</doc> c:type="GstWebRTCPriorityType"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/webrtc_fwd.h" - line="300">GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low -GST_WEBRTC_PRIORITY_TYPE_LOW: low -GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium -GST_WEBRTC_PRIORITY_TYPE_HIGH: high -See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc> + 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" - version="1.16" parent="Gst.Object" glib:type-name="GstWebRTCRTPReceiver" glib:get-type="gst_webrtc_rtp_receiver_get_type" glib:type-struct="WebRTCRTPReceiverClass"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtpreceiver.h" - line="38">An object to track the receiving aspect of the stream - -Mostly matches the WebRTC RTCRtpReceiver interface.</doc> - <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" line="63"/> - <constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new"> - <source-position filename="gst-libs/gst/webrtc/rtpreceiver.h" - line="66"/> - <return-value transfer-ownership="none"> - <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/> - </return-value> - </constructor> - <field name="parent"> - <type name="Gst.Object" c:type="GstObject"/> - </field> - <field name="transport"> + <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.h" - line="40">The transport for RTP packets</doc> - <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + 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/rtpreceiver.h" line="63"/> - <field name="parent_class"> - <type name="Gst.ObjectClass" c:type="GstObjectClass"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + <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" - version="1.16" parent="Gst.Object" glib:type-name="GstWebRTCRTPSender" glib:get-type="gst_webrtc_rtp_sender_get_type" glib:type-struct="WebRTCRTPSenderClass"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtpsender.h" - line="38">An object to track the sending aspect of the stream - -Mostly matches the WebRTC RTCRtpSender interface.</doc> - <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="75"/> - <constructor name="new" c:identifier="gst_webrtc_rtp_sender_new"> - <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="78"/> - <return-value transfer-ownership="none"> - <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/> - </return-value> - </constructor> + <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="59">Sets the content of the IPv4 Type of Service (ToS), also known as DSCP + 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="81"/> + <source-position filename="gst-libs/gst/webrtc/rtpsender.h" line="39"/> <return-value transfer-ownership="none"> <type name="none" c:type="void"/> </return-value> @@ -1716,13 +1162,13 @@ This also sets the Traffic Class field of IPv6.</doc> <instance-parameter name="sender" transfer-ownership="none"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/rtpsender.c" - line="61">a #GstWebRTCRTPSender</doc> + 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="62">The priority of this sender</doc> + line="64">The priority of this sender</doc> <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/> </parameter> </parameters> @@ -1733,74 +1179,77 @@ This also sets the Traffic Class field of IPv6.</doc> transfer-ownership="none"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/rtpsender.c" - line="136">The priority from which to set the DSCP field on packets</doc> + line="143">The priority from which to set the DSCP field on packets</doc> <type name="WebRTCPriorityType"/> </property> - <field name="parent"> - <type name="Gst.Object" c:type="GstObject"/> - </field> - <field name="transport"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtpsender.h" - line="40">The transport for RTP packets</doc> - <type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/> - </field> - <field name="send_encodings"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtpsender.h" - line="41">Unused</doc> - <array name="GLib.Array" c:type="GArray*"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> - <field name="priority" version="1.20"> + <property name="transport" version="1.20" transfer-ownership="none"> <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtpsender.h" - line="50">The priority of the stream</doc> - <type name="WebRTCPriorityType" c:type="GstWebRTCPriorityType"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + 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/rtpsender.h" line="75"/> - <field name="parent_class"> - <type name="Gst.ObjectClass" c:type="GstObjectClass"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + <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" - version="1.16" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCRTPTransceiver" glib:get-type="gst_webrtc_rtp_transceiver_get_type" glib:type-struct="WebRTCRTPTransceiverClass"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="39">Mostly matches the WebRTC RTCRtpTransceiver interface.</doc> - <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="96"/> + <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="188">Direction of the transceiver.</doc> + 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" @@ -1819,93 +1268,12 @@ This also sets the Traffic Class field of IPv6.</doc> transfer-ownership="none"> <type name="WebRTCRTPSender"/> </property> - <field name="parent"> - <type name="Gst.Object" c:type="GstObject"/> - </field> - <field name="mline"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="41">the mline number this transceiver corresponds to</doc> - <type name="guint" c:type="guint"/> - </field> - <field name="mid"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="42">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*"/> - </field> - <field name="stopped"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="48">Indicates whether or not sending and receiving using the paired -#GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled, -either due to SDP offer/answer</doc> - <type name="gboolean" c:type="gboolean"/> - </field> - <field name="sender"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="51">The #GstWebRTCRTPSender object responsible sending data to the -remote peer</doc> - <type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/> - </field> - <field name="receiver"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="53">The #GstWebRTCRTPReceiver object responsible for receiver data from -the remote peer.</doc> - <type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/> - </field> - <field name="direction"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="55">The transceiver's desired direction.</doc> - <type name="WebRTCRTPTransceiverDirection" - c:type="GstWebRTCRTPTransceiverDirection"/> - </field> - <field name="current_direction"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="56">The transceiver's current direction (read-only)</doc> - <type name="WebRTCRTPTransceiverDirection" - c:type="GstWebRTCRTPTransceiverDirection"/> - </field> - <field name="codec_preferences"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="57">A caps representing the codec preferences (read-only)</doc> - <type name="Gst.Caps" c:type="GstCaps*"/> - </field> - <field name="kind" version="1.20"> - <doc xml:space="preserve" - filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="64">Type of media</doc> - <type name="WebRTCKind" c:type="GstWebRTCKind"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> </class> <record name="WebRTCRTPTransceiverClass" c:type="GstWebRTCRTPTransceiverClass" + disguised="1" glib:is-gtype-struct-for="WebRTCRTPTransceiver"> - <source-position filename="gst-libs/gst/webrtc/rtptransceiver.h" - line="96"/> - <field name="parent_class"> - <type name="Gst.ObjectClass" c:type="GstObjectClass"/> - </field> - <field name="_padding"> - <array zero-terminated="0" fixed-size="4"> - <type name="gpointer" c:type="gpointer"/> - </array> - </field> + <source-position filename="gst-libs/gst/webrtc/webrtc_fwd.h" line="60"/> </record> <enumeration name="WebRTCRTPTransceiverDirection" glib:type-name="GstWebRTCRTPTransceiverDirection" @@ -1959,30 +1327,38 @@ the remote peer.</doc> c:type="GstWebRTCSCTPTransportState"> <doc xml:space="preserve" filename="gst-libs/gst/webrtc/webrtc_fwd.h" - line="281">GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new -GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting -GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected -GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed -See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate></doc> + 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" |