update gir-files from gst master

This commit is contained in:
Guillaume Desmottes 2020-06-02 15:47:15 +02:00
parent 27f694f995
commit ba98d9cece
10 changed files with 2166 additions and 387 deletions

File diff suppressed because it is too large Load diff

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@ -898,12 +898,15 @@ of their segment.
a SEGMENT_START have posted a SEGMENT_DONE.
* GST_MESSAGE_DURATION_CHANGED: Is posted by an element that detected a change
in the stream duration. The default bin behaviour is to clear any
cached duration values so that the next duration query will perform
a full duration recalculation. The duration change is posted to the
in the stream duration. The duration change is posted to the
application so that it can refetch the new duration with a duration
query. Note that these messages can be posted before the bin is
prerolled, in which case the duration query might fail.
prerolled, in which case the duration query might fail. Note also that
there might be a discrepancy (due to internal buffering/queueing) between the
stream being currently displayed and the returned duration query.
Applications might want to also query for duration (and changes) by
listening to the GST_MESSAGE_STREAM_START message, signaling the active start
of a (new) stream.
* GST_MESSAGE_CLOCK_LOST: This message is posted by an element when it
can no longer provide a clock. The default bin behaviour is to
@ -928,11 +931,8 @@ the parent of the bin.
A #GstBin implements the following default behaviour for answering to a
#GstQuery:
* GST_QUERY_DURATION:If the query has been asked before with the same format
and the bin is a toplevel bin (ie. has no parent),
use the cached previous value. If no previous value was cached, the
query is sent to all sink elements in the bin and the MAXIMUM of all
values is returned. If the bin is a toplevel bin the value is cached.
* GST_QUERY_DURATION: The bin will forward the query to all sink
elements contained within and will return the maximum value.
If no sinks are available in the bin, the query fails.
* GST_QUERY_POSITION:The query is sent to all sink elements in the bin and the
@ -9408,7 +9408,7 @@ not be extended or allocated using a custom allocator.</doc>
</member>
<member name="tai" value="3" c:identifier="GST_CLOCK_TYPE_TAI" glib:nick="tai">
<doc xml:space="preserve">time since Epoch, but using International Atomic Time
as reference (Since 1.18)</doc>
as reference (Since: 1.18)</doc>
</member>
</enumeration>
<record name="Context" c:type="GstContext" version="1.2" glib:type-name="GstContext" glib:get-type="gst_context_get_type" c:symbol-prefix="context">
@ -17817,6 +17817,25 @@ check the name of a custom event.</doc>
</parameter>
</parameters>
</method>
<method name="has_name_id" c:identifier="gst_event_has_name_id" version="1.18">
<doc xml:space="preserve">Checks if @event has the given @name. This function is usually used to
check the name of a custom event.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if @name matches the name of the event structure.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="event" transfer-ownership="none">
<doc xml:space="preserve">The #GstEvent.</doc>
<type name="Event" c:type="GstEvent*"/>
</instance-parameter>
<parameter name="name" transfer-ownership="none">
<doc xml:space="preserve">name to check as a GQuark</doc>
<type name="GLib.Quark" c:type="GQuark"/>
</parameter>
</parameters>
</method>
<method name="parse_buffer_size" c:identifier="gst_event_parse_buffer_size">
<doc xml:space="preserve">Get the format, minsize, maxsize and async-flag in the buffersize event.</doc>
@ -33473,7 +33492,9 @@ producer of the value and the consumer of the value.
A #GstPromise is created with gst_promise_new() by the consumer and passed
to the producer to avoid thread safety issues with the change callback.
A #GstPromise can be replied to with a value (or an error) by the producer
with gst_promise_reply(). gst_promise_interrupt() is for the consumer to
with gst_promise_reply(). The exact value returned is defined by the API
contract of the producer and %NULL may be a valid reply.
gst_promise_interrupt() is for the consumer to
indicate to the producer that the value is not needed anymore and producing
that value can stop. The @GST_PROMISE_RESULT_EXPIRED state set by a call
to gst_promise_expire() indicates to the consumer that a value will never
@ -33570,7 +33591,7 @@ message is handled and/or destroyed (possibly unanswered).</doc>
<doc xml:space="preserve">Retrieve the reply set on @promise. @promise must be in
%GST_PROMISE_RESULT_REPLIED and the returned structure is owned by @promise</doc>
<return-value transfer-ownership="none">
<return-value transfer-ownership="none" nullable="1">
<doc xml:space="preserve">The reply set on @promise</doc>
<type name="Structure" c:type="const GstStructure*"/>
</return-value>
@ -33612,7 +33633,7 @@ is not visible to the consumer.</doc>
<doc xml:space="preserve">a #GstPromise</doc>
<type name="Promise" c:type="GstPromise*"/>
</instance-parameter>
<parameter name="s" transfer-ownership="full">
<parameter name="s" transfer-ownership="full" nullable="1" allow-none="1">
<doc xml:space="preserve">a #GstStructure with the the reply contents</doc>
<type name="Structure" c:type="GstStructure*"/>
</parameter>
@ -37662,45 +37683,90 @@ info to stream time (which is always between 0 and the duration of the stream).<
<type name="SegmentFlags" c:type="GstSegmentFlags"/>
</field>
<field name="rate" writable="1">
<doc xml:space="preserve">the playback rate of the segment</doc>
<doc xml:space="preserve">the playback rate of the segment is set in response to a seek
event and, without any seek, the value should be `1.0`. This
value is used by elements that synchronize buffer [running
times](additional/design/synchronisation.md#running-time) on
the clock (usually the sink elements), leading to consuming
buffers faster (for a value `&gt; 1.0`) or slower (for `0.0 &lt;
value &lt; 1.0`) than normal playback speed. The rate also
defines the playback direction, meaning that when the value is
lower than `0.0`, the playback happens in reverse, and the
[stream-time](additional/design/synchronisation.md#stream-time)
is going backward. The `rate` value should never be `0.0`.</doc>
<type name="gdouble" c:type="gdouble"/>
</field>
<field name="applied_rate" writable="1">
<doc xml:space="preserve">the already applied rate to the segment</doc>
<doc xml:space="preserve">The applied rate is the rate that has been applied to the stream.
The effective/resulting playback rate of a stream is
`rate * applied_rate`.
The applied rate can be set by source elements when a server is
sending the stream with an already modified playback speed
rate. Filter elements that modify the stream in a way that
modifies the playback speed should also modify the applied
rate. For example the #videorate element when its
#videorate:rate property is set will set the applied rate of
the segment it pushed downstream. Also #scaletempo applies the
input segment rate to the stream and outputs a segment with
rate=1.0 and applied_rate=&lt;inputsegment.rate&gt;.</doc>
<type name="gdouble" c:type="gdouble"/>
</field>
<field name="format" writable="1">
<doc xml:space="preserve">the format of the segment values</doc>
<doc xml:space="preserve">the unit used for all of the segment's values.</doc>
<type name="Format" c:type="GstFormat"/>
</field>
<field name="base" writable="1">
<doc xml:space="preserve">the running time (plus elapsed time, see offset) of the segment start</doc>
<doc xml:space="preserve">the running time (plus elapsed time, see offset) of the
segment [start](GstSegment.start) ([stop](GstSegment.stop) if
rate &lt; 0.0).</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="offset" writable="1">
<doc xml:space="preserve">the amount (in buffer timestamps) that has already been elapsed in
the segment</doc>
<doc xml:space="preserve">the offset expresses the elapsed time (in buffer timestamps)
before a seek with its start (stop if rate &lt; 0.0) seek type
set to #GST_SEEK_TYPE_NONE, the value is set to the position
of the segment at the time of the seek.</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="start" writable="1">
<doc xml:space="preserve">the start of the segment in buffer timestamp time (PTS)</doc>
<doc xml:space="preserve">the start time of the segment (in buffer timestamps)
[(PTS)](GstBuffer.pts), that is the timestamp of the first
buffer to output inside the segment (last one during
reverse playback). For example decoders will
[clip](gst_segment_clip) out the buffers before the start
time.</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="stop" writable="1">
<doc xml:space="preserve">the stop of the segment in buffer timestamp time (PTS)</doc>
<doc xml:space="preserve">the stop time of the segment (in buffer timestamps)
[(PTS)](GstBuffer.pts), that is the timestamp of the last
buffer to output inside the segment (first one during
reverse playback). For example decoders will
[clip](gst_segment_clip) out buffers after the stop time.</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="time" writable="1">
<doc xml:space="preserve">the stream time of the segment start</doc>
<doc xml:space="preserve">the stream time of the segment [start](GstSegment.start)
([stop](GstSegment.stop) if rate &lt; 0.0).</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="position" writable="1">
<doc xml:space="preserve">the buffer timestamp position in the segment (used internally by
elements such as sources, demuxers or parsers to track progress)</doc>
<doc xml:space="preserve">the buffer timestamp position in the segment is supposed to be
updated by elements such as sources, demuxers or parsers to
track progress by setting it to the last pushed buffer' end time
([timestamp](GstBuffer.pts) + #GstBuffer.duration) for that
specific segment. The position is used when reconfiguring the
segment with #gst_segment_do_seek when the seek is only
updating the segment (see [offset](GstSegment.offset)).</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="duration" writable="1">
<doc xml:space="preserve">the duration of the stream</doc>
<doc xml:space="preserve">the duration of the segment is the maximum absolute difference
between #GstSegment.start and #GstSegment.stop if stop is not
set, otherwise it should be the difference between those
two values. This should be set by elements that know the
overall stream duration (like demuxers) and will be used when
seeking with #GST_SEEK_TYPE_END.</doc>
<type name="guint64" c:type="guint64"/>
</field>
<field name="_gst_reserved" readable="0" private="1">

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@ -8552,17 +8552,6 @@ All scheduling of samples and timestamps is done in this base class
together with #GstAudioBaseSink using a default implementation of a
#GstAudioRingBuffer that uses threads.</doc>
<virtual-method name="clear_all">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="close">
<return-value transfer-ownership="none">
@ -8839,6 +8828,13 @@ together with #GstAudioBaseSink using a default implementation of a
</parameters>
</callback>
</field>
<field name="extension">
<doc xml:space="preserve">class extension structure. Since: 1.18</doc>
<type name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension*"/>
</field>
</record>
<record name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension">
<field name="clear_all">
<callback name="clear_all">
@ -8852,11 +8848,6 @@ together with #GstAudioBaseSink using a default implementation of a
</parameters>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" fixed-size="0">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<class name="AudioSrc" c:symbol-prefix="audio_src" c:type="GstAudioSrc" parent="AudioBaseSrc" glib:type-name="GstAudioSrc" glib:get-type="gst_audio_src_get_type" glib:type-struct="AudioSrcClass">
<doc xml:space="preserve">This is the most simple base class for audio sources that only requires
@ -10161,6 +10152,22 @@ string is not a known format.</doc>
</parameter>
</parameters>
</function>
<function name="audio_formats_raw" c:identifier="gst_audio_formats_raw" version="1.18">
<doc xml:space="preserve">Return all the raw audio formats supported by GStreamer.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">an array of #GstAudioFormat</doc>
<array length="0" zero-terminated="0" c:type="const GstAudioFormat*">
<type name="AudioFormat" c:type="GstAudioFormat"/>
</array>
</return-value>
<parameters>
<parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">the number of elements in the returned array</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="audio_get_channel_reorder_map" c:identifier="gst_audio_get_channel_reorder_map">
<doc xml:space="preserve">Returns a reorder map for @from to @to that can be used in
custom channel reordering code, e.g. to convert from or to the
@ -10259,6 +10266,32 @@ otherwise.</doc>
</parameter>
</parameters>
</function>
<function name="audio_make_raw_caps" c:identifier="gst_audio_make_raw_caps" version="1.18">
<doc xml:space="preserve">Return a generic raw audio caps for formats defined in @formats.
If @formats is %NULL returns a caps for all the supported raw audio formats,
see gst_audio_formats_raw().</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">an audio @GstCaps</doc>
<type name="Gst.Caps" c:type="GstCaps*"/>
</return-value>
<parameters>
<parameter name="formats" transfer-ownership="none" nullable="1" allow-none="1">
<doc xml:space="preserve">an array of raw #GstAudioFormat, or %NULL</doc>
<array length="1" zero-terminated="0" c:type="const GstAudioFormat*">
<type name="AudioFormat" c:type="GstAudioFormat"/>
</array>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:space="preserve">the size of @formats</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="layout" transfer-ownership="none">
<doc xml:space="preserve">the layout of audio samples</doc>
<type name="AudioLayout" c:type="GstAudioLayout"/>
</parameter>
</parameters>
</function>
<function name="audio_meta_api_get_type" c:identifier="gst_audio_meta_api_get_type">
<return-value transfer-ownership="none">

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@ -1487,7 +1487,7 @@ downstream.</doc>
<type name="Aggregator" c:type="GstAggregator*"/>
</instance-parameter>
<parameter name="segment" transfer-ownership="none">
<type name="Gst.Segment" c:type="GstSegment*"/>
<type name="Gst.Segment" c:type="const GstSegment*"/>
</parameter>
</parameters>
</method>

View file

@ -2347,7 +2347,7 @@ MT safe.</doc>
<doc xml:space="preserve">A "crank" consists of three steps:
1: Wait for a #GstClockID to be registered with the #GstTestClock.
2: Advance the #GstTestClock to the time the #GstClockID is waiting, unless
the clock time is already passed the clock id (Since 1.18).
the clock time is already passed the clock id (Since: 1.18).
3: Release the #GstClockID wait.
A "crank" can be though of as the notion of
manually driving the clock forward to its next logical step.</doc>

View file

@ -1054,7 +1054,7 @@ function to allocate and OpenGL resources needed for your application</doc>
to the texture</doc>
</member>
</bitfield>
<class name="GLBaseSrc" c:symbol-prefix="gl_base_src" c:type="GstGLBaseSrc" parent="GstBase.PushSrc" abstract="1" glib:type-name="GstGLBaseSrc" glib:get-type="gst_gl_base_src_get_type" glib:type-struct="GLBaseSrcClass">
<class name="GLBaseSrc" c:symbol-prefix="gl_base_src" c:type="GstGLBaseSrc" version="1.18" parent="GstBase.PushSrc" abstract="1" glib:type-name="GstGLBaseSrc" glib:get-type="gst_gl_base_src_get_type" glib:type-struct="GLBaseSrcClass">
<doc xml:space="preserve">#GstGLBaseSrc handles the nitty gritty details of retrieving an OpenGL
context. It also provided some wrappers around #GstBaseSrc's `start()` and
`stop()` virtual methods that ensure an OpenGL context is available and
@ -1131,7 +1131,7 @@ current in the calling thread.</doc>
<type name="GLBaseSrcPrivate" c:type="GstGLBaseSrcPrivate*"/>
</field>
</class>
<record name="GLBaseSrcClass" c:type="GstGLBaseSrcClass" glib:is-gtype-struct-for="GLBaseSrc">
<record name="GLBaseSrcClass" c:type="GstGLBaseSrcClass" glib:is-gtype-struct-for="GLBaseSrc" version="1.18">
<doc xml:space="preserve">The base class for GStreamer GL Video sources.</doc>
<field name="parent_class">
@ -3315,10 +3315,12 @@ single input and producing a single output with a #GstGLFramebuffer</doc>
<doc xml:space="preserve">a #GstGLFilter</doc>
<type name="GLFilter" c:type="GstGLFilter*"/>
</instance-parameter>
<parameter name="in_tex" transfer-ownership="none">
<parameter name="input" transfer-ownership="none">
<doc xml:space="preserve">an input buffer</doc>
<type name="GLMemory" c:type="GstGLMemory*"/>
</parameter>
<parameter name="out_tex" transfer-ownership="none">
<parameter name="output" transfer-ownership="none">
<doc xml:space="preserve">an output buffer</doc>
<type name="GLMemory" c:type="GstGLMemory*"/>
</parameter>
</parameters>
@ -3399,11 +3401,11 @@ any shaders, viewport state, etc must be setup by the caller.</doc>
<doc xml:space="preserve">a #GstGLFilter</doc>
<type name="GLFilter" c:type="GstGLFilter*"/>
</instance-parameter>
<parameter name="inbuf" transfer-ownership="none">
<parameter name="input" transfer-ownership="none">
<doc xml:space="preserve">an input buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="outbuf" transfer-ownership="none">
<parameter name="output" transfer-ownership="none">
<doc xml:space="preserve">an output buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
@ -3584,10 +3586,12 @@ See also: gst_gl_filter_render_to_target()</doc>
<doc xml:space="preserve">a #GstGLFilter</doc>
<type name="GLFilter" c:type="GstGLFilter*"/>
</parameter>
<parameter name="in_tex" transfer-ownership="none">
<parameter name="input" transfer-ownership="none">
<doc xml:space="preserve">an input buffer</doc>
<type name="GLMemory" c:type="GstGLMemory*"/>
</parameter>
<parameter name="out_tex" transfer-ownership="none">
<parameter name="output" transfer-ownership="none">
<doc xml:space="preserve">an output buffer</doc>
<type name="GLMemory" c:type="GstGLMemory*"/>
</parameter>
</parameters>
@ -7141,6 +7145,7 @@ gst_gl_upload_set_caps() creating a new #GstBuffer in @outbuf_ptr.</doc>
<doc xml:space="preserve">This element requires a reconfiguration.</doc>
</member>
<member name="unshared_gl_context" value="-100" c:identifier="GST_GL_UPLOAD_UNSHARED_GL_CONTEXT" glib:nick="unshared-gl-context">
<doc xml:space="preserve">private return value.</doc>
</member>
</enumeration>
<record name="GLVideoAllocationParams" c:type="GstGLVideoAllocationParams" glib:type-name="GstGLVideoAllocationParams" glib:get-type="gst_gl_video_allocation_params_get_type" c:symbol-prefix="gl_video_allocation_params">
@ -8295,25 +8300,32 @@ have been executed when this function returns.</doc>
</parameter>
</parameters>
</method>
<method name="send_scroll_event" c:identifier="gst_gl_window_send_scroll_event">
<method name="send_scroll_event" c:identifier="gst_gl_window_send_scroll_event" version="1.18">
<doc xml:space="preserve">Notify a @window about a scroll event. A scroll signal holding the event
coordinates will be emitted.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="window" transfer-ownership="none">
<doc xml:space="preserve">a #GstGLWindow</doc>
<type name="GLWindow" c:type="GstGLWindow*"/>
</instance-parameter>
<parameter name="posx" transfer-ownership="none">
<doc xml:space="preserve">x position of the mouse cursor</doc>
<type name="gdouble" c:type="double"/>
</parameter>
<parameter name="posy" transfer-ownership="none">
<doc xml:space="preserve">y position of the mouse cursor</doc>
<type name="gdouble" c:type="double"/>
</parameter>
<parameter name="delta_x" transfer-ownership="none">
<doc xml:space="preserve">the x offset of the scroll event</doc>
<type name="gdouble" c:type="double"/>
</parameter>
<parameter name="delta_y" transfer-ownership="none">
<doc xml:space="preserve">the y offset of the scroll event</doc>
<type name="gdouble" c:type="double"/>
</parameter>
</parameters>
@ -9177,6 +9189,10 @@ user-defined purposes.</doc>
</parameter>
</parameters>
</function-macro>
<constant name="GL_COLOR_CONVERT_EXT_FORMATS" value=", BGR10A2_LE, RGB10A2_LE, P010_10LE, P012_LE, P016_LE, Y212_LE, Y412_LE" c:type="GST_GL_COLOR_CONVERT_EXT_FORMATS" introspectable="0">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="GL_COLOR_CONVERT_FORMATS" value="{ RGBA, RGB, RGBx, BGR, BGRx, BGRA, xRGB, xBGR, ARGB, ABGR, Y444, I420, YV12, Y42B, Y41B, NV12, NV21, NV16, NV61, YUY2, UYVY, Y210, AYUV, VUYA, Y410, GRAY8, GRAY16_LE, GRAY16_BE, RGB16, BGR16, ARGB64 " c:type="GST_GL_COLOR_CONVERT_FORMATS">
<doc xml:space="preserve">The currently supported formats that can be converted</doc>
@ -9469,6 +9485,10 @@ user-defined purposes.</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="GL_MEMORY_VIDEO_EXT_FORMATS" value=", BGR10A2_LE, RGB10A2_LE, P010_10LE, P012_LE, P016_LE, Y212_LE, Y412_LE" c:type="GST_GL_MEMORY_VIDEO_EXT_FORMATS" introspectable="0">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="GL_MEMORY_VIDEO_FORMATS_STR" value="{ RGBA, BGRA, RGBx, BGRx, ARGB, ABGR, xRGB, xBGR, RGB, BGR, RGB16, BGR16, AYUV, VUYA, Y410, I420, YV12, NV12, NV21, NV16, NV61, YUY2, UYVY, Y210, Y41B, Y42B, Y444, GRAY8, GRAY16_LE, GRAY16_BE, ARGB64" c:type="GST_GL_MEMORY_VIDEO_FORMATS_STR">
<doc xml:space="preserve">List of video formats that are supported by #GstGLMemory</doc>

View file

@ -3510,15 +3510,13 @@ g_strfreev() when no longer needed.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_request_uri_with_control" c:identifier="gst_rtsp_url_get_request_uri_with_control">
<method name="get_request_uri_with_control" c:identifier="gst_rtsp_url_get_request_uri_with_control" version="1.18">
<doc xml:space="preserve">Get a newly allocated string describing the request URI for @url
combined with the control path for @control_path</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a string with the request URI combined with the control path.
g_free() after usage.
Since 1.18</doc>
g_free() after usage.</doc>
<type name="utf8" c:type="gchar*"/>
</return-value>
<parameters>

View file

@ -3657,6 +3657,20 @@ g_object_unref() after usage.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_dscp_qos" c:identifier="gst_rtsp_media_get_dscp_qos" version="1.18">
<doc xml:space="preserve">Get the configured DSCP QoS of attached media.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the DSCP QoS value of attached streams or -1 if disabled.</doc>
<type name="gint" c:type="gint"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMedia</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_element" c:identifier="gst_rtsp_media_get_element">
<doc xml:space="preserve">Get the element that was used when constructing @media.</doc>
@ -4297,6 +4311,23 @@ INADDR_ANY.</doc>
</parameter>
</parameters>
</method>
<method name="set_dscp_qos" c:identifier="gst_rtsp_media_set_dscp_qos" version="1.18">
<doc xml:space="preserve">Configure the dscp qos of attached streams to @dscp_qos.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMedia</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
<parameter name="dscp_qos" transfer-ownership="none">
<doc xml:space="preserve">a new dscp qos value (0-63, or -1 to disable)</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</method>
<method name="set_eos_shutdown" c:identifier="gst_rtsp_media_set_eos_shutdown">
<doc xml:space="preserve">Set or unset if an EOS event will be sent to the pipeline for @media before
it is unprepared.</doc>
@ -4733,6 +4764,9 @@ when the media was not in the suspended state.</doc>
<property name="clock" writable="1" transfer-ownership="none">
<type name="Gst.Clock"/>
</property>
<property name="dscp-qos" writable="1" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="element" writable="1" construct-only="1" transfer-ownership="none">
<type name="Gst.Element"/>
</property>
@ -5416,6 +5450,20 @@ of all medias created from this factory.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_dscp_qos" c:identifier="gst_rtsp_media_factory_get_dscp_qos" version="1.18">
<doc xml:space="preserve">Get the configured media DSCP QoS.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the media DSCP QoS value or -1 if disabled.</doc>
<type name="gint" c:type="gint"/>
</return-value>
<parameters>
<instance-parameter name="factory" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMediaFactory</doc>
<type name="RTSPMediaFactory" c:type="GstRTSPMediaFactory*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_latency" c:identifier="gst_rtsp_media_factory_get_latency">
<doc xml:space="preserve">Get the latency that is used for receiving media</doc>
@ -5728,6 +5776,23 @@ receiving media</doc>
</parameter>
</parameters>
</method>
<method name="set_dscp_qos" c:identifier="gst_rtsp_media_factory_set_dscp_qos" version="1.18">
<doc xml:space="preserve">Configure the media dscp qos to @dscp_qos.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="factory" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMediaFactory</doc>
<type name="RTSPMediaFactory" c:type="GstRTSPMediaFactory*"/>
</instance-parameter>
<parameter name="dscp_qos" transfer-ownership="none">
<doc xml:space="preserve">a new dscp qos value (0-63, or -1 to disable)</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</method>
<method name="set_eos_shutdown" c:identifier="gst_rtsp_media_factory_set_eos_shutdown">
<doc xml:space="preserve">Configure if media created from this factory will have an EOS sent to the
pipeline before shutdown.</doc>
@ -6005,6 +6070,9 @@ when a client disconnects without sending TEARDOWN.</doc>
<property name="clock" writable="1" transfer-ownership="none">
<type name="Gst.Clock"/>
</property>
<property name="dscp-qos" writable="1" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="eos-shutdown" writable="1" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
@ -6315,6 +6383,23 @@ and called when a message has been sent on the transport.</doc>
</parameter>
</parameters>
</callback>
<callback name="RTSPMessageSentFuncFull" c:type="GstRTSPMessageSentFuncFull" version="1.18">
<doc xml:space="preserve">Function registered with gst_rtsp_stream_transport_set_message_sent_full()
and called when a message has been sent on the transport.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="trans" transfer-ownership="none">
<type name="RTSPStreamTransport" c:type="GstRTSPStreamTransport*"/>
</parameter>
<parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="1">
<doc xml:space="preserve">user data</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
</parameters>
</callback>
<class name="RTSPMountPoints" c:symbol-prefix="rtsp_mount_points" c:type="GstRTSPMountPoints" parent="GObject.Object" glib:type-name="GstRTSPMountPoints" glib:get-type="gst_rtsp_mount_points_get_type" glib:type-struct="RTSPMountPointsClass">
<doc xml:space="preserve">Creates a #GstRTSPMediaFactory object for a given url.</doc>
@ -10358,7 +10443,7 @@ It remains valid for as long as @trans is valid.</doc>
</parameters>
</method>
<method name="message_sent" c:identifier="gst_rtsp_stream_transport_message_sent" version="1.16">
<doc xml:space="preserve">Signal the installed message_sent callback for @trans.</doc>
<doc xml:space="preserve">Signal the installed message_sent / message_sent_full callback for @trans.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
@ -10594,6 +10679,31 @@ to a client. This is usually used when sending RTP/RTCP over TCP.</doc>
</parameter>
</parameters>
</method>
<method name="set_message_sent_full" c:identifier="gst_rtsp_stream_transport_set_message_sent_full" version="1.18">
<doc xml:space="preserve">Install a callback that will be called when a message has been sent on @trans.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="trans" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPStreamTransport</doc>
<type name="RTSPStreamTransport" c:type="GstRTSPStreamTransport*"/>
</instance-parameter>
<parameter name="message_sent" transfer-ownership="none" scope="notified" closure="1" destroy="2">
<doc xml:space="preserve">a callback called when a message has been sent</doc>
<type name="RTSPMessageSentFuncFull" c:type="GstRTSPMessageSentFuncFull"/>
</parameter>
<parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1">
<doc xml:space="preserve">user data passed to callback</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="notify" transfer-ownership="none" nullable="1" allow-none="1" scope="async">
<doc xml:space="preserve">called with the user_data when no longer needed</doc>
<type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
</parameter>
</parameters>
</method>
<method name="set_timed_out" c:identifier="gst_rtsp_stream_transport_set_timed_out">
<doc xml:space="preserve">Set the timed out state of @trans to @timedout</doc>

View file

@ -1485,7 +1485,7 @@ gst_navigation_event_parse_command() to extract the details from the event.</doc
<member name="mouse_scroll" value="7" c:identifier="GST_NAVIGATION_EVENT_MOUSE_SCROLL" glib:nick="mouse-scroll">
<doc xml:space="preserve">A mouse scroll event. Use
gst_navigation_event_parse_mouse_scroll_event() to extract the details from
the event.</doc>
the event. (Since: 1.18)</doc>
</member>
</enumeration>
<record name="NavigationInterface" c:type="GstNavigationInterface" glib:is-gtype-struct-for="Navigation">
@ -3457,7 +3457,7 @@ and SMPTE ST2016-1</doc>
</return-value>
</function>
</record>
<enumeration name="VideoAFDSpec" glib:type-name="GstVideoAFDSpec" glib:get-type="gst_video_afd_spec_get_type" c:type="GstVideoAFDSpec">
<enumeration name="VideoAFDSpec" version="1.18" glib:type-name="GstVideoAFDSpec" glib:get-type="gst_video_afd_spec_get_type" c:type="GstVideoAFDSpec">
<doc xml:space="preserve">Enumeration of the different standards that may apply to AFD data:
0) ETSI/DVB:
@ -3466,9 +3466,7 @@ https://www.etsi.org/deliver/etsi_ts/101100_101199/101154/02.01.01_60/ts_101154v
1) ATSC A/53:
https://www.atsc.org/wp-content/uploads/2015/03/a_53-Part-4-2009.pdf
2) SMPTE ST2016-1:
Since 1.18</doc>
2) SMPTE ST2016-1:</doc>
<member name="dvb_etsi" value="0" c:identifier="GST_VIDEO_AFD_SPEC_DVB_ETSI" glib:nick="dvb-etsi">
<doc xml:space="preserve">AFD value is from DVB/ETSI standard</doc>
</member>
@ -15819,6 +15817,22 @@ the #GstVideoFormat if there is one, or NULL otherwise.</doc>
</parameter>
</parameters>
</function>
<function name="video_formats_raw" c:identifier="gst_video_formats_raw" version="1.18">
<doc xml:space="preserve">Return all the raw video formats supported by GStreamer.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">an array of #GstVideoFormat</doc>
<array length="0" zero-terminated="0" c:type="const GstVideoFormat*">
<type name="VideoFormat" c:type="GstVideoFormat"/>
</array>
</return-value>
<parameters>
<parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">the number of elements in the returned array</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="video_gl_texture_upload_meta_api_get_type" c:identifier="gst_video_gl_texture_upload_meta_api_get_type">
<return-value transfer-ownership="none">
@ -15891,6 +15905,55 @@ recognised, and %FALSE otherwise.</doc>
</parameter>
</parameters>
</function>
<function name="video_make_raw_caps" c:identifier="gst_video_make_raw_caps" version="1.18">
<doc xml:space="preserve">Return a generic raw video caps for formats defined in @formats.
If @formats is %NULL returns a caps for all the supported raw video formats,
see gst_video_formats_raw().</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a video @GstCaps</doc>
<type name="Gst.Caps" c:type="GstCaps*"/>
</return-value>
<parameters>
<parameter name="formats" transfer-ownership="none" nullable="1" allow-none="1">
<doc xml:space="preserve">an array of raw #GstVideoFormat, or %NULL</doc>
<array length="1" zero-terminated="0" c:type="const GstVideoFormat*">
<type name="VideoFormat" c:type="GstVideoFormat"/>
</array>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:space="preserve">the size of @formats</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="video_make_raw_caps_with_features" c:identifier="gst_video_make_raw_caps_with_features" version="1.18">
<doc xml:space="preserve">Return a generic raw video caps for formats defined in @formats with features
@features.
If @formats is %NULL returns a caps for all the supported video formats,
see gst_video_formats_raw().</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a video @GstCaps</doc>
<type name="Gst.Caps" c:type="GstCaps*"/>
</return-value>
<parameters>
<parameter name="formats" transfer-ownership="none" nullable="1" allow-none="1">
<doc xml:space="preserve">an array of raw #GstVideoFormat, or %NULL</doc>
<array length="1" zero-terminated="0" c:type="const GstVideoFormat*">
<type name="VideoFormat" c:type="GstVideoFormat"/>
</array>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:space="preserve">the size of @formats</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="features" transfer-ownership="full" nullable="1" allow-none="1">
<doc xml:space="preserve">the #GstCapsFeatures to set on the caps</doc>
<type name="Gst.CapsFeatures" c:type="GstCapsFeatures*"/>
</parameter>
</parameters>
</function>
<function name="video_mastering_display_info_from_string" c:identifier="gst_video_mastering_display_info_from_string" moved-to="VideoMasteringDisplayInfo.from_string" version="1.18">
<doc xml:space="preserve">Extract #GstVideoMasteringDisplayInfo from @mastering</doc>

View file

@ -8,6 +8,20 @@ and/or use gtk-doc annotations. -->
<package name="gstreamer-webrtc-1.0"/>
<c:include name="gst/webrtc/webrtc.h"/>
<namespace name="GstWebRTC" version="1.0" shared-library="libgstwebrtc-1.0.so.0" c:identifier-prefixes="Gst" c:symbol-prefixes="gst">
<function-macro name="IS_WEBRTC_DATA_CHANNEL" c:identifier="GST_IS_WEBRTC_DATA_CHANNEL" introspectable="0">
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_DATA_CHANNEL_CLASS" c:identifier="GST_IS_WEBRTC_DATA_CHANNEL_CLASS" introspectable="0">
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_DTLS_TRANSPORT" c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT" introspectable="0">
<parameters>
@ -78,6 +92,41 @@ and/or use gtk-doc annotations. -->
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DATA_CHANNEL" c:identifier="GST_WEBRTC_DATA_CHANNEL" introspectable="0">
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DATA_CHANNEL_CLASS" c:identifier="GST_WEBRTC_DATA_CHANNEL_CLASS" introspectable="0">
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DATA_CHANNEL_GET_CLASS" c:identifier="GST_WEBRTC_DATA_CHANNEL_GET_CLASS" introspectable="0">
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DATA_CHANNEL_LOCK" c:identifier="GST_WEBRTC_DATA_CHANNEL_LOCK" introspectable="0">
<parameters>
<parameter name="channel">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DATA_CHANNEL_UNLOCK" c:identifier="GST_WEBRTC_DATA_CHANNEL_UNLOCK" introspectable="0">
<parameters>
<parameter name="channel">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DTLS_TRANSPORT" c:identifier="GST_WEBRTC_DTLS_TRANSPORT" introspectable="0">
<parameters>
@ -322,6 +371,407 @@ for more information.</doc>
<doc xml:space="preserve">connected</doc>
</member>
</enumeration>
<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">
<virtual-method name="close" invoker="close">
<doc xml:space="preserve">Close the @channel.</doc>
<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">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">Send @data as a data message over @channel.</doc>
<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">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">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">Send @str as a string message over @channel.</doc>
<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">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">a string or %NULL</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</virtual-method>
<method name="close" c:identifier="gst_webrtc_data_channel_close">
<doc xml:space="preserve">Close the @channel.</doc>
<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">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">Signal that the data channel reached a low buffered amount. Should only be used by subclasses.</doc>
<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">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">Signal that the data channel was closed. Should only be used by subclasses.</doc>
<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">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">Signal that the data channel had an error. Should only be used by subclasses.</doc>
<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">a #GstWebRTCDataChannel</doc>
<type name="WebRTCDataChannel" c:type="GstWebRTCDataChannel*"/>
</instance-parameter>
<parameter name="error" transfer-ownership="full">
<doc xml:space="preserve">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">Signal that the data channel received a data message. Should only be used by subclasses.</doc>
<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">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">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">Signal that the data channel received a string message. Should only be used by subclasses.</doc>
<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">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">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">Signal that the data channel was opened. Should only be used by subclasses.</doc>
<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">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">Send @data as a data message over @channel.</doc>
<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">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">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">Send @str as a string message over @channel.</doc>
<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">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">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>
<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">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">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">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">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">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">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" glib:is-gtype-struct-for="WebRTCDataChannel" version="1.18">
<field name="parent_class">
<type name="GObject.ObjectClass" c:type="GObjectClass"/>
</field>
<field name="send_data">
<callback name="send_data">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="channel" transfer-ownership="none">
<doc xml:space="preserve">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">a #GBytes or %NULL</doc>
<type name="GLib.Bytes" c:type="GBytes*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="send_string">
<callback name="send_string">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="channel" transfer-ownership="none">
<doc xml:space="preserve">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">a string or %NULL</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="close">
<callback name="close">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="channel" transfer-ownership="none">
<doc xml:space="preserve">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>
</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">GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection