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https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
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update gir-files from gst master
This commit is contained in:
Guillaume Desmottes
parent
27f694f995
commit
ba98d9cece
10 changed files with 2166 additions and 387 deletions
File diff suppressed because it is too large
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@ -898,12 +898,15 @@ of their segment.
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a SEGMENT_START have posted a SEGMENT_DONE.
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* GST_MESSAGE_DURATION_CHANGED: Is posted by an element that detected a change
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in the stream duration. The default bin behaviour is to clear any
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cached duration values so that the next duration query will perform
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a full duration recalculation. The duration change is posted to the
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in the stream duration. The duration change is posted to the
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application so that it can refetch the new duration with a duration
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query. Note that these messages can be posted before the bin is
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prerolled, in which case the duration query might fail.
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prerolled, in which case the duration query might fail. Note also that
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there might be a discrepancy (due to internal buffering/queueing) between the
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stream being currently displayed and the returned duration query.
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Applications might want to also query for duration (and changes) by
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listening to the GST_MESSAGE_STREAM_START message, signaling the active start
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of a (new) stream.
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* GST_MESSAGE_CLOCK_LOST: This message is posted by an element when it
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can no longer provide a clock. The default bin behaviour is to
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@ -928,11 +931,8 @@ the parent of the bin.
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A #GstBin implements the following default behaviour for answering to a
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#GstQuery:
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* GST_QUERY_DURATION:If the query has been asked before with the same format
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and the bin is a toplevel bin (ie. has no parent),
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use the cached previous value. If no previous value was cached, the
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query is sent to all sink elements in the bin and the MAXIMUM of all
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values is returned. If the bin is a toplevel bin the value is cached.
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* GST_QUERY_DURATION: The bin will forward the query to all sink
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elements contained within and will return the maximum value.
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If no sinks are available in the bin, the query fails.
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* GST_QUERY_POSITION:The query is sent to all sink elements in the bin and the
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@ -9408,7 +9408,7 @@ not be extended or allocated using a custom allocator.</doc>
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</member>
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<member name="tai" value="3" c:identifier="GST_CLOCK_TYPE_TAI" glib:nick="tai">
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<doc xml:space="preserve">time since Epoch, but using International Atomic Time
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as reference (Since 1.18)</doc>
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as reference (Since: 1.18)</doc>
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</member>
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</enumeration>
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<record name="Context" c:type="GstContext" version="1.2" glib:type-name="GstContext" glib:get-type="gst_context_get_type" c:symbol-prefix="context">
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@ -17817,6 +17817,25 @@ check the name of a custom event.</doc>
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</parameter>
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</parameters>
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</method>
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<method name="has_name_id" c:identifier="gst_event_has_name_id" version="1.18">
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<doc xml:space="preserve">Checks if @event has the given @name. This function is usually used to
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check the name of a custom event.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">%TRUE if @name matches the name of the event structure.</doc>
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<type name="gboolean" c:type="gboolean"/>
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</return-value>
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<parameters>
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<instance-parameter name="event" transfer-ownership="none">
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<doc xml:space="preserve">The #GstEvent.</doc>
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<type name="Event" c:type="GstEvent*"/>
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</instance-parameter>
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<parameter name="name" transfer-ownership="none">
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<doc xml:space="preserve">name to check as a GQuark</doc>
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<type name="GLib.Quark" c:type="GQuark"/>
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</parameter>
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</parameters>
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</method>
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<method name="parse_buffer_size" c:identifier="gst_event_parse_buffer_size">
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<doc xml:space="preserve">Get the format, minsize, maxsize and async-flag in the buffersize event.</doc>
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@ -33473,7 +33492,9 @@ producer of the value and the consumer of the value.
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A #GstPromise is created with gst_promise_new() by the consumer and passed
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to the producer to avoid thread safety issues with the change callback.
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A #GstPromise can be replied to with a value (or an error) by the producer
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with gst_promise_reply(). gst_promise_interrupt() is for the consumer to
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with gst_promise_reply(). The exact value returned is defined by the API
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contract of the producer and %NULL may be a valid reply.
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gst_promise_interrupt() is for the consumer to
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indicate to the producer that the value is not needed anymore and producing
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that value can stop. The @GST_PROMISE_RESULT_EXPIRED state set by a call
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to gst_promise_expire() indicates to the consumer that a value will never
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@ -33570,7 +33591,7 @@ message is handled and/or destroyed (possibly unanswered).</doc>
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<doc xml:space="preserve">Retrieve the reply set on @promise. @promise must be in
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%GST_PROMISE_RESULT_REPLIED and the returned structure is owned by @promise</doc>
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<return-value transfer-ownership="none">
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<return-value transfer-ownership="none" nullable="1">
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<doc xml:space="preserve">The reply set on @promise</doc>
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<type name="Structure" c:type="const GstStructure*"/>
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</return-value>
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@ -33612,7 +33633,7 @@ is not visible to the consumer.</doc>
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<doc xml:space="preserve">a #GstPromise</doc>
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<type name="Promise" c:type="GstPromise*"/>
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</instance-parameter>
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<parameter name="s" transfer-ownership="full">
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<parameter name="s" transfer-ownership="full" nullable="1" allow-none="1">
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<doc xml:space="preserve">a #GstStructure with the the reply contents</doc>
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<type name="Structure" c:type="GstStructure*"/>
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</parameter>
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@ -37662,45 +37683,90 @@ info to stream time (which is always between 0 and the duration of the stream).<
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<type name="SegmentFlags" c:type="GstSegmentFlags"/>
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</field>
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<field name="rate" writable="1">
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<doc xml:space="preserve">the playback rate of the segment</doc>
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<doc xml:space="preserve">the playback rate of the segment is set in response to a seek
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event and, without any seek, the value should be `1.0`. This
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value is used by elements that synchronize buffer [running
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times](additional/design/synchronisation.md#running-time) on
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the clock (usually the sink elements), leading to consuming
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buffers faster (for a value `> 1.0`) or slower (for `0.0 <
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value < 1.0`) than normal playback speed. The rate also
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defines the playback direction, meaning that when the value is
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lower than `0.0`, the playback happens in reverse, and the
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[stream-time](additional/design/synchronisation.md#stream-time)
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is going backward. The `rate` value should never be `0.0`.</doc>
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<type name="gdouble" c:type="gdouble"/>
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</field>
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<field name="applied_rate" writable="1">
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<doc xml:space="preserve">the already applied rate to the segment</doc>
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<doc xml:space="preserve">The applied rate is the rate that has been applied to the stream.
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The effective/resulting playback rate of a stream is
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`rate * applied_rate`.
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The applied rate can be set by source elements when a server is
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sending the stream with an already modified playback speed
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rate. Filter elements that modify the stream in a way that
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modifies the playback speed should also modify the applied
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rate. For example the #videorate element when its
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#videorate:rate property is set will set the applied rate of
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the segment it pushed downstream. Also #scaletempo applies the
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input segment rate to the stream and outputs a segment with
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rate=1.0 and applied_rate=<inputsegment.rate>.</doc>
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<type name="gdouble" c:type="gdouble"/>
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</field>
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<field name="format" writable="1">
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<doc xml:space="preserve">the format of the segment values</doc>
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<doc xml:space="preserve">the unit used for all of the segment's values.</doc>
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<type name="Format" c:type="GstFormat"/>
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</field>
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<field name="base" writable="1">
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<doc xml:space="preserve">the running time (plus elapsed time, see offset) of the segment start</doc>
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<doc xml:space="preserve">the running time (plus elapsed time, see offset) of the
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segment [start](GstSegment.start) ([stop](GstSegment.stop) if
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rate < 0.0).</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="offset" writable="1">
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<doc xml:space="preserve">the amount (in buffer timestamps) that has already been elapsed in
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the segment</doc>
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<doc xml:space="preserve">the offset expresses the elapsed time (in buffer timestamps)
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before a seek with its start (stop if rate < 0.0) seek type
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set to #GST_SEEK_TYPE_NONE, the value is set to the position
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of the segment at the time of the seek.</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="start" writable="1">
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<doc xml:space="preserve">the start of the segment in buffer timestamp time (PTS)</doc>
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<doc xml:space="preserve">the start time of the segment (in buffer timestamps)
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[(PTS)](GstBuffer.pts), that is the timestamp of the first
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buffer to output inside the segment (last one during
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reverse playback). For example decoders will
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[clip](gst_segment_clip) out the buffers before the start
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time.</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="stop" writable="1">
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<doc xml:space="preserve">the stop of the segment in buffer timestamp time (PTS)</doc>
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<doc xml:space="preserve">the stop time of the segment (in buffer timestamps)
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[(PTS)](GstBuffer.pts), that is the timestamp of the last
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buffer to output inside the segment (first one during
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reverse playback). For example decoders will
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[clip](gst_segment_clip) out buffers after the stop time.</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="time" writable="1">
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<doc xml:space="preserve">the stream time of the segment start</doc>
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<doc xml:space="preserve">the stream time of the segment [start](GstSegment.start)
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([stop](GstSegment.stop) if rate < 0.0).</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="position" writable="1">
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<doc xml:space="preserve">the buffer timestamp position in the segment (used internally by
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elements such as sources, demuxers or parsers to track progress)</doc>
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<doc xml:space="preserve">the buffer timestamp position in the segment is supposed to be
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updated by elements such as sources, demuxers or parsers to
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track progress by setting it to the last pushed buffer' end time
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([timestamp](GstBuffer.pts) + #GstBuffer.duration) for that
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specific segment. The position is used when reconfiguring the
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segment with #gst_segment_do_seek when the seek is only
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updating the segment (see [offset](GstSegment.offset)).</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="duration" writable="1">
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<doc xml:space="preserve">the duration of the stream</doc>
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<doc xml:space="preserve">the duration of the segment is the maximum absolute difference
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between #GstSegment.start and #GstSegment.stop if stop is not
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set, otherwise it should be the difference between those
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two values. This should be set by elements that know the
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overall stream duration (like demuxers) and will be used when
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seeking with #GST_SEEK_TYPE_END.</doc>
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<type name="guint64" c:type="guint64"/>
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</field>
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<field name="_gst_reserved" readable="0" private="1">
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|
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@ -691,7 +691,7 @@ return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).</do
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</parameter>
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</parameters>
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</function-macro>
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<constant name="AUDIO_FORMATS_ALL" value=" { S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }" c:type="GST_AUDIO_FORMATS_ALL">
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<constant name="AUDIO_FORMATS_ALL" value="{ S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }" c:type="GST_AUDIO_FORMATS_ALL">
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<doc xml:space="preserve">List of all audio formats, for use in template caps strings.</doc>
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<type name="utf8" c:type="gchar*"/>
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@ -8552,17 +8552,6 @@ All scheduling of samples and timestamps is done in this base class
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together with #GstAudioBaseSink using a default implementation of a
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#GstAudioRingBuffer that uses threads.</doc>
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<virtual-method name="clear_all">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="sink" transfer-ownership="none">
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<type name="AudioSink" c:type="GstAudioSink*"/>
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</instance-parameter>
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</parameters>
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</virtual-method>
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<virtual-method name="close">
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<return-value transfer-ownership="none">
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@ -8839,6 +8828,13 @@ together with #GstAudioBaseSink using a default implementation of a
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</parameters>
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</callback>
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</field>
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<field name="extension">
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<doc xml:space="preserve">class extension structure. Since: 1.18</doc>
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<type name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension*"/>
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</field>
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</record>
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<record name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension">
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<field name="clear_all">
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<callback name="clear_all">
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@ -8852,11 +8848,6 @@ together with #GstAudioBaseSink using a default implementation of a
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</parameters>
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</callback>
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</field>
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<field name="_gst_reserved" readable="0" private="1">
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<array zero-terminated="0" fixed-size="0">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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</record>
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<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">
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<doc xml:space="preserve">This is the most simple base class for audio sources that only requires
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@ -10161,6 +10152,22 @@ string is not a known format.</doc>
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</parameter>
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</parameters>
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</function>
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<function name="audio_formats_raw" c:identifier="gst_audio_formats_raw" version="1.18">
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<doc xml:space="preserve">Return all the raw audio formats supported by GStreamer.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">an array of #GstAudioFormat</doc>
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<array length="0" zero-terminated="0" c:type="const GstAudioFormat*">
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<type name="AudioFormat" c:type="GstAudioFormat"/>
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</array>
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</return-value>
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<parameters>
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<parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full">
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<doc xml:space="preserve">the number of elements in the returned array</doc>
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<type name="guint" c:type="guint*"/>
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</parameter>
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</parameters>
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</function>
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<function name="audio_get_channel_reorder_map" c:identifier="gst_audio_get_channel_reorder_map">
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<doc xml:space="preserve">Returns a reorder map for @from to @to that can be used in
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custom channel reordering code, e.g. to convert from or to the
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@ -10259,6 +10266,32 @@ otherwise.</doc>
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</parameter>
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</parameters>
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</function>
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<function name="audio_make_raw_caps" c:identifier="gst_audio_make_raw_caps" version="1.18">
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<doc xml:space="preserve">Return a generic raw audio caps for formats defined in @formats.
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If @formats is %NULL returns a caps for all the supported raw audio formats,
|
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see gst_audio_formats_raw().</doc>
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<return-value transfer-ownership="full">
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<doc xml:space="preserve">an audio @GstCaps</doc>
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<type name="Gst.Caps" c:type="GstCaps*"/>
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</return-value>
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<parameters>
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<parameter name="formats" transfer-ownership="none" nullable="1" allow-none="1">
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<doc xml:space="preserve">an array of raw #GstAudioFormat, or %NULL</doc>
|
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<array length="1" zero-terminated="0" c:type="const GstAudioFormat*">
|
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<type name="AudioFormat" c:type="GstAudioFormat"/>
|
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</array>
|
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</parameter>
|
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<parameter name="len" transfer-ownership="none">
|
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<doc xml:space="preserve">the size of @formats</doc>
|
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<type name="guint" c:type="guint"/>
|
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</parameter>
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<parameter name="layout" transfer-ownership="none">
|
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<doc xml:space="preserve">the layout of audio samples</doc>
|
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<type name="AudioLayout" c:type="GstAudioLayout"/>
|
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</parameter>
|
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</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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|
|
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|
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@ -1487,7 +1487,7 @@ downstream.</doc>
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<type name="Aggregator" c:type="GstAggregator*"/>
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</instance-parameter>
|
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<parameter name="segment" transfer-ownership="none">
|
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<type name="Gst.Segment" c:type="GstSegment*"/>
|
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<type name="Gst.Segment" c:type="const GstSegment*"/>
|
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</parameter>
|
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</parameters>
|
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</method>
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|
|
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|
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@ -2347,7 +2347,7 @@ MT safe.</doc>
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<doc xml:space="preserve">A "crank" consists of three steps:
|
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1: Wait for a #GstClockID to be registered with the #GstTestClock.
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2: Advance the #GstTestClock to the time the #GstClockID is waiting, unless
|
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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.
|
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A "crank" can be though of as the notion of
|
||||
manually driving the clock forward to its next logical step.</doc>
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|
|
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|
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@ -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">
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<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>
|
||||
|
||||
|
|
|
|||
|
|
@ -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>
|
||||
|
|
|
|||
|
|
@ -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>
|
||||
|
||||
|
|
|
|||
|
|
@ -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>
|
||||
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
Loading…
Reference in a new issue