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docs: fix broken xrefs
This commit is contained in:
Stefan Kost
parent
2ba8b37d15
commit
a9a6c0f0cd
2 changed files with 86 additions and 78 deletions
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@ -37,7 +37,7 @@
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* #GstBaseSink provides support for exactly one sink pad, which should be
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* named "sink". A sink implementation (subclass of #GstBaseSink) should
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* install a pad template in its base_init function, like so:
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* <programlisting>
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* |[
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* static void
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* my_element_base_init (gpointer g_class)
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* {
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@ -50,34 +50,35 @@
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* // see #GstElementDetails
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* gst_element_class_set_details (gstelement_class, &details);
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* }
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* </programlisting>
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* ]|
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*
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* #GstBaseSink will handle the prerolling correctly. This means that it will
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* return #GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
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* buffer arrives in this element. The base class will call the
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* #GstBaseSink::preroll vmethod with this preroll buffer and will then commit
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* the state change to the next asynchronously pending state.
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* #GstBaseSinkClass.preroll() vmethod with this preroll buffer and will then
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* commit the state change to the next asynchronously pending state.
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*
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* When the element is set to PLAYING, #GstBaseSink will synchronise on the
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* clock using the times returned from ::get_times. If this function returns
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* #GST_CLOCK_TIME_NONE for the start time, no synchronisation will be done.
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* Synchronisation can be disabled entirely by setting the object "sync"
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* property to %FALSE.
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* clock using the times returned from #GstBaseSinkClass.get_times(). If this
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* function returns #GST_CLOCK_TIME_NONE for the start time, no synchronisation
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* will be done. Synchronisation can be disabled entirely by setting the object
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* #GstBaseSink:sync property to %FALSE.
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*
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* After synchronisation the virtual method #GstBaseSink::render will be called.
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* Subclasses should minimally implement this method.
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* After synchronisation the virtual method #GstBaseSinkClass.render() will be
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* called. Subclasses should minimally implement this method.
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*
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* Since 0.10.3 subclasses that synchronise on the clock in the ::render method
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* are supported as well. These classes typically receive a buffer in the render
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* method and can then potentially block on the clock while rendering. A typical
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* example is an audiosink. Since 0.10.11 these subclasses can use
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* gst_base_sink_wait_preroll() to perform the blocking wait.
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* Since 0.10.3 subclasses that synchronise on the clock in the
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* #GstBaseSinkClass.render() method are supported as well. These classes
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* typically receive a buffer in the render method and can then potentially
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* block on the clock while rendering. A typical example is an audiosink.
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* Since 0.10.11 these subclasses can use gst_base_sink_wait_preroll() to
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* perform the blocking wait.
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*
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* Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait
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* for the clock to reach the time indicated by the stop time of the last
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* ::get_times call before posting an EOS message. When the element receives
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* EOS in PAUSED, preroll completes, the event is queued and an EOS message is
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* posted when going to PLAYING.
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* #GstBaseSinkClass.get_times() call before posting an EOS message. When the
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* element receives EOS in PAUSED, preroll completes, the event is queued and an
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* EOS message is posted when going to PLAYING.
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*
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* #GstBaseSink will internally use the #GST_EVENT_NEWSEGMENT events to schedule
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* synchronisation and clipping of buffers. Buffers that fall completely outside
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@ -90,49 +91,53 @@
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* If no clock has been set on the element, the query will be forwarded
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* upstream.
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*
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* The ::set_caps function will be called when the subclass should configure
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* itself to process a specific media type.
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* The #GstBaseSinkClass.set_caps() function will be called when the subclass
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* should configure itself to process a specific media type.
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*
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* The ::start and ::stop virtual methods will be called when resources should
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* be allocated. Any ::preroll, ::render and ::set_caps function will be
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* called between the ::start and ::stop calls.
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* The #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() virtual methods
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* will be called when resources should be allocated. Any
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* #GstBaseSinkClass.preroll(), #GstBaseSinkClass.render() and
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* #GstBaseSinkClass.set_caps() function will be called between the
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* #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() calls.
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*
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* The ::event virtual method will be called when an event is received by
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* #GstBaseSink. Normally this method should only be overriden by very specific
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* elements (such as file sinks) which need to handle the newsegment event
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* specially.
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* The #GstBaseSinkClass.event() virtual method will be called when an event is
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* received by #GstBaseSink. Normally this method should only be overriden by
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* very specific elements (such as file sinks) which need to handle the
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* newsegment event specially.
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*
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* #GstBaseSink provides an overridable ::buffer_alloc function that can be
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* used by sinks that want to do reverse negotiation or to provide
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* custom buffers (hardware buffers for example) to upstream elements.
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* #GstBaseSink provides an overridable #GstBaseSinkClass.buffer_alloc()
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* function that can be used by sinks that want to do reverse negotiation or to
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* provide custom buffers (hardware buffers for example) to upstream elements.
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*
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* The ::unlock method is called when the elements should unblock any blocking
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* operations they perform in the ::render method. This is mostly useful when
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* the ::render method performs a blocking write on a file descriptor, for
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* example.
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* The #GstBaseSinkClass.unlock() method is called when the elements should
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* unblock any blocking operations they perform in the
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* #GstBaseSinkClass.render() method. This is mostly useful when the
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* #GstBaseSinkClass.render() method performs a blocking write on a file
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* descriptor, for example.
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*
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* The max-lateness property affects how the sink deals with buffers that
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* arrive too late in the sink. A buffer arrives too late in the sink when
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* the presentation time (as a combination of the last segment, buffer
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* The #GstBaseSink:max-lateness property affects how the sink deals with
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* buffers that arrive too late in the sink. A buffer arrives too late in the
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* sink when the presentation time (as a combination of the last segment, buffer
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* timestamp and element base_time) plus the duration is before the current
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* time of the clock.
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* If the frame is later than max-lateness, the sink will drop the buffer
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* without calling the render method.
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* This feature is disabled if sync is disabled, the ::get-times method does
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* not return a valid start time or max-lateness is set to -1 (the default).
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* This feature is disabled if sync is disabled, the
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* #GstBaseSinkClass.get_times() method does not return a valid start time or
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* max-lateness is set to -1 (the default).
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* Subclasses can use gst_base_sink_set_max_lateness() to configure the
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* max-lateness value.
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*
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* The qos property will enable the quality-of-service features of the basesink
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* which gather statistics about the real-time performance of the clock
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* synchronisation. For each buffer received in the sink, statistics are
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* The #GstBaseSink:qos property will enable the quality-of-service features of
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* the basesink which gather statistics about the real-time performance of the
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* clock synchronisation. For each buffer received in the sink, statistics are
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* gathered and a QOS event is sent upstream with these numbers. This
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* information can then be used by upstream elements to reduce their processing
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* rate, for example.
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*
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* Since 0.10.15 the async property can be used to instruct the sink to never
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* perform an ASYNC state change. This feature is mostly usable when dealing
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* with non-synchronized streams or sparse streams.
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* Since 0.10.15 the #GstBaseSink:async property can be used to instruct the
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* sink to never perform an ASYNC state change. This feature is mostly usable
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* when dealing with non-synchronized streams or sparse streams.
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*
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* Last reviewed on 2007-08-29 (0.10.15)
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*/
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@ -1931,8 +1936,9 @@ gst_base_sink_adjust_time (GstBaseSink * basesink, GstClockTime time)
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* is no clock, no synchronisation is done and #GST_CLOCK_BADTIME is returned.
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*
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* This function should only be called with the PREROLL_LOCK held, like when
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* receiving an EOS event in the ::event vmethod or when receiving a buffer in
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* the ::render vmethod.
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* receiving an EOS event in the #GstBaseSinkClass.event() vmethod or when
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* receiving a buffer in
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* the #GstBaseSinkClass.render() vmethod.
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*
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* The @time argument should be the running_time of when this method should
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* return and is not adjusted with any latency or offset configured in the
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@ -2006,9 +2012,9 @@ no_clock:
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* gst_base_sink_wait_preroll:
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* @sink: the sink
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*
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* If the #GstBaseSinkClass::render method performs its own synchronisation against
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* the clock it must unblock when going from PLAYING to the PAUSED state and call
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* this method before continuing to render the remaining data.
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* If the #GstBaseSinkClass.render() method performs its own synchronisation
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* against the clock it must unblock when going from PLAYING to the PAUSED state
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* and call this method before continuing to render the remaining data.
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*
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* This function will block until a state change to PLAYING happens (in which
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* case this function returns #GST_FLOW_OK) or the processing must be stopped due
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@ -43,12 +43,12 @@
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* <itemizedlist>
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* <listitem><para>The format is set to #GST_FORMAT_BYTES (default).</para>
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* </listitem>
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* <listitem><para>#GstBaseSrc::is_seekable returns %TRUE.</para>
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* <listitem><para>#GstBaseSrcClass.is_seekable() returns %TRUE.</para>
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* </listitem>
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* </itemizedlist>
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*
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* Since 0.10.9, any #GstBaseSrc can enable pull based scheduling at any
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* time by overriding #GstBaseSrc::check_get_range so that it returns %TRUE.
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* Since 0.10.9, any #GstBaseSrc can enable pull based scheduling at any time
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* by overriding #GstBaseSrcClass.check_get_range() so that it returns %TRUE.
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*
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* If all the conditions are met for operating in pull mode, #GstBaseSrc is
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* automatically seekable in push mode as well. The following conditions must
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@ -56,23 +56,24 @@
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* #GST_FORMAT_BYTES:
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* <itemizedlist>
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* <listitem><para>
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* #GstBaseSrc::is_seekable returns %TRUE.
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* #GstBaseSrcClass.is_seekable() returns %TRUE.
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* </para></listitem>
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* <listitem><para>
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* #GstBaseSrc::query can convert all supported seek formats to the
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* #GstBaseSrc:Class.query() can convert all supported seek formats to the
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* internal format as set with gst_base_src_set_format().
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* </para></listitem>
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* <listitem><para>
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* #GstBaseSrc::do_seek is implemented, performs the seek and returns %TRUE.
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* #GstBaseSrcClass.do_seek() is implemented, performs the seek and returns
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* %TRUE.
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* </para></listitem>
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* </itemizedlist>
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*
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* When the element does not meet the requirements to operate in pull mode,
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* the offset and length in the #GstBaseSrc::create method should be ignored.
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* When the element does not meet the requirements to operate in pull mode, the
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* offset and length in the #GstBaseSrcClass.create() method should be ignored.
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* It is recommended to subclass #GstPushSrc instead, in this situation. If the
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* element can operate in pull mode but only with specific offsets and
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* lengths, it is allowed to generate an error when the wrong values are passed
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* to the #GstBaseSrc::create function.
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* to the #GstBaseSrcClass.create() function.
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*
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* #GstBaseSrc has support for live sources. Live sources are sources that when
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* paused discard data, such as audio or video capture devices. A typical live
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@ -81,9 +82,10 @@
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* Use gst_base_src_set_live() to activate the live source mode.
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*
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* A live source does not produce data in the PAUSED state. This means that the
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* #GstBaseSrc::create method will not be called in PAUSED but only in PLAYING.
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* To signal the pipeline that the element will not produce data, the return
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* value from the READY to PAUSED state will be #GST_STATE_CHANGE_NO_PREROLL.
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* #GstBaseSrcClass.create() method will not be called in PAUSED but only in
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* PLAYING. To signal the pipeline that the element will not produce data, the
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* return value from the READY to PAUSED state will be
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* #GST_STATE_CHANGE_NO_PREROLL.
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*
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* A typical live source will timestamp the buffers it creates with the
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* current running time of the pipeline. This is one reason why a live source
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@ -91,17 +93,17 @@
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* distributed and running.
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*
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* Live sources that synchronize and block on the clock (an audio source, for
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* example) can since 0.10.12 use gst_base_src_wait_playing() when the ::create
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* function was interrupted by a state change to PAUSED.
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* example) can since 0.10.12 use gst_base_src_wait_playing() when the
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* #GstBaseSrcClass.create() function was interrupted by a state change to
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* PAUSED.
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*
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* The #GstBaseSrc::get_times method can be used to implement pseudo-live
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* sources.
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* It only makes sense to implement the ::get_times function if the source is
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* a live source. The ::get_times function should return timestamps starting
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* from 0, as if it were a non-live source. The base class will make sure that
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* the timestamps are transformed into the current running_time.
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* The base source will then wait for the calculated running_time before pushing
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* out the buffer.
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* The #GstBaseSrcClass.get_times() method can be used to implement pseudo-live
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* sources. It only makes sense to implement the #GstBaseSrcClass.get_times()
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* function if the source is a live source. The #GstBaseSrcClass.get_times()
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* function should return timestamps starting from 0, as if it were a non-live
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* source. The base class will make sure that the timestamps are transformed
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* into the current running_time. The base source will then wait for the
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* calculated running_time before pushing out the buffer.
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*
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* For live sources, the base class will by default report a latency of 0.
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* For pseudo live sources, the base class will by default measure the difference
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@ -113,7 +115,7 @@
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* There is only support in #GstBaseSrc for exactly one source pad, which
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* should be named "src". A source implementation (subclass of #GstBaseSrc)
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* should install a pad template in its class_init function, like so:
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* <programlisting>
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* |[
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* static void
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* my_element_class_init (GstMyElementClass *klass)
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* {
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@ -125,7 +127,7 @@
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* // see #GstElementDetails
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* gst_element_class_set_details (gstelement_class, &details);
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* }
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* </programlisting>
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* ]|
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*
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* <refsect2>
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* <title>Controlled shutdown of live sources in applications</title>
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@ -453,9 +455,9 @@ gst_base_src_finalize (GObject * object)
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* gst_base_src_wait_playing:
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* @src: the src
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*
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* If the #GstBaseSrcClass::create method performs its own synchronisation against
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* the clock it must unblock when going from PLAYING to the PAUSED state and call
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* this method before continuing to produce the remaining data.
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* If the #GstBaseSrcClass.create() method performs its own synchronisation
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* against the clock it must unblock when going from PLAYING to the PAUSED state
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* and call this method before continuing to produce the remaining data.
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*
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* This function will block until a state change to PLAYING happens (in which
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* case this function returns #GST_FLOW_OK) or the processing must be stopped due
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@ -544,7 +546,7 @@ gst_base_src_is_live (GstBaseSrc * src)
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* for sending NEW_SEGMENT events and for performing seeks.
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*
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* If a format of GST_FORMAT_BYTES is set, the element will be able to
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* operate in pull mode if the #GstBaseSrc::is_seekable returns TRUE.
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* operate in pull mode if the #GstBaseSrc.is_seekable() returns TRUE.
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*
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* Since: 0.10.1
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*/
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