Commit 76cd3ff1 authored by Nicolas Dufresne's avatar Nicolas Dufresne Committed by GStreamer Marge Bot
Browse files

doc: base: Fix reference to virtual function

The hotdoc syntax is #ClassName::function, but the code was using
without anything before.

Part-of: <!808>
parent 9efd519c
Pipeline #318128 passed with stages
in 30 minutes and 23 seconds
......@@ -30,12 +30,12 @@
* Control is given to the subclass when all pads have data.
*
* * Base class for mixers and muxers. Subclasses should at least implement
* the #GstAggregatorClass.aggregate() virtual method.
* the #GstAggregatorClass::aggregate virtual method.
*
* * Installs a #GstPadChainFunction, a #GstPadEventFullFunction and a
* #GstPadQueryFunction to queue all serialized data packets per sink pad.
* Subclasses should not overwrite those, but instead implement
* #GstAggregatorClass.sink_event() and #GstAggregatorClass.sink_query() as
* #GstAggregatorClass::sink_event and #GstAggregatorClass::sink_query as
* needed.
*
* * When data is queued on all pads, the aggregate vmethod is called.
......@@ -1282,7 +1282,7 @@ gst_aggregator_negotiate_unlocked (GstAggregator * self)
*
* Negotiates src pad caps with downstream elements.
* Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
* if #GstAggregatorClass.negotiate() fails.
* if #GstAggregatorClass::negotiate fails.
*
* Returns: %TRUE if the negotiation succeeded, else %FALSE.
*
......@@ -3664,7 +3664,7 @@ gst_aggregator_get_allocator (GstAggregator * self,
* gst_aggregator_simple_get_next_time:
* @self: A #GstAggregator
*
* This is a simple #GstAggregatorClass.get_next_time() implementation that
* This is a simple #GstAggregatorClass::get_next_time implementation that
* just looks at the #GstSegment on the srcpad of the aggregator and bases
* the next time on the running time there.
*
......
......@@ -47,10 +47,10 @@
*
* ## Set-up phase
*
* * #GstBaseParse calls #GstBaseParseClass.start() to inform subclass
* * #GstBaseParse calls #GstBaseParseClass::start to inform subclass
* that data processing is about to start now.
*
* * #GstBaseParse class calls #GstBaseParseClass.set_sink_caps() to
* * #GstBaseParse class calls #GstBaseParseClass::set_sink_caps to
* inform the subclass about incoming sinkpad caps. Subclass could
* already set the srcpad caps accordingly, but this might be delayed
* until calling gst_base_parse_finish_frame() with a non-queued frame.
......@@ -69,11 +69,11 @@
* #GstAdapter.
*
* * A buffer of (at least) min_frame_size bytes is passed to subclass
* with #GstBaseParseClass.handle_frame(). Subclass checks the contents
* with #GstBaseParseClass::handle_frame. Subclass checks the contents
* and can optionally return #GST_FLOW_OK along with an amount of data
* to be skipped to find a valid frame (which will result in a
* subsequent DISCONT). If, otherwise, the buffer does not hold a
* complete frame, #GstBaseParseClass.handle_frame() can merely return
* complete frame, #GstBaseParseClass::handle_frame can merely return
* and will be called again when additional data is available. In push
* mode this amounts to an additional input buffer (thus minimal
* additional latency), in pull mode this amounts to some arbitrary
......@@ -88,7 +88,7 @@
* data while simultaneously providing custom output data. Note that
* baseclass performs some processing (such as tracking overall consumed
* data rate versus duration) for each finished frame, but other state
* is only updated upon each call to #GstBaseParseClass.handle_frame()
* is only updated upon each call to #GstBaseParseClass::handle_frame
* (such as tracking upstream input timestamp).
*
* Subclass is also responsible for setting the buffer metadata
......@@ -99,30 +99,30 @@
* duration obtained from configuration (see below), and offset
* if meaningful (in pull mode).
*
* Note that #GstBaseParseClass.handle_frame() might receive any small
* Note that #GstBaseParseClass::handle_frame might receive any small
* amount of input data when leftover data is being drained (e.g. at
* EOS).
*
* * As part of finish frame processing, just prior to actually pushing
* the buffer in question, it is passed to
* #GstBaseParseClass.pre_push_frame() which gives subclass yet one last
* #GstBaseParseClass::pre_push_frame which gives subclass yet one last
* chance to examine buffer metadata, or to send some custom (tag)
* events, or to perform custom (segment) filtering.
*
* * During the parsing process #GstBaseParseClass will handle both srcpad
* and sinkpad events. They will be passed to subclass if
* #GstBaseParseClass.sink_event() or #GstBaseParseClass.src_event()
* #GstBaseParseClass::sink_event or #GstBaseParseClass::src_event
* implementations have been provided.
*
* ## Shutdown phase
*
* * #GstBaseParse class calls #GstBaseParseClass.stop() to inform the
* * #GstBaseParse class calls #GstBaseParseClass::stop to inform the
* subclass that data parsing will be stopped.
*
* Subclass is responsible for providing pad template caps for source and
* sink pads. The pads need to be named "sink" and "src". It also needs to
* set the fixed caps on srcpad, when the format is ensured (e.g. when
* base class calls subclass' #GstBaseParseClass.set_sink_caps() function).
* base class calls subclass' #GstBaseParseClass::set_sink_caps function).
*
* This base class uses %GST_FORMAT_DEFAULT as a meaning of frames. So,
* subclass conversion routine needs to know that conversion from
......@@ -142,7 +142,7 @@
* * Inform base class how big data chunks should be retrieved. This is
* done with gst_base_parse_set_min_frame_size() function.
* * Examine data chunks passed to subclass with
* #GstBaseParseClass.handle_frame() and pass proper frame(s) to
* #GstBaseParseClass::handle_frame and pass proper frame(s) to
* gst_base_parse_finish_frame(), and setting src pad caps and timestamps
* on frame.
* * Provide conversion functions
......@@ -158,7 +158,7 @@
* metadata (which will be taken from upstream as much as
* possible). Internally keeping track of frame durations and respective
* sizes that have been pushed provides #GstBaseParse with an estimated
* bitrate. A default #GstBaseParseClass.convert() (used if not
* bitrate. A default #GstBaseParseClass::convert (used if not
* overridden) will then use these rates to perform obvious conversions.
* These rates are also used to update (estimated) duration at regular
* frame intervals.
......@@ -1742,7 +1742,7 @@ gst_base_parse_src_event_default (GstBaseParse * parse, GstEvent * event)
* @dest_format: #GstFormat defining the converted format.
* @dest_value: (out): Pointer where the conversion result will be put.
*
* Default implementation of #GstBaseParseClass.convert().
* Default implementation of #GstBaseParseClass::convert.
*
* Returns: %TRUE if conversion was successful.
*/
......@@ -4014,10 +4014,10 @@ gst_base_parse_set_syncable (GstBaseParse * parse, gboolean syncable)
* Set if the nature of the format or configuration does not allow (much)
* parsing, and the parser should operate in passthrough mode (which only
* applies when operating in push mode). That is, incoming buffers are
* pushed through unmodified, i.e. no #GstBaseParseClass.handle_frame()
* will be invoked, but #GstBaseParseClass.pre_push_frame() will still be
* pushed through unmodified, i.e. no #GstBaseParseClass::handle_frame
* will be invoked, but #GstBaseParseClass::pre_push_frame will still be
* invoked, so subclass can perform as much or as little is appropriate for
* passthrough semantics in #GstBaseParseClass.pre_push_frame().
* passthrough semantics in #GstBaseParseClass::pre_push_frame.
*/
void
gst_base_parse_set_passthrough (GstBaseParse * parse, gboolean passthrough)
......
......@@ -55,8 +55,8 @@ G_BEGIN_DECLS
* GST_BASE_PARSE_FLOW_DROPPED:
*
* A #GstFlowReturn that can be returned from
* #GstBaseParseClass.handle_frame() to indicate that no output buffer was
* generated, or from #GstBaseParseClass.pre_push_frame() to to forego
* #GstBaseParseClass::handle_frame to indicate that no output buffer was
* generated, or from #GstBaseParseClass::pre_push_frame to to forego
* pushing buffer.
*/
#define GST_BASE_PARSE_FLOW_DROPPED GST_FLOW_CUSTOM_SUCCESS
......
......@@ -59,19 +59,19 @@
* #GstBaseSink will handle the prerolling correctly. This means that it will
* return %GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
* buffer arrives in this element. The base class will call the
* #GstBaseSinkClass.preroll() vmethod with this preroll buffer and will then
* #GstBaseSinkClass::preroll vmethod with this preroll buffer and will then
* commit the state change to the next asynchronously pending state.
*
* When the element is set to PLAYING, #GstBaseSink will synchronise on the
* clock using the times returned from #GstBaseSinkClass.get_times(). If this
* clock using the times returned from #GstBaseSinkClass::get_times. If this
* function returns %GST_CLOCK_TIME_NONE for the start time, no synchronisation
* will be done. Synchronisation can be disabled entirely by setting the object
* #GstBaseSink:sync property to %FALSE.
*
* After synchronisation the virtual method #GstBaseSinkClass.render() will be
* After synchronisation the virtual method #GstBaseSinkClass::render will be
* called. Subclasses should minimally implement this method.
*
* Subclasses that synchronise on the clock in the #GstBaseSinkClass.render()
* Subclasses that synchronise on the clock in the #GstBaseSinkClass::render
* method are supported as well. These classes typically receive a buffer in
* the render method and can then potentially block on the clock while
* rendering. A typical example is an audiosink.
......@@ -80,7 +80,7 @@
*
* Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait
* for the clock to reach the time indicated by the stop time of the last
* #GstBaseSinkClass.get_times() call before posting an EOS message. When the
* #GstBaseSinkClass::get_times call before posting an EOS message. When the
* element receives EOS in PAUSED, preroll completes, the event is queued and an
* EOS message is posted when going to PLAYING.
*
......@@ -95,24 +95,24 @@
* If no clock has been set on the element, the query will be forwarded
* upstream.
*
* The #GstBaseSinkClass.set_caps() function will be called when the subclass
* The #GstBaseSinkClass::set_caps function will be called when the subclass
* should configure itself to process a specific media type.
*
* The #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() virtual methods
* The #GstBaseSinkClass::start and #GstBaseSinkClass::stop virtual methods
* will be called when resources should be allocated. Any
* #GstBaseSinkClass.preroll(), #GstBaseSinkClass.render() and
* #GstBaseSinkClass.set_caps() function will be called between the
* #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() calls.
* #GstBaseSinkClass::preroll, #GstBaseSinkClass::render and
* #GstBaseSinkClass::set_caps function will be called between the
* #GstBaseSinkClass::start and #GstBaseSinkClass::stop calls.
*
* The #GstBaseSinkClass.event() virtual method will be called when an event is
* The #GstBaseSinkClass::event virtual method will be called when an event is
* received by #GstBaseSink. Normally this method should only be overridden by
* very specific elements (such as file sinks) which need to handle the
* newsegment event specially.
*
* The #GstBaseSinkClass.unlock() method is called when the elements should
* The #GstBaseSinkClass::unlock method is called when the elements should
* unblock any blocking operations they perform in the
* #GstBaseSinkClass.render() method. This is mostly useful when the
* #GstBaseSinkClass.render() method performs a blocking write on a file
* #GstBaseSinkClass::render method. This is mostly useful when the
* #GstBaseSinkClass::render method performs a blocking write on a file
* descriptor, for example.
*
* The #GstBaseSink:max-lateness property affects how the sink deals with
......@@ -123,7 +123,7 @@
* If the frame is later than max-lateness, the sink will drop the buffer
* without calling the render method.
* This feature is disabled if sync is disabled, the
* #GstBaseSinkClass.get_times() method does not return a valid start time or
* #GstBaseSinkClass::get_times method does not return a valid start time or
* max-lateness is set to -1 (the default).
* Subclasses can use gst_base_sink_set_max_lateness() to configure the
* max-lateness value.
......@@ -2305,9 +2305,9 @@ gst_base_sink_adjust_time (GstBaseSink * basesink, GstClockTime time)
* is no clock, no synchronisation is done and %GST_CLOCK_BADTIME is returned.
*
* This function should only be called with the PREROLL_LOCK held, like when
* receiving an EOS event in the #GstBaseSinkClass.event() vmethod or when
* receiving an EOS event in the #GstBaseSinkClass::event vmethod or when
* receiving a buffer in
* the #GstBaseSinkClass.render() vmethod.
* the #GstBaseSinkClass::render vmethod.
*
* The @time argument should be the running_time of when this method should
* return and is not adjusted with any latency or offset configured in the
......@@ -2395,14 +2395,14 @@ no_clock:
* gst_base_sink_wait_preroll:
* @sink: the sink
*
* If the #GstBaseSinkClass.render() method performs its own synchronisation
* If the #GstBaseSinkClass::render method performs its own synchronisation
* against the clock it must unblock when going from PLAYING to the PAUSED state
* and call this method before continuing to render the remaining data.
*
* If the #GstBaseSinkClass.render() method can block on something else than
* If the #GstBaseSinkClass::render method can block on something else than
* the clock, it must also be ready to unblock immediately on
* the #GstBaseSinkClass.unlock() method and cause the
* #GstBaseSinkClass.render() method to immediately call this function.
* the #GstBaseSinkClass::unlock method and cause the
* #GstBaseSinkClass::render method to immediately call this function.
* In this case, the subclass must be prepared to continue rendering where it
* left off if this function returns %GST_FLOW_OK.
*
......
......@@ -125,10 +125,10 @@ struct _GstBaseSink {
* @unlock: Unlock any pending access to the resource. Subclasses should
* unblock any blocked function ASAP and call gst_base_sink_wait_preroll()
* @unlock_stop: Clear the previous unlock request. Subclasses should clear
* any state they set during #GstBaseSinkClass.unlock(), and be ready to
* any state they set during #GstBaseSinkClass::unlock, and be ready to
* continue where they left off after gst_base_sink_wait_preroll(),
* gst_base_sink_wait() or gst_wait_sink_wait_clock() return or
* #GstBaseSinkClass.render() is called again.
* #GstBaseSinkClass::render is called again.
* @query: perform a #GstQuery on the element.
* @event: Override this to handle events arriving on the sink pad
* @wait_event: Override this to implement custom logic to wait for the event
......
......@@ -42,25 +42,25 @@
* conditions are met, it also supports pull mode scheduling:
*
* * The format is set to %GST_FORMAT_BYTES (default).
* * #GstBaseSrcClass.is_seekable() returns %TRUE.
* * #GstBaseSrcClass::is_seekable returns %TRUE.
*
* If all the conditions are met for operating in pull mode, #GstBaseSrc is
* automatically seekable in push mode as well. The following conditions must
* be met to make the element seekable in push mode when the format is not
* %GST_FORMAT_BYTES:
*
* * #GstBaseSrcClass.is_seekable() returns %TRUE.
* * #GstBaseSrcClass.query() can convert all supported seek formats to the
* * #GstBaseSrcClass::is_seekable returns %TRUE.
* * #GstBaseSrcClass::query can convert all supported seek formats to the
* internal format as set with gst_base_src_set_format().
* * #GstBaseSrcClass.do_seek() is implemented, performs the seek and returns
* * #GstBaseSrcClass::do_seek is implemented, performs the seek and returns
* %TRUE.
*
* When the element does not meet the requirements to operate in pull mode, the
* offset and length in the #GstBaseSrcClass.create() method should be ignored.
* offset and length in the #GstBaseSrcClass::create method should be ignored.
* It is recommended to subclass #GstPushSrc instead, in this situation. If the
* element can operate in pull mode but only with specific offsets and
* lengths, it is allowed to generate an error when the wrong values are passed
* to the #GstBaseSrcClass.create() function.
* to the #GstBaseSrcClass::create function.
*
* #GstBaseSrc has support for live sources. Live sources are sources that when
* paused discard data, such as audio or video capture devices. A typical live
......@@ -69,7 +69,7 @@
* Use gst_base_src_set_live() to activate the live source mode.
*
* A live source does not produce data in the PAUSED state. This means that the
* #GstBaseSrcClass.create() method will not be called in PAUSED but only in
* #GstBaseSrcClass::create method will not be called in PAUSED but only in
* PLAYING. To signal the pipeline that the element will not produce data, the
* return value from the READY to PAUSED state will be
* %GST_STATE_CHANGE_NO_PREROLL.
......@@ -81,12 +81,12 @@
*
* Live sources that synchronize and block on the clock (an audio source, for
* example) can use gst_base_src_wait_playing() when the
* #GstBaseSrcClass.create() function was interrupted by a state change to
* #GstBaseSrcClass::create function was interrupted by a state change to
* PAUSED.
*
* The #GstBaseSrcClass.get_times() method can be used to implement pseudo-live
* sources. It only makes sense to implement the #GstBaseSrcClass.get_times()
* function if the source is a live source. The #GstBaseSrcClass.get_times()
* The #GstBaseSrcClass::get_times method can be used to implement pseudo-live
* sources. It only makes sense to implement the #GstBaseSrcClass::get_times
* function if the source is a live source. The #GstBaseSrcClass::get_times
* function should return timestamps starting from 0, as if it were a non-live
* source. The base class will make sure that the timestamps are transformed
* into the current running_time. The base source will then wait for the
......@@ -533,7 +533,7 @@ flushing:
* gst_base_src_wait_playing:
* @src: the src
*
* If the #GstBaseSrcClass.create() method performs its own synchronisation
* If the #GstBaseSrcClass::create method performs its own synchronisation
* against the clock it must unblock when going from PLAYING to the PAUSED state
* and call this method before continuing to produce the remaining data.
*
......@@ -614,7 +614,7 @@ gst_base_src_is_live (GstBaseSrc * src)
* for sending SEGMENT events and for performing seeks.
*
* If a format of GST_FORMAT_BYTES is set, the element will be able to
* operate in pull mode if the #GstBaseSrcClass.is_seekable() returns %TRUE.
* operate in pull mode if the #GstBaseSrcClass::is_seekable returns %TRUE.
*
* This function must only be called in states < %GST_STATE_PAUSED.
*/
......@@ -659,7 +659,7 @@ gst_base_src_set_dynamic_size (GstBaseSrc * src, gboolean dynamic)
* When @src operates in %GST_FORMAT_TIME, #GstBaseSrc will send an EOS
* when a buffer outside of the currently configured segment is pushed if
* @automatic_eos is %TRUE. Since 1.16, if @automatic_eos is %FALSE an
* EOS will be pushed only when the #GstBaseSrcClass.create() implementation
* EOS will be pushed only when the #GstBaseSrcClass::create implementation
* returns %GST_FLOW_EOS.
*
* Since: 1.4
......@@ -3462,10 +3462,10 @@ gst_base_src_negotiate_unlocked (GstBaseSrc * basesrc)
*
* Negotiates src pad caps with downstream elements.
* Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
* if #GstBaseSrcClass.negotiate() fails.
* if #GstBaseSrcClass::negotiate fails.
*
* Do not call this in the #GstBaseSrcClass.fill() vmethod. Call this in
* #GstBaseSrcClass.create() or in #GstBaseSrcClass.alloc(), _before_ any
* Do not call this in the #GstBaseSrcClass::fill vmethod. Call this in
* #GstBaseSrcClass::create or in #GstBaseSrcClass::alloc, _before_ any
* buffer is allocated.
*
* Returns: %TRUE if the negotiation succeeded, else %FALSE.
......
......@@ -135,7 +135,7 @@ struct _GstBaseSrc {
* gst_base_src_set_format().
* @is_seekable: Check if the source can seek
* @prepare_seek_segment: Prepare the #GstSegment that will be passed to the
* #GstBaseSrcClass.do_seek() vmethod for executing a seek
* #GstBaseSrcClass::do_seek vmethod for executing a seek
* request. Sub-classes should override this if they support seeking in
* formats other than the configured native format. By default, it tries to
* convert the seek arguments to the configured native format and prepare a
......@@ -144,11 +144,11 @@ struct _GstBaseSrc {
* @unlock: Unlock any pending access to the resource. Subclasses should unblock
* any blocked function ASAP. In particular, any `create()` function in
* progress should be unblocked and should return GST_FLOW_FLUSHING. Any
* future #GstBaseSrcClass.create() function call should also return
* GST_FLOW_FLUSHING until the #GstBaseSrcClass.unlock_stop() function has
* future #GstBaseSrcClass::create function call should also return
* GST_FLOW_FLUSHING until the #GstBaseSrcClass::unlock_stop function has
* been called.
* @unlock_stop: Clear the previous unlock request. Subclasses should clear any
* state they set during #GstBaseSrcClass.unlock(), such as clearing command
* state they set during #GstBaseSrcClass::unlock, such as clearing command
* queues.
* @query: Handle a requested query.
* @event: Override this to implement custom event handling.
......@@ -158,7 +158,7 @@ struct _GstBaseSrc {
* buffer should be returned when the return value is different from
* GST_FLOW_OK. A return value of GST_FLOW_EOS signifies that the end of
* stream is reached. The default implementation will call
* #GstBaseSrcClass.alloc() and then call #GstBaseSrcClass.fill().
* #GstBaseSrcClass::alloc and then call #GstBaseSrcClass::fill.
* @alloc: Ask the subclass to allocate a buffer with for offset and size. The
* default implementation will create a new buffer from the negotiated allocator.
* @fill: Ask the subclass to fill the buffer with data for offset and size. The
......
......@@ -1452,15 +1452,15 @@ done:
* marked as needing reconfiguring. Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in
* any case. But marks it again if negotiation fails.
*
* Do not call this in the #GstBaseTransformClass.transform() or
* #GstBaseTransformClass.transform_ip() vmethod. Call this in
* #GstBaseTransformClass.submit_input_buffer(),
* #GstBaseTransformClass.prepare_output_buffer() or in
* #GstBaseTransformClass.generate_output() _before_ any output buffer is
* Do not call this in the #GstBaseTransformClass::transform or
* #GstBaseTransformClass::transform_ip vmethod. Call this in
* #GstBaseTransformClass::submit_input_buffer,
* #GstBaseTransformClass::prepare_output_buffer or in
* #GstBaseTransformClass::generate_output _before_ any output buffer is
* allocated.
*
* It will be default be called when handling an ALLOCATION query or at the
* very beginning of the default #GstBaseTransformClass.submit_input_buffer()
* very beginning of the default #GstBaseTransformClass::submit_input_buffer
* implementation.
*
* Returns: %TRUE if the negotiation succeeded, else %FALSE.
......
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