gnomemm r1426 - in gstreamermm/trunk: . gstreamer/src gstreamerbase/src
- From: jaalburqu svn gnome org
- To: svn-commits-list gnome org
- Subject: gnomemm r1426 - in gstreamermm/trunk: . gstreamer/src gstreamerbase/src
- Date: Thu, 27 Mar 2008 00:01:27 +0000 (GMT)
Author: jaalburqu
Date: Thu Mar 27 00:01:26 2008
New Revision: 1426
URL: http://svn.gnome.org/viewvc/gnomemm?rev=1426&view=rev
Log:
2008-03-26 Josà Alburquerque <jaalburqu svn gnome org>
* gstreamer/src/gst_docs.xml:
* gstreamerbase/src/gstbase_docs.xml: Generated docs with all
directories that includes API that may be wrapped
Modified:
gstreamermm/trunk/ChangeLog
gstreamermm/trunk/gstreamer/src/gst_docs.xml
gstreamermm/trunk/gstreamerbase/src/gstbase_docs.xml
Modified: gstreamermm/trunk/gstreamer/src/gst_docs.xml
==============================================================================
--- gstreamermm/trunk/gstreamer/src/gst_docs.xml (original)
+++ gstreamermm/trunk/gstreamer/src/gst_docs.xml Thu Mar 27 00:01:26 2008
@@ -133,19 +133,27 @@
</return>
</function>
-<function name="gst_element_factory_get_description">
+<function name="gst_buffer_join">
<description>
-Gets the description for this factory.
+Create a new buffer that is the concatenation of the two source
+buffers, and unrefs the original source buffers.
+
+If the buffers point to contiguous areas of memory, the buffer
+is created without copying the data.
</description>
<parameters>
-<parameter name="factory">
-<parameter_description> a #GstElementFactory
+<parameter name="buf1">
+<parameter_description> the first source #GstBuffer.
+</parameter_description>
+</parameter>
+<parameter name="buf2">
+<parameter_description> the second source #GstBuffer.
</parameter_description>
</parameter>
</parameters>
-<return> the description
+<return> the new #GstBuffer which is the concatenation of the source buffers.
</return>
</function>
@@ -315,6 +323,27 @@
</return>
</function>
+<function name="gst_object_set_controller">
+<description>
+Sets the controller on the given GObject
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that should get the controller
+</parameter_description>
+</parameter>
+<parameter name="controller">
+<parameter_description> the controller object to plug in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if the GObject already has an controller, %TRUE otherwise
+Since: 0.9
+</return>
+</function>
+
<function name="gst_trace_read_tsc">
<description>
Read a platform independent timer value that can be used in
@@ -627,27 +656,50 @@
</return>
</function>
-<function name="gst_buffer_join">
+<function name="gst_element_factory_get_description">
<description>
-Create a new buffer that is the concatenation of the two source
-buffers, and unrefs the original source buffers.
+Gets the description for this factory.
-If the buffers point to contiguous areas of memory, the buffer
-is created without copying the data.
+</description>
+<parameters>
+<parameter name="factory">
+<parameter_description> a #GstElementFactory
+</parameter_description>
+</parameter>
+</parameters>
+<return> the description
+</return>
+</function>
+
+<function name="gst_adapter_peek">
+<description>
+Gets the first @size bytes stored in the @adapter. The returned pointer is
+valid until the next function is called on the adapter.
+
+Note that setting the returned pointer as the data of a #GstBuffer is
+incorrect for general-purpose plugins. The reason is that if a downstream
+element stores the buffer so that it has access to it outside of the bounds
+of its chain function, the buffer will have an invalid data pointer after
+your element flushes the bytes. In that case you should use
+gst_adapter_take(), which returns a freshly-allocated buffer that you can set
+as #GstBuffer malloc_data or the potentially more performant
+gst_adapter_take_buffer().
+
+Returns: a pointer to the first @size bytes of data, or NULL.
</description>
<parameters>
-<parameter name="buf1">
-<parameter_description> the first source #GstBuffer.
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
</parameter_description>
</parameter>
-<parameter name="buf2">
-<parameter_description> the second source #GstBuffer.
+<parameter name="size">
+<parameter_description> the number of bytes to peek
</parameter_description>
</parameter>
</parameters>
-<return> the new #GstBuffer which is the concatenation of the source buffers.
+<return> a pointer to the first @size bytes of data, or NULL.
</return>
</function>
@@ -741,6 +793,83 @@
</return>
</function>
+<function name="gst_control_source_get_value_array">
+<description>
+Gets an array of values for one element property.
+
+All fields of @value_array must be filled correctly. Especially the
+ value_array-&gt;values array must be big enough to keep the requested amount
+of values.
+
+The type of the values in the array is the same as the property's type.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstControlSource object
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+<parameter name="value_array">
+<parameter_description> array to put control-values in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the given array could be filled, %FALSE otherwise
+</return>
+</function>
+
+<function name="gst_controller_set_from_list">
+<description>
+Sets multiple timed values at once.
+
+Deprecated: Use #GstControlSource, for example #GstInterpolationControlSource
+directly.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object which handles the properties
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property to set
+</parameter_description>
+</parameter>
+<parameter name="timedvalues">
+<parameter_description> a list with #GstTimedValue items
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if the values couldn't be set (ex : properties not handled by controller), %TRUE otherwise
+</return>
+</function>
+
+<function name="gst_controller_new_valist">
+<description>
+Creates a new GstController for the given object's properties
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object of which some properties should be controlled
+</parameter_description>
+</parameter>
+<parameter name="var_args">
+<parameter_description> %NULL terminated list of property names that should be controlled
+</parameter_description>
+</parameter>
+</parameters>
+<return> the new controller.
+</return>
+</function>
+
<function name="gst_plugin_get_module">
<description>
Gets the #GModule of the plugin. If the plugin isn't loaded yet, NULL is
@@ -823,6 +952,39 @@
</return>
</function>
+<function name="gst_adapter_copy">
+<description>
+Copies @size bytes of data starting at @offset out of the buffers
+contained in @GstAdapter into an array @dest provided by the caller.
+
+The array @dest should be large enough to contain @size bytes.
+The user should check that the adapter has (@offset + @size) bytes
+available before calling this function.
+
+Since: 0.10.12
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+<parameter name="dest">
+<parameter_description> the memory where to copy to
+</parameter_description>
+</parameter>
+<parameter name="offset">
+<parameter_description> the bytes offset in the adapter to start from
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the number of bytes to copy
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_parse_bin_from_description">
<description>
This is a convenience wrapper around gst_parse_launch() to create a
@@ -883,6 +1045,45 @@
</return>
</function>
+<function name="gst_type_find_helper">
+<description>
+Tries to find what type of data is flowing from the given source #GstPad.
+
+Returns #NULL if no #GstCaps matches the data stream.
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> A source #GstPad
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> The length in bytes
+</parameter_description>
+</parameter>
+</parameters>
+<return>#NULL if no #GstCaps matches the data stream.
+</return>
+</function>
+
+<function name="gst_base_transform_is_passthrough">
+<description>
+See if @trans is configured as a passthrough transform.
+
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> the #GstBaseTransform to query
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE is the transform is configured in passthrough mode.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_segtrap_set_enabled">
<description>
Applications might want to disable/enable the SIGSEGV handling of
@@ -900,6 +1101,40 @@
<return></return>
</function>
+<function name="gst_dp_packet_from_caps">
+<description>
+Creates a GDP packet from the given caps.
+
+Deprecated: use a #GstDPPacketizer
+
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description> a #GstCaps to create a packet for
+</parameter_description>
+</parameter>
+<parameter name="flags">
+<parameter_description> the #GDPHeaderFlags to create the header with
+</parameter_description>
+</parameter>
+<parameter name="length">
+<parameter_description> a guint pointer to store the header length in
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> a guint8 pointer to store a newly allocated header byte array in
+</parameter_description>
+</parameter>
+<parameter name="payload">
+<parameter_description> a guint8 pointer to store a newly allocated payload byte array in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the packet was successfully created.
+</return>
+</function>
+
<function name="gst_message_new_state_changed">
<description>
Create a state change message. This message is posted whenever an element
@@ -1056,6 +1291,30 @@
</return>
</function>
+<function name="gst_interpolation_control_source_set">
+<description>
+Set the value of given controller-handled property at a certain time.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource object
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time the control-change is scheduled for
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> the control-value
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if the values couldn't be set, TRUE otherwise.
+</return>
+</function>
+
<function name="gst_static_pad_template_get">
<description>
Converts a #GstStaticPadTemplate into a #GstPadTemplate.
@@ -1127,41 +1386,25 @@
<return></return>
</function>
-<function name="gst_clock_id_wait">
+<function name="gst_value_take_mini_object">
<description>
-Perform a blocking wait on @id.
- id should have been created with gst_clock_new_single_shot_id()
-or gst_clock_new_periodic_id() and should not have been unscheduled
-with a call to gst_clock_id_unschedule().
-
-If the @jitter argument is not NULL and this function returns #GST_CLOCK_OK
-or #GST_CLOCK_EARLY, it will contain the difference
-against the clock and the time of @id when this method was
-called.
-Positive values indicate how late @id was relative to the clock
-(in which case this function will return #GST_CLOCK_EARLY).
-Negative values indicate how much time was spent waiting on the clock
-before this function returned.
-
+Set the contents of a %GST_TYPE_MINI_OBJECT derived #GValue to
+ mini_object
+Takes over the ownership of the caller's reference to @mini_object;
+the caller doesn't have to unref it any more.
</description>
<parameters>
-<parameter name="id">
-<parameter_description> The #GstClockID to wait on
+<parameter name="value">
+<parameter_description> a valid #GValue of %GST_TYPE_MINI_OBJECT derived type
</parameter_description>
</parameter>
-<parameter name="jitter">
-<parameter_description> A pointer that will contain the jitter, can be NULL.
+<parameter name="mini_object">
+<parameter_description> mini object value to take
</parameter_description>
</parameter>
</parameters>
-<return> the result of the blocking wait. #GST_CLOCK_EARLY will be returned
-if the current clock time is past the time of @id, #GST_CLOCK_OK if
- id was scheduled in time. #GST_CLOCK_UNSCHEDULED if @id was
-unscheduled with gst_clock_id_unschedule().
-
-MT safe.
-</return>
+<return></return>
</function>
<function name="gst_query_set_formatsv">
@@ -1208,36 +1451,54 @@
</return>
</function>
-<function name="gst_bin_iterate_sorted">
+<function name="gst_poll_restart">
<description>
-Gets an iterator for the elements in this bin in topologically
-sorted order. This means that the elements are returned from
-the most downstream elements (sinks) to the sources.
+Restart any gst_poll_wait() that is in progress. This function is typically
+used after adding or removing descriptors to @set.
-This function is used internally to perform the state changes
-of the bin elements and for clock selection.
-
-Each element yielded by the iterator will have its refcount increased, so
-unref after use.
-
-MT safe. Caller owns returned value.
+If @set is not controllable, then this call will have no effect.
+Since: 0.10.18
</description>
<parameters>
-<parameter name="bin">
-<parameter_description> a #GstBin
+<parameter name="set">
+<parameter_description> a #GstPoll.
</parameter_description>
</parameter>
</parameters>
-<return> a #GstIterator of #GstElement, or NULL
-</return>
+<return></return>
</function>
-<function name="gst_structure_from_string">
+<function name="gst_object_replace">
<description>
-Creates a #GstStructure from a string representation.
-If end is not NULL, a pointer to the place inside the given string
+Unrefs the #GstObject pointed to by @oldobj, refs @newobj and
+puts @newobj in * oldobj Be carefull when calling this
+function, it does not take any locks. You might want to lock
+the object owning @oldobj pointer before calling this
+function.
+
+Make sure not to LOCK @oldobj because it might be unreffed
+which could cause a deadlock when it is disposed.
+
+</description>
+<parameters>
+<parameter name="oldobj">
+<parameter_description> pointer to a place of a #GstObject to replace
+</parameter_description>
+</parameter>
+<parameter name="newobj">
+<parameter_description> a new #GstObject
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_structure_from_string">
+<description>
+Creates a #GstStructure from a string representation.
+If end is not NULL, a pointer to the place inside the given string
where parsing ended will be returned.
@@ -1356,6 +1617,29 @@
</return>
</function>
+<function name="gst_base_src_wait_playing">
+<description>
+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.
+
+This function will block until a state change to PLAYING happens (in which
+case this function returns #GST_FLOW_OK) or the processing must be stopped due
+to a state change to READY or a FLUSH event (in which case this function
+Returns: #GST_FLOW_OK if @src is PLAYING and processing can
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> the src
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_FLOW_OK if @src is PLAYING and processing can
+continue. Any other return value should be returned from the create vmethod.
+</return>
+</function>
+
<function name="gst_element_query_convert">
<description>
Queries an element to convert @src_val in @src_format to @dest_format.
@@ -1388,6 +1672,19 @@
</return>
</function>
+<function name="gst_check_drop_buffers">
+<description>
+Unref and remove all buffers that are in the global @buffers GList,
+emptying the list.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_tag_list_insert">
<description>
Inserts the tags of the second list into the first list using the given mode.
@@ -1430,6 +1727,64 @@
<return></return>
</function>
+<function name="gst_dp_packet_from_event">
+<description>
+Creates a GDP packet from the given event.
+
+Deprecated: use a #GstDPPacketizer
+
+
+</description>
+<parameters>
+<parameter name="event">
+<parameter_description> a #GstEvent to create a packet for
+</parameter_description>
+</parameter>
+<parameter name="flags">
+<parameter_description> the #GDPHeaderFlags to create the header with
+</parameter_description>
+</parameter>
+<parameter name="length">
+<parameter_description> a guint pointer to store the header length in
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> a guint8 pointer to store a newly allocated header byte array in
+</parameter_description>
+</parameter>
+<parameter name="payload">
+<parameter_description> a guint8 pointer to store a newly allocated payload byte array in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the packet was successfully created.
+</return>
+</function>
+
+<function name="gst_dp_validate_packet">
+<description>
+Validates the given packet by checking version information and checksums.
+
+
+</description>
+<parameters>
+<parameter name="header_length">
+<parameter_description> the length of the packet header
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> the byte array of the packet header
+</parameter_description>
+</parameter>
+<parameter name="payload">
+<parameter_description> the byte array of the packet payload
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the packet validates.
+</return>
+</function>
+
<function name="gst_tag_merge_strings_with_comma">
<description>
This is a convenience function for the func argument of gst_tag_register().
@@ -1489,6 +1844,25 @@
</return>
</function>
+<function name="gst_base_sink_is_qos_enabled">
+<description>
+Checks if @sink is currently configured to send Quality-of-Service events
+upstream.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the sink is configured to perform Quality-of-Service.
+
+Since: 0.10.5
+</return>
+</function>
+
<function name="gst_type_find_factory_get_caps">
<description>
Gets the #GstCaps associated with a typefind factory.
@@ -1809,6 +2183,22 @@
</return>
</function>
+<function name="gst_dp_header_payload_length">
+<description>
+Get the length of the payload described by @header.
+
+
+</description>
+<parameters>
+<parameter name="header">
+<parameter_description> the byte header of the packet array
+</parameter_description>
+</parameter>
+</parameters>
+<return> the length of the payload this header describes.
+</return>
+</function>
+
<function name="gst_uri_handler_set_uri">
<description>
Tries to set the URI of the given handler.
@@ -1872,6 +2262,21 @@
</return>
</function>
+<function name="gst_interpolation_control_source_unset_all">
+<description>
+Used to remove all time-stamped values of given controller-handled property
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource object
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_is_tag_list">
<description>
Checks if the given pointer is a taglist.
@@ -1888,6 +2293,49 @@
</return>
</function>
+<function name="gst_base_sink_query_latency">
+<description>
+Query the sink for the latency parameters. The latency will be queried from
+the upstream elements. @live will be TRUE if @sink is configured to
+synchronize against the clock. @upstream_live will be TRUE if an upstream
+element is live.
+
+If both @live and @upstream_live are TRUE, the sink will want to compensate
+for the latency introduced by the upstream elements by setting the
+ min_latency to a strictly possitive value.
+
+This function is mostly used by subclasses.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="live">
+<parameter_description> if the sink is live
+</parameter_description>
+</parameter>
+<parameter name="upstream_live">
+<parameter_description> if an upstream element is live
+</parameter_description>
+</parameter>
+<parameter name="min_latency">
+<parameter_description> the min latency of the upstream elements
+</parameter_description>
+</parameter>
+<parameter name="max_latency">
+<parameter_description> the max latency of the upstream elements
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the query succeeded.
+
+Since: 0.10.12
+</return>
+</function>
+
<function name="gst_plugin_get_version">
<description>
get the version of the plugin
@@ -2045,6 +2493,29 @@
</return>
</function>
+<function name="gst_base_sink_set_async_enabled">
+<description>
+Configures @sink to perform all state changes asynchronusly. When async is
+disabled, the sink will immediatly go to PAUSED instead of waiting for a
+preroll buffer. This feature is usefull if the sink does not synchronize
+against the clock or when it is dealing with sparse streams.
+
+Since: 0.10.15
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="enabled">
+<parameter_description> the new async value.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_error_get_message">
<description>
Get a string describing the error message in the current locale.
@@ -2145,6 +2616,44 @@
</return>
</function>
+<function name="gst_poll_remove_fd">
+<description>
+Remove a file descriptor from the file descriptor set.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the file descriptor was successfully removed from the set.
+
+Since: 0.10.18
+</return>
+</function>
+
+<function name="gst_interpolation_control_source_get_count">
+<description>
+Returns: the number of control points that are set.
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource to get the number of values from
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of control points that are set.
+
+</return>
+</function>
+
<function name="gst_buffer_make_metadata_writable">
<description>
Similar to gst_buffer_make_writable, but does not ensure that the buffer
@@ -2185,6 +2694,38 @@
</return>
</function>
+<function name="gst_controller_get_value_array">
+<description>
+Function to be able to get an array of values for one element property.
+
+All fields of @value_array must be filled correctly. Especially the
+ value_array-&gt;values array must be big enough to keep the requested amount
+of values.
+
+The type of the values in the array is the same as the property's type.
+
+&lt;note&gt;&lt;para&gt;This doesn't modify the controlled GObject property!&lt;/para&gt;&lt;/note&gt;
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller that handles the values
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+<parameter name="value_array">
+<parameter_description> array to put control-values in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the given array could be filled, %FALSE otherwise
+</return>
+</function>
+
<function name="gst_message_new_clock_lost">
<description>
Create a clock lost message. This message is posted whenever the
@@ -2212,6 +2753,43 @@
</return>
</function>
+<function name="gst_type_find_helper_get_range">
+<description>
+Utility function to do pull-based typefinding. Unlike gst_type_find_helper()
+however, this function will use the specified function @func to obtain the
+data needed by the typefind functions, rather than operating on a given
+source pad. This is useful mostly for elements like tag demuxers which
+strip off data at the beginning and/or end of a file and want to typefind
+the stripped data stream before adding their own source pad (the specified
+callback can then call the upstream peer pad with offsets adjusted for the
+tag size, for example).
+
+Returns #NULL if no #GstCaps matches the data stream.
+
+</description>
+<parameters>
+<parameter name="obj">
+<parameter_description> A #GstObject that will be passed as first argument to @func
+</parameter_description>
+</parameter>
+<parameter name="func">
+<parameter_description> A generic #GstTypeFindHelperGetRangeFunction that will be used
+to access data at random offsets when doing the typefinding
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> The length in bytes
+</parameter_description>
+</parameter>
+<parameter name="prob">
+<parameter_description> location to store the probability of the found caps, or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return>#NULL if no #GstCaps matches the data stream.
+</return>
+</function>
+
<function name="gst_structure_get_fraction">
<description>
Sets the integers pointed to by @value_numerator and @value_denominator
@@ -2665,6 +3243,27 @@
<return></return>
</function>
+<function name="gst_object_sync_values">
+<description>
+Convenience function for GObject
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+</parameters>
+<return> same thing as gst_controller_sync_values()
+Since: 0.9
+</return>
+</function>
+
<function name="gst_alloc_trace_list">
<description>
Get a list of all registered alloc trace objects.
@@ -2936,6 +3535,27 @@
</return>
</function>
+<function name="gst_adapter_flush">
+<description>
+Flushes the first @flush bytes in the @adapter. The caller must ensure that
+at least this many bytes are available.
+
+See also: gst_adapter_peek().
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+<parameter name="flush">
+<parameter_description> the number of bytes to flush
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_ghost_pad_new">
<description>
Create a new ghostpad with @target as the target. The direction will be taken
@@ -2982,9 +3602,35 @@
</return>
</function>
-<function name="gst_init">
+<function name="gst_object_control_properties">
<description>
-Initializes the GStreamer library, setting up internal path lists,
+Convenience function for GObject
+
+Creates a GstController that allows you to dynamically control one, or more, GObject properties.
+If the given GObject already has a GstController, it adds the given properties to the existing
+controller and returns that controller.
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object of which some properties should be controlled
+</parameter_description>
+</parameter>
+<parameter name="Varargs">
+<parameter_description> %NULL terminated list of property names that should be controlled
+</parameter_description>
+</parameter>
+</parameters>
+<return> The GstController with which the user can control the given properties dynamically or NULL if
+one or more of the given properties aren't available, or cannot be controlled, for the given element.
+Since: 0.9
+</return>
+</function>
+
+<function name="gst_init">
+<description>
+Initializes the GStreamer library, setting up internal path lists,
registering built-in elements, and loading standard plugins.
This function should be called before calling any other GLib functions. If
@@ -3064,6 +3710,33 @@
</return>
</function>
+<function name="gst_dp_buffer_from_header">
+<description>
+Creates a newly allocated #GstBuffer from the given header.
+The buffer data needs to be copied into it before validating.
+
+Use this function if you want to pre-allocate a buffer based on the
+packet header to read the packet payload in to.
+
+This function does not check the header passed to it, use
+gst_dp_validate_header() first if the header data is unchecked.
+
+
+</description>
+<parameters>
+<parameter name="header_length">
+<parameter_description> the length of the packet header
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> the byte array of the packet header
+</parameter_description>
+</parameter>
+</parameters>
+<return> A #GstBuffer if the buffer was successfully created, or NULL.
+</return>
+</function>
+
<function name="gst_bus_create_watch">
<description>
Create watch for this bus. The GSource will be dispatched whenever
@@ -3082,6 +3755,24 @@
</return>
</function>
+<function name="gst_controller_suggest_next_sync">
+<description>
+Returns: Returns the suggested timestamp or %GST_CLOCK_TIME_NONE
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller that handles the values
+</parameter_description>
+</parameter>
+</parameters>
+<return> Returns the suggested timestamp or %GST_CLOCK_TIME_NONE
+if no control-rate was set.
+
+Since: 0.10.13
+</return>
+</function>
+
<function name="gst_value_array_get_value">
<description>
Gets the value that is a member of the array contained in @value and
@@ -3103,6 +3794,49 @@
</return>
</function>
+<function name="gst_base_sink_get_last_buffer">
+<description>
+Get the last buffer that arrived in the sink and was used for preroll or for
+rendering. This property can be used to generate thumbnails.
+
+The #GstCaps on the buffer can be used to determine the type of the buffer.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstBuffer. gst_buffer_unref() after usage. This function returns
+NULL when no buffer has arrived in the sink yet or when the sink is not in
+PAUSED or PLAYING.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_interpolation_control_source_set_from_list">
+<description>
+Sets multiple timed values at once.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource object
+</parameter_description>
+</parameter>
+<parameter name="timedvalues">
+<parameter_description> a list with #GstTimedValue items
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if the values couldn't be set, TRUE otherwise.
+</return>
+</function>
+
<function name="gst_query_type_register">
<description>
Create a new GstQueryType based on the nick or return an
@@ -3125,6 +3859,29 @@
</return>
</function>
+<function name="gst_data_queue_pop">
+<description>
+Retrieves the first @item available on the @queue. If the queue is currently
+empty, the call will block until at least one item is available, OR the
+ queue is set to the flushing state.
+MT safe.
+
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> a #GstDataQueue.
+</parameter_description>
+</parameter>
+<parameter name="item">
+<parameter_description> pointer to store the returned #GstDataQueueItem.
+</parameter_description>
+</parameter>
+</parameters>
+<return> #TRUE if an @item was successfully retrieved from the @queue.
+</return>
+</function>
+
<function name="gst_pad_proxy_setcaps">
<description>
Calls gst_pad_set_caps() for every other pad belonging to the
@@ -3487,6 +4244,25 @@
</return>
</function>
+<function name="gst_adapter_take">
+<description>
+Returns: oven-fresh hot data, or #NULL if @nbytes bytes are not available
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+<parameter name="nbytes">
+<parameter_description> the number of bytes to take
+</parameter_description>
+</parameter>
+</parameters>
+<return> oven-fresh hot data, or #NULL if @nbytes bytes are not available
+</return>
+</function>
+
<function name="gst_pad_set_query_function">
<description>
Set the given query function for the pad.
@@ -3553,6 +4329,33 @@
<return></return>
</function>
+<function name="gst_check_element_push_buffer">
+<description>
+Create an @element with the factory with the name and push the
+ buffer_in to this element. The element should create one buffer
+and this will be compared with @buffer_out. We only check the caps
+and the data of the buffers. This function unrefs the buffers.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="element_name">
+<parameter_description> name of the element that needs to be created
+</parameter_description>
+</parameter>
+<parameter name="buffer_in">
+<parameter_description> push this buffer to the element
+</parameter_description>
+</parameter>
+<parameter name="buffer_out">
+<parameter_description> compare the result with this buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_message_parse_segment_done">
<description>
Extracts the position and format from the segment start message.
@@ -3681,6 +4484,34 @@
<return></return>
</function>
+<function name="gst_net_time_packet_receive">
+<description>
+Receives a #GstNetTimePacket over a socket. Handles interrupted system calls,
+but otherwise returns NULL on error. See recvfrom(2) for more information on
+how to interpret @sockaddr.
+
+MT safe. Caller owns return value (g_free to free).
+
+
+</description>
+<parameters>
+<parameter name="fd">
+<parameter_description> a file descriptor created by socket(2)
+</parameter_description>
+</parameter>
+<parameter name="addr">
+<parameter_description> a pointer to a sockaddr to hold the address of the sender
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> a pointer to the size of the data pointed to by @addr
+</parameter_description>
+</parameter>
+</parameters>
+<return> The new #GstNetTimePacket.
+</return>
+</function>
+
<function name="gst_tag_list_foreach">
<description>
Calls the given function for each tag inside the tag list. Note that if there
@@ -3704,6 +4535,32 @@
<return></return>
</function>
+<function name="gst_buffer_straw_get_buffer">
+<description>
+Get one buffer from @pad. Implemented via buffer probes. This function will
+block until the pipeline passes a buffer over @pad, so for robust behavior
+in unit tests, you need to use check's timeout to fail out in the case that a
+buffer never arrives.
+
+You must have previously called gst_buffer_straw_start_pipeline() on
+ pipeline and @pad.
+
+
+</description>
+<parameters>
+<parameter name="bin">
+<parameter_description> the pipeline previously started via gst_buffer_straw_start_pipeline()
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> the pad previously passed to gst_buffer_straw_start_pipeline()
+</parameter_description>
+</parameter>
+</parameters>
+<return> the captured #GstBuffer.
+</return>
+</function>
+
<function name="gst_bus_pop_filtered">
<description>
Get a message matching @type from the bus. Will discard all messages on
@@ -3921,19 +4778,22 @@
</return>
</function>
-<function name="gst_value_set_date">
+<function name="gst_pad_set_setcaps_function">
<description>
-Sets the contents of @value to coorespond to @date. The actual
-#GDate structure is copied before it is used.
+Sets the given setcaps function for the pad. The setcaps function
+will be called whenever a buffer with a new media type is pushed or
+pulled from the pad. The pad/element needs to update its internal
+structures to process the new media type. If this new type is not
+acceptable, the setcaps function should return FALSE.
</description>
<parameters>
-<parameter name="value">
-<parameter_description> a GValue initialized to GST_TYPE_DATE
+<parameter name="pad">
+<parameter_description> a #GstPad.
</parameter_description>
</parameter>
-<parameter name="date">
-<parameter_description> the date to set the value to
+<parameter name="setcaps">
+<parameter_description> the #GstPadSetCapsFunction to set.
</parameter_description>
</parameter>
</parameters>
@@ -4242,6 +5102,28 @@
</return>
</function>
+<function name="gst_base_src_set_do_timestamp">
+<description>
+Configure @src to automatically timestamp outgoing buffers based on the
+current running_time of the pipeline. This property is mostly useful for live
+sources.
+
+Since: 0.10.15
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> the source
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> enable or disable timestamping
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_pad_pull_range">
<description>
Pulls a @buffer from the peer pad.
@@ -4445,6 +5327,52 @@
</return>
</function>
+<function name="gst_dp_packetizer_new">
+<description>
+Creates a new packetizer.
+
+
+</description>
+<parameters>
+<parameter name="version">
+<parameter_description> the #GstDPVersion of the protocol to packetize for.
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly allocated #GstDPPacketizer
+</return>
+</function>
+
+<function name="gst_object_get_value_array">
+<description>
+Function to be able to get an array of values for one element properties
+
+If the GstValueArray-&gt;values array is NULL, it will be created by the function.
+The type of the values in the array are the same as the property's type.
+
+The g_object_* functions are just convenience functions for GObject
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+<parameter name="value_array">
+<parameter_description> array to put control-values in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the given array(s) could be filled, %FALSE otherwise
+Since: 0.9
+</return>
+</function>
+
<function name="gst_value_set_fraction_range">
<description>
Sets @value to the range specified by @start and @end.
@@ -4534,6 +5462,20 @@
</return>
</function>
+<function name="gst_adapter_clear">
+<description>
+Removes all buffers from @adapter.
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_xml_parse_doc">
<description>
Fills the GstXML object with the elements from the
@@ -4596,6 +5538,28 @@
</return>
</function>
+<function name="gst_poll_fd_can_read">
+<description>
+Check if @fd in @set has data to be read.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the descriptor has data to be read.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_tag_list_get_uchar">
<description>
Copies the contents for the given tag into the value, merging multiple values
@@ -4638,6 +5602,38 @@
</return>
</function>
+<function name="gst_object_get_value_arrays">
+<description>
+Function to be able to get an array of values for one or more given element
+properties.
+
+If the GstValueArray-&gt;values array in list nodes is NULL, it will be created
+by the function.
+The type of the values in the array are the same as the property's type.
+
+The g_object_* functions are just convenience functions for GObject
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+<parameter name="value_arrays">
+<parameter_description> list to return the control-values in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the given array(s) could be filled, %FALSE otherwise
+Since: 0.9
+</return>
+</function>
+
<function name="gst_registry_binary_read_cache">
<description>
Read the contents of the binary cache file at @location into @registry.
@@ -5129,7 +6125,23 @@
</return>
</function>
-<function name="gst_element_state_get_name">
+<function name="gst_dp_header_payload_type">
+<description>
+Get the type of the payload described by @header.
+
+
+</description>
+<parameters>
+<parameter name="header">
+<parameter_description> the byte header of the packet array
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstDPPayloadType the payload this header describes.
+</return>
+</function>
+
+<function name="gst_element_state_get_name">
<description>
Gets a string representing the given state.
@@ -5286,6 +6298,62 @@
<return></return>
</function>
+<function name="gst_base_sink_wait_eos">
+<description>
+This function will block until @time is reached. It is usually called by
+subclasses that use their own internal synchronisation but want to let the
+EOS be handled by the base class.
+
+This function should only be called with the PREROLL_LOCK held, like when
+receiving an EOS event in the ::event vmethod.
+
+Since 0.10.15
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="time">
+<parameter_description> the running_time to be reached
+</parameter_description>
+</parameter>
+<parameter name="jitter">
+<parameter_description> the jitter to be filled with time diff (can be NULL)
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GstFlowReturn
+</return>
+</function>
+
+<function name="gst_dp_validate_payload">
+<description>
+Validates the given packet payload using the given packet header
+by checking the CRC checksum.
+
+
+</description>
+<parameters>
+<parameter name="header_length">
+<parameter_description> the length of the packet header
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> the byte array of the packet header
+</parameter_description>
+</parameter>
+<parameter name="payload">
+<parameter_description> the byte array of the packet payload
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the CRC matches, or no CRC checksum is present.
+</return>
+</function>
+
<function name="gst_pad_use_fixed_caps">
<description>
A helper function you can use that sets the
@@ -5346,6 +6414,27 @@
</return>
</function>
+<function name="gst_net_time_packet_serialize">
+<description>
+Serialized a #GstNetTimePacket into a newly-allocated sequence of
+#GST_NET_TIME_PACKET_SIZE bytes, in network byte order. The value returned is
+suitable for passing to write(2) or sendto(2) for communication over the
+network.
+
+MT safe. Caller owns return value (g_free to free).
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> the #GstNetTimePacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated sequence of #GST_NET_TIME_PACKET_SIZE bytes.
+</return>
+</function>
+
<function name="gst_xml_make_element">
<description>
Load the element from the XML description
@@ -5368,19 +6457,13 @@
<function name="SECTION">
<description>
-GStreamer pipelines can be saved to xml files using gst_xml_write_file().
-They can be loaded back using gst_xml_parse_doc() / gst_xml_parse_file() /
-gst_xml_parse_memory().
-Additionally one can load saved pipelines into the gst-editor to inspect the
-graph.
-
-#GstElement implementations need to override gst_object_save_thyself() and
-gst_object_restore_thyself().
+These macros and functions are for internal use of the unit tests found
+inside the 'check' directories of various GStreamer packages.
</description>
<parameters>
<parameter name="short_description">
-<parameter_description> XML save/restore operations of pipelines
+<parameter_description> Common code for GStreamer unit tests
</parameter_description>
</parameter>
</parameters>
@@ -5528,6 +6611,28 @@
</return>
</function>
+<function name="gst_base_transform_set_qos_enabled">
+<description>
+Enable or disable QoS handling in the transform.
+
+Since: 0.10.5
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> a #GstBaseTransform
+</parameter_description>
+</parameter>
+<parameter name="enabled">
+<parameter_description> new state
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_bus_set_sync_handler">
<description>
Sets the synchronous handler on the bus. The function will be called
@@ -5749,6 +6854,30 @@
</return>
</function>
+<function name="gst_base_src_set_format">
+<description>
+Sets the default format of the source. This will be the format used
+for sending NEW_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 #GstBaseSrc::is_seekable returns TRUE.
+
+Since: 0.10.1
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> base source instance
+</parameter_description>
+</parameter>
+<parameter name="format">
+<parameter_description> the format to use
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_bus_add_watch_full">
<description>
Adds a bus watch to the default main context with the given @priority.
@@ -5790,24 +6919,20 @@
</return>
</function>
-<function name="gst_element_provide_clock">
+<function name="gst_index_factory_make">
<description>
-Get the clock provided by the given element.
-&lt;note&gt;An element is only required to provide a clock in the PAUSED
-state. Some elements can provide a clock in other states.&lt;/note&gt;
+Create a new #GstIndex instance from the
+indexfactory with the given name.
</description>
<parameters>
-<parameter name="element">
-<parameter_description> a #GstElement to query
+<parameter name="name">
+<parameter_description> the name of the factory used to create the instance
</parameter_description>
</parameter>
</parameters>
-<return> the GstClock provided by the element or %NULL
-if no clock could be provided. Unref after usage.
-
-MT safe.
+<return> A new #GstIndex instance.
</return>
</function>
@@ -6029,6 +7154,22 @@
</return>
</function>
+<function name="gst_data_queue_flush">
+<description>
+Flushes all the contents of the @queue. Any call to #gst_data_queue_pull and
+#gst_data_queue_pop will be released.
+MT safe.
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> a #GstDataQueue.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_query_type_get_by_nick">
<description>
Get the query type registered with @nick.
@@ -6089,6 +7230,36 @@
</return>
</function>
+<function name="gst_controller_set_interpolation_mode">
+<description>
+Sets the given interpolation mode on the given property.
+
+&lt;note&gt;&lt;para&gt;User interpolation is not yet available and quadratic interpolation
+is deprecated and maps to cubic interpolation.&lt;/para&gt;&lt;/note&gt;
+
+Deprecated: Use #GstControlSource, for example #GstInterpolationControlSource
+directly.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property for which to change the interpolation
+</parameter_description>
+</parameter>
+<parameter name="mode">
+<parameter_description> interpolation mode
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the property is handled by the controller, %FALSE otherwise
+</return>
+</function>
+
<function name="gst_pipeline_set_clock">
<description>
Set the clock for @pipeline. The clock will be distributed
@@ -6217,25 +7388,41 @@
</return>
</function>
-<function name="gst_value_take_mini_object">
+<function name="gst_clock_id_wait">
<description>
-Set the contents of a %GST_TYPE_MINI_OBJECT derived #GValue to
- mini_object
-Takes over the ownership of the caller's reference to @mini_object;
-the caller doesn't have to unref it any more.
+Perform a blocking wait on @id.
+ id should have been created with gst_clock_new_single_shot_id()
+or gst_clock_new_periodic_id() and should not have been unscheduled
+with a call to gst_clock_id_unschedule().
+
+If the @jitter argument is not NULL and this function returns #GST_CLOCK_OK
+or #GST_CLOCK_EARLY, it will contain the difference
+against the clock and the time of @id when this method was
+called.
+Positive values indicate how late @id was relative to the clock
+(in which case this function will return #GST_CLOCK_EARLY).
+Negative values indicate how much time was spent waiting on the clock
+before this function returned.
+
</description>
<parameters>
-<parameter name="value">
-<parameter_description> a valid #GValue of %GST_TYPE_MINI_OBJECT derived type
+<parameter name="id">
+<parameter_description> The #GstClockID to wait on
</parameter_description>
</parameter>
-<parameter name="mini_object">
-<parameter_description> mini object value to take
+<parameter name="jitter">
+<parameter_description> A pointer that will contain the jitter, can be NULL.
</parameter_description>
</parameter>
</parameters>
-<return></return>
+<return> the result of the blocking wait. #GST_CLOCK_EARLY will be returned
+if the current clock time is past the time of @id, #GST_CLOCK_OK if
+ id was scheduled in time. #GST_CLOCK_UNSCHEDULED if @id was
+unscheduled with gst_clock_id_unschedule().
+
+MT safe.
+</return>
</function>
<function name="gst_alloc_trace_print">
@@ -6344,6 +7531,47 @@
<return></return>
</function>
+<function name="gst_object_get_controller">
+<description>
+Gets the controller for the given GObject
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+</parameters>
+<return> the controller handling some of the given element's properties, %NULL if no controller
+Since: 0.9
+</return>
+</function>
+
+<function name="gst_collect_pads_set_flushing">
+<description>
+Change the flushing state of all the pads in the collection. No pad
+is able to accept anymore data when @flushing is %TRUE. Calling this
+function with @flushing %FALSE makes @pads accept data again.
+
+MT safe.
+
+Since: 0.10.7.
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+<parameter name="flushing">
+<parameter_description> desired state of the pads
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_child_proxy_get_property">
<description>
Gets a single property using the GstChildProxy mechanism.
@@ -6510,6 +7738,50 @@
<return></return>
</function>
+<function name="gst_controller_remove_properties_list">
+<description>
+Removes the given object properties from the controller
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object from which some properties should be removed
+</parameter_description>
+</parameter>
+<parameter name="list">
+<parameter_description> #GList of property names that should be removed
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if one of the given property isn't handled by the controller, %TRUE otherwise
+</return>
+</function>
+
+<function name="gst_base_transform_set_passthrough">
+<description>
+Set passthrough mode for this filter by default. This is mostly
+useful for filters that do not care about negotiation.
+
+Always TRUE for filters which don't implement either a transform
+or transform_ip method.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> the #GstBaseTransform to set
+</parameter_description>
+</parameter>
+<parameter name="passthrough">
+<parameter_description> boolean indicating passthrough mode.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_structure_id_set">
<description>
Identical to gst_structure_set, except that field names are
@@ -6565,6 +7837,40 @@
</return>
</function>
+<function name="gst_type_find_helper_for_buffer">
+<description>
+Tries to find what type of data is contained in the given #GstBuffer, the
+assumption being that the buffer represents the beginning of the stream or
+file.
+
+All available typefinders will be called on the data in order of rank. If
+a typefinding function returns a probability of #GST_TYPE_FIND_MAXIMUM,
+typefinding is stopped immediately and the found caps will be returned
+right away. Otherwise, all available typefind functions will the tried,
+and the caps with the highest probability will be returned, or #NULL if
+the content of the buffer could not be identified.
+
+
+</description>
+<parameters>
+<parameter name="obj">
+<parameter_description> object doing the typefinding, or NULL (used for logging)
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> a #GstBuffer with data to typefind
+</parameter_description>
+</parameter>
+<parameter name="prob">
+<parameter_description> location to store the probability of the found caps, or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> The #GstCaps corresponding to the data, or #NULL if no type could
+be found. The caller should free the caps returned with gst_caps_unref().
+</return>
+</function>
+
<function name="gst_message_parse_async_start">
<description>
Extract the new_base_time from the async_start message.
@@ -6646,13 +7952,41 @@
</return>
</function>
-<function name="gst_util_dump_mem">
+<function name="gst_data_queue_push">
<description>
-Dumps the memory block into a hex representation. Useful for debugging.
+Pushes a #GstDataQueueItem (or a structure that begins with the same fields)
+on the @queue. If the @queue is full, the call will block until space is
+available, OR the @queue is set to flushing state.
+MT safe.
+
+Note that this function has slightly different semantics than gst_pad_push()
+and gst_pad_push_event(): this function only takes ownership of @item and
+the #GstMiniObject contained in @item if the push was successful. If FALSE
+is returned, the caller is responsible for freeing @item and its contents.
+
</description>
<parameters>
-<parameter name="mem">
+<parameter name="queue">
+<parameter_description> a #GstDataQueue.
+</parameter_description>
+</parameter>
+<parameter name="item">
+<parameter_description> a #GstDataQueueItem.
+</parameter_description>
+</parameter>
+</parameters>
+<return> #TRUE if the @item was successfully pushed on the @queue.
+</return>
+</function>
+
+<function name="gst_util_dump_mem">
+<description>
+Dumps the memory block into a hex representation. Useful for debugging.
+
+</description>
+<parameters>
+<parameter name="mem">
<parameter_description> a pointer to the memory to dump
</parameter_description>
</parameter>
@@ -6683,6 +8017,26 @@
<return></return>
</function>
+<function name="gst_controller_remove_properties_valist">
+<description>
+Removes the given object properties from the controller
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object from which some properties should be removed
+</parameter_description>
+</parameter>
+<parameter name="var_args">
+<parameter_description> %NULL terminated list of property names that should be removed
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if one of the given property isn't handled by the controller, %TRUE otherwise
+</return>
+</function>
+
<function name="gst_ghost_pad_set_target">
<description>
Set the new target of the ghostpad @gpad. Any existing target
@@ -6772,21 +8126,30 @@
</return>
</function>
-<function name="gst_iterator_resync">
+<function name="gst_bin_iterate_sorted">
<description>
-Resync the iterator. this function is mostly called
-after gst_iterator_next() returned %GST_ITERATOR_RESYNC.
+Gets an iterator for the elements in this bin in topologically
+sorted order. This means that the elements are returned from
+the most downstream elements (sinks) to the sources.
+
+This function is used internally to perform the state changes
+of the bin elements and for clock selection.
+
+Each element yielded by the iterator will have its refcount increased, so
+unref after use.
+
+MT safe. Caller owns returned value.
-MT safe.
</description>
<parameters>
-<parameter name="it">
-<parameter_description> The #GstIterator to resync
+<parameter name="bin">
+<parameter_description> a #GstBin
</parameter_description>
</parameter>
</parameters>
-<return></return>
+<return> a #GstIterator of #GstElement, or NULL
+</return>
</function>
<function name="gst_plugin_name_filter">
@@ -6810,6 +8173,44 @@
</return>
</function>
+<function name="gst_base_sink_get_latency">
+<description>
+Get the currently configured latency.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> The configured latency.
+
+Since: 0.10.12
+</return>
+</function>
+
+<function name="gst_adapter_take_buffer">
+<description>
+Returns: a #GstBuffer containing the first @nbytes of the adapter,
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+<parameter name="nbytes">
+<parameter_description> the number of bytes to take
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstBuffer containing the first @nbytes of the adapter,
+or #NULL if @nbytes bytes are not available
+</return>
+</function>
+
<function name="_gst_debug_init">
<description>
Initializes the debugging system.
@@ -6930,6 +8331,50 @@
</return>
</function>
+<function name="gst_value_dup_mini_object">
+<description>
+Get the contents of a %GST_TYPE_MINI_OBJECT derived #GValue,
+increasing its reference count.
+
+
+</description>
+<parameters>
+<parameter name="value">
+<parameter_description> a valid #GValue of %GST_TYPE_MINI_OBJECT derived type
+</parameter_description>
+</parameter>
+</parameters>
+<return> mini object contents of @value
+
+Since: 0.10.19
+</return>
+</function>
+
+<function name="gst_controller_unset_all">
+<description>
+Used to remove all time-stamped values of given controller-handled property
+
+Deprecated: Use #GstControlSource, for example #GstInterpolationControlSource
+directly.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object which handles the properties
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property to unset
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if the values couldn't be unset (ex : properties not handled
+by controller), %TRUE otherwise
+Since: 0.10.5
+</return>
+</function>
+
<function name="gst_value_subtract">
<description>
Subtracts @subtrahend from @minuend and stores the result in @dest.
@@ -6955,6 +8400,31 @@
</return>
</function>
+<function name="gst_collect_pads_peek">
+<description>
+Peek at the buffer currently queued in @data. This function
+should be called with the @pads LOCK held, such as in the callback
+handler.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to peek
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to use
+</parameter_description>
+</parameter>
+</parameters>
+<return> The buffer in @data or NULL if no buffer is queued.
+should unref the buffer after usage.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_pad_get_caps">
<description>
Gets the capabilities this pad can produce or consume.
@@ -7042,6 +8512,30 @@
</return>
</function>
+<function name="gst_object_uncontrol_properties">
+<description>
+Convenience function for GObject
+
+Removes the given element's properties from it's controller
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object of which some properties should not be controlled anymore
+</parameter_description>
+</parameter>
+<parameter name="Varargs">
+<parameter_description> %NULL terminated list of property names that should be controlled
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if one of the given property names isn't handled by the
+controller, %TRUE otherwise
+Since: 0.9
+</return>
+</function>
+
<function name="GstIndex">
<description>
Is emitted when a new entry is added to the index.
@@ -7181,6 +8675,32 @@
<return></return>
</function>
+<function name="gst_object_get_control_rate">
+<description>
+Obtain the control-rate for this @object. Audio processing #GstElement
+objects will use this rate to sub-divide their processing loop and call
+gst_object_sync_values() inbetween. The length of the processing segment
+should be up to @control-rate nanoseconds.
+
+If the @object is not under property control, this will return
+%GST_CLOCK_TIME_NONE. This allows the element to avoid the sub-dividing.
+
+The control-rate is not expected to change if the element is in
+%GST_STATE_PAUSED or %GST_STATE_PLAYING.
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+</parameters>
+<return> the control rate in nanoseconds
+Since: 0.10.10
+</return>
+</function>
+
<function name="gst_value_get_int_range_max">
<description>
Gets the maximum of the range specified by @value.
@@ -7197,6 +8717,35 @@
</return>
</function>
+<function name="gst_dp_event_from_packet">
+<description>
+Creates a newly allocated #GstEvent from the given packet.
+
+This function does not check the arguments passed to it, use
+gst_dp_validate_packet() first if the header and payload data are
+unchecked.
+
+
+</description>
+<parameters>
+<parameter name="header_length">
+<parameter_description> the length of the packet header
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> the byte array of the packet header
+</parameter_description>
+</parameter>
+<parameter name="payload">
+<parameter_description> the byte array of the packet payload
+</parameter_description>
+</parameter>
+</parameters>
+<return> A #GstEvent if the event was successfully created,
+or NULL if an event could not be read from the payload.
+</return>
+</function>
+
<function name="gst_message_parse_clock_provide">
<description>
Extracts the clock and ready flag from the GstMessage.
@@ -7411,6 +8960,36 @@
</return>
</function>
+<function name="gst_collect_pads_flush">
+<description>
+Flush @size bytes from the pad @data.
+
+This function should be called with @pads LOCK held, such as
+in the callback.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to query
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to use
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the number of bytes to flush
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of bytes flushed. This can be less than @size and
+is 0 if the pad was end-of-stream.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_index_factory_new">
<description>
Create a new indexfactory with the given parameters
@@ -7435,6 +9014,39 @@
</return>
</function>
+<function name="gst_controller_get_value_arrays">
+<description>
+Function to be able to get an array of values for one or more given element
+properties.
+
+All fields of the %GstValueArray in the list must be filled correctly.
+Especially the GstValueArray-&gt;values arrays must be big enough to keep
+the requested amount of values.
+
+The types of the values in the array are the same as the property's type.
+
+&lt;note&gt;&lt;para&gt;This doesn't modify the controlled GObject properties!&lt;/para&gt;&lt;/note&gt;
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller that handles the values
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+<parameter name="value_arrays">
+<parameter_description> list to return the control-values in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the given array(s) could be filled, %FALSE otherwise
+</return>
+</function>
+
<function name="gst_bin_iterate_elements">
<description>
Gets an iterator for the elements in this bin.
@@ -7569,6 +9181,28 @@
</return>
</function>
+<function name="gst_poll_fd_has_closed">
+<description>
+Check if @fd in @set has closed the connection.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the connection was closed.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="GstChildProxy">
<description>
Will be emitted after the @object was removed from the @child_proxy.
@@ -7806,6 +9440,32 @@
</return>
</function>
+<function name="gst_object_set_control_rate">
+<description>
+Change the control-rate for this @object. Audio processing #GstElement
+objects will use this rate to sub-divide their processing loop and call
+gst_object_sync_values() inbetween. The length of the processing segment
+should be up to @control-rate nanoseconds.
+
+The control-rate should not change if the element is in %GST_STATE_PAUSED or
+%GST_STATE_PLAYING.
+
+Since: 0.10.10
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+<parameter name="control_rate">
+<parameter_description> the new control-rate in nanoseconds.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_value_set_fraction">
<description>
Sets @value to the fraction specified by @numerator over @denominator.
@@ -7923,29 +9583,51 @@
</return>
</function>
-<function name="gst_element_get_compatible_pad">
+<function name="gst_buffer_straw_stop_pipeline">
<description>
-Looks for an unlinked pad to which the given pad can link. It is not
-guaranteed that linking the pads will work, though it should work in most
-cases.
+Set @bin to #GST_STATE_NULL and release resource allocated in
+gst_buffer_straw_start_pipeline().
+You must have previously called gst_buffer_straw_start_pipeline() on
+ pipeline and @pad.
</description>
<parameters>
-<parameter name="element">
-<parameter_description> a #GstElement in which the pad should be found.
+<parameter name="bin">
+<parameter_description> the pipeline previously started via gst_buffer_straw_start_pipeline()
</parameter_description>
</parameter>
<parameter name="pad">
-<parameter_description> the #GstPad to find a compatible one for.
-</parameter_description>
-</parameter>
-<parameter name="caps">
-<parameter_description> the #GstCaps to use as a filter.
+<parameter_description> the pad previously passed to gst_buffer_straw_start_pipeline()
</parameter_description>
</parameter>
</parameters>
-<return> the #GstPad to which a link can be made, or %NULL if one cannot be
+<return></return>
+</function>
+
+<function name="gst_element_get_compatible_pad">
+<description>
+Looks for an unlinked pad to which the given pad can link. It is not
+guaranteed that linking the pads will work, though it should work in most
+cases.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> a #GstElement in which the pad should be found.
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> the #GstPad to find a compatible one for.
+</parameter_description>
+</parameter>
+<parameter name="caps">
+<parameter_description> the #GstCaps to use as a filter.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstPad to which a link can be made, or %NULL if one cannot be
found. gst_object_unref() after usage.
</return>
</function>
@@ -8078,6 +9760,50 @@
</return>
</function>
+<function name="gst_collect_pads_pop">
+<description>
+Pop the buffer currently queued in @data. This function
+should be called with the @pads LOCK held, such as in the callback
+handler.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to pop
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to use
+</parameter_description>
+</parameter>
+</parameters>
+<return> The buffer in @data or NULL if no buffer was queued.
+You should unref the buffer after usage.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_value_set_date">
+<description>
+Sets the contents of @value to coorespond to @date. The actual
+#GDate structure is copied before it is used.
+
+</description>
+<parameters>
+<parameter name="value">
+<parameter_description> a GValue initialized to GST_TYPE_DATE
+</parameter_description>
+</parameter>
+<parameter name="date">
+<parameter_description> the date to set the value to
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_structure_get_fourcc">
<description>
Sets the #GstFourcc pointed to by @value corresponding to the value of the
@@ -8224,6 +9950,52 @@
</return>
</function>
+<function name="gst_buffer_straw_start_pipeline">
+<description>
+Sets up a pipeline for buffer sucking. This will allow you to call
+gst_buffer_straw_get_buffer() to access buffers as they pass over @pad.
+
+This function is normally used in unit tests that want to verify that a
+particular element is outputting correct buffers. For example, you would make
+a pipeline via gst_parse_launch(), pull out the pad you want to monitor, then
+call gst_buffer_straw_get_buffer() to get the buffers that pass through @pad.
+The pipeline will block until you have sucked off the buffers.
+
+This function will set the state of @bin to PLAYING; to clean up, be sure to
+call gst_buffer_straw_stop_pipeline().
+
+Note that you may not start two buffer straws at the same time. This function
+is intended for unit tests, not general API use. In fact it calls fail_if
+from libcheck, so you cannot use it outside unit tests.
+
+</description>
+<parameters>
+<parameter name="bin">
+<parameter_description> the pipeline to run
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> a pad on an element in @bin
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_dp_packetizer_free">
+<description>
+Free the given packetizer.
+
+</description>
+<parameters>
+<parameter name="packetizer">
+<parameter_description> the #GstDPPacketizer to free.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_message_new_element">
<description>
Create a new element-specific message. This is meant as a generic way of
@@ -8396,6 +10168,24 @@
</return>
</function>
+<function name="gst_adapter_available">
+<description>
+Gets the maximum amount of bytes available, that is it returns the maximum
+value that can be supplied to gst_adapter_peek() without that function
+returning NULL.
+
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+</parameters>
+<return> number of bytes available in @adapter
+</return>
+</function>
+
<function name="gst_query_parse_convert">
<description>
Parse a convert query answer. Any of @src_format, @src_value, @dest_format,
@@ -8544,6 +10334,35 @@
</return>
</function>
+<function name="gst_collect_pads_collect_range">
+<description>
+Collect data with @offset and @length on all pads. This function
+is typically called in the getrange function of an element.
+
+This function is currently not implemented.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+<parameter name="offset">
+<parameter_description> the offset to collect
+</parameter_description>
+</parameter>
+<parameter name="length">
+<parameter_description> the length to collect
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GstFlowReturn of the operation.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_plugin_feature_get_name">
<description>
Gets the name of a plugin feature.
@@ -8625,6 +10444,21 @@
</return>
</function>
+<function name="gst_data_queue_limits_changed">
+<description>
+Inform the queue that the limits for the fullness check have changed and that
+any blocking gst_data_queue_push() should be unblocked to recheck the limts.
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> The #GstDataQueue
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_tag_list_add_valist">
<description>
Sets the values for the given tags using the specified mode.
@@ -8964,6 +10798,22 @@
</return>
</function>
+<function name="gst_poll_free">
+<description>
+Free a file descriptor set.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_type_find_peek">
<description>
returns and must not be freed.
@@ -9150,6 +11000,26 @@
</return>
</function>
+<function name="gst_controller_new">
+<description>
+Creates a new GstController for the given object's properties
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object of which some properties should be controlled
+</parameter_description>
+</parameter>
+<parameter name="Varargs">
+<parameter_description> %NULL terminated list of property names that should be controlled
+</parameter_description>
+</parameter>
+</parameters>
+<return> the new controller.
+</return>
+</function>
+
<function name="gst_bin_iterate_sources">
<description>
Gets an iterator for all elements in the bin that have no sinkpads and have
@@ -9242,6 +11112,42 @@
</return>
</function>
+<function name="gst_collect_pads_read">
+<description>
+Get a pointer in @bytes where @size bytes can be read from the
+given pad @data.
+
+This function should be called with @pads LOCK held, such as
+in the callback.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to query
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to use
+</parameter_description>
+</parameter>
+<parameter name="bytes">
+<parameter_description> a pointer to a byte array
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the number of bytes to read
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of bytes available for consumption in the
+memory pointed to by @bytes. This can be less than @size and
+is 0 if the pad is end-of-stream.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_tag_list_get_pointer_index">
<description>
Gets the value that is at the given index for the given tag in the given
@@ -9312,19 +11218,19 @@
</return>
</function>
-<function name="gst_element_factory_get_author">
+<function name="gst_structure_get_name">
<description>
-Gets the author for this factory.
+Get the name of @structure as a string.
</description>
<parameters>
-<parameter name="factory">
-<parameter_description> a #GstElementFactory
+<parameter name="structure">
+<parameter_description> a #GstStructure
</parameter_description>
</parameter>
</parameters>
-<return> the author
+<return> the name of the structure.
</return>
</function>
@@ -9427,6 +11333,24 @@
</return>
</function>
+<function name="gst_data_queue_get_level">
+<description>
+Get the current level of the queue.
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> The #GstDataQueue
+</parameter_description>
+</parameter>
+<parameter name="level">
+<parameter_description> the location to store the result
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_event_new_latency">
<description>
Create a new latency event. The event is sent upstream from the sinks and
@@ -9556,6 +11480,41 @@
<return></return>
</function>
+<function name="gst_base_src_query_latency">
+<description>
+Query the source for the latency parameters. @live will be TRUE when @src is
+configured as a live source. @min_latency will be set to the difference
+between the running time and the timestamp of the first buffer.
+ max_latency is always the undefined value of -1.
+
+This function is mostly used by subclasses.
+
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> the source
+</parameter_description>
+</parameter>
+<parameter name="live">
+<parameter_description> if the source is live
+</parameter_description>
+</parameter>
+<parameter name="min_latency">
+<parameter_description> the min latency of the source
+</parameter_description>
+</parameter>
+<parameter name="max_latency">
+<parameter_description> the max latency of the source
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the query succeeded.
+
+Since: 0.10.13
+</return>
+</function>
+
<function name="gst_type_find_factory_call_function">
<description>
Calls the #GstTypeFindFunction associated with this factory.
@@ -9751,6 +11710,26 @@
<return></return>
</function>
+<function name="gst_dp_validate_header">
+<description>
+Validates the given packet header by checking the CRC checksum.
+
+
+</description>
+<parameters>
+<parameter name="header_length">
+<parameter_description> the length of the packet header
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> the byte array of the packet header
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the CRC matches, or no CRC checksum is present.
+</return>
+</function>
+
<function name="gst_element_link_pads">
<description>
Links the two named pads of the source and destination elements.
@@ -9826,6 +11805,25 @@
</return>
</function>
+<function name="gst_adapter_push">
+<description>
+Adds the data from @buf to the data stored inside @adapter and takes
+ownership of the buffer.
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> a #GstBuffer to add to queue in the adapter
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_caps_load_thyself">
<description>
Creates a #GstCaps from its XML serialization.
@@ -10352,29 +12350,59 @@
</return>
</function>
-<function name="gst_object_replace">
+<function name="gst_net_client_clock_new">
<description>
-Unrefs the #GstObject pointed to by @oldobj, refs @newobj and
-puts @newobj in * oldobj Be carefull when calling this
-function, it does not take any locks. You might want to lock
-the object owning @oldobj pointer before calling this
-function.
+Create a new #GstNetClientClock that will report the time
+provided by the #GstNetClockProvider on @remote_address and
+ remote_port
-Make sure not to LOCK @oldobj because it might be unreffed
-which could cause a deadlock when it is disposed.
</description>
<parameters>
-<parameter name="oldobj">
-<parameter_description> pointer to a place of a #GstObject to replace
+<parameter name="name">
+<parameter_description> a name for the clock
</parameter_description>
</parameter>
-<parameter name="newobj">
-<parameter_description> a new #GstObject
+<parameter name="remote_address">
+<parameter_description> the address of the remote clock provider
+</parameter_description>
+</parameter>
+<parameter name="remote_port">
+<parameter_description> the port of the remote clock provider
+</parameter_description>
+</parameter>
+<parameter name="base_time">
+<parameter_description> initial time of the clock
</parameter_description>
</parameter>
</parameters>
-<return></return>
+<return> a new #GstClock that receives a time from the remote
+clock.
+</return>
+</function>
+
+<function name="gst_control_source_get_value">
+<description>
+Gets the value for this #GstControlSource at a given timestamp.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstControlSource object
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time for which the value should be returned
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> the value
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if the value couldn't be returned, TRUE otherwise.
+</return>
</function>
<function name="gst_object_ref">
@@ -10537,6 +12565,23 @@
</return>
</function>
+<function name="GstDataQueue">
+<description>
+Reports that the queue became full (full).
+A queue is full if the total amount of data inside it (num-visible, time,
+size) is higher than the boundary values which can be set through the GObject
+properties.
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> the queue instance
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_query_new_seeking">
<description>
Constructs a new query object for querying seeking properties of
@@ -10588,6 +12633,29 @@
</return>
</function>
+<function name="gst_net_time_packet_new">
+<description>
+Creates a new #GstNetTimePacket from a buffer received over the network. The
+caller is responsible for ensuring that @buffer is at least
+#GST_NET_TIME_PACKET_SIZE bytes long.
+
+If @buffer is #NULL, the local and remote times will be set to
+#GST_CLOCK_TIME_NONE.
+
+MT safe. Caller owns return value (g_free to free).
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a buffer from which to construct the packet, or NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> The new #GstNetTimePacket.
+</return>
+</function>
+
<function name="gst_pad_new">
<description>
Creates a new pad with the given name in the given direction.
@@ -10690,6 +12758,30 @@
</return>
</function>
+<function name="gst_poll_set_controllable">
+<description>
+When @controllable is %TRUE, this function ensures that future calls to
+gst_poll_wait() will be affected by gst_poll_restart() and
+gst_poll_set_flushing().
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a #GstPoll.
+</parameter_description>
+</parameter>
+<parameter name="controllable">
+<parameter_description> new controllable state.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the controllability of @set could be updated.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_pipeline_new">
<description>
Create a new pipeline with the given name.
@@ -10816,35 +12908,21 @@
</return>
</function>
-<function name="gst_filter_run">
+<function name="gst_iterator_resync">
<description>
-Iterates over the elements in @list, calling @func with the
-list item data for each item. If @func returns TRUE, @data is
-prepended to the list of results returned. If @first is true,
-the search is halted after the first result is found.
+Resync the iterator. this function is mostly called
+after gst_iterator_next() returned %GST_ITERATOR_RESYNC.
+MT safe.
</description>
<parameters>
-<parameter name="list">
-<parameter_description> a linked list
-</parameter_description>
-</parameter>
-<parameter name="func">
-<parameter_description> the function to execute for each item
-</parameter_description>
-</parameter>
-<parameter name="first">
-<parameter_description> flag to stop execution after a successful item
-</parameter_description>
-</parameter>
-<parameter name="user_data">
-<parameter_description> user data
+<parameter name="it">
+<parameter_description> The #GstIterator to resync
</parameter_description>
</parameter>
</parameters>
-<return> the list of results
-</return>
+<return></return>
</function>
<function name="gst_element_is_locked_state">
@@ -10946,6 +13024,26 @@
</return>
</function>
+<function name="gst_base_transform_is_qos_enabled">
+<description>
+Queries if the transform will handle QoS.
+
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> a #GstBaseTransform
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if QoS is enabled.
+
+Since: 0.10.5
+
+MT safe.
+</return>
+</function>
+
<function name="gst_tag_list_get_float_index">
<description>
Gets the value that is at the given index for the given tag in the given
@@ -11105,24 +13203,20 @@
</return>
</function>
-<function name="gst_pad_set_getrange_function">
+<function name="gst_element_factory_get_longname">
<description>
-Sets the given getrange function for the pad. The getrange function is
-called to produce a new #GstBuffer to start the processing pipeline. see
-#GstPadGetRangeFunction for a description of the getrange function.
+Gets the longname for this factory
+
</description>
<parameters>
-<parameter name="pad">
-<parameter_description> a source #GstPad.
-</parameter_description>
-</parameter>
-<parameter name="get">
-<parameter_description> the #GstPadGetRangeFunction to set.
+<parameter name="factory">
+<parameter_description> a #GstElementFactory
</parameter_description>
</parameter>
</parameters>
-<return></return>
+<return> the longname
+</return>
</function>
<function name="gst_value_list_append_value">
@@ -11167,6 +13261,27 @@
</return>
</function>
+<function name="gst_interpolation_control_source_unset">
+<description>
+Used to remove the value of given controller-handled property at a certain
+time.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource object
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time the control-change should be removed from
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if the value couldn't be unset (i.e. not found, TRUE otherwise.
+</return>
+</function>
+
<function name="gst_pad_add_data_probe">
<description>
Adds a "data probe" to a pad. This function will be called whenever data
@@ -11211,6 +13326,23 @@
</return>
</function>
+<function name="gst_collect_pads_stop">
+<description>
+Stops the processing of data in the collect_pads. this function
+will also unblock any blocking operations.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_caps_append_structure">
<description>
Appends @structure to @caps. The structure is not copied; @caps
@@ -11241,6 +13373,24 @@
</return>
</function>
+<function name="gst_base_src_get_do_timestamp">
+<description>
+Query if @src timestamps outgoing buffers based on the current running_time.
+
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> the source
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the base class will automatically timestamp outgoing buffers.
+
+Since: 0.10.15
+</return>
+</function>
+
<function name="gst_registry_feature_filter">
<description>
Runs a filter against all features of the plugins in the registry
@@ -11396,6 +13546,28 @@
</return>
</function>
+<function name="gst_poll_fd_can_write">
+<description>
+Check if @fd in @set can be used for writing.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the descriptor can be used for writing.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_structure_fixate_field_boolean">
<description>
Fixates a #GstStructure by changing the given @field_name field to the given
@@ -11499,6 +13671,26 @@
</return>
</function>
+<function name="gst_controller_new_list">
+<description>
+Creates a new GstController for the given object's properties
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object of which some properties should be controlled
+</parameter_description>
+</parameter>
+<parameter name="list">
+<parameter_description> list of property names that should be controlled
+</parameter_description>
+</parameter>
+</parameters>
+<return> the new controller.
+</return>
+</function>
+
<function name="gst_clock_id_get_time">
<description>
Get the time of the clock ID
@@ -11592,20 +13784,49 @@
</return>
</function>
-<function name="gst_element_factory_get_longname">
+<function name="gst_pad_set_getrange_function">
<description>
-Gets the longname for this factory
+Sets the given getrange function for the pad. The getrange function is
+called to produce a new #GstBuffer to start the processing pipeline. see
+#GstPadGetRangeFunction for a description of the getrange function.
+
+</description>
+<parameters>
+<parameter name="pad">
+<parameter_description> a source #GstPad.
+</parameter_description>
+</parameter>
+<parameter name="get">
+<parameter_description> the #GstPadGetRangeFunction to set.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_base_src_set_live">
+<description>
+If the element listens to a live source, @live should
+be set to %TRUE.
+A live source will not produce data in the PAUSED state and
+will therefore not be able to participate in the PREROLL phase
+of a pipeline. To signal this fact to the application and the
+pipeline, the state change return value of the live source will
+be GST_STATE_CHANGE_NO_PREROLL.
</description>
<parameters>
-<parameter name="factory">
-<parameter_description> a #GstElementFactory
+<parameter name="src">
+<parameter_description> base source instance
+</parameter_description>
+</parameter>
+<parameter name="live">
+<parameter_description> new live-mode
</parameter_description>
</parameter>
</parameters>
-<return> the longname
-</return>
+<return></return>
</function>
<function name="gst_plugin_get_package">
@@ -11663,6 +13884,54 @@
</return>
</function>
+<function name="gst_collect_pads_add_pad_full">
+<description>
+Add a pad to the collection of collect pads. The pad has to be
+a sinkpad. The refcount of the pad is incremented. Use
+gst_collect_pads_remove_pad() to remove the pad from the collection
+again.
+
+You specify a size for the returned #GstCollectData structure
+so that you can use it to store additional information.
+
+You can also specify a #GstCollectDataDestroyNotify that will be called
+just before the #GstCollectData structure is freed. It is passed the
+pointer to the structure and should free any custom memory and resources
+allocated for it.
+
+The pad will be automatically activated in push mode when @pads is
+started.
+
+Since: 0.10.12
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> the pad to add
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of the returned #GstCollectData structure
+</parameter_description>
+</parameter>
+<parameter name="destroy_notify">
+<parameter_description> function to be called before the returned #GstCollectData
+structure is freed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstCollectData to identify the new pad. Or NULL
+if wrong parameters are supplied.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_mini_object_is_writable">
<description>
Checks if a mini-object is writable. A mini-object is writable
@@ -11684,6 +13953,25 @@
</return>
</function>
+<function name="gst_data_queue_new">
+<description>
+
+</description>
+<parameters>
+<parameter name="checkfull">
+<parameter_description> the callback used to tell if the element considers the queue full
+or not.
+</parameter_description>
+</parameter>
+<parameter name="checkdata">
+<parameter_description> a #gpointer that will be given in the @checkfull callback.
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstDataQueue.
+</return>
+</function>
+
<function name="gst_index_entry_free">
<description>
Free the memory used by the given entry.
@@ -12014,23 +14302,6 @@
</return>
</function>
-<function name="gst_value_get_mini_object">
-<description>
-Get the contents of a %GST_TYPE_MINI_OBJECT derived #GValue.
-Does not increase the refcount of the returned object.
-
-
-</description>
-<parameters>
-<parameter name="value">
-<parameter_description> a valid #GValue of %GST_TYPE_MINI_OBJECT derived type
-</parameter_description>
-</parameter>
-</parameters>
-<return> mini object contents of @value
-</return>
-</function>
-
<function name="GstXML">
<description>
Signals that a new object has been deserialized.
@@ -12254,6 +14525,27 @@
<return></return>
</function>
+<function name="gst_control_source_bind">
+<description>
+Binds a #GstControlSource to a specific property. This must be called only once for a
+#GstControlSource.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstControlSource object
+</parameter_description>
+</parameter>
+<parameter name="pspec">
+<parameter_description> #GParamSpec for the property for which this #GstControlSource should generate values.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the #GstControlSource was bound correctly, %FALSE otherwise.
+</return>
+</function>
+
<function name="gst_value_get_fourcc">
<description>
Gets the #guint32 fourcc contained in @value.
@@ -12317,6 +14609,36 @@
<return></return>
</function>
+<function name="gst_base_transform_update_qos">
+<description>
+Set the QoS parameters in the transform.
+
+Since: 0.10.5
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> a #GstBaseTransform
+</parameter_description>
+</parameter>
+<parameter name="proportion">
+<parameter_description> the proportion
+</parameter_description>
+</parameter>
+<parameter name="diff">
+<parameter_description> the diff against the clock
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the timestamp of the buffer generating the QoS
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_pipeline_auto_clock">
<description>
Let @pipeline select a clock automatically. This is the default
@@ -12399,6 +14721,39 @@
<return></return>
</function>
+<function name="gst_collect_pads_take_buffer">
+<description>
+Get a subbuffer of @size bytes from the given pad @data. Flushes the amount
+of read bytes.
+
+This function should be called with @pads LOCK held, such as in the callback.
+
+Since: 0.10.18
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to query
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to use
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the number of bytes to read
+</parameter_description>
+</parameter>
+</parameters>
+<return> A sub buffer. The size of the buffer can be less that requested.
+A return of NULL signals that the pad is end-of-stream.
+Unref the buffer after use.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_element_factory_get_element_type">
<description>
Get the #GType for elements managed by this factory. The type can
@@ -12770,6 +15125,25 @@
</return>
</function>
+<function name="gst_base_sink_get_sync">
+<description>
+Checks if @sink is currently configured to synchronize against the
+clock.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the sink is configured to synchronize against the clock.
+
+Since: 0.10.4
+</return>
+</function>
+
<function name="gst_pad_set_activatepull_function">
<description>
Sets the given activate_pull function for the pad. An activate_pull function
@@ -12986,6 +15360,31 @@
</return>
</function>
+<function name="gst_net_time_provider_new">
+<description>
+Allows network clients to get the current time of @clock.
+
+
+</description>
+<parameters>
+<parameter name="clock">
+<parameter_description> a #GstClock to export over the network
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> an address to bind on as a dotted quad (xxx.xxx.xxx.xxx), or NULL
+to bind to all addresses
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> a port to bind on, or 0 to let the kernel choose
+</parameter_description>
+</parameter>
+</parameters>
+<return> the new #GstNetTimeProvider, or NULL on error
+</return>
+</function>
+
<function name="gst_task_set_lock">
<description>
Set the mutex used by the task. The mutex will be acquired before
@@ -13118,6 +15517,22 @@
<return></return>
</function>
+<function name="gst_base_src_is_live">
+<description>
+Check if an element is in live mode.
+
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> base source instance
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if element is in live mode.
+</return>
+</function>
+
<function name="gst_bus_post">
<description>
Post a message on the given bus. Ownership of the message
@@ -13163,28 +15578,20 @@
</return>
</function>
-<function name="gst_index_entry_assoc_map">
+<function name="gst_value_get_mini_object">
<description>
-Gets alternative formats associated with the indexentry.
+Get the contents of a %GST_TYPE_MINI_OBJECT derived #GValue.
+Does not increase the refcount of the returned object.
</description>
<parameters>
-<parameter name="entry">
-<parameter_description> the index to search
-</parameter_description>
-</parameter>
-<parameter name="format">
-<parameter_description> the format of the value the find
-</parameter_description>
-</parameter>
<parameter name="value">
-<parameter_description> a pointer to store the value
+<parameter_description> a valid #GValue of %GST_TYPE_MINI_OBJECT derived type
</parameter_description>
</parameter>
</parameters>
-<return> TRUE if there was a value associated with the given
-format.
+<return> mini object contents of @value
</return>
</function>
@@ -13367,6 +15774,24 @@
</return>
</function>
+<function name="gst_base_transform_is_in_place">
+<description>
+See if @trans is configured as a in_place transform.
+
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> the #GstBaseTransform to query
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE is the transform is configured in in_place mode.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_bus_timed_pop_filtered">
<description>
Get a message from the bus whose type matches the message type mask @types,
@@ -13434,6 +15859,27 @@
</return>
</function>
+<function name="gst_base_sink_get_max_lateness">
+<description>
+Gets the max lateness value. See gst_base_sink_set_max_lateness for
+more details.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> The maximum time in nanoseconds that a buffer can be late
+before it is dropped and not rendered. A value of -1 means an
+unlimited time.
+
+Since: 0.10.4
+</return>
+</function>
+
<function name="gst_plugin_get_origin">
<description>
get the URL where the plugin comes from
@@ -13618,6 +16064,20 @@
</return>
</function>
+<function name="gst_collect_pads_new">
+<description>
+Create a new instance of #GstCollectsPads.
+
+
+</description>
+<parameters>
+</parameters>
+<return> a new #GstCollectPads, or NULL in case of an error.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_message_parse_warning">
<description>
Extracts the GError and debug string from the GstMessage. The values returned
@@ -13643,6 +16103,18 @@
<return></return>
</function>
+<function name="gst_interpolation_control_source_new">
+<description>
+This returns a new, unbound #GstInterpolationControlSource.
+
+
+</description>
+<parameters>
+</parameters>
+<return> a new, unbound #GstInterpolationControlSource.
+</return>
+</function>
+
<function name="gst_message_type_get_name">
<description>
Get a printable name for the given message type. Do not modify or free.
@@ -13659,6 +16131,30 @@
</return>
</function>
+<function name="gst_collect_pads_available">
+<description>
+Query how much bytes can be read from each queued buffer. This means
+that the result of this call is the maximum number of bytes that can
+be read from each of the pads.
+
+This function should be called with @pads LOCK held, such as
+in the callback.
+
+returns 0 if a pad has no queued buffer.
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to query
+</parameter_description>
+</parameter>
+</parameters>
+<return>0 if a pad has no queued buffer.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_caps_new_full">
<description>
Creates a new #GstCaps and adds all the structures listed as
@@ -13703,6 +16199,29 @@
</return>
</function>
+<function name="gst_base_sink_set_ts_offset">
+<description>
+Adjust the synchronisation of @sink with @offset. A negative value will
+render buffers earlier than their timestamp. A positive value will delay
+rendering. This function can be used to fix playback of badly timestamped
+buffers.
+
+Since: 0.10.15
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="offset">
+<parameter_description> the new offset
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_pad_get_negotiated_caps">
<description>
Gets the capabilities of the media type that currently flows through @pad
@@ -14020,6 +16539,34 @@
</return>
</function>
+<function name="gst_object_set_control_source">
+<description>
+Sets the #GstControlSource for @property_name. If there already was a #GstControlSource
+for this property it will be unreferenced.
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the controller object
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> name of the property for which the #GstControlSource should be set
+</parameter_description>
+</parameter>
+<parameter name="csource">
+<parameter_description> the #GstControlSource that should be used for the property
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if the given property isn't handled by the controller or the new #GstControlSource
+couldn't be bound to the property, %TRUE if everything worked as expected.
+
+Since: 0.10.14
+</return>
+</function>
+
<function name="gst_value_set_caps">
<description>
Sets the contents of @value to @caps. The actual
@@ -14244,31 +16791,58 @@
</return>
</function>
-<function name="gst_query_new_application">
+<function name="gst_poll_fd_ctl_write">
<description>
-Constructs a new custom application query object. Use gst_query_unref()
-when done with it.
+Control whether the descriptor @fd in @set will be monitored for
+writability.
</description>
<parameters>
-<parameter name="type">
-<parameter_description> the query type
+<parameter name="set">
+<parameter_description> a file descriptor set.
</parameter_description>
</parameter>
-<parameter name="structure">
-<parameter_description> a structure for the query
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+<parameter name="active">
+<parameter_description> a new status.
</parameter_description>
</parameter>
</parameters>
-<return> a #GstQuery
+<return> %TRUE if the descriptor was successfully updated.
+
+Since: 0.10.18
</return>
</function>
-<function name="gst_type_find_register">
+<function name="gst_query_new_application">
<description>
-Registers a new typefind function to be used for typefinding. After
-registering this function will be available for typefinding.
+Constructs a new custom application query object. Use gst_query_unref()
+when done with it.
+
+
+</description>
+<parameters>
+<parameter name="type">
+<parameter_description> the query type
+</parameter_description>
+</parameter>
+<parameter name="structure">
+<parameter_description> a structure for the query
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstQuery
+</return>
+</function>
+
+<function name="gst_type_find_register">
+<description>
+Registers a new typefind function to be used for typefinding. After
+registering this function will be available for typefinding.
This function is typically called during an element's plugin initialization.
@@ -14315,6 +16889,28 @@
</return>
</function>
+<function name="gst_poll_fd_has_error">
+<description>
+Check if @fd in @set has an error.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the descriptor has an error.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_structure_set_value">
<description>
Sets the field with the given name @field to @value. If the field
@@ -14365,6 +16961,33 @@
</return>
</function>
+<function name="gst_poll_fd_ignored">
+<description>
+Mark @fd as ignored so that the next call to gst_poll_wait() will yield
+the same result for @fd as last time. This function must be called if no
+operation (read/write/recv/send/etc.) will be performed on @fd before
+the next call to gst_poll_wait().
+
+The reason why this is needed is because the underlying implementation
+might not allow querying the fd more than once between calls to one of
+the re-enabling operations.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_clock_get_calibration">
<description>
Gets the internal rate and reference time of @clock. See
@@ -14468,6 +17091,30 @@
</return>
</function>
+<function name="gst_data_queue_set_flushing">
+<description>
+Sets the queue to flushing state if @flushing is #TRUE. If set to flushing
+state, any incoming data on the @queue will be discarded. Any call currently
+blocking on #gst_data_queue_push or #gst_data_queue_pop will return straight
+away with a return value of #FALSE. While the @queue is in flushing state,
+all calls to those two functions will return #FALSE.
+
+MT Safe.
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> a #GstDataQueue.
+</parameter_description>
+</parameter>
+<parameter name="flushing">
+<parameter_description> a #gboolean stating if the queue will be flushing or not.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_ghost_pad_get_target">
<description>
Get the target pad of #gpad. Unref target pad after usage.
@@ -14650,6 +17297,37 @@
<return></return>
</function>
+<function name="GstLFOControlSource">
+<description>
+Specifies the offset for the waveform of this #GstLFOControlSource.
+
+It should be given as a #GValue with a type that can be transformed
+to the type of the bound property.
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_base_sink_get_ts_offset">
+<description>
+Get the synchronisation offset of @sink.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> The synchronisation offset.
+
+Since: 0.10.15
+</return>
+</function>
+
<function name="gst_tag_list_get_string">
<description>
Copies the contents for the given tag into the value, possibly merging
@@ -14710,6 +17388,80 @@
</return>
</function>
+<function name="gst_controller_set_disabled">
+<description>
+This function is used to disable all properties of the #GstController
+for some time, i.e. gst_controller_sync_values() will do nothing.
+
+Since: 0.10.14
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstController which should be disabled
+</parameter_description>
+</parameter>
+<parameter name="disabled">
+<parameter_description> boolean that specifies whether to disable the controller
+or not.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_collect_pads_read_buffer">
+<description>
+Get a subbuffer of @size bytes from the given pad @data.
+
+This function should be called with @pads LOCK held, such as in the callback.
+
+Since: 0.10.18
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to query
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to use
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the number of bytes to read
+</parameter_description>
+</parameter>
+</parameters>
+<return> A sub buffer. The size of the buffer can be less that requested.
+A return of NULL signals that the pad is end-of-stream.
+Unref the buffer after use.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_base_sink_is_async_enabled">
+<description>
+Checks if @sink is currently configured to perform asynchronous state
+changes to PAUSED.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the sink is configured to perform asynchronous state
+changes.
+
+Since: 0.10.15
+</return>
+</function>
+
<function name="gst_debug_add_log_function">
<description>
Adds the logging function to the list of logging functions.
@@ -14747,6 +17499,26 @@
</return>
</function>
+<function name="gst_base_sink_set_qos_enabled">
+<description>
+Configures @sink to send Quality-of-Service events upstream.
+
+Since: 0.10.5
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="enabled">
+<parameter_description> the new qos value.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_query_new_duration">
<description>
Constructs a new stream duration query object to query in the given format.
@@ -14848,6 +17620,34 @@
</return>
</function>
+<function name="gst_controller_set_control_source">
+<description>
+Sets the #GstControlSource for @property_name. If there already was a #GstControlSource
+for this property it will be unreferenced.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> name of the property for which the #GstControlSource should be set
+</parameter_description>
+</parameter>
+<parameter name="csource">
+<parameter_description> the #GstControlSource that should be used for the property
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if the given property isn't handled by the controller or the new #GstControlSource
+couldn't be bound to the property, %TRUE if everything worked as expected.
+
+Since: 0.10.14
+</return>
+</function>
+
<function name="gst_caps_is_equal_fixed">
<description>
Tests if two #GstCaps are equal. This function only works on fixed
@@ -15394,20 +18194,24 @@
</return>
</function>
-<function name="gst_index_factory_make">
+<function name="gst_element_provide_clock">
<description>
-Create a new #GstIndex instance from the
-indexfactory with the given name.
+Get the clock provided by the given element.
+&lt;note&gt;An element is only required to provide a clock in the PAUSED
+state. Some elements can provide a clock in other states.&lt;/note&gt;
</description>
<parameters>
-<parameter name="name">
-<parameter_description> the name of the factory used to create the instance
+<parameter name="element">
+<parameter_description> a #GstElement to query
</parameter_description>
</parameter>
</parameters>
-<return> A new #GstIndex instance.
+<return> the GstClock provided by the element or %NULL
+if no clock could be provided. Unref after usage.
+
+MT safe.
</return>
</function>
@@ -15486,6 +18290,33 @@
</return>
</function>
+<function name="gst_poll_fd_ctl_read">
+<description>
+Control whether the descriptor @fd in @set will be monitored for
+readability.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+<parameter name="active">
+<parameter_description> a new status.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the descriptor was successfully updated.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_message_new_segment_done">
<description>
Create a new segment done message. This message is posted by elements that
@@ -15533,6 +18364,28 @@
</return>
</function>
+<function name="gst_dp_crc">
+<description>
+Calculate a CRC for the given buffer over the given number of bytes.
+This is only provided for verification purposes; typical GDP users
+will not need this function.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> array of bytes
+</parameter_description>
+</parameter>
+<parameter name="length">
+<parameter_description> the length of @buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> a two-byte CRC checksum.
+</return>
+</function>
+
<function name="gst_element_implements_interface">
<description>
Test whether the given element implements a certain interface of type
@@ -15664,28 +18517,28 @@
</return>
</function>
-<function name="gst_pad_activate_push">
+<function name="gst_index_entry_assoc_map">
<description>
-Activates or deactivates the given pad in push mode via dispatching to the
-pad's activatepushfunc. For use from within pad activation functions only.
-
-If you don't know what this is, you probably don't want to call it.
+Gets alternative formats associated with the indexentry.
</description>
<parameters>
-<parameter name="pad">
-<parameter_description> the #GstPad to activate or deactivate.
+<parameter name="entry">
+<parameter_description> the index to search
</parameter_description>
</parameter>
-<parameter name="active">
-<parameter_description> whether the pad should be active or not.
+<parameter name="format">
+<parameter_description> the format of the value the find
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> a pointer to store the value
</parameter_description>
</parameter>
</parameters>
-<return> %TRUE if the operation was successful.
-
-MT safe.
+<return> TRUE if there was a value associated with the given
+format.
</return>
</function>
@@ -15832,25 +18685,66 @@
</return>
</function>
-<function name="gst_element_get_pad_from_template">
+<function name="gst_collect_pads_set_function">
<description>
-Gets a pad from @element described by @templ. If the presence of @templ is
-#GST_PAD_REQUEST, requests a new pad. Can return %NULL for #GST_PAD_SOMETIMES
-templates.
+Set the callback function and user data that will be called when
+all the pads added to the collection have buffers queued.
+MT safe.
</description>
<parameters>
-<parameter name="element">
-<parameter_description> a #GstElement.
+<parameter name="pads">
+<parameter_description> the collectspads to use
</parameter_description>
</parameter>
-<parameter name="templ">
-<parameter_description> a #GstPadTemplate belonging to @element.
+<parameter name="func">
+<parameter_description> the function to set
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> user data passed to the function
</parameter_description>
</parameter>
</parameters>
-<return> the #GstPad, or NULL if one could not be found or created.
+<return></return>
+</function>
+
+<function name="gst_collect_pads_start">
+<description>
+Starts the processing of data in the collect_pads.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_element_get_pad_from_template">
+<description>
+Gets a pad from @element described by @templ. If the presence of @templ is
+#GST_PAD_REQUEST, requests a new pad. Can return %NULL for #GST_PAD_SOMETIMES
+templates.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> a #GstElement.
+</parameter_description>
+</parameter>
+<parameter name="templ">
+<parameter_description> a #GstPadTemplate belonging to @element.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstPad, or NULL if one could not be found or created.
</return>
</function>
@@ -15872,6 +18766,24 @@
</return>
</function>
+<function name="gst_data_queue_is_full">
+<description>
+Queries if @queue is full. This check will be done using the
+#GstDataQueueCheckFullCallback registered with @queue.
+MT safe.
+
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> a #GstDataQueue.
+</parameter_description>
+</parameter>
+</parameters>
+<return> #TRUE if @queue is full.
+</return>
+</function>
+
<function name="gst_util_set_object_arg">
<description>
Convertes the string value to the type of the objects argument and
@@ -15911,6 +18823,37 @@
</return>
</function>
+<function name="gst_controller_set">
+<description>
+Set the value of given controller-handled property at a certain time.
+
+Deprecated: Use #GstControlSource, for example #GstInterpolationControlSource
+directly.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object which handles the properties
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property to set
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time the control-change is schedules for
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> the control-value
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if the values couldn't be set (ex : properties not handled by controller), TRUE otherwise
+</return>
+</function>
+
<function name="gst_value_fraction_subtract">
<description>
Subtracts the @subtrahend from the @minuend and sets @dest to the result.
@@ -16002,19 +18945,38 @@
</return>
</function>
-<function name="gst_element_factory_get_klass">
+<function name="gst_element_link_pads_filtered">
<description>
-Gets the class for this factory.
+Links the two named pads of the source and destination elements. Side effect
+is that if one of the pads has no parent, it becomes a child of the parent of
+the other element. If they have different parents, the link fails. If @caps
+is not #NULL, makes sure that the caps of the link is a subset of @caps.
</description>
<parameters>
-<parameter name="factory">
-<parameter_description> a #GstElementFactory
+<parameter name="src">
+<parameter_description> a #GstElement containing the source pad.
+</parameter_description>
+</parameter>
+<parameter name="srcpadname">
+<parameter_description> the name of the #GstPad in source element or NULL for any pad.
+</parameter_description>
+</parameter>
+<parameter name="dest">
+<parameter_description> the #GstElement containing the destination pad.
+</parameter_description>
+</parameter>
+<parameter name="destpadname">
+<parameter_description> the name of the #GstPad in destination element or NULL for any pad.
+</parameter_description>
+</parameter>
+<parameter name="filter">
+<parameter_description> the #GstCaps to filter the link, or #NULL for no filter.
</parameter_description>
</parameter>
</parameters>
-<return> the class
+<return> TRUE if the pads could be linked, FALSE otherwise.
</return>
</function>
@@ -16041,6 +19003,29 @@
</return>
</function>
+<function name="gst_poll_set_flushing">
+<description>
+When @flushing is %TRUE, this function ensures that current and future calls
+to gst_poll_wait() will return -1, with errno set to EBUSY.
+
+Unsetting the flushing state will restore normal operation of @set.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a #GstPoll.
+</parameter_description>
+</parameter>
+<parameter name="flushing">
+<parameter_description> new flushing state.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_caps_new_any">
<description>
Creates a new #GstCaps that indicates that it is compatible with
@@ -16143,26 +19128,53 @@
</return>
</function>
-<function name="gst_pad_set_setcaps_function">
+<function name="gst_collect_pads_is_active">
<description>
-Sets the given setcaps function for the pad. The setcaps function
-will be called whenever a buffer with a new media type is pushed or
-pulled from the pad. The pad/element needs to update its internal
-structures to process the new media type. If this new type is not
-acceptable, the setcaps function should return FALSE.
+Check if a pad is active.
+
+This function is currently not implemented.
+
</description>
<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
<parameter name="pad">
-<parameter_description> a #GstPad.
+<parameter_description> the pad to check
</parameter_description>
</parameter>
-<parameter name="setcaps">
-<parameter_description> the #GstPadSetCapsFunction to set.
+</parameters>
+<return> %TRUE if the pad is active.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_pad_activate_push">
+<description>
+Activates or deactivates the given pad in push mode via dispatching to the
+pad's activatepushfunc. For use from within pad activation functions only.
+
+If you don't know what this is, you probably don't want to call it.
+
+
+</description>
+<parameters>
+<parameter name="pad">
+<parameter_description> the #GstPad to activate or deactivate.
+</parameter_description>
+</parameter>
+<parameter name="active">
+<parameter_description> whether the pad should be active or not.
</parameter_description>
</parameter>
</parameters>
-<return></return>
+<return> %TRUE if the operation was successful.
+
+MT safe.
+</return>
</function>
<function name="gst_event_parse_new_segment_full">
@@ -16211,6 +19223,42 @@
<return></return>
</function>
+<function name="gst_base_sink_set_max_lateness">
+<description>
+Sets the new max lateness value to @max_lateness. This value is
+used to decide if a buffer should be dropped or not based on the
+buffer timestamp and the current clock time. A value of -1 means
+an unlimited time.
+
+Since: 0.10.4
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="max_lateness">
+<parameter_description> the new max lateness value.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_dp_init">
+<description>
+Initialize GStreamer Data Protocol library.
+
+Should be called before using these functions from source linking
+to this source file.
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_object_set_name_prefix">
<description>
Sets the name prefix of @object to @name_prefix.
@@ -16373,19 +19421,19 @@
<return></return>
</function>
-<function name="gst_structure_get_name">
+<function name="gst_element_factory_get_author">
<description>
-Get the name of @structure as a string.
+Gets the author for this factory.
</description>
<parameters>
-<parameter name="structure">
-<parameter_description> a #GstStructure
+<parameter name="factory">
+<parameter_description> a #GstElementFactory
</parameter_description>
</parameter>
</parameters>
-<return> the name of the structure.
+<return> the author
</return>
</function>
@@ -16407,6 +19455,35 @@
</return>
</function>
+<function name="GstBaseSink">
+<description>
+The last buffer that arrived in the sink and was used for preroll or for
+rendering. This property can be used to generate thumbnails. This property
+can be NULL when the sink has not yet received a bufer.
+
+Since: 0.10.15
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_interpolation_control_source_get_all">
+<description>
+Returns: a copy of the list, or %NULL if the property isn't handled by the controller
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource to get the list from
+</parameter_description>
+</parameter>
+</parameters>
+<return> a copy of the list, or %NULL if the property isn't handled by the controller
+</return>
+</function>
+
<function name="gst_tag_setter_set_tag_merge_mode">
<description>
Sets the given merge mode that is used for adding tags from events to tags
@@ -16427,6 +19504,18 @@
<return></return>
</function>
+<function name="gst_adapter_new">
+<description>
+Creates a new #GstAdapter. Free with g_object_unref().
+
+
+</description>
+<parameters>
+</parameters>
+<return> a new #GstAdapter
+</return>
+</function>
+
<function name="gst_child_proxy_get_children_count">
<description>
Gets the number of child objects this parent contains.
@@ -16498,6 +19587,29 @@
</return>
</function>
+<function name="gst_interpolation_control_source_set_interpolation_mode">
+<description>
+Sets the given interpolation mode.
+
+&lt;note&gt;&lt;para&gt;User interpolation is not yet available and quadratic interpolation
+is deprecated and maps to cubic interpolation.&lt;/para&gt;&lt;/note&gt;
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstInterpolationControlSource object
+</parameter_description>
+</parameter>
+<parameter name="mode">
+<parameter_description> interpolation mode
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the interpolation mode could be set, %FALSE otherwise
+</return>
+</function>
+
<function name="gst_message_new_warning">
<description>
Create a new warning message. The message will make copies of @error and
@@ -16544,6 +19656,30 @@
</return>
</function>
+<function name="gst_base_sink_set_sync">
+<description>
+Configures @sink to synchronize on the clock or not. When
+ sync is FALSE, incomming samples will be played as fast as
+possible. If @sync is TRUE, the timestamps of the incomming
+buffers will be used to schedule the exact render time of its
+contents.
+
+Since: 0.10.4
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+<parameter name="sync">
+<parameter_description> the new sync value.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_structure_set_valist">
<description>
va_list form of gst_structure_set().
@@ -16594,21 +19730,70 @@
<return></return>
</function>
-<function name="gst_version_string">
+<function name="gst_index_set_resolver_full">
<description>
-This function returns a string that is useful for describing this version
-of GStreamer to the outside world: user agent strings, logging, ...
+Lets the app register a custom function to map index
+ids to writer descriptions.
+Since: 0.10.18
</description>
<parameters>
-</parameters>
-<return> a newly allocated string describing this version of GStreamer.
-</return>
-</function>
-
-<function name="gst_message_new_info">
-<description>
+<parameter name="index">
+<parameter_description> the index to register the resolver on
+</parameter_description>
+</parameter>
+<parameter name="resolver">
+<parameter_description> the resolver to register
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> data passed to the resolver function
+</parameter_description>
+</parameter>
+<parameter name="user_data_destroy">
+<parameter_description> destroy function for @user_data
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_controller_remove_properties">
+<description>
+Removes the given object properties from the controller
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object from which some properties should be removed
+</parameter_description>
+</parameter>
+<parameter name="Varargs">
+<parameter_description> %NULL terminated list of property names that should be removed
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if one of the given property isn't handled by the controller, %TRUE otherwise
+</return>
+</function>
+
+<function name="gst_version_string">
+<description>
+This function returns a string that is useful for describing this version
+of GStreamer to the outside world: user agent strings, logging, ...
+
+
+</description>
+<parameters>
+</parameters>
+<return> a newly allocated string describing this version of GStreamer.
+</return>
+</function>
+
+<function name="gst_message_new_info">
+<description>
Create a new info message. The message will make copies of @error and
@debug.
@@ -16721,6 +19906,31 @@
<return></return>
</function>
+<function name="gst_controller_get">
+<description>
+Gets the value for the given controller-handled property at the requested
+time.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object which handles the properties
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property to get
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time the control-change should be read from
+</parameter_description>
+</parameter>
+</parameters>
+<return> the GValue of the property at the given time, or %NULL if the property isn't handled by the controller
+</return>
+</function>
+
<function name="gst_event_new_buffer_size">
<description>
Create a new buffersize event. The event is sent downstream and notifies
@@ -16752,6 +19962,23 @@
</return>
</function>
+<function name="gst_data_queue_is_empty">
+<description>
+Queries if there are any items in the @queue.
+MT safe.
+
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> a #GstDataQueue.
+</parameter_description>
+</parameter>
+</parameters>
+<return> #TRUE if @queue is empty.
+</return>
+</function>
+
<function name="gst_structure_remove_field">
<description>
Removes the field with the given name. If the field with the given
@@ -16871,6 +20098,32 @@
</return>
</function>
+<function name="gst_base_transform_set_in_place">
+<description>
+Determines whether a non-writable buffer will be copied before passing
+to the transform_ip function.
+&lt;itemizedlist&gt;
+&lt;listitem&gt;Always TRUE if no transform function is implemented.&lt;/listitem&gt;
+&lt;listitem&gt;Always FALSE if ONLY transform function is implemented.&lt;/listitem&gt;
+&lt;/itemizedlist&gt;
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> the #GstBaseTransform to modify
+</parameter_description>
+</parameter>
+<parameter name="in_place">
+<parameter_description> Boolean value indicating that we would like to operate
+on in_place buffers.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_task_start">
<description>
Starts @task. The @task must have a lock associated with it using
@@ -16921,6 +20174,35 @@
<return></return>
</function>
+<function name="gst_dp_caps_from_packet">
+<description>
+Creates a newly allocated #GstCaps from the given packet.
+
+This function does not check the arguments passed to it, use
+gst_dp_validate_packet() first if the header and payload data are
+unchecked.
+
+
+</description>
+<parameters>
+<parameter name="header_length">
+<parameter_description> the length of the packet header
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> the byte array of the packet header
+</parameter_description>
+</parameter>
+<parameter name="payload">
+<parameter_description> the byte array of the packet payload
+</parameter_description>
+</parameter>
+</parameters>
+<return> A #GstCaps containing the caps represented in the packet,
+or NULL if the packet could not be converted.
+</return>
+</function>
+
<function name="gst_xml_get_topelements">
<description>
Retrieve a list of toplevel elements.
@@ -17371,6 +20653,46 @@
</return>
</function>
+<function name="gst_controller_unset">
+<description>
+Used to remove the value of given controller-handled property at a certain
+time.
+
+Deprecated: Use #GstControlSource, for example #GstInterpolationControlSource
+directly.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object which handles the properties
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property to unset
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time the control-change should be removed from
+</parameter_description>
+</parameter>
+</parameters>
+<return> %FALSE if the values couldn't be unset (ex : properties not handled by controller), %TRUE otherwise
+</return>
+</function>
+
+<function name="gst_lfo_control_source_new">
+<description>
+This returns a new, unbound #GstLFOControlSource.
+
+
+</description>
+<parameters>
+</parameters>
+<return> a new, unbound #GstLFOControlSource.
+</return>
+</function>
+
<function name="gst_element_seek_simple">
<description>
Simple API to perform a seek on the given element, meaning it just seeks
@@ -17435,6 +20757,27 @@
</return>
</function>
+<function name="gst_controller_init">
+<description>
+Initializes the use of the controller library. Suggested to be called right
+after gst_init().
+
+
+</description>
+<parameters>
+<parameter name="argc">
+<parameter_description> pointer to the commandline argument count
+</parameter_description>
+</parameter>
+<parameter name="argv">
+<parameter_description> pointer to the commandline argument values
+</parameter_description>
+</parameter>
+</parameters>
+<return> the %TRUE for success.
+</return>
+</function>
+
<function name="gst_tag_list_get_value_index">
<description>
Gets the value that is at the given index for the given tag in the given
@@ -17642,6 +20985,34 @@
</return>
</function>
+<function name="gst_poll_wait">
+<description>
+Wait for activity on the file descriptors in @set. This function waits up to
+the specified @timeout. A timeout of #GST_CLOCK_TIME_NONE waits forever.
+
+When this function is called from multiple threads, -1 will be returned with
+errno set to EPERM.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a #GstPoll.
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout in nanoseconds.
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of #GstPollFD in @set that have activity or 0 when no
+activity was detected after @timeout. If an error occurs, -1 is returned
+and errno is set.
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_caps_ref">
<description>
Add a reference to a #GstCaps object.
@@ -17827,6 +21198,25 @@
</return>
</function>
+<function name="gst_controller_get_all">
+<description>
+Returns: a copy of the list, or %NULL if the property isn't handled by the controller
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller to get the list from
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> the name of the property to get the list for
+</parameter_description>
+</parameter>
+</parameters>
+<return> a copy of the list, or %NULL if the property isn't handled by the controller
+</return>
+</function>
+
<function name="gst_structure_id_set_valist">
<description>
va_list form of gst_structure_id_set().
@@ -17893,6 +21283,30 @@
<return></return>
</function>
+<function name="gst_controller_get_control_source">
+<description>
+Gets the corresponding #GstControlSource for the property. This should be unreferenced
+again after use.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller object
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> name of the property for which the #GstControlSource should be get
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstControlSource for @property_name or NULL if the property is not
+controlled by this controller or no #GstControlSource was assigned yet.
+
+Since: 0.10.14
+</return>
+</function>
+
<function name="gst_pad_proxy_getcaps">
<description>
Calls gst_pad_get_allowed_caps() for every other pad belonging to the
@@ -18347,6 +21761,44 @@
<return></return>
</function>
+<function name="gst_filter_run">
+<description>
+Iterates over the elements in @list, calling @func with the
+list item data for each item. If @func returns TRUE, @data is
+prepended to the list of results returned. If @first is true,
+the search is halted after the first result is found.
+
+Since gst_filter_run() knows nothing about the type of @data, no
+reference will be taken (if @data refers to an object) and no copy of
+ data wil be made in any other way when prepending @data to the list of
+results.
+
+
+</description>
+<parameters>
+<parameter name="list">
+<parameter_description> a linked list
+</parameter_description>
+</parameter>
+<parameter name="func">
+<parameter_description> the function to execute for each item
+</parameter_description>
+</parameter>
+<parameter name="first">
+<parameter_description> flag to stop execution after a successful item
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> user data
+</parameter_description>
+</parameter>
+</parameters>
+<return> the list of results. Free with g_list_free() when no longer needed
+(the data contained in the list is a flat copy and does need to be
+unreferenced or freed).
+</return>
+</function>
+
<function name="gst_element_change_state">
<description>
Perform @transition on @element.
@@ -18519,6 +21971,45 @@
</return>
</function>
+<function name="gst_collect_pads_add_pad">
+<description>
+Add a pad to the collection of collect pads. The pad has to be
+a sinkpad. The refcount of the pad is incremented. Use
+gst_collect_pads_remove_pad() to remove the pad from the collection
+again.
+
+You specify a size for the returned #GstCollectData structure
+so that you can use it to store additional information.
+
+The pad will be automatically activated in push mode when @pads is
+started.
+
+This function calls gst_collect_pads_add_pad() passing a value of NULL
+for destroy_notify.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> the pad to add
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of the returned #GstCollectData structure
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstCollectData to identify the new pad. Or NULL
+if wrong parameters are supplied.
+
+MT safe.
+</return>
+</function>
+
<function name="gst_mini_object_make_writable">
<description>
Checks if a mini-object is writable. If not, a writable copy is made and
@@ -18587,22 +22078,103 @@
</return>
</function>
-<function name="gst_object_restore_thyself">
+<function name="gst_object_get_control_source">
<description>
-Restores @object with the data from the parent XML node.
+Gets the corresponding #GstControlSource for the property. This should be unreferenced
+again after use.
+
</description>
<parameters>
<parameter name="object">
-<parameter_description> a #GstObject to load into
+<parameter_description> the object
</parameter_description>
</parameter>
-<parameter name="self">
-<parameter_description> The XML node to load @object from
+<parameter name="property_name">
+<parameter_description> name of the property for which the #GstControlSource should be get
</parameter_description>
</parameter>
</parameters>
-<return></return>
+<return> the #GstControlSource for @property_name or NULL if the property is not
+controlled by this controller or no #GstControlSource was assigned yet.
+
+Since: 0.10.14
+</return>
+</function>
+
+<function name="gst_object_restore_thyself">
+<description>
+Restores @object with the data from the parent XML node.
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> a #GstObject to load into
+</parameter_description>
+</parameter>
+<parameter name="self">
+<parameter_description> The XML node to load @object from
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_collect_pads_remove_pad">
+<description>
+Remove a pad from the collection of collect pads. This function will also
+free the #GstCollectData and all the resources that were allocated with
+gst_collect_pads_add_pad().
+
+The pad will be deactivated automatically when @pads is stopped.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> the pad to remove
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the pad could be removed.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_net_time_packet_send">
+<description>
+Sends a #GstNetTimePacket over a socket. Essentially a thin wrapper around
+sendto(2) and gst_net_time_packet_serialize().
+
+MT safe.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> the #GstNetTimePacket
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor created by socket(2)
+</parameter_description>
+</parameter>
+<parameter name="addr">
+<parameter_description> a pointer to a sockaddr to hold the address of the sender
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the size of the data pointed to by @addr
+</parameter_description>
+</parameter>
+</parameters>
+<return> The return value of sendto(2).
+</return>
</function>
<function name="gst_query_set_position">
@@ -18627,6 +22199,26 @@
<return></return>
</function>
+<function name="gst_data_queue_drop_head">
+<description>
+Pop and unref the head-most #GstMiniObject with the given #GType.
+
+
+</description>
+<parameters>
+<parameter name="queue">
+<parameter_description> The #GstDataQueue to drop an item from.
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> The #GType of the item to drop.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if an element was removed.
+</return>
+</function>
+
<function name="gst_index_set_resolver">
<description>
Lets the app register a custom function to map index
@@ -18671,6 +22263,57 @@
</return>
</function>
+<function name="gst_collect_pads_collect">
+<description>
+Collect data on all pads. This function is usually called
+from a #GstTask function in an element.
+
+This function is currently not implemented.
+
+
+</description>
+<parameters>
+<parameter name="pads">
+<parameter_description> the collectspads to use
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GstFlowReturn of the operation.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_dp_header_from_buffer">
+<description>
+Creates a GDP header from the given buffer.
+
+Deprecated: use a #GstDPPacketizer
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a #GstBuffer to create a header for
+</parameter_description>
+</parameter>
+<parameter name="flags">
+<parameter_description> the #GDPHeaderFlags to create the header with
+</parameter_description>
+</parameter>
+<parameter name="length">
+<parameter_description> a guint pointer to store the header length in
+</parameter_description>
+</parameter>
+<parameter name="header">
+<parameter_description> a guint8 * pointer to store a newly allocated header byte array in
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the header was successfully created.
+</return>
+</function>
+
<function name="gst_element_found_tags">
<description>
Posts a message to the bus that new tags were found, and pushes an event
@@ -18693,38 +22336,38 @@
<return></return>
</function>
-<function name="gst_element_link_pads_filtered">
+<function name="gst_element_factory_get_klass">
<description>
-Links the two named pads of the source and destination elements. Side effect
-is that if one of the pads has no parent, it becomes a child of the parent of
-the other element. If they have different parents, the link fails. If @caps
-is not #NULL, makes sure that the caps of the link is a subset of @caps.
+Gets the class for this factory.
</description>
<parameters>
-<parameter name="src">
-<parameter_description> a #GstElement containing the source pad.
-</parameter_description>
-</parameter>
-<parameter name="srcpadname">
-<parameter_description> the name of the #GstPad in source element or NULL for any pad.
-</parameter_description>
-</parameter>
-<parameter name="dest">
-<parameter_description> the #GstElement containing the destination pad.
-</parameter_description>
-</parameter>
-<parameter name="destpadname">
-<parameter_description> the name of the #GstPad in destination element or NULL for any pad.
+<parameter name="factory">
+<parameter_description> a #GstElementFactory
</parameter_description>
</parameter>
-<parameter name="filter">
-<parameter_description> the #GstCaps to filter the link, or #NULL for no filter.
+</parameters>
+<return> the class
+</return>
+</function>
+
+<function name="gst_adapter_available_fast">
+<description>
+Gets the maximum number of bytes that are immediately available without
+requiring any expensive operations (like copying the data into a
+temporary buffer).
+
+
+</description>
+<parameters>
+<parameter name="adapter">
+<parameter_description> a #GstAdapter
</parameter_description>
</parameter>
</parameters>
-<return> TRUE if the pads could be linked, FALSE otherwise.
+<return> number of bytes that are available in @adapter without expensive
+operations
</return>
</function>
@@ -18780,6 +22423,27 @@
</return>
</function>
+<function name="gst_poll_new">
+<description>
+Create a new file descriptor set. If @controllable, it
+is possible to restart or flush a call to gst_poll_wait() with
+gst_poll_restart() and gst_poll_set_flushing() respectively.
+
+
+</description>
+<parameters>
+<parameter name="controllable">
+<parameter_description> whether it should be possible to control a wait.
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstPoll, or %NULL in case of an error. Free with
+gst_poll_free().
+
+Since: 0.10.18
+</return>
+</function>
+
<function name="gst_pad_add_event_probe">
<description>
Adds a probe that will be called for all events passing through a pad. See
@@ -18805,6 +22469,23 @@
</return>
</function>
+<function name="gst_poll_fd_init">
+<description>
+Initializes @fd. Alternatively you can initialize it with
+#GST_POLL_FD_INIT.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="fd">
+<parameter_description> a #GstPollFD
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_xml_write">
<description>
Converts the given element into an XML presentation.
@@ -18872,6 +22553,83 @@
</return>
</function>
+<function name="gst_controller_sync_values">
+<description>
+Sets the properties of the element, according to the controller that (maybe)
+handles them and for the given timestamp.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the controller that handles the values
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the time that should be processed
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the controller values could be applied to the object
+properties, %FALSE otherwise
+</return>
+</function>
+
+<function name="gst_poll_add_fd">
+<description>
+Add a file descriptor to the file descriptor set.
+
+
+</description>
+<parameters>
+<parameter name="set">
+<parameter_description> a file descriptor set.
+</parameter_description>
+</parameter>
+<parameter name="fd">
+<parameter_description> a file descriptor.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the file descriptor was successfully added to the set.
+
+Since: 0.10.18
+</return>
+</function>
+
+<function name="gst_object_suggest_next_sync">
+<description>
+Convenience function for GObject
+
+
+</description>
+<parameters>
+<parameter name="object">
+<parameter_description> the object that has controlled properties
+</parameter_description>
+</parameter>
+</parameters>
+<return> same thing as gst_controller_suggest_next_sync()
+Since: 0.10.13
+</return>
+</function>
+
+<function name="gst_check_caps_equal">
+<description>
+ caps1: first caps to compare
+ caps2: second caps to compare
+
+Compare two caps with gst_caps_is_equal and fail unless they are
+equal.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_child_proxy_get_child_by_index">
<description>
Fetches a child by its number.
@@ -18895,6 +22653,29 @@
</return>
</function>
+<function name="gst_base_sink_wait_preroll">
+<description>
+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.
+
+This function will block until a state change to PLAYING happens (in which
+case this function returns #GST_FLOW_OK) or the processing must be stopped due
+to a state change to READY or a FLUSH event (in which case this function
+Returns: #GST_FLOW_OK if the preroll completed and processing can
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> the sink
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_FLOW_OK if the preroll completed and processing can
+continue. Any other return value should be returned from the render vmethod.
+</return>
+</function>
+
<function name="gst_bus_poll">
<description>
Poll the bus for messages. Will block while waiting for messages to come.
@@ -19083,28 +22864,33 @@
<return></return>
</function>
-<function name="gst_tag_list_merge">
+<function name="gst_element_register">
<description>
-Merges the two given lists into a new list. If one of the lists is NULL, a
-copy of the other is returned. If both lists are NULL, NULL is returned.
+Create a new elementfactory capable of instantiating objects of the
+ type and add the factory to @plugin.
</description>
<parameters>
-<parameter name="list1">
-<parameter_description> first list to merge
+<parameter name="plugin">
+<parameter_description> #GstPlugin to register the element with, or NULL for a static
+element (note that passing NULL only works in GStreamer 0.10.13 and later)
</parameter_description>
</parameter>
-<parameter name="list2">
-<parameter_description> second list to merge
+<parameter name="name">
+<parameter_description> name of elements of this type
</parameter_description>
</parameter>
-<parameter name="mode">
-<parameter_description> the mode to use
+<parameter name="rank">
+<parameter_description> rank of element (higher rank means more importance when autoplugging)
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> GType of element to register
</parameter_description>
</parameter>
</parameters>
-<return> the new list
+<return> TRUE, if the registering succeeded, FALSE on error
</return>
</function>
@@ -19155,6 +22941,60 @@
</return>
</function>
+<function name="gst_base_transform_set_gap_aware">
+<description>
+If @gap_aware is %FALSE (the default), output buffers will have the
+%GST_BUFFER_FLAG_GAP flag unset.
+
+If set to %TRUE, the element must handle output buffers with this flag set
+correctly, i.e. it can assume that the buffer contains neutral data but must
+unset the flag if the output is no neutral data.
+
+Since: 0.10.16
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> a #GstBaseTransform
+</parameter_description>
+</parameter>
+<parameter name="gap_aware">
+<parameter_description> New state
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_controller_set_property_disabled">
+<description>
+This function is used to disable the #GstController on a property for
+some time, i.e. gst_controller_sync_values() will do nothing for the
+property.
+
+Since: 0.10.14
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> the #GstController which should be disabled
+</parameter_description>
+</parameter>
+<parameter name="property_name">
+<parameter_description> property to disable
+</parameter_description>
+</parameter>
+<parameter name="disabled">
+<parameter_description> boolean that specifies whether to disable the controller
+or not.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
<function name="gst_index_add_format">
<description>
Adds a format entry into the index. This function is
@@ -19181,33 +23021,28 @@
</return>
</function>
-<function name="gst_element_register">
+<function name="gst_tag_list_merge">
<description>
-Create a new elementfactory capable of instantiating objects of the
- type and add the factory to @plugin.
+Merges the two given lists into a new list. If one of the lists is NULL, a
+copy of the other is returned. If both lists are NULL, NULL is returned.
</description>
<parameters>
-<parameter name="plugin">
-<parameter_description> #GstPlugin to register the element with, or NULL for a static
-element (note that passing NULL only works in GStreamer 0.10.13 and later)
-</parameter_description>
-</parameter>
-<parameter name="name">
-<parameter_description> name of elements of this type
+<parameter name="list1">
+<parameter_description> first list to merge
</parameter_description>
</parameter>
-<parameter name="rank">
-<parameter_description> rank of element (higher rank means more importance when autoplugging)
+<parameter name="list2">
+<parameter_description> second list to merge
</parameter_description>
</parameter>
-<parameter name="type">
-<parameter_description> GType of element to register
+<parameter name="mode">
+<parameter_description> the mode to use
</parameter_description>
</parameter>
</parameters>
-<return> TRUE, if the registering succeeded, FALSE on error
+<return> the new list
</return>
</function>
Modified: gstreamermm/trunk/gstreamerbase/src/gstbase_docs.xml
==============================================================================
--- gstreamermm/trunk/gstreamerbase/src/gstbase_docs.xml (original)
+++ gstreamermm/trunk/gstreamerbase/src/gstbase_docs.xml Thu Mar 27 00:01:26 2008
@@ -1,2 +1,9599 @@
<root>
+<function name="gst_rtsp_find_header_field">
+<description>
+Convert @header to a #GstRTSPHeaderField.
+
+
+</description>
+<parameters>
+<parameter name="header">
+<parameter_description> a header string
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPHeaderField for @header or #GST_RTSP_HDR_INVALID if the
+header field is unknown.
+</return>
+</function>
+
+<function name="gst_ring_buffer_set_sample">
+<description>
+Make sure that the next sample written to the device is
+accounted for as being the @sample sample written to the
+device. This value will be used in reporting the current
+sample position of the ringbuffer.
+
+This function will also clear the buffer with silence.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to use
+</parameter_description>
+</parameter>
+<parameter name="sample">
+<parameter_description> the sample number to set
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_audio_default_registry_mixer_filter">
+<description>
+Utility function to find audio mixer elements.
+
+Will traverse the default plugin registry in order of plugin rank and
+find usable audio mixer elements. The caller may optionally fine-tune
+the selection by specifying a filter function.
+
+
+</description>
+<parameters>
+<parameter name="filter_func">
+<parameter_description> filter function, or #NULL
+</parameter_description>
+</parameter>
+<parameter name="first">
+<parameter_description> set to #TRUE if you only want the first suitable mixer element
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> user data to pass to the filter function
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GList of audio mixer #GstElement&lt;!-- --&gt;s. You must free each
+element in the list by setting it to NULL state and calling
+gst_object_unref(). After that the list itself should be freed
+using g_list_free().
+
+Since: 0.10.2
+</return>
+</function>
+
+<function name="gst_tuner_list_channels">
+<description>
+Retrieve a list of channels (e.g. 'composite', 's-video', ...)
+from the given tuner object.
+
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) to get the channels from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> a list of channels available on this tuner.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_attribute">
+<description>
+Add the attribute with @key and @value to @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> the key
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> the value
+</parameter_description>
+</parameter>
+</parameters>
+<return> @GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_sdp_media_set_information">
+<description>
+Set the media information of @media to @information.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="information">
+<parameter_description> the media information
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_mixer_option_changed">
+<description>
+This function is called by the mixer implementation to produce
+a notification message on the bus indicating that the given options
+object has changed state.
+
+This function only works for GstElements that are implementing the
+GstMixer interface, and the element needs to have been provided a bus.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the options
+</parameter_description>
+</parameter>
+<parameter name="opts">
+<parameter_description> the GstMixerOptions that has changed value.
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> the new value of the GstMixerOptions.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_new_allocate_len">
+<description>
+Create a new #GstBuffer that can hold an RTP packet that is exactly
+ packet_len long. The length of the payload depends on @pad_len and
+ csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len().
+All RTP header fields will be set to 0/FALSE.
+
+
+</description>
+<parameters>
+<parameter name="packet_len">
+<parameter_description> the total length of the packet
+</parameter_description>
+</parameter>
+<parameter name="pad_len">
+<parameter_description> the amount of padding
+</parameter_description>
+</parameter>
+<parameter name="csrc_count">
+<parameter_description> the number of CSRC entries
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer that can hold an RTP packet of @packet_len.
+</return>
+</function>
+
+<function name="gst_missing_uri_source_installer_detail_new">
+<description>
+Returns: a newly-allocated detail string, or NULL on error. Free string
+
+</description>
+<parameters>
+<parameter name="protocol">
+<parameter_description> the URI protocol the missing source needs to implement,
+e.g. "http" or "mms"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated detail string, or NULL on error. Free string
+with g_free() when not needed any longer.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_seq">
+<description>
+Set the sequence number of the RTP packet in @buffer to @seq.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="seq">
+<parameter_description> the new sequence number
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_message_get_header">
+<description>
+Get the @indx header value with key @field from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="field">
+<parameter_description> a #GstRTSPHeaderField
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> pointer to hold the result
+</parameter_description>
+</parameter>
+<parameter name="indx">
+<parameter_description> the index of the header
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK when @field was found, #GST_RTSP_ENOTIMPL if the key
+was not found.
+</return>
+</function>
+
+<function name="gst_sdp_media_init">
+<description>
+Initialize @media so that its contents are as if it was freshly allocated
+with gst_sdp_media_new(). This function is mostly used to initialize a media
+allocated on the stack. gst_sdp_media_uninit() undoes this operation.
+
+When this function is invoked on newly allocated data (with malloc or on the
+stack), its contents should be set to 0 before calling this function.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_ring_buffer_debug_spec_buff">
+<description>
+Print debug info about the buffer sized in @spec to the debug log.
+
+</description>
+<parameters>
+<parameter name="spec">
+<parameter_description> the spec to debug
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_fft_s16_fft">
+<description>
+This performs the FFT on @timedata and puts the result in @freqdata.
+
+ timedata must have as many samples as specified with the @len parameter while
+allocating the #GstFFTS16 instance with gst_fft_s16_new().
+
+ freqdata must be large enough to hold @len/2 + 1 #GstFFTS16Complex frequency
+domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS16 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Buffer of the samples in the time domain
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Target buffer for the samples in the frequency domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_netaddress_get_net_type">
+<description>
+Get the type of address stored in @naddr.
+
+
+</description>
+<parameters>
+<parameter name="naddr">
+<parameter_description> a network address
+</parameter_description>
+</parameter>
+</parameters>
+<return> the network type stored in @naddr.
+</return>
+</function>
+
+<function name="gst_video_orientation_get_hcenter">
+<description>
+Get the horizontal centering offset from the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="center">
+<parameter_description> return location for the result
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports centering
+</return>
+</function>
+
+<function name="gst_fft_f64_inverse_fft">
+<description>
+This performs the inverse FFT on @freqdata and puts the result in @timedata.
+
+ freqdata must have @len/2 + 1 samples, where @len is the parameter specified
+while allocating the #GstFFTF64 instance with gst_fft_f64_new().
+
+ timedata must be large enough to hold @len time domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF64 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Buffer of the samples in the frequency domain
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Target buffer for the samples in the time domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_packet_add_rb">
+<description>
+Add a new report block to @packet with the given values.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SR or RR #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> data source being reported
+</parameter_description>
+</parameter>
+<parameter name="fractionlost">
+<parameter_description> fraction lost since last SR/RR
+</parameter_description>
+</parameter>
+<parameter name="packetslost">
+<parameter_description> the cumululative number of packets lost
+</parameter_description>
+</parameter>
+<parameter name="exthighestseq">
+<parameter_description> the extended last sequence number received
+</parameter_description>
+</parameter>
+<parameter name="jitter">
+<parameter_description> the interarrival jitter
+</parameter_description>
+</parameter>
+<parameter name="lsr">
+<parameter_description> the last SR packet from this source
+</parameter_description>
+</parameter>
+<parameter name="dlsr">
+<parameter_description> the delay since last SR packet
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the packet was created. This function can return %FALSE if
+the max MTU is exceeded or the number of report blocks is greater than
+#GST_RTCP_MAX_RB_COUNT.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_next_entry">
+<description>
+Move to the next SDES entry in the current item.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if there was a next entry.
+</return>
+</function>
+
+<function name="gst_rtsp_transport_as_text">
+<description>
+Convert @transport into a string that can be used to signal the transport in
+an RTSP SETUP response.
+
+
+</description>
+<parameters>
+<parameter name="transport">
+<parameter_description> a #GstRTSPTransport
+</parameter_description>
+</parameter>
+</parameters>
+<return> a string describing the RTSP transport or #NULL when the transport
+is invalid.
+</return>
+</function>
+
+<function name="gst_missing_decoder_message_new">
+<description>
+Creates a missing-plugin message for @element to notify the application
+that a decoder element for a particular set of (fixed) caps is missing.
+This function is mainly for use in plugins.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> the #GstElement posting the message
+</parameter_description>
+</parameter>
+<parameter name="decode_caps">
+<parameter_description> the (fixed) caps for which a decoder element is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstMessage, or NULL on error
+</return>
+</function>
+
+<function name="gst_mixer_set_mute">
+<description>
+Mutes or unmutes the given channel. To find out whether a
+track is currently muted, use GST_MIXER_TRACK_HAS_FLAG ().
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the track.
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> the #GstMixerTrack to operate on.
+</parameter_description>
+</parameter>
+<parameter name="mute">
+<parameter_description> a boolean value indicating whether to turn on or off
+muting.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_ring_buffer_acquire">
+<description>
+Allocate the resources for the ringbuffer. This function fills
+in the data pointer of the ring buffer with a valid #GstBuffer
+to which samples can be written.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to acquire
+</parameter_description>
+</parameter>
+<parameter name="spec">
+<parameter_description> the specs of the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be acquired, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_mixer_options_get_values">
+<description>
+Get the values for the mixer option.
+
+
+</description>
+<parameters>
+<parameter name="mixer_options">
+<parameter_description> The #GstMixerOptions item that owns the values.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A list of strings with all the possible values for the mixer
+option. You must not free or modify the list or its contents, it belongs
+to the @mixer_options object.
+</return>
+</function>
+
+<function name="gst_basertppayload_is_filled">
+<description>
+Check if the packet with @size and @duration would exceed the configure
+maximum size.
+
+
+</description>
+<parameters>
+<parameter name="payload">
+<parameter_description> a #GstBaseRTPPayload
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of the packet
+</parameter_description>
+</parameter>
+<parameter name="duration">
+<parameter_description> the duration of the packet
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the packet of @size and @duration would exceed the
+configured MTU or max_ptime.
+</return>
+</function>
+
+<function name="gst_mixer_set_record">
+<description>
+Enables or disables recording on the given track. Note that
+this is only possible on input tracks, not on output tracks
+(see GST_MIXER_TRACK_HAS_FLAG () and the GST_MIXER_TRACK_INPUT
+flag).
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> The #GstMixer (a #GstElement) that owns the track.
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> the #GstMixerTrack to operate on.
+</parameter_description>
+</parameter>
+<parameter name="record">
+<parameter_description> a boolean value that indicates whether to turn on
+or off recording.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_tag_to_vorbis_comments">
+<description>
+Creates a new tag list that contains the information parsed out of a
+vorbiscomment packet.
+
+
+</description>
+<parameters>
+<parameter name="list">
+<parameter_description> a #GstTagList
+</parameter_description>
+</parameter>
+<parameter name="tag">
+<parameter_description> a GStreamer tag identifier, such as #GST_TAG_ARTIST
+</parameter_description>
+</parameter>
+</parameters>
+<return> A #GList of newly-allowcated key=value strings. Free with
+g_list_foreach (list, (GFunc) g_free, NULL) plus g_list_free (list)
+</return>
+</function>
+
+<function name="gst_property_probe_needs_probe">
+<description>
+Checks whether a property needs a probe. This might be because
+the property wasn't initialized before, or because host setup
+changed. This might be, for example, because a new device was
+added, and thus device probing needs to be refreshed to display
+the new device.
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe object to which the given property belongs.
+</parameter_description>
+</parameter>
+<parameter name="pspec">
+<parameter_description> a #GParamSpec that identifies the property to check.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the property needs a new probe, FALSE if not.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_set_auth">
+<description>
+Configure @conn for authentication mode @method with @user and @pass as the
+user and password respectively.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="method">
+<parameter_description> authentication method
+</parameter_description>
+</parameter>
+<parameter name="user">
+<parameter_description> the user
+</parameter_description>
+</parameter>
+<parameter name="pass">
+<parameter_description> the password
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_rtcp_buffer_new_take_data">
+<description>
+Create a new buffer and set the data and size of the buffer to @data and @len
+respectively. @data will be freed when the buffer is unreffed, so this
+function transfers ownership of @data to the new buffer.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> data for the new buffer
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of data
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer with @data and of size @len.
+</return>
+</function>
+
+<function name="gst_x_overlay_prepare_xwindow_id">
+<description>
+This will post a "prepare-xwindow-id" element message on the bus
+to give applications an opportunity to call
+gst_x_overlay_set_xwindow_id() before a plugin creates its own
+window.
+
+This function should only be used by video overlay plugin developers.
+
+</description>
+<parameters>
+<parameter name="overlay">
+<parameter_description> a #GstXOverlay which does not yet have an XWindow.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_add_attribute">
+<description>
+Add the attribute with @key and @value to @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> a key
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> a value
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_tuner_set_frequency">
+<description>
+Sets a tuning frequency on the given tuner/channel. Note that this
+requires the given channel to be a "tuning" channel, which can be
+checked using GST_TUNER_CHANNEL_HAS_FLAG (), with the proper flag
+being GST_TUNER_CHANNEL_FREQUENCY.
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #Gsttuner (a #GstElement) that owns the given channel.
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> the #GstTunerChannel to set the frequency on.
+</parameter_description>
+</parameter>
+<parameter name="frequency">
+<parameter_description> the frequency to tune in to.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_calc_packet_len">
+<description>
+Calculate the total length of an RTP packet with a payload size of @payload_len,
+a padding of @pad_len and a @csrc_count CSRC entries.
+
+
+</description>
+<parameters>
+<parameter name="payload_len">
+<parameter_description> the length of the payload
+</parameter_description>
+</parameter>
+<parameter name="pad_len">
+<parameter_description> the amount of padding
+</parameter_description>
+</parameter>
+<parameter name="csrc_count">
+<parameter_description> the number of CSRC entries
+</parameter_description>
+</parameter>
+</parameters>
+<return> The total length of an RTP header with given parameters.
+</return>
+</function>
+
+<function name="gst_rtsp_message_get_type">
+<description>
+Get the message type of @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the message type.
+</return>
+</function>
+
+<function name="gst_rtcp_buffer_validate_data">
+<description>
+Check if the @data and @size point to the data of a valid RTCP (compound)
+packet.
+Use this function to validate a packet before using the other functions in
+this module.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> the data to validate
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of @data to validate
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the data points to a valid RTCP packet.
+</return>
+</function>
+
+<function name="gst_ring_buffer_clear_all">
+<description>
+Fill the ringbuffer with silence.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to clear
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_format_is_rgb">
+<description>
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @format represents RGB video
+</return>
+</function>
+
+<function name="gst_rtsp_message_init">
+<description>
+Initialize @msg. This function is mostly used when @msg is allocated on the
+stack. The reverse operation of this is gst_rtsp_message_unset().
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_mixer_options_list_changed">
+<description>
+This function is called by the mixer implementation to produce
+a notification message on the bus indicating that the list of possible
+options of a given options object has changed.
+
+The new options are not contained in the message on purpose. Applications
+should call gst_mixer_option_get_values() on @opts to make @opts update
+its internal state and obtain the new list of values.
+
+This function only works for GstElements that are implementing the
+GstMixer interface, and the element needs to have been provided a bus
+for this to work.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the options
+</parameter_description>
+</parameter>
+<parameter name="opts">
+<parameter_description> the GstMixerOptions whose list of values has changed
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_set_payload_type">
+<description>
+Set the payload type of the RTP packet in @buffer to @payload_type.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="payload_type">
+<parameter_description> the new type
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_phones_len">
+<description>
+Get the number of phones in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of phones in @msg.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sr_get_sender_info">
+<description>
+Parse the SR sender info and store the values.
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SR #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> result SSRC
+</parameter_description>
+</parameter>
+<parameter name="ntptime">
+<parameter_description> result NTP time
+</parameter_description>
+</parameter>
+<parameter name="rtptime">
+<parameter_description> result RTP time
+</parameter_description>
+</parameter>
+<parameter name="packet_count">
+<parameter_description> result packet count
+</parameter_description>
+</parameter>
+<parameter name="octet_count">
+<parameter_description> result octect count
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_buffer_get_first_packet">
+<description>
+Initialize a new #GstRTCPPacket pointer that points to the first packet in
+ buffer
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a valid RTCP buffer
+</parameter_description>
+</parameter>
+<parameter name="packet">
+<parameter_description> a #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the packet existed in @buffer.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_key">
+<description>
+Get the encryption information from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPKey.
+</return>
+</function>
+
+<function name="gst_sdp_media_free">
+<description>
+Free all resources allocated by @media. @media should not be used anymore after
+this function. This function should be used when @media was dynamically
+allocated with gst_sdp_media_new().
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_sdp_media_set_proto">
+<description>
+Set the media transport protocol of @media to @proto.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="proto">
+<parameter_description> the media transport protocol
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_time">
+<description>
+Add time information @start and @stop to @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="start">
+<parameter_description> the start time
+</parameter_description>
+</parameter>
+<parameter name="stop">
+<parameter_description> the stop time
+</parameter_description>
+</parameter>
+<parameter name="repeat">
+<parameter_description> the repeat times
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_csrc">
+<description>
+Get the CSRC at index @idx in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the index of the CSRC to get
+</parameter_description>
+</parameter>
+</parameters>
+<return> the CSRC at index @idx in host order.
+</return>
+</function>
+
+<function name="gst_video_format_get_pixel_stride">
+<description>
+Calculates the pixel stride (number of bytes from one pixel to the
+pixel to its immediate left) for the video component with an index
+of @component. See @gst_video_format_get_row_stride for a description
+of the component index.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="component">
+<parameter_description> the component index
+</parameter_description>
+</parameter>
+</parameters>
+<return> pixel stride of component @component
+</return>
+</function>
+
+<function name="gst_rtp_buffer_new_take_data">
+<description>
+Create a new buffer and set the data and size of the buffer to @data and @len
+respectively. @data will be freed when the buffer is unreffed, so this
+function transfers ownership of @data to the new buffer.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> data for the new buffer
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of data
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer with @data and of size @len.
+</return>
+</function>
+
+<function name="gst_rtsp_message_add_header">
+<description>
+Add a header with key @field and @value to @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="field">
+<parameter_description> a #GstRTSPHeaderField
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> the value of the header
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_ring_buffer_samples_done">
+<description>
+Get the number of samples that were processed by the ringbuffer
+since it was last started. This does not include the number of samples not
+yet processed (see gst_ring_buffer_delay()).
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to query
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of samples processed by the ringbuffer.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_property_probe_get_values_name">
+<description>
+Same as gst_property_probe_get_values ().
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe object.
+</parameter_description>
+</parameter>
+<parameter name="name">
+<parameter_description> the name of the property to get values for.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A list of valid values for the given property.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_key">
+<description>
+Get the encryption information from @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPKey.
+</return>
+</function>
+
+<function name="gst_ring_buffer_is_acquired">
+<description>
+Check if the ringbuffer is acquired and ready to use.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to check
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the ringbuffer is acquired, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_zone">
+<description>
+Get time zone information with index @idx from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the zone index
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPZone.
+</return>
+</function>
+
+<function name="gst_sdp_message_dump">
+<description>
+Dump the parsed contents of @msg to stdout.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_pb_utils_get_element_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="factory_name">
+<parameter_description> the name of the element, e.g. "gnomevfssrc"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_tag_id3_genre_count">
+<description>
+Gets the number of ID3v1 genres that can be identified. Winamp genres are
+included.
+
+
+</description>
+<parameters>
+</parameters>
+<return> the number of ID3v1 genres that can be identified
+</return>
+</function>
+
+<function name="gst_tag_to_vorbis_tag">
+<description>
+Looks up the vorbiscomment tag for a GStreamer tag.
+
+
+</description>
+<parameters>
+<parameter name="gst_tag">
+<parameter_description> GStreamer tag to convert to vorbiscomment tag
+</parameter_description>
+</parameter>
+</parameters>
+<return> The corresponding vorbiscomment tag or NULL if none exists.
+</return>
+</function>
+
+<function name="gst_rtsp_find_method">
+<description>
+Convert @method to a #GstRTSPMethod.
+
+
+</description>
+<parameters>
+<parameter name="method">
+<parameter_description> a method
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPMethod for @method or #GST_RTSP_INVALID if the
+method is unknown.
+</return>
+</function>
+
+<function name="gst_base_audio_sink_set_slave_method">
+<description>
+Controls how clock slaving will be performed in @sink.
+
+Since: 0.10.16
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> a #GstBaseAudioSink
+</parameter_description>
+</parameter>
+<parameter name="method">
+<parameter_description> the new slave method
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_fft_s16_window">
+<description>
+This calls the window function @window on the @timedata sample buffer.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS16 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Time domain samples
+</parameter_description>
+</parameter>
+<parameter name="window">
+<parameter_description> Window function to apply
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_set_session_name">
+<description>
+Set the session name in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="session_name">
+<parameter_description> the session name
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_calc_header_len">
+<description>
+Calculate the header length of an RTP packet with @csrc_count CSRC entries.
+An RTP packet can have at most 15 CSRC entries.
+
+
+</description>
+<parameters>
+<parameter name="csrc_count">
+<parameter_description> the number of CSRC entries
+</parameter_description>
+</parameter>
+</parameters>
+<return> The length of an RTP header with @csrc_count CSRC entries.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_copy_entry">
+<description>
+This function is like gst_rtcp_packet_sdes_get_entry() but it returns a
+null-terminated copy of the data instead. use g_free() after usage.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> result of the entry type
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> result length of the entry data
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> result entry data
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if there was valid data.
+</return>
+</function>
+
+<function name="gst_missing_plugin_message_get_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a missing-plugin #GstMessage of type #GST_MESSAGE_ELEMENT
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_email">
+<description>
+Add @email to the list of emails in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="email">
+<parameter_description> an email
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtsp_range_free">
+<description>
+Free the memory alocated by @range.
+
+</description>
+<parameters>
+<parameter name="range">
+<parameter_description> a #GstRTSPTimeRange
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_free">
+<description>
+Free all resources allocated by @msg. @msg should not be used anymore after
+this function. This function should be used when @msg was dynamically
+allocated with gst_sdp_message_new().
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_set_samplebits_options">
+<description>
+Sets the options for sample based audio codecs.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="basertpaudiopayload">
+<parameter_description> a pointer to the element.
+</parameter_description>
+</parameter>
+<parameter name="sample_size">
+<parameter_description> Size per sample in bits.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_tag_list_to_vorbiscomment_buffer">
+<description>
+Creates a new vorbiscomment buffer from a tag list.
+
+
+</description>
+<parameters>
+<parameter name="list">
+<parameter_description> tag list to convert
+</parameter_description>
+</parameter>
+<parameter name="id_data">
+<parameter_description> identification data at start of stream
+</parameter_description>
+</parameter>
+<parameter name="id_data_length">
+<parameter_description> length of identification data, may be 0 if @id_data is NULL
+</parameter_description>
+</parameter>
+<parameter name="vendor_string">
+<parameter_description> string that describes the vendor string or NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> A new #GstBuffer containing a vorbiscomment buffer with all tags
+that could be converted from the given tag list.
+</return>
+</function>
+
+<function name="gst_tag_list_new_from_id3v1">
+<description>
+Parses the data containing an ID3v1 tag and returns a #GstTagList from the
+parsed data.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> 128 bytes of data containing the ID3v1 tag
+</parameter_description>
+</parameter>
+</parameters>
+<return> A new tag list or NULL if the data was not an ID3v1 tag.
+</return>
+</function>
+
+<function name="gst_Tuner_get_channel">
+<description>
+Retrieve the current channel from the tuner.
+
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) to get the current channel from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the current channel of the tuner object.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_connection">
+<description>
+Get the connection of @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPConnection. The result remains valid as long as @msg is valid.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_bye_get_reason">
+<description>
+Get the reason in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The reason for the BYE @packet or NULL if the packet did not contain
+a reason string. The string must be freed with g_free() after usage.
+</return>
+</function>
+
+<function name="gst_netaddress_get_ip6_address">
+<description>
+Get the IPv6 address stored in @naddr into @address.
+
+
+</description>
+<parameters>
+<parameter name="naddr">
+<parameter_description> a network address
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> a location to store the result.
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> a location to store the port.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the address could be retrieved.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_marker">
+<description>
+Set the marker bit on the RTP packet in @buffer to @marker.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="marker">
+<parameter_description> the new marker
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_set_version">
+<description>
+Set the version of the RTP packet in @buffer to @version.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="version">
+<parameter_description> the new version
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_base64_decode_ip">
+<description>
+Decode the base64 string pointed to by @data in-place. When @len is not #NULL
+it will contain the length of the decoded data.
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> the base64 encoded data
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> location for output length or NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_transport_free">
+<description>
+Free the memory used by @transport.
+
+
+</description>
+<parameters>
+<parameter name="transport">
+<parameter_description> a #GstRTSPTransport
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_property_probe_probe_and_get_values_name">
+<description>
+Same as gst_property_probe_probe_and_get_values ().
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe object.
+</parameter_description>
+</parameter>
+<parameter name="name">
+<parameter_description> the name of the property to get values for.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the list of valid values for this property.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_add_item">
+<description>
+Add a new SDES item for @ssrc to @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> the SSRC of the new item to add
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the item could be added, %FALSE if the maximum amount of
+items has been exceeded for the SDES packet or the MTU has been reached.
+</return>
+</function>
+
+<function name="gst_ring_buffer_may_start">
+<description>
+Tell the ringbuffer that it is allowed to start playback when
+the ringbuffer is filled with samples.
+
+Since: 0.10.6
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer
+</parameter_description>
+</parameter>
+<parameter name="allowed">
+<parameter_description> the new value
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_orientation_get_vcenter">
+<description>
+Get the vertical centering offset from the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="center">
+<parameter_description> return location for the result
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports centering
+</return>
+</function>
+
+<function name="gst_rtcp_buffer_end">
+<description>
+Finish @buffer after being constructured. This function is usually called
+after gst_rtcp_buffer_new() and after adding the RTCP items to the new buffer.
+
+The function adjusts the size of @buffer with the total length of all the
+added packets.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a buffer with an RTCP packet
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_install_plugins_context_new">
+<description>
+Creates a new #GstInstallPluginsContext.
+
+
+</description>
+<parameters>
+</parameters>
+<return> a new #GstInstallPluginsContext. Free with
+gst_install_plugins_context_free() when no longer needed
+
+Since: 0.10.12
+</return>
+</function>
+
+<function name="gst_mixer_mute_toggled">
+<description>
+This function is called by the mixer implementation to produce
+a notification message on the bus indicating that the given track
+has changed mute state.
+
+This function only works for GstElements that are implementing the
+GstMixer interface, and the element needs to have been provided a bus.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the track
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> the GstMixerTrack that has change mute state.
+</parameter_description>
+</parameter>
+<parameter name="mute">
+<parameter_description> the new state of the mute flag on the track
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_buffer_new_copy_data">
+<description>
+Create a new buffer and set the data to a copy of @len
+bytes of @data and the size to @len. The data will be freed when the buffer
+is freed.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> data for the new buffer
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of data
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer with a copy of @data and of size @len.
+</return>
+</function>
+
+<function name="gst_fft_f32_window">
+<description>
+This calls the window function @window on the @timedata sample buffer.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF32 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Time domain samples
+</parameter_description>
+</parameter>
+<parameter name="window">
+<parameter_description> Window function to apply
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_riff_parse_info">
+<description>
+Parses stream metadata from input data.
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging/error).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input data to be used for parsing, stripped from header.
+</parameter_description>
+</parameter>
+<parameter name="taglist">
+<parameter_description> a pointer to a taglist (returned by this function)
+containing information about this stream. May be
+NULL if no supported tags were found.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_get_padding">
+<description>
+Check if the padding bit is set on the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @buffer has the padding bit set.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_session_name">
+<description>
+Get the session name in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtcp_buffer_validate">
+<description>
+Check if the data pointed to by @buffer is a valid RTCP packet using
+gst_rtcp_buffer_validate_data().
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer to validate
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @buffer is a valid RTCP packet.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_attribute">
+<description>
+Get the attribute at position @idx in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstSDPAttribute at position @idx.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_attribute_val_n">
+<description>
+Get the @nth attribute value for @key in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> a key
+</parameter_description>
+</parameter>
+<parameter name="nth">
+<parameter_description> an index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the @nth attribute value.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_bye_set_reason">
+<description>
+Set the reason string to @reason in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="reason">
+<parameter_description> a reason string
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the string could be set.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_csrc_count">
+<description>
+Get the CSRC count of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> the CSRC count of @buffer.
+</return>
+</function>
+
+<function name="gst_missing_encoder_installer_detail_new">
+<description>
+Returns: a newly-allocated detail string, or NULL on error. Free string
+
+</description>
+<parameters>
+<parameter name="encode_caps">
+<parameter_description> the (fixed) caps for which an encoder element is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated detail string, or NULL on error. Free string
+with g_free() when not needed any longer.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_missing_plugin_message_get_installer_detail">
+<description>
+Returns: a newly-allocated detail string, or NULL on error. Free string
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a missing-plugin #GstMessage of type #GST_MESSAGE_ELEMENT
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated detail string, or NULL on error. Free string
+with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_flush">
+<description>
+Start or stop the flushing action on @conn. When flushing, all current
+and future actions on @conn will return #GST_RTSP_EINTR until the connection
+is set to non-flushing mode again.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="flush">
+<parameter_description> start or stop the flush
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_audio_frame_length">
+<description>
+Calculate length of buffer in frames.
+
+
+</description>
+<parameters>
+<parameter name="pad">
+<parameter_description> the #GstPad to get the caps from
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> the #GstBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> 0 if there's an error, or the number of frames if everything's ok
+</return>
+</function>
+
+<function name="gst_netbuffer_new">
+<description>
+Create a new network buffer.
+
+
+</description>
+<parameters>
+</parameters>
+<return> a new #GstNetBuffer.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_time">
+<description>
+Get time information with index @idx from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the time index
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPTime.
+</return>
+</function>
+
+<function name="gst_vorbis_tag_add">
+<description>
+Convenience function using gst_tag_from_vorbis_tag(), parsing
+a vorbis comment string into the right type and adding it to the
+given taglist @list.
+
+Unknown vorbiscomment tags will be added to the tag list in form
+of a #GST_TAG_EXTENDED_COMMENT (since 0.10.10 at least).
+
+</description>
+<parameters>
+<parameter name="list">
+<parameter_description> a #GstTagList
+</parameter_description>
+</parameter>
+<parameter name="tag">
+<parameter_description> a vorbiscomment tag string (key in key=value), must be valid UTF-8
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> a vorbiscomment value string (value in key=value), must be valid UTF-8
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_format_from_rgb32_masks">
+<description>
+Converts red, green, blue bit masks into the corresponding
+#GstVideoFormat.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="red_mask">
+<parameter_description> red bit mask
+</parameter_description>
+</parameter>
+<parameter name="green_mask">
+<parameter_description> green bit mask
+</parameter_description>
+</parameter>
+<parameter name="blue_mask">
+<parameter_description> blue bit mask
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstVideoFormat corresponding to the bit masks
+</return>
+</function>
+
+<function name="gst_missing_encoder_message_new">
+<description>
+Creates a missing-plugin message for @element to notify the application
+that an encoder element for a particular set of (fixed) caps is missing.
+This function is mainly for use in plugins.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> the #GstElement posting the message
+</parameter_description>
+</parameter>
+<parameter name="encode_caps">
+<parameter_description> the (fixed) caps for which an encoder element is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstMessage, or NULL on error
+</return>
+</function>
+
+<function name="gst_video_format_from_fourcc">
+<description>
+Converts a FOURCC value into the corresponding #GstVideoFormat.
+If the FOURCC cannot be represented by #GstVideoFormat,
+#GST_VIDEO_FORMAT_UNKNOWN is returned.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="fourcc">
+<parameter_description> a FOURCC value representing raw YUV video
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstVideoFormat describing the FOURCC value
+</return>
+</function>
+
+<function name="gst_rtsp_message_init_data">
+<description>
+Initialize a new data #GstRTSPMessage for @channel.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> a channel
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_fft_f32_free">
+<description>
+This frees the memory allocated for @self.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF32 instance for this call
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_netaddress_equal">
+<description>
+Compare two #GstNetAddress structures
+
+
+</description>
+<parameters>
+<parameter name="naddr1">
+<parameter_description> The first #GstNetAddress
+</parameter_description>
+</parameter>
+<parameter name="naddr2">
+<parameter_description> The second #GstNetAddress
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if they are identical, FALSE otherwise
+
+Since: 0.10.18
+</return>
+</function>
+
+<function name="gst_rtsp_transport_parse">
+<description>
+Parse the RTSP transport string @str into @transport.
+
+
+</description>
+<parameters>
+<parameter name="str">
+<parameter_description> a transport string
+</parameter_description>
+</parameter>
+<parameter name="transport">
+<parameter_description> a #GstRTSPTransport
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_video_format_to_fourcc">
+<description>
+Converts a #GstVideoFormat value into the corresponding FOURCC. Only
+a few YUV formats have corresponding FOURCC values. If @format has
+no corresponding FOURCC value, 0 is returned.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat video format
+</parameter_description>
+</parameter>
+</parameters>
+<return> the FOURCC corresponding to @format
+</return>
+</function>
+
+<function name="gst_fft_s32_inverse_fft">
+<description>
+This performs the inverse FFT on @freqdata and puts the result in @timedata.
+
+ freqdata must have @len/2 + 1 samples, where @len is the parameter specified
+while allocating the #GstFFTS32 instance with gst_fft_s32_new().
+
+ timedata must be large enough to hold @len time domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS32 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Buffer of the samples in the frequency domain
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Target buffer for the samples in the time domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_get_information">
+<description>
+Get the information of @media
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the information of @media.
+</return>
+</function>
+
+<function name="gst_pb_utils_init">
+<description>
+Initialises the base utils support library. This function is not
+thread-safe. Applications should call it after calling gst_init(),
+plugins should call it from their plugin_init function.
+
+This function may be called multiple times. It will do nothing if the
+library has already been initialised.
+
+Since: 0.10.12
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_add_bandwidth">
+<description>
+Add the bandwidth information with @bwtype and @bandwidth to @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="bwtype">
+<parameter_description> the bandwidth modifier type
+</parameter_description>
+</parameter>
+<parameter name="bandwidth">
+<parameter_description> the bandwidth in kilobits per second
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_x_overlay_handle_events">
+<description>
+Tell an overlay that it should handle events from the window system. These
+events are forwared upstream as navigation events. In some window system,
+events are not propagated in the window hierarchy if a client is listening
+for them. This method allows you to disable events handling completely
+from the XOverlay.
+
+</description>
+<parameters>
+<parameter name="overlay">
+<parameter_description> a #GstXOverlay to expose.
+</parameter_description>
+</parameter>
+<parameter name="handle_events">
+<parameter_description> a #gboolean indicating if events should be handled or not.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_orientation_get_hflip">
+<description>
+Get the horizontal flipping state (%TRUE for flipped) from the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="flip">
+<parameter_description> return location for the result
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports flipping
+</return>
+</function>
+
+<function name="gst_video_format_is_yuv">
+<description>
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @format represents YUV video
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_get_item_count">
+<description>
+Get the number of items in the SDES packet @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of items in @packet.
+</return>
+</function>
+
+<function name="gst_rtsp_transport_get_manager">
+<description>
+Get the #GStreamer element that can handle the buffers transported over
+ trans
+
+It is possible that there are several managers available, use @option to
+selected one.
+
+ manager will contain an element name or #NULL when no manager is
+needed/available for @trans.
+
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> a #GstRTSPTransMode
+</parameter_description>
+</parameter>
+<parameter name="manager">
+<parameter_description> location to hold the result
+</parameter_description>
+</parameter>
+<parameter name="option">
+<parameter_description> option index.
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_bandwidth">
+<description>
+Get the bandwidth at position @idx in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> an index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstSDPBandwidth at position @idx.
+</return>
+</function>
+
+<function name="gst_tuner_set_norm">
+<description>
+Changes the video norm on this tuner to the given norm.
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) to set the norm on.
+</parameter_description>
+</parameter>
+<parameter name="norm">
+<parameter_description> the norm to use for the current channel.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_get_port">
+<description>
+Get the port number for @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the port number of @media.
+</return>
+</function>
+
+<function name="gst_sdp_message_uninit">
+<description>
+Free all resources allocated in @msg. @msg should not be used anymore after
+this function. This function should be used when @msg was allocated on the
+stack and initialized with gst_sdp_message_init().
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_ring_buffer_device_is_open">
+<description>
+Checks the status of the device associated with the ring buffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device was open, FALSE if it was closed.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_sdp_message_medias_len">
+<description>
+Get the number of media descriptions in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of media descriptions in @msg.
+</return>
+</function>
+
+<function name="gst_fft_s16_inverse_fft">
+<description>
+This performs the inverse FFT on @freqdata and puts the result in @timedata.
+
+ freqdata must have @len/2 + 1 samples, where @len is the parameter specified
+while allocating the #GstFFTS16 instance with gst_fft_s16_new().
+
+ timedata must be large enough to hold @len time domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS16 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Buffer of the samples in the frequency domain
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Target buffer for the samples in the time domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_tag_freeform_string_to_utf8">
+<description>
+Convenience function to read a string with unknown character encoding. If
+the string is already in UTF-8 encoding, it will be returned right away.
+Otherwise, the environment will be searched for a number of environment
+variables (whose names are specified in the NULL-terminated string array
+ env_vars) containing a list of character encodings to try/use. If none
+are specified, the current locale will be tried. If that also doesn't work,
+ISO-8859-1 is assumed (which will almost always succeed).
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> string data
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> length of string data, or -1 if the string is NUL-terminated
+</parameter_description>
+</parameter>
+<parameter name="env_vars">
+<parameter_description> a NULL-terminated string array of environment variable names,
+or NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated string in UTF-8 encoding, or NULL
+
+Since: 0.10.13
+</return>
+</function>
+
+<function name="gst_tag_from_id3_user_tag">
+<description>
+Looks up the GStreamer tag for an ID3v2 user tag (e.g. description in
+TXXX frame or owner in UFID frame).
+
+
+</description>
+<parameters>
+<parameter name="type">
+<parameter_description> the type of ID3v2 user tag (e.g. "TXXX" or "UDIF")
+</parameter_description>
+</parameter>
+<parameter name="id3_user_tag">
+<parameter_description> ID3v2 user tag to convert to GStreamer tag
+</parameter_description>
+</parameter>
+</parameters>
+<return> The corresponding GStreamer tag or NULL if none exists.
+</return>
+</function>
+
+<function name="gst_sdp_message_attributes_len">
+<description>
+Get the number of attributes in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of attributes in @msg.
+</return>
+</function>
+
+<function name="gst_pb_utils_add_codec_description_to_tag_list">
+<description>
+Adds a codec tag describing the format specified by @caps to @taglist.
+
+
+</description>
+<parameters>
+<parameter name="taglist">
+<parameter_description> a #GstTagList
+</parameter_description>
+</parameter>
+<parameter name="codec_tag">
+<parameter_description> a GStreamer codec tag such as #GST_TAG_AUDIO_CODEC,
+#GST_TAG_VIDEO_CODEC or #GST_TAG_CODEC
+</parameter_description>
+</parameter>
+<parameter name="caps">
+<parameter_description> the (fixed) #GstCaps for which a codec tag should be added.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if a codec tag was added, FALSE otherwise.
+</return>
+</function>
+
+<function name="gst_pb_utils_get_source_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="protocol">
+<parameter_description> the protocol the source element needs to handle, e.g. "http"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_fft_s32_free">
+<description>
+This frees the memory allocated for @self.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS32 instance for this call
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_add_connection">
+<description>
+Add the given connection parameters to @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="nettype">
+<parameter_description> the type of network. "IN" is defined to have the meaning
+"Internet".
+</parameter_description>
+</parameter>
+<parameter name="addrtype">
+<parameter_description> the type of address.
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> the address
+</parameter_description>
+</parameter>
+<parameter name="ttl">
+<parameter_description> the time to live of the address
+</parameter_description>
+</parameter>
+<parameter name="addr_number">
+<parameter_description> the number of layers
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_riff_create_video_caps_with_data">
+<description>
+
+</description>
+<parameters>
+<parameter name="codec_fcc">
+<parameter_description> fourCC codec for this codec.
+</parameter_description>
+</parameter>
+<parameter name="strh">
+<parameter_description> pointer to the strh stream header structure.
+</parameter_description>
+</parameter>
+<parameter name="strf">
+<parameter_description> pointer to the strf stream header structure, including any
+data that is within the range of strf.size, but excluding any
+additional data withint this chunk but outside strf.size.
+</parameter_description>
+</parameter>
+<parameter name="strf_data">
+<parameter_description> a #GstBuffer containing the additional data in the strf
+chunk outside reach of strf.size. Ususally a palette.
+</parameter_description>
+</parameter>
+<parameter name="strd_data">
+<parameter_description> a #GstBuffer containing the data in the strd stream header
+chunk. Usually codec initialization data.
+</parameter_description>
+</parameter>
+<parameter name="codec_name">
+<parameter_description> if given, will be filled with a human-readable codec name.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_new">
+<description>
+Allocate a new GstSDPMedia and store the result in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> pointer to new #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtsp_message_dump">
+<description>
+Dump the contents of @msg to stdout.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_sdp_message_parse_buffer">
+<description>
+Parse the contents of @size bytes pointed to by @data and store the result in
+ msg
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> the start of the buffer
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of the buffer
+</parameter_description>
+</parameter>
+<parameter name="msg">
+<parameter_description> the result #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK on success.
+</return>
+</function>
+
+<function name="gst_sdp_message_emails_len">
+<description>
+Get the number of emails in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of emails in @msg.
+</return>
+</function>
+
+<function name="gst_rtsp_message_new_request">
+<description>
+Create a new #GstRTSPMessage with @method and @uri and store the result
+request message in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a location for the new #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="method">
+<parameter_description> the request method to use
+</parameter_description>
+</parameter>
+<parameter name="uri">
+<parameter_description> the uri of the request
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult. Free with gst_rtsp_message_free().
+</return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_set_frame_options">
+<description>
+Sets the options for frame based audio codecs.
+
+
+</description>
+<parameters>
+<parameter name="basertpaudiopayload">
+<parameter_description> a pointer to the element.
+</parameter_description>
+</parameter>
+<parameter name="frame_duration">
+<parameter_description> The duraction of an audio frame in milliseconds.
+</parameter_description>
+</parameter>
+<parameter name="frame_size">
+<parameter_description> The size of an audio frame in bytes.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_message_new_response">
+<description>
+Create a new response #GstRTSPMessage with @code and @reason and store the
+result message in @msg.
+
+When @reason is #NULL, the default reason for @code will be used.
+
+When @request is not #NULL, the relevant headers will be copied to the new
+response message.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a location for the new #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="code">
+<parameter_description> the status code
+</parameter_description>
+</parameter>
+<parameter name="reason">
+<parameter_description> the status reason or #NULL
+</parameter_description>
+</parameter>
+<parameter name="request">
+<parameter_description> the request that triggered the response or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult. Free with gst_rtsp_message_free().
+</return>
+</function>
+
+<function name="gst_rtsp_connection_close">
+<description>
+Close the connected @conn.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_video_format_get_size">
+<description>
+Calculates the total number of bytes in the raw video format. This
+number should be used when allocating a buffer for raw video.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> the width of video
+</parameter_description>
+</parameter>
+<parameter name="height">
+<parameter_description> the height of video
+</parameter_description>
+</parameter>
+</parameters>
+<return> size (in bytes) of raw video format
+</return>
+</function>
+
+<function name="gst_rtsp_message_steal_body">
+<description>
+Take the body of @msg and store it in @data and @size. After this method,
+the body and size of @msg will be set to #NULL and 0 respectively.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> location for the data
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> location for the size of @data
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_x_overlay_set_xwindow_id">
+<description>
+This will call the video overlay's set_xwindow_id method. You should
+use this method to tell to a XOverlay to display video output to a
+specific XWindow. Passing 0 as the xwindow_id will tell the overlay to
+stop using that window and create an internal one.
+
+</description>
+<parameters>
+<parameter name="overlay">
+<parameter_description> a #GstXOverlay to set the XWindow on.
+</parameter_description>
+</parameter>
+<parameter name="xwindow_id">
+<parameter_description> a #XID referencing the XWindow.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_base_audio_sink_set_provide_clock">
+<description>
+Controls whether @sink will provide a clock or not. If @provide is %TRUE,
+gst_element_provide_clock() will return a clock that reflects the datarate
+of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
+
+Since: 0.10.16
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> a #GstBaseAudioSink
+</parameter_description>
+</parameter>
+<parameter name="provide">
+<parameter_description> new state
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_ring_buffer_close_device">
+<description>
+Close the audio device associated with the ring buffer. The ring buffer
+should already have been released via gst_ring_buffer_release().
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be closed, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_ring_buffer_clear">
+<description>
+Clear the given segment of the buffer with silence samples.
+This function is used by subclasses.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to clear
+</parameter_description>
+</parameter>
+<parameter name="segment">
+<parameter_description> the segment to clear
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_packet_get_length">
+<description>
+Get the length field of @packet. This is the length of the packet in
+32-bit words minus one.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The length field of @packet.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_csrc">
+<description>
+Modify the CSRC at index @idx in @buffer to @csrc.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the CSRC index to set
+</parameter_description>
+</parameter>
+<parameter name="csrc">
+<parameter_description> the CSRC in host order to set at @idx
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_set_connection">
+<description>
+Configure the SDP connection in @msg with the given parameters.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="nettype">
+<parameter_description> the type of network. "IN" is defined to have the meaning
+"Internet".
+</parameter_description>
+</parameter>
+<parameter name="addrtype">
+<parameter_description> the type of address.
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> the address
+</parameter_description>
+</parameter>
+<parameter name="ttl">
+<parameter_description> the time to live of the address
+</parameter_description>
+</parameter>
+<parameter name="addr_number">
+<parameter_description> the number of layers
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_tag_id3_genre_get">
+<description>
+Gets the ID3v1 genre name for a given ID.
+
+
+</description>
+<parameters>
+<parameter name="id">
+<parameter_description> ID of genre to query
+</parameter_description>
+</parameter>
+</parameters>
+<return> the genre or NULL if no genre is associated with that ID.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_extension">
+<description>
+Set the extension bit on the RTP packet in @buffer to @extension.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="extension">
+<parameter_description> the new extension
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_get_extension">
+<description>
+Check if the extension bit is set on the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @buffer has the extension bit set.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_rr_set_ssrc">
+<description>
+Set the ssrc field of the RR @packet.
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid RR #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> the SSRC to set
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_get_format">
+<description>
+Get the format information at position @idx in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> an index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the format at position @idx.
+</return>
+</function>
+
+<function name="SECTION">
+<description>
+&lt;refsect2&gt;
+&lt;para&gt;
+This library contains some helper functions and includes the
+videosink and videofilter base classes.
+&lt;/para&gt;
+&lt;/refsect2&gt;
+
+</description>
+<parameters>
+<parameter name="short_description">
+<parameter_description> Support library for video operations
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_set_version">
+<description>
+Set the version in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="version">
+<parameter_description> the version
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_attribute_val">
+<description>
+Get the first attribute value for @key in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> a key
+</parameter_description>
+</parameter>
+</parameters>
+<return> the first attribute value for @key.
+</return>
+</function>
+
+<function name="gst_mixer_volume_changed">
+<description>
+This function is called by the mixer implementation to produce
+a notification message on the bus indicating that the volume(s) for the
+given track have changed.
+
+This function only works for GstElements that are implementing the
+GstMixer interface, and the element needs to have been provided a bus.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the track
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> the GstMixerTrack that has changed.
+</parameter_description>
+</parameter>
+<parameter name="volumes">
+<parameter_description> Array of volume values, one per channel on the mixer track.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_audio_set_channel_positions">
+<description>
+Adds a "channel-positions" field to the given #GstStructure,
+which will represent the channel positions as given in the
+provided #GstAudioChannelPosition array.
+
+</description>
+<parameters>
+<parameter name="str">
+<parameter_description> A #GstStructure to set channel positions on.
+</parameter_description>
+</parameter>
+<parameter name="pos">
+<parameter_description> an array of channel positions. The number of members
+in this array should be equal to the (fixed!) number
+of the "channels" field in the given #GstStructure.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_payload_info_for_pt">
+<description>
+Get the #GstRTPPayloadInfo for @payload_type. This function is
+mostly used to get the default clock-rate and bandwidth for static payload
+types specified with @payload_type.
+
+
+</description>
+<parameters>
+<parameter name="payload_type">
+<parameter_description> the payload_type to find
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTPPayloadInfo or NULL when no info could be found.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_phone">
+<description>
+Add @phone to the list of phones in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="phone">
+<parameter_description> a phone
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="GstMixerTrack">
+<description>
+The untranslated label of the mixer track, if available. Mixer track
+implementations must set this at construct time. Applications may find
+this useful to determine icons for various kind of tracks. However,
+applications mustn't make any assumptions about the naming of tracks,
+the untranslated labels are purely informational and may change.
+
+Since: 0.10.13
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_format_get_component_width">
+<description>
+Calculates the width of the component. See
+ gst_video_format_get_row_stride for a description
+of the component index.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="component">
+<parameter_description> the component index
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> the width of video
+</parameter_description>
+</parameter>
+</parameters>
+<return> width of component @component
+</return>
+</function>
+
+<function name="gst_fft_s32_new">
+<description>
+This returns a new #GstFFTS32 instance with the given parameters. It makes
+sense to keep one instance for several calls for speed reasons.
+
+ len must be even and to get the best performance a product of
+2, 3 and 5. To get the next number with this characteristics use
+gst_fft_next_fast_length().
+
+
+</description>
+<parameters>
+<parameter name="len">
+<parameter_description> Length of the FFT in the time domain
+</parameter_description>
+</parameter>
+<parameter name="inverse">
+<parameter_description> %TRUE if the #GstFFTS32 instance should be used for the inverse FFT
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstFFTS32 instance.
+</return>
+</function>
+
+<function name="gst_tuner_get_norms_list">
+<description>
+Retrieve a list of available norms on the currently tuned channel
+from the given tuner object.
+
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (*a #GstElement) to get the list of norms from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A list of norms available on the current channel for this
+tuner object.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_add_entry">
+<description>
+Add a new SDES entry to the current item in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> the #GstRTCPSDESType of the SDES entry
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the data length
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the item could be added, %FALSE if the MTU has been
+reached.
+</return>
+</function>
+
+<function name="gst_x_overlay_got_xwindow_id">
+<description>
+This will post a "have-xwindow-id" element message on the bus.
+
+This function should only be used by video overlay plugin developers.
+
+</description>
+<parameters>
+<parameter name="overlay">
+<parameter_description> a #GstXOverlay which got a XWindow.
+</parameter_description>
+</parameter>
+<parameter name="xwindow_id">
+<parameter_description> a #XID referencing the XWindow.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_get_packet_len">
+<description>
+Return the total length of the packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The total length of the packet in @buffer.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_get_count">
+<description>
+Get the count field in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The count field in @packet or -1 if @packet does not point to a
+valid packet.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_packet_len">
+<description>
+Set the total @buffer size to @len. The data in the buffer will be made
+larger if needed. Any padding will be removed from the packet.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the new packet length
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_property_probe_get_values">
+<description>
+Gets the possible (probed) values for the given property,
+requires the property to have been probed before.
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe object.
+</parameter_description>
+</parameter>
+<parameter name="pspec">
+<parameter_description> the #GParamSpec property identifier.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A list of valid values for the given property.
+</return>
+</function>
+
+<function name="gst_sdp_message_bandwidths_len">
+<description>
+Get the number of bandwidth information in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of bandwidth information in @msg.
+</return>
+</function>
+
+<function name="gst_rtsp_message_remove_header">
+<description>
+Remove the @indx header with key @field from @msg. If @indx equals -1, all
+headers will be removed.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="field">
+<parameter_description> a #GstRTSPHeaderField
+</parameter_description>
+</parameter>
+<parameter name="indx">
+<parameter_description> the index of the header
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_ssrc">
+<description>
+Get the SSRC of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> the SSRC of @buffer in host order.
+</return>
+</function>
+
+<function name="gst_base_audio_src_set_provide_clock">
+<description>
+Controls whether @src will provide a clock or not. If @provide is %TRUE,
+gst_element_provide_clock() will return a clock that reflects the datarate
+of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
+
+Since: 0.10.16
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> a #GstBaseAudioSrc
+</parameter_description>
+</parameter>
+<parameter name="provide">
+<parameter_description> new state
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_mixer_get_volume">
+<description>
+Get the current volume(s) on the given track.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the track
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> the GstMixerTrack to get the volume from.
+</parameter_description>
+</parameter>
+<parameter name="volumes">
+<parameter_description> a pre-allocated array of integers (of size
+track-&gt;num_channels) to store the current volume
+of each channel in the given track in.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_format_get_row_stride">
+<description>
+Calculates the row stride (number of bytes from one row of pixels to
+the next) for the video component with an index of @component. For
+YUV video, Y, U, and V have component indices of 0, 1, and 2,
+respectively. For RGB video, R, G, and B have component indicies of
+0, 1, and 2, respectively. Alpha channels, if present, have a component
+index of 3. The @width parameter always represents the width of the
+video, not the component.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="component">
+<parameter_description> the component index
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> the width of video
+</parameter_description>
+</parameter>
+</parameters>
+<return> row stride of component @component
+</return>
+</function>
+
+<function name="gst_riff_parse_strf_iavs">
+<description>
+Parses a interleaved (also known as "complex") streamÂs strf
+structure plus optionally some extradata from input data. This
+function takes ownership of @buf.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging/error).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input data to be used for parsing, stripped from header.
+</parameter_description>
+</parameter>
+<parameter name="strf">
+<parameter_description> a pointer (returned by this function) to a filled-in
+strf/iavs structure. Caller should free it.
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> a pointer (returned by this function) to a buffer
+containing extradata for this particular stream (e.g.
+codec initialization data).
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if parsing succeeded, otherwise FALSE.
+</return>
+</function>
+
+<function name="gst_ring_buffer_read">
+<description>
+Read @len samples from the ringbuffer into the memory pointed
+to by @data.
+The first sample should be read from position @sample in
+the ringbuffer.
+
+ len should not be a multiple of the segment size of the ringbuffer
+although it is recommended.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to read from
+</parameter_description>
+</parameter>
+<parameter name="sample">
+<parameter_description> the sample position of the data
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> where the data should be read
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the number of samples in data to read
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of samples read from the ringbuffer or -1 on
+error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_new_allocate">
+<description>
+Allocate a new #Gstbuffer with enough data to hold an RTP packet with @csrc_count CSRCs,
+a payload length of @payload_len and padding of @pad_len.
+All other RTP header fields will be set to 0/FALSE.
+
+
+</description>
+<parameters>
+<parameter name="payload_len">
+<parameter_description> the length of the payload
+</parameter_description>
+</parameter>
+<parameter name="pad_len">
+<parameter_description> the amount of padding
+</parameter_description>
+</parameter>
+<parameter name="csrc_count">
+<parameter_description> the number of CSRC entries
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer that can hold an RTP packet with given
+parameters.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_reset_timeout">
+<description>
+Reset the timeout of @conn.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_rtcp_buffer_new">
+<description>
+Create a new buffer for constructing RTCP packets. The packet will have a
+maximum size of @mtu.
+
+
+</description>
+<parameters>
+<parameter name="mtu">
+<parameter_description> the maximum mtu size.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer.
+</return>
+</function>
+
+<function name="gst_rtsp_version_as_text">
+<description>
+Convert @version to a string.
+
+
+</description>
+<parameters>
+<parameter name="version">
+<parameter_description> a #GstRTSPVersion
+</parameter_description>
+</parameter>
+</parameters>
+<return> a string representation of @version.
+</return>
+</function>
+
+<function name="gst_sdp_media_connections_len">
+<description>
+Get the number of connection fields in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of connections in @media.
+</return>
+</function>
+
+<function name="gst_missing_uri_sink_message_new">
+<description>
+Creates a missing-plugin message for @element to notify the application
+that a sink element for a particular URI protocol is missing. This
+function is mainly for use in plugins.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> the #GstElement posting the message
+</parameter_description>
+</parameter>
+<parameter name="protocol">
+<parameter_description> the URI protocol the missing sink needs to implement,
+e.g. "http" or "smb"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstMessage, or NULL on error
+</return>
+</function>
+
+<function name="gst_ring_buffer_set_flushing">
+<description>
+Set the ringbuffer to flushing mode or normal mode.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to flush
+</parameter_description>
+</parameter>
+<parameter name="flushing">
+<parameter_description> the new mode
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_orientation_set_vflip">
+<description>
+Set the vertical flipping state (%TRUE for flipped) for the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="flip">
+<parameter_description> use flipping
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports flipping
+</return>
+</function>
+
+<function name="gst_rtp_buffer_new_copy_data">
+<description>
+Create a new buffer and set the data to a copy of @len
+bytes of @data and the size to @len. The data will be freed when the buffer
+is freed.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> data for the new buffer
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of data
+</parameter_description>
+</parameter>
+</parameters>
+<return> A newly allocated buffer with a copy of @data and of size @len.
+</return>
+</function>
+
+<function name="gst_video_parse_caps_framerate">
+<description>
+Extracts the frame rate from @caps and places the values in the locations
+pointed to by @fps_n and @fps_d. Returns TRUE if the values could be
+parsed correctly, FALSE if not.
+
+This function can be used with #GstCaps that have any media type; it
+is not limited to formats handled by #GstVideoFormat.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description>
+</parameter_description>
+</parameter>
+<parameter name="fps_n">
+<parameter_description> pointer to numerator of frame rate (output)
+</parameter_description>
+</parameter>
+<parameter name="fps_d">
+<parameter_description> pointer to denominator of frame rate (output)
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @caps was parsed correctly.
+</return>
+</function>
+
+<function name="gst_install_plugins_supported">
+<description>
+Checks whether plugin installation is likely to be supported by the
+current environment. This currently only checks whether the helper script
+that is to be provided by the distribution or operating system vendor
+exists.
+
+
+</description>
+<parameters>
+</parameters>
+<return> TRUE if plugin installation is likely to be supported.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_extension_data">
+<description>
+Set the extension bit of the rtp buffer and fill in the @bits and @length of the
+extension header. It will refuse to set the extension data if the buffer is not
+large enough.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="bits">
+<parameter_description> the bits specific for the extension
+</parameter_description>
+</parameter>
+<parameter name="length">
+<parameter_description> the length that counts the number of 32-bit words in
+the extension, excluding the extension header ( therefore zero is a valid length)
+</parameter_description>
+</parameter>
+</parameters>
+<return> True if done.
+
+Since : 0.10.18
+</return>
+</function>
+
+<function name="gst_rtsp_url_free">
+<description>
+Free the memory used by @url.
+
+</description>
+<parameters>
+<parameter name="url">
+<parameter_description> a #GstRTSPUrl
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="GstBaseRTPDepayload">
+<description>
+Control the amount of packets to buffer.
+
+Deprecated: Use a jitterbuffer or RTP session manager to delay packet
+playback. This property has no effect anymore since 0.10.15.
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_default_clock_rate">
+<description>
+Get the default clock-rate for the static payload type @payload_type.
+
+
+</description>
+<parameters>
+<parameter name="payload_type">
+<parameter_description> the static payload type
+</parameter_description>
+</parameter>
+</parameters>
+<return> the default clock rate or -1 if the payload type is not static or
+the clock-rate is undefined.
+
+Since: 0.10.13
+</return>
+</function>
+
+<function name="gst_sdp_media_attributes_len">
+<description>
+Get the number of attribute fields in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of attributes in @media.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_get_type">
+<description>
+Get the packet type of the packet pointed to by @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The packet type.
+</return>
+</function>
+
+<function name="gst_video_calculate_display_ratio">
+<description>
+Given the Pixel Aspect Ratio and size of an input video frame, and the
+pixel aspect ratio of the intended display device, calculates the actual
+display ratio the video will be rendered with.
+
+
+</description>
+<parameters>
+<parameter name="dar_n">
+<parameter_description> Numerator of the calculated display_ratio
+</parameter_description>
+</parameter>
+<parameter name="dar_d">
+<parameter_description> Denominator of the calculated display_ratio
+</parameter_description>
+</parameter>
+<parameter name="video_width">
+<parameter_description> Width of the video frame in pixels
+</parameter_description>
+</parameter>
+<parameter name="video_height">
+<parameter_description> Height of the video frame in pixels
+</parameter_description>
+</parameter>
+<parameter name="video_par_n">
+<parameter_description> Numerator of the pixel aspect ratio of the input video.
+</parameter_description>
+</parameter>
+<parameter name="video_par_d">
+<parameter_description> Denominator of the pixel aspect ratio of the input video.
+</parameter_description>
+</parameter>
+<parameter name="display_par_n">
+<parameter_description> Numerator of the pixel aspect ratio of the display device
+</parameter_description>
+</parameter>
+<parameter name="display_par_d">
+<parameter_description> Denominator of the pixel aspect ratio of the display device
+</parameter_description>
+</parameter>
+</parameters>
+<return> A boolean indicating success and a calculated Display Ratio in the
+dar_n and dar_d parameters.
+The return value is FALSE in the case of integer overflow or other error.
+
+Since: 0.10.7
+</return>
+</function>
+
+<function name="gst_rtsp_transport_new">
+<description>
+Allocate a new initialized #GstRTSPTransport. Use gst_rtsp_transport_free()
+after usage.
+
+
+</description>
+<parameters>
+<parameter name="transport">
+<parameter_description> location to hold the new #GstRTSPTransport
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="GstBaseRTPAudioPayload">
+<description>
+Minimum duration of the packet data in ns (can't go above MTU)
+
+Since: 0.10.13
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_get_payload_len">
+<description>
+Get the length of the payload of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The length of the payload in @buffer.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_num_ports">
+<description>
+Get the number of ports for @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of ports for @media.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_origin">
+<description>
+Get the origin of @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPOrigin. The result remains valid as long as @msg is valid.
+</return>
+</function>
+
+<function name="gst_install_plugins_return_get_name">
+<description>
+Convenience function to return the descriptive string associated
+with a status code. This function returns English strings and
+should not be used for user messages. It is here only to assist
+in debugging.
+
+
+</description>
+<parameters>
+<parameter name="ret">
+<parameter_description> the return status code
+</parameter_description>
+</parameter>
+</parameters>
+<return> a descriptive string for the status code in @ret
+
+Since: 0.10.12
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_get_entry">
+<description>
+Get the data of the current SDES item entry. @type (when not NULL) will
+contain the type of the entry. @data (when not NULL) will point to @len
+bytes.
+
+When @type refers to a text item, @data will point to a UTF8 string. Note
+that this UTF8 string is NOT null-terminated. Use
+gst_rtcp_packet_sdes_copy_entry() to get a null-termined copy of the entry.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> result of the entry type
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> result length of the entry data
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> result entry data
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if there was valid data.
+</return>
+</function>
+
+<function name="gst_audio_buffer_clip">
+<description>
+Clip the the buffer to the given %GstSegment.
+
+After calling this function the caller does not own a reference to
+ buffer anymore.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> The buffer to clip.
+</parameter_description>
+</parameter>
+<parameter name="segment">
+<parameter_description> Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.
+</parameter_description>
+</parameter>
+<parameter name="rate">
+<parameter_description> sample rate.
+</parameter_description>
+</parameter>
+<parameter name="frame_size">
+<parameter_description> size of one audio frame in bytes.
+</parameter_description>
+</parameter>
+</parameters>
+<return> %NULL if the buffer is completely outside the configured segment,
+otherwise the clipped buffer is returned.
+
+If the buffer has no timestamp, it is assumed to be inside the segment and
+is not clipped
+
+Since: 0.10.14
+</return>
+</function>
+
+<function name="gst_rtp_buffer_ext_timestamp">
+<description>
+Update the @exttimestamp field with @timestamp. For the first call of the
+method, @exttimestamp should point to a location with a value of -1.
+
+This function makes sure that the returned value is a constantly increasing
+value even in the case where there is a timestamp wraparound.
+
+
+</description>
+<parameters>
+<parameter name="exttimestamp">
+<parameter_description> a previous extended timestamp
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> a new timestamp
+</parameter_description>
+</parameter>
+</parameters>
+<return> The extended timestamp of @timestamp.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_rtcp_packet_bye_get_ssrc_count">
+<description>
+Get the number of SSRC fields in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of SSRC fields in @packet.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_get_rb_count">
+<description>
+Get the number of report blocks in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SR or RR #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of report blocks in @packet.
+</return>
+</function>
+
+<function name="gst_audio_set_structure_channel_positions_list">
+<description>
+Sets a (possibly non-fixed) list of possible audio channel
+positions (given in pos) on the given structure. The
+structure, after this function has been called, will contain
+a "channel-positions" field with an array of the size of
+the "channels" field value in the given structure (note
+that this means that the channels field in the provided
+structure should be fixed!). Each value in the array will
+contain each of the values given in the pos array.
+
+</description>
+<parameters>
+<parameter name="str">
+<parameter_description> #GstStructure to set the list of channel positions
+on.
+</parameter_description>
+</parameter>
+<parameter name="pos">
+<parameter_description> the array containing one or more possible audio
+channel positions that we should add in each value
+of the array in the given structure.
+</parameter_description>
+</parameter>
+<parameter name="num_positions">
+<parameter_description> the number of values in pos.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_mixer_message_parse_record_toggled">
+<description>
+Extracts the contents of a record-toggled bus message. Reads
+the GstMixerTrack that has changed, and the new value of the
+recording flag.
+
+The GstMixerTrack remains valid until the message is freed.
+
+Since: 0.10.14
+
+</description>
+<parameters>
+<parameter name="message">
+<parameter_description> A record-toggled change notification message.
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> Pointer to hold a GstMixerTrack object, or NULL.
+</parameter_description>
+</parameter>
+<parameter name="record">
+<parameter_description> A pointer to a gboolean variable, or NULL.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_mixer_set_option">
+<description>
+Sets a name/value option in the mixer to the requested value.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> The #GstMixer (a #GstElement) that owns the optionlist.
+</parameter_description>
+</parameter>
+<parameter name="opts">
+<parameter_description> The #GstMixerOptions that we operate on.
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> The requested new option value.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_get_adapter">
+<description>
+Gets the internal adapter used by the depayloader.
+
+
+</description>
+<parameters>
+<parameter name="basertpaudiopayload">
+<parameter_description> a #GstBaseRTPAudioPayload
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstAdapter.
+
+Since: 0.10.13
+</return>
+</function>
+
+<function name="gst_base_audio_sink_get_provide_clock">
+<description>
+Queries whether @sink will provide a clock or not. See also
+gst_base_audio_sink_set_provide_clock.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> a #GstBaseAudioSink
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if @sink will provide a clock.
+
+Since: 0.10.16
+</return>
+</function>
+
+<function name="gst_base_audio_src_get_provide_clock">
+<description>
+Queries whether @src will provide a clock or not. See also
+gst_base_audio_src_set_provide_clock.
+
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> a #GstBaseAudioSrc
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if @src will provide a clock.
+
+Since: 0.10.16
+</return>
+</function>
+
+<function name="gst_video_format_new_caps">
+<description>
+Creates a new #GstCaps object based on the parameters provided.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> the #GstVideoFormat describing the raw video format
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> width of video
+</parameter_description>
+</parameter>
+<parameter name="height">
+<parameter_description> height of video
+</parameter_description>
+</parameter>
+<parameter name="framerate_n">
+<parameter_description> numerator of frame rate
+</parameter_description>
+</parameter>
+<parameter name="framerate_d">
+<parameter_description> denominator of frame rate
+</parameter_description>
+</parameter>
+<parameter name="par_n">
+<parameter_description> numerator of pixel aspect ratio
+</parameter_description>
+</parameter>
+<parameter name="par_d">
+<parameter_description> denominator of pixel aspect ratio
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstCaps object, or NULL if there was an error
+</return>
+</function>
+
+<function name="gst_audio_fixate_channel_positions">
+<description>
+Custom fixate function. Elements that implement some sort of
+channel conversion algorithm should use this function for
+fixating on GstAudioChannelPosition properties. It will take
+care of equal channel positioning (left/right). Caller g_free()s
+the return value. The input properties may be (and are supposed
+to be) unfixed.
+Note that this function is mostly a hack because we currently
+have no way to add default fixation functions for new GTypes.
+
+
+</description>
+<parameters>
+<parameter name="str">
+<parameter_description> a #GstStructure containing a (possibly unfixed)
+"channel-positions" field.
+</parameter_description>
+</parameter>
+</parameters>
+<return> fixed values that the caller could use as a fixed
+set of #GstAudioChannelPosition values.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_zone">
+<description>
+Add time zone information to @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="adj_time">
+<parameter_description> the NTP time that a time zone adjustment happens
+</parameter_description>
+</parameter>
+<parameter name="typed_time">
+<parameter_description> the offset from the time when the session was first scheduled
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_set_sample_options">
+<description>
+Sets the options for sample based audio codecs.
+
+
+</description>
+<parameters>
+<parameter name="basertpaudiopayload">
+<parameter_description> a pointer to the element.
+</parameter_description>
+</parameter>
+<parameter name="sample_size">
+<parameter_description> Size per sample in bytes.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_get_email">
+<description>
+Get the email with number @idx from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> an email index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the email at position @idx.
+</return>
+</function>
+
+<function name="gst_rtp_payload_info_for_name">
+<description>
+Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is
+mostly used to get the default clock-rate and bandwidth for dynamic payload
+types specified with @media and @encoding name.
+
+The search for @encoding_name will be performed in a case insensitve way.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> the media to find
+</parameter_description>
+</parameter>
+<parameter name="encoding_name">
+<parameter_description> the encoding name to find
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTPPayloadInfo or NULL when no info could be found.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_payload_subbuffer">
+<description>
+Create a subbuffer of the payload of the RTP packet in @buffer. @offset bytes
+are skipped in the payload and the subbuffer will be of size @len.
+If @len is -1 the total payload starting from @offset if subbuffered.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="offset">
+<parameter_description> the offset in the payload
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length in the payload
+</parameter_description>
+</parameter>
+</parameters>
+<return> A new buffer with the specified data of the payload.
+
+Since: 0.10.10
+</return>
+</function>
+
+<function name="gst_riff_parse_strf_auds">
+<description>
+Parses an audio streamÂs strf structure plus optionally some
+extradata from input data. This function takes ownership of @buf.
+use.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging/error).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input data to be used for parsing, stripped from header.
+</parameter_description>
+</parameter>
+<parameter name="strf">
+<parameter_description> a pointer (returned by this function) to a filled-in
+strf/auds structure. Caller should free it.
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> a pointer (returned by this function) to a buffer
+containing extradata for this particular stream (e.g.
+codec initialization data).
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if parsing succeeded, otherwise FALSE. The stream
+should be skipped on error, but it is not fatal.
+</return>
+</function>
+
+<function name="gst_netaddress_get_ip4_address">
+<description>
+Get the IPv4 address stored in @naddr into @address.
+
+
+</description>
+<parameters>
+<parameter name="naddr">
+<parameter_description> a network address
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> a location to store the address.
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> a location to store the port.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the address could be retrieved.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_extension_data">
+<description>
+Get the extension data. @bits will contain the extension 16 bits of custom
+data. @data will point to the data in the extension and @wordlen will contain
+the length of @data in 32 bits words.
+
+If @buffer did not contain an extension, this function will return %FALSE
+with @bits, @data and @wordlen unchanged.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="bits">
+<parameter_description> location for result bits
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> location for data
+</parameter_description>
+</parameter>
+<parameter name="wordlen">
+<parameter_description> location for length of @data in 32 bits words
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @buffer had the extension bit set.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_rtsp_connection_connect">
+<description>
+Attempt to connect to the url of @conn made with
+gst_rtsp_connection_create(). If @timeout is #NULL this function can block
+forever. If @timeout contains a valid timeout, this function will return
+#GST_RTSP_ETIMEOUT after the timeout expired.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a #GTimeVal timeout
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK when a connection could be made.
+</return>
+</function>
+
+<function name="gst_pb_utils_get_encoder_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description> the (fixed) #GstCaps for which an encoder description is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_payload_buffer">
+<description>
+Create a buffer of the payload of the RTP packet in @buffer. This function
+will internally create a subbuffer of @buffer so that a memcpy can be
+avoided.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> A new buffer with the data of the payload.
+</return>
+</function>
+
+<function name="gst_rtcp_buffer_get_packet_count">
+<description>
+Get the number of RTCP packets in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a valid RTCP buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of RTCP packets in @buffer.
+</return>
+</function>
+
+<function name="gst_mixer_list_tracks">
+<description>
+Returns: A #GList consisting of zero or more #GstMixerTracks.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) to get the tracks from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A #GList consisting of zero or more #GstMixerTracks.
+</return>
+</function>
+
+<function name="gst_property_probe_probe_and_get_values">
+<description>
+Check whether the given property requires a new probe. If so,
+fo the probe. After that, retrieve a value list. Meant as a
+utility function that wraps the above functions.
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe object.
+</parameter_description>
+</parameter>
+<parameter name="pspec">
+<parameter_description> The #GParamSpec property identifier.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the list of valid values for this property.
+</return>
+</function>
+
+<function name="gst_tag_to_id3_tag">
+<description>
+Looks up the ID3v2 tag for a GStreamer tag.
+
+
+</description>
+<parameters>
+<parameter name="gst_tag">
+<parameter_description> GStreamer tag to convert to vorbiscomment tag
+</parameter_description>
+</parameter>
+</parameters>
+<return> The corresponding ID3v2 tag or NULL if none exists.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_header_len">
+<description>
+Return the total length of the header in @buffer. This include the length of
+the fixed header, the CSRC list and the extension header.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The total length of the header in @buffer.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_timestamp">
+<description>
+Set the timestamp of the RTP packet in @buffer to @timestamp.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> the new timestamp
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_audio_structure_set_int">
+<description>
+Do not use anymore.
+
+Deprecated: use gst_structure_set()
+
+</description>
+<parameters>
+<parameter name="structure">
+<parameter_description> a #GstStructure
+</parameter_description>
+</parameter>
+<parameter name="flag">
+<parameter_description> a set of #GstAudioFieldFlag
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_ring_buffer_set_callback">
+<description>
+Sets the given callback function on the buffer. This function
+will be called every time a segment has been written to a device.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to set the callback on
+</parameter_description>
+</parameter>
+<parameter name="cb">
+<parameter_description> the callback to set
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> user data passed to the callback
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_packet_bye_add_ssrcs">
+<description>
+Adds @len SSRCs in @ssrc to BYE @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> an array of SSRCs to add
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> number of elements in @ssrc
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the all the SSRCs were added. This function can return %FALSE if
+the max MTU is exceeded or the number of sources blocks is greater than
+#GST_RTCP_MAX_BYE_SSRC_COUNT.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_set_rb">
+<description>
+Set the @nth new report block in @packet with the given values.
+
+Note: Not implemented.
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SR or RR #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="nth">
+<parameter_description> the nth report block to set
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> data source being reported
+</parameter_description>
+</parameter>
+<parameter name="fractionlost">
+<parameter_description> fraction lost since last SR/RR
+</parameter_description>
+</parameter>
+<parameter name="packetslost">
+<parameter_description> the cumululative number of packets lost
+</parameter_description>
+</parameter>
+<parameter name="exthighestseq">
+<parameter_description> the extended last sequence number received
+</parameter_description>
+</parameter>
+<parameter name="jitter">
+<parameter_description> the interarrival jitter
+</parameter_description>
+</parameter>
+<parameter name="lsr">
+<parameter_description> the last SR packet from this source
+</parameter_description>
+</parameter>
+<parameter name="dlsr">
+<parameter_description> the delay since last SR packet
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_buffer_add_packet">
+<description>
+Add a new packet of @type to @buffer. @packet will point to the newly created
+packet.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a valid RTCP buffer
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> the #GstRTCPType of the new packet
+</parameter_description>
+</parameter>
+<parameter name="packet">
+<parameter_description> pointer to new packet
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the packet could be created. This function returns %FALSE
+if the max mtu is exceeded for the buffer.
+</return>
+</function>
+
+<function name="gst_mixer_message_parse_mute_toggled">
+<description>
+Extracts the contents of a mute-toggled bus message. Reads
+the GstMixerTrack that has changed, and the new value of the mute
+flag.
+
+The GstMixerTrack remains valid until the message is freed.
+
+Since: 0.10.14
+
+</description>
+<parameters>
+<parameter name="message">
+<parameter_description> A mute-toggled change notification message.
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> Pointer to hold a GstMixerTrack object, or NULL.
+</parameter_description>
+</parameter>
+<parameter name="mute">
+<parameter_description> A pointer to a gboolean variable, or NULL.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_packet_remove">
+<description>
+Removes the packet pointed to by @packet.
+
+Note: Not implemented.
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_riff_parse_file_header">
+<description>
+Reads the first few bytes from the provided buffer, checks
+if this stream is a RIFF stream, and determines document type.
+This function takes ownership of @buf so it should not be used anymore
+after calling this function.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging/error).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input buffer from which the file header will be parsed,
+should be at least 12 bytes long.
+</parameter_description>
+</parameter>
+<parameter name="doctype">
+<parameter_description> a fourcc (returned by this function) to indicate the
+type of document (according to the header).
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if this is not a RIFF stream (in which case the
+caller should error out; we already throw an error), or TRUE
+if it is.
+</return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_set_sample_based">
+<description>
+Tells #GstBaseRTPAudioPayload that the child element is for a sample based
+audio codec
+
+
+</description>
+<parameters>
+<parameter name="basertpaudiopayload">
+<parameter_description> a pointer to the element.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_basertppayload_set_options">
+<description>
+Set the rtp options of the payloader. These options will be set in the caps
+of the payloader. Subclasses must call this method before calling
+gst_basertppayload_push() or gst_basertppayload_set_outcaps().
+
+</description>
+<parameters>
+<parameter name="payload">
+<parameter_description> a #GstBaseRTPPayload
+</parameter_description>
+</parameter>
+<parameter name="media">
+<parameter_description> the media type (typically "audio" or "video")
+</parameter_description>
+</parameter>
+<parameter name="dynamic">
+<parameter_description> if the payload type is dynamic
+</parameter_description>
+</parameter>
+<parameter name="encoding_name">
+<parameter_description> the encoding name
+</parameter_description>
+</parameter>
+<parameter name="clock_rate">
+<parameter_description> the clock rate of the media
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_message_parse_data">
+<description>
+Parse the data message @msg and store the channel in @channel.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> location to hold the channel
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_read">
+<description>
+Attempt to read @size bytes into @data from the connected @conn, blocking up to
+the specified @timeout. @timeout can be #NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to read
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of @data
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout value or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_rtsp_message_init_response">
+<description>
+Initialize @msg with @code and @reason.
+
+When @reason is #NULL, the default reason for @code will be used.
+
+When @request is not #NULL, the relevant headers will be copied to the new
+response message.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="code">
+<parameter_description> the status code
+</parameter_description>
+</parameter>
+<parameter name="reason">
+<parameter_description> the status reason or #NULL
+</parameter_description>
+</parameter>
+<parameter name="request">
+<parameter_description> the request that triggered the response or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_ring_buffer_commit">
+<description>
+Same as gst_ring_buffer_commit_full() but with a in_samples and out_samples
+equal to @len, ignoring accum.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to commit
+</parameter_description>
+</parameter>
+<parameter name="sample">
+<parameter_description> the sample position of the data
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to commit
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the number of samples in the data to commit
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of samples written to the ringbuffer or -1 on
+error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_sdp_media_set_media">
+<description>
+Set the media description of @media to @med.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="med">
+<parameter_description> the media description
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_rtsp_url_get_port">
+<description>
+Get the port number of @url.
+
+
+</description>
+<parameters>
+<parameter name="url">
+<parameter_description> a #GstRTSPUrl
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> location to hold the port
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_rtsp_transport_init">
+<description>
+Initialize @transport so that it can be used.
+
+
+</description>
+<parameters>
+<parameter name="transport">
+<parameter_description> a #GstRTSPTransport
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_audio_is_buffer_framed">
+<description>
+Check if the buffer size is a whole multiple of the frame size.
+
+
+</description>
+<parameters>
+<parameter name="pad">
+<parameter_description> the #GstPad to get the caps from
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> the #GstBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if buffer size is multiple.
+</return>
+</function>
+
+<function name="gst_rtsp_url_set_port">
+<description>
+Set the port number in @url to @port.
+
+
+</description>
+<parameters>
+<parameter name="url">
+<parameter_description> a #GstRTSPUrl
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> the port
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_rtsp_header_as_text">
+<description>
+Convert @field to a string.
+
+
+</description>
+<parameters>
+<parameter name="field">
+<parameter_description> a #GstRTSPHeaderField
+</parameter_description>
+</parameter>
+</parameters>
+<return> a string representation of @field.
+</return>
+</function>
+
+<function name="gst_sdp_message_set_information">
+<description>
+Set the information in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="information">
+<parameter_description> the information
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_sdp_media_bandwidths_len">
+<description>
+Get the number of bandwidth fields in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of bandwidths in @media.
+</return>
+</function>
+
+<function name="gst_video_orientation_set_hflip">
+<description>
+Set the horizontal flipping state (%TRUE for flipped) for the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="flip">
+<parameter_description> use flipping
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports flipping
+</return>
+</function>
+
+<function name="gst_mixer_message_get_type">
+<description>
+Check a bus message to see if it is a GstMixer notification
+message and return the GstMixerMessageType identifying which
+type of notification it is.
+
+
+</description>
+<parameters>
+<parameter name="message">
+<parameter_description> A GstMessage to inspect.
+</parameter_description>
+</parameter>
+</parameters>
+<return> The type of the GstMixerMessage, or GST_MIXER_MESSAGE_NONE
+if the message is not a GstMixer notification.
+
+Since: 0.10.14
+</return>
+</function>
+
+<function name="gst_sdp_message_get_information">
+<description>
+Get the information in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_payload">
+<description>
+Get a pointer to the payload data in @buffer. This pointer is valid as long
+as a reference to @buffer is held.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> A pointer to the payload data in @buffer.
+</return>
+</function>
+
+<function name="gst_sdp_media_set_port_info">
+<description>
+Set the port information in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> the port number
+</parameter_description>
+</parameter>
+<parameter name="num_ports">
+<parameter_description> the number of ports
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_audio_clock_new">
+<description>
+Create a new #GstAudioClock instance. Whenever the clock time should be
+calculated it will call @func with @user_data. When @func returns
+#GST_CLOCK_TIME_NONE, the clock will return the last reported time.
+
+
+</description>
+<parameters>
+<parameter name="name">
+<parameter_description> the name of the clock
+</parameter_description>
+</parameter>
+<parameter name="func">
+<parameter_description> a function
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> user data
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstAudioClock casted to a #GstClock.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_rr_get_ssrc">
+<description>
+Get the ssrc field of the RR @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid RR #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> the ssrc.
+</return>
+</function>
+
+<function name="gst_mixer_message_parse_options_list_changed">
+<description>
+Extracts the GstMixerOptions whose value list has changed from an
+options-list-changed bus notification message.
+
+The options object returned remains valid until the message is freed. You
+do not need to unref it.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="message">
+<parameter_description> A volume-changed change notification message.
+</parameter_description>
+</parameter>
+<parameter name="options">
+<parameter_description> Pointer to hold a GstMixerOptions object, or NULL.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_fft_f32_fft">
+<description>
+This performs the FFT on @timedata and puts the result in @freqdata.
+
+ timedata must have as many samples as specified with the @len parameter while
+allocating the #GstFFTF32 instance with gst_fft_f32_new().
+
+ freqdata must be large enough to hold @len/2 + 1 #GstFFTF32Complex frequency
+domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF32 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Buffer of the samples in the time domain
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Target buffer for the samples in the frequency domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_base_audio_sink_create_ringbuffer">
+<description>
+Create and return the #GstRingBuffer for @sink. This function will call the
+::create_ringbuffer vmethod and will set @sink as the parent of the returned
+buffer (see gst_object_set_parent()).
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> a #GstBaseAudioSink.
+</parameter_description>
+</parameter>
+</parameters>
+<return> The new ringbuffer of @sink.
+</return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_push">
+<description>
+Create an RTP buffer and store @payload_len bytes of @data as the
+payload. Set the timestamp on the new buffer to @timestamp before pushing
+the buffer downstream.
+
+
+</description>
+<parameters>
+<parameter name="baseaudiopayload">
+<parameter_description> a #GstBaseRTPPayload
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> data to set as payload
+</parameter_description>
+</parameter>
+<parameter name="payload_len">
+<parameter_description> length of payload
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> a #GstClockTime
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstFlowReturn
+
+Since: 0.10.13
+</return>
+</function>
+
+<function name="gst_ring_buffer_open_device">
+<description>
+Open the audio device associated with the ring buffer. Does not perform any
+setup on the device. You must open the device before acquiring the ring
+buffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be opened, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_rtsp_message_new">
+<description>
+Create a new initialized #GstRTSPMessage.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a location for the new #GstRTSPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult. Free with gst_rtsp_message_free().
+</return>
+</function>
+
+<function name="gst_ring_buffer_debug_spec_caps">
+<description>
+Print debug info about the parsed caps in @spec to the debug log.
+
+</description>
+<parameters>
+<parameter name="spec">
+<parameter_description> the spec to debug
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_audio_duration_from_pad_buffer">
+<description>
+Calculate length in nanoseconds of audio buffer @buf based on capabilities of
+ pad
+
+Return: the length.
+
+</description>
+<parameters>
+<parameter name="pad">
+<parameter_description> the #GstPad to get the caps from
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> the #GstBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_fft_f64_window">
+<description>
+This calls the window function @window on the @timedata sample buffer.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF64 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Time domain samples
+</parameter_description>
+</parameter>
+<parameter name="window">
+<parameter_description> Window function to apply
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_pad_to">
+<description>
+Set the amount of padding in the RTP packet in @buffer to
+ len If @len is 0, the padding is removed.
+
+NOTE: This function does not work correctly.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the new amount of padding
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_strresult">
+<description>
+Convert @result in a human readable string.
+
+
+</description>
+<parameters>
+<parameter name="result">
+<parameter_description> a #GstRTSPResult
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly allocated string. g_free() after usage.
+</return>
+</function>
+
+<function name="gst_fft_s32_fft">
+<description>
+This performs the FFT on @timedata and puts the result in @freqdata.
+
+ timedata must have as many samples as specified with the @len parameter while
+allocating the #GstFFTS32 instance with gst_fft_s32_new().
+
+ freqdata must be large enough to hold @len/2 + 1 #GstFFTS32Complex frequency
+domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS32 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Buffer of the samples in the time domain
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Target buffer for the samples in the frequency domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_range_parse">
+<description>
+Parse @rangestr to a #GstRTSPTimeRange.
+
+
+</description>
+<parameters>
+<parameter name="rangestr">
+<parameter_description> a range string to parse
+</parameter_description>
+</parameter>
+<parameter name="range">
+<parameter_description> location to hold the #GstRTSPTimeRange result
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_base_rtp_depayload_push_ts">
+<description>
+Push @out_buf to the peer of @filter. This function takes ownership of
+ out_buf
+
+Unlike gst_base_rtp_depayload_push(), this function will apply @timestamp
+on the outgoing buffer, using the configured clock_rate to convert the
+timestamp to a valid GStreamer clock time.
+
+
+</description>
+<parameters>
+<parameter name="filter">
+<parameter_description> a #GstBaseRTPDepayload
+</parameter_description>
+</parameter>
+<parameter name="timestamp">
+<parameter_description> an RTP timestamp to apply
+</parameter_description>
+</parameter>
+<parameter name="out_buf">
+<parameter_description> a #GstBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstFlowReturn.
+</return>
+</function>
+
+<function name="gst_property_probe_probe_property_name">
+<description>
+Runs a probe on the given property.
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe to check.
+</parameter_description>
+</parameter>
+<parameter name="name">
+<parameter_description> name of the property to return.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_connection_free">
+<description>
+Close and free @conn.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_video_format_get_component_height">
+<description>
+Calculates the height of the component. See
+ gst_video_format_get_row_stride for a description
+of the component index.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="component">
+<parameter_description> the component index
+</parameter_description>
+</parameter>
+<parameter name="height">
+<parameter_description> the height of video
+</parameter_description>
+</parameter>
+</parameters>
+<return> height of component @component
+</return>
+</function>
+
+<function name="gst_rtsp_connection_create">
+<description>
+Create a newly allocated #GstRTSPConnection from @url and store it in @conn.
+The connection will not yet attempt to connect to @url, use
+gst_rtsp_connection_connect().
+
+
+</description>
+<parameters>
+<parameter name="url">
+<parameter_description> a #GstRTSPUrl
+</parameter_description>
+</parameter>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK when @conn contains a valid connection.
+</return>
+</function>
+
+<function name="gst_tag_register_musicbrainz_tags">
+<description>
+Registers additional musicbrainz-specific tags with the GStreamer tag
+system. Plugins and applications that use these tags should call this
+function before using them. Can be called multiple times.
+
+</description>
+<parameters>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_get_seq">
+<description>
+Get the sequence number of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The sequence number in host order.
+</return>
+</function>
+
+<function name="gst_video_format_has_alpha">
+<description>
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @format has an alpha channel
+</return>
+</function>
+
+<function name="gst_tuner_get_signal_strength">
+<description>
+get the strength of the signal on this channel. Note that this
+requires the current channel to be a "tuning" channel, e.g. a
+channel on which frequency can be set. This can be checked using
+GST_TUNER_CHANNEL_HAS_FLAG (), and the appropriate flag to check
+for is GST_TUNER_CHANNEL_FREQUENCY.
+
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) that owns the given channel.
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> the #GstTunerChannel to get the signal strength from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> signal strength, or 0 on error.
+</return>
+</function>
+
+<function name="gst_rtsp_status_as_text">
+<description>
+Convert @code to a string.
+
+
+</description>
+<parameters>
+<parameter name="code">
+<parameter_description> a #GstRTSPStatusCode
+</parameter_description>
+</parameter>
+</parameters>
+<return> a string representation of @code.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_first_item">
+<description>
+Move to the first SDES item in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if there was a first item.
+</return>
+</function>
+
+<function name="gst_rtsp_message_parse_request">
+<description>
+Parse the request message @msg and store the values @method, @uri and
+ version The result locations can be #NULL if one is not interested in its
+value.
+
+ uri remains valid for as long as @msg is valid and unchanged.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="method">
+<parameter_description> location to hold the method
+</parameter_description>
+</parameter>
+<parameter name="uri">
+<parameter_description> location to hold the uri
+</parameter_description>
+</parameter>
+<parameter name="version">
+<parameter_description> location to hold the version
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_sdp_message_init">
+<description>
+Initialize @msg so that its contents are as if it was freshly allocated
+with gst_sdp_message_new(). This function is mostly used to initialize a message
+allocated on the stack. gst_sdp_message_uninit() undoes this operation.
+
+When this function is invoked on newly allocated data (with malloc or on the
+stack), its contents should be set to 0 before calling this function.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_sdp_message_times_len">
+<description>
+Get the number of time information entries in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of time information entries in @msg.
+</return>
+</function>
+
+<function name="gst_fft_f32_inverse_fft">
+<description>
+This performs the inverse FFT on @freqdata and puts the result in @timedata.
+
+ freqdata must have @len/2 + 1 samples, where @len is the parameter specified
+while allocating the #GstFFTF32 instance with gst_fft_f32_new().
+
+ timedata must be large enough to hold @len time domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF32 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Buffer of the samples in the frequency domain
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Target buffer for the samples in the time domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_format_get_component_offset">
+<description>
+Calculates the offset (in bytes) of the first pixel of the component
+with index @component. For packed formats, this will typically be a
+small integer (0, 1, 2, 3). For planar formats, this will be a
+(relatively) large offset to the beginning of the second or third
+component planes. See @gst_video_format_get_row_stride for a description
+of the component index.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="component">
+<parameter_description> the component index
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> the width of video
+</parameter_description>
+</parameter>
+<parameter name="height">
+<parameter_description> the height of video
+</parameter_description>
+</parameter>
+</parameters>
+<return> offset of component @component
+</return>
+</function>
+
+<function name="gst_rtsp_message_take_body">
+<description>
+Set the body of @msg to @data and @size. This method takes ownership of
+ data
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of @data
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_video_sink_center_rect">
+<description>
+Takes @src rectangle and position it at the center of @dst rectangle with or
+without @scaling. It handles clipping if the @src rectangle is bigger than
+the @dst one and @scaling is set to FALSE.
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> the #GstVideoRectangle describing the source area
+</parameter_description>
+</parameter>
+<parameter name="dst">
+<parameter_description> the #GstVideoRectangle describing the destination area
+</parameter_description>
+</parameter>
+<parameter name="result">
+<parameter_description> a pointer to a #GstVideoRectangle which will receive the result area
+</parameter_description>
+</parameter>
+<parameter name="scaling">
+<parameter_description> a #gboolean indicating if scaling should be applied or not
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_riff_read_chunk">
+<description>
+Reads a single chunk of data. Since 0.10.8 'JUNK' chunks
+are skipped automatically.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging).
+</parameter_description>
+</parameter>
+<parameter name="pad">
+<parameter_description> pad to pull data from.
+</parameter_description>
+</parameter>
+<parameter name="offset">
+<parameter_description> offset to pull from, incremented by this function.
+</parameter_description>
+</parameter>
+<parameter name="tag">
+<parameter_description> fourcc of the chunk (returned by this function).
+</parameter_description>
+</parameter>
+<parameter name="chunk_data">
+<parameter_description> buffer (returned by this function).
+</parameter_description>
+</parameter>
+</parameters>
+<return> flow status.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_validate_data">
+<description>
+Check if the @data and @size point to the data of a valid RTP packet.
+This function checks the length, version and padding of the packet data.
+Use this function to validate a packet before using the other functions in
+this module.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> the data to validate
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of @data to validate
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the data points to a valid RTP packet.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_poll">
+<description>
+Wait up to the specified @timeout for the connection to become available for
+at least one of the operations specified in @events. When the function returns
+with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
+ conn
+
+ timeout can be #NULL, in which case this function might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="events">
+<parameter_description> a bitmask of #GstRTSPEvent flags to check
+</parameter_description>
+</parameter>
+<parameter name="revents">
+<parameter_description> location for result flags
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_sdp_message_get_phone">
+<description>
+Get the phone with number @idx from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> a phone index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the phone at position @idx.
+</return>
+</function>
+
+<function name="gst_pb_utils_get_codec_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description> the (fixed) #GstCaps for which an format description is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_ring_buffer_parse_caps">
+<description>
+Parse @caps into @spec.
+
+
+</description>
+<parameters>
+<parameter name="spec">
+<parameter_description> a spec
+</parameter_description>
+</parameter>
+<parameter name="caps">
+<parameter_description> a #GstCaps
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the caps could be parsed.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sr_set_sender_info">
+<description>
+Set the given values in the SR packet @packet.
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SR #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> the SSRC
+</parameter_description>
+</parameter>
+<parameter name="ntptime">
+<parameter_description> the NTP time
+</parameter_description>
+</parameter>
+<parameter name="rtptime">
+<parameter_description> the RTP time
+</parameter_description>
+</parameter>
+<parameter name="packet_count">
+<parameter_description> the packet count
+</parameter_description>
+</parameter>
+<parameter name="octet_count">
+<parameter_description> the octect count
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_video_format_convert">
+<description>
+Converts among various #GstFormat types. This function handles
+GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
+raw video, GST_FORMAT_DEFAULT corresponds to video frames. This
+function can be to handle pad queries of the type GST_QUERY_CONVERT.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="format">
+<parameter_description> a #GstVideoFormat
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> the width of video
+</parameter_description>
+</parameter>
+<parameter name="height">
+<parameter_description> the height of video
+</parameter_description>
+</parameter>
+<parameter name="fps_n">
+<parameter_description> frame rate numerator
+</parameter_description>
+</parameter>
+<parameter name="fps_d">
+<parameter_description> frame rate denominator
+</parameter_description>
+</parameter>
+<parameter name="src_format">
+<parameter_description> #GstFormat of the @src_value
+</parameter_description>
+</parameter>
+<parameter name="src_value">
+<parameter_description> value to convert
+</parameter_description>
+</parameter>
+<parameter name="dest_format">
+<parameter_description> #GstFormat of the @dest_value
+</parameter_description>
+</parameter>
+<parameter name="dest_value">
+<parameter_description> pointer to destination value
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the conversion was successful.
+</return>
+</function>
+
+<function name="gst_fft_s16_new">
+<description>
+This returns a new #GstFFTS16 instance with the given parameters. It makes
+sense to keep one instance for several calls for speed reasons.
+
+ len must be even and to get the best performance a product of
+2, 3 and 5. To get the next number with this characteristics use
+gst_fft_next_fast_length().
+
+
+</description>
+<parameters>
+<parameter name="len">
+<parameter_description> Length of the FFT in the time domain
+</parameter_description>
+</parameter>
+<parameter name="inverse">
+<parameter_description> %TRUE if the #GstFFTS16 instance should be used for the inverse FFT
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstFFTS16 instance.
+</return>
+</function>
+
+<function name="gst_video_parse_caps_pixel_aspect_ratio">
+<description>
+Extracts the pixel aspect ratio from @caps and places the values in
+the locations pointed to by @par_n and @par_d. Returns TRUE if the
+values could be parsed correctly, FALSE if not.
+
+This function can be used with #GstCaps that have any media type; it
+is not limited to formats handled by #GstVideoFormat.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description>
+</parameter_description>
+</parameter>
+<parameter name="par_n">
+<parameter_description> pointer to numerator of pixel aspect ratio (output)
+</parameter_description>
+</parameter>
+<parameter name="par_d">
+<parameter_description> pointer to denominator of pixel aspect ratio (output)
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @caps was parsed correctly.
+</return>
+</function>
+
+<function name="gst_install_plugins_context_set_xid">
+<description>
+This function is for X11-based applications (such as most Gtk/Qt
+applications on linux/unix) only. You can use it to tell the external
+the XID of your main application window, so the installer can make its
+own window transient to your application windonw during the installation.
+
+If set, the XID will be passed to the installer via a --transient-for=XID
+command line option.
+
+Gtk+/Gnome application should be able to obtain the XID of the top-level
+window like this:
+&lt;programlisting&gt;
+##include &lt;gtk/gtk.h&gt;
+##ifdef GDK_WINDOWING_X11
+##include &lt;gdk/gdkx.h&gt;
+##endif
+...
+##ifdef GDK_WINDOWING_X11
+xid = GDK_WINDOW_XWINDOW (GTK_WIDGET (application_window)-&gt;window);
+##endif
+...
+&lt;/programlisting&gt;
+
+Since: 0.10.12
+
+</description>
+<parameters>
+<parameter name="ctx">
+<parameter_description> a #GstInstallPluginsContext
+</parameter_description>
+</parameter>
+<parameter name="xid">
+<parameter_description> the XWindow ID (XID) of the top-level application
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_base_rtp_audio_payload_set_frame_based">
+<description>
+Tells #GstBaseRTPAudioPayload that the child element is for a frame based
+audio codec
+
+
+</description>
+<parameters>
+<parameter name="basertpaudiopayload">
+<parameter_description> a pointer to the element.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_allocate_data">
+<description>
+Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs,
+a payload length of @payload_len and padding of @pad_len.
+MALLOCDATA of @buffer will be overwritten and will not be freed.
+All other RTP header fields will be set to 0/FALSE.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> a #GstBuffer
+</parameter_description>
+</parameter>
+<parameter name="payload_len">
+<parameter_description> the length of the payload
+</parameter_description>
+</parameter>
+<parameter name="pad_len">
+<parameter_description> the amount of padding
+</parameter_description>
+</parameter>
+<parameter name="csrc_count">
+<parameter_description> the number of CSRC entries
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_audio_set_caps_channel_positions_list">
+<description>
+Sets a (possibly non-fixed) list of possible audio channel
+positions (given in pos) on the given caps. Each of the
+structures of the caps, after this function has been called,
+will contain a "channel-positions" field with an array.
+Each value in the array will contain each of the values given
+in the pos array. Note that the size of the caps might be
+increased by this, since each structure with a "channel-
+positions" field needs to have a fixed "channels" field.
+The input caps is not required to have this.
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description> #GstCaps to set the list of channel positions on.
+</parameter_description>
+</parameter>
+<parameter name="pos">
+<parameter_description> the array containing one or more possible audio
+channel positions that we should add in each value
+of the array in the given structure.
+</parameter_description>
+</parameter>
+<parameter name="num_positions">
+<parameter_description> the number of values in pos.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_audio_get_channel_positions">
+<description>
+Retrieves a number of (fixed!) audio channel positions from
+the provided #GstStructure and returns it as a newly allocated
+array. The caller should g_free () this array. The caller
+should also check that the members in this #GstStructure are
+indeed "fixed" before calling this function.
+
+
+</description>
+<parameters>
+<parameter name="str">
+<parameter_description> A #GstStructure to retrieve channel positions from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly allocated array containing the channel
+positions as provided in the given #GstStructure. Returns
+NULL on error.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_first_entry">
+<description>
+Move to the first SDES entry in the current item.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if there was a first entry.
+</return>
+</function>
+
+<function name="gst_sdp_media_set_key">
+<description>
+Adds the encryption information to @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> the encryption type
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the encryption data
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_tag_list_from_vorbiscomment_buffer">
+<description>
+Creates a new tag list that contains the information parsed out of a
+vorbiscomment packet.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> buffer to convert
+</parameter_description>
+</parameter>
+<parameter name="id_data">
+<parameter_description> identification data at start of stream
+</parameter_description>
+</parameter>
+<parameter name="id_data_length">
+<parameter_description> length of identification data
+</parameter_description>
+</parameter>
+<parameter name="vendor_string">
+<parameter_description> pointer to a string that should take the vendor string
+of this vorbis comment or NULL if you don't need it.
+</parameter_description>
+</parameter>
+</parameters>
+<return> A new #GstTagList with all tags that could be extracted from the
+given vorbiscomment buffer or NULL on error.
+</return>
+</function>
+
+<function name="gst_rtcp_unix_to_ntp">
+<description>
+Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should
+pass a value with nanoseconds since 1970. The NTP time will, in the upper
+32 bits, contain the number of seconds since 1900 and, in the lower 32
+bits, the fractional seconds. The resulting value can be used as an ntptime
+for constructing SR RTCP packets.
+
+
+</description>
+<parameters>
+<parameter name="unixtime">
+<parameter_description> an UNIX timestamp in nanoseconds
+</parameter_description>
+</parameter>
+</parameters>
+<return> the NTP time for @gsttime.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_payload_type">
+<description>
+Get the payload type of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The payload type.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_receive">
+<description>
+Attempt to read into @message from the connected @conn, blocking up to
+the specified @timeout. @timeout can be #NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="message">
+<parameter_description> the message to read
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout value or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_video_orientation_set_vcenter">
+<description>
+Set the vertical centering offset for the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="center">
+<parameter_description> centering offset
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports centering
+</return>
+</function>
+
+<function name="gst_video_orientation_set_hcenter">
+<description>
+Set the horizontal centering offset for the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="center">
+<parameter_description> centering offset
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports centering
+</return>
+</function>
+
+<function name="gst_ring_buffer_advance">
+<description>
+Subclasses should call this function to notify the fact that
+ advance segments are now processed by the device.
+
+MT safe.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to advance
+</parameter_description>
+</parameter>
+<parameter name="advance">
+<parameter_description> the number of segments written
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_ring_buffer_release">
+<description>
+Free the resources of the ringbuffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to release
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be released, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_padding">
+<description>
+Set the padding bit on the RTP packet in @buffer to @padding.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="padding">
+<parameter_description> the new padding
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_ring_buffer_delay">
+<description>
+Get the number of samples queued in the audio device. This is
+usually less than the segment size but can be bigger when the
+implementation uses another internal buffer between the audio
+device.
+
+For playback ringbuffers this is the amount of samples transfered from the
+ringbuffer to the device but still not played.
+
+For capture ringbuffers this is the amount of samples in the device that are
+not yet transfered to the ringbuffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to query
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of samples queued in the audio device.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_rtsp_message_set_body">
+<description>
+Set the body of @msg to a copy of @data.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of @data
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_ring_buffer_start">
+<description>
+Start processing samples from the ringbuffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to start
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be started, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="read_packet_header">
+<description>
+Read the packet headers for the packet pointed to by @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a packet
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @packet pointed to a valid header.
+</return>
+</function>
+
+<function name="gst_fft_f64_new">
+<description>
+This returns a new #GstFFTF64 instance with the given parameters. It makes
+sense to keep one instance for several calls for speed reasons.
+
+ len must be even and to get the best performance a product of
+2, 3 and 5. To get the next number with this characteristics use
+gst_fft_next_fast_length().
+
+
+</description>
+<parameters>
+<parameter name="len">
+<parameter_description> Length of the FFT in the time domain
+</parameter_description>
+</parameter>
+<parameter name="inverse">
+<parameter_description> %TRUE if the #GstFFTF64 instance should be used for the inverse FFT
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstFFTF64 instance.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_bandwidth">
+<description>
+Add the specified bandwidth information to @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="bwtype">
+<parameter_description> the bandwidth modifier type
+</parameter_description>
+</parameter>
+<parameter name="bandwidth">
+<parameter_description> the bandwidth in kilobits per second
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtsp_message_parse_response">
+<description>
+Parse the response message @msg and store the values @code, @reason and
+ version The result locations can be #NULL if one is not interested in its
+value.
+
+ reason remains valid for as long as @msg is valid and unchanged.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="code">
+<parameter_description> location to hold the status code
+</parameter_description>
+</parameter>
+<parameter name="reason">
+<parameter_description> location to hold the status reason
+</parameter_description>
+</parameter>
+<parameter name="version">
+<parameter_description> location to hold the version
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_sdp_message_set_uri">
+<description>
+Set the URI in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="uri">
+<parameter_description> the URI
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_sdp_media_formats_len">
+<description>
+Get the number of formats in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of formats in @media.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_compare_seqnum">
+<description>
+Compare two sequence numbers, taking care of wraparounds.
+
+
+</description>
+<parameters>
+<parameter name="seqnum1">
+<parameter_description> a sequence number
+</parameter_description>
+</parameter>
+<parameter name="seqnum2">
+<parameter_description> a sequence number
+</parameter_description>
+</parameter>
+</parameters>
+<return> -1 if @seqnum1 is before @seqnum2, 0 if they are equal or 1 if
+ seqnum1 is bigger than @segnum2.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_ring_buffer_stop">
+<description>
+Stop processing samples from the ringbuffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to stop
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be stopped, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_install_plugins_installation_in_progress">
+<description>
+Checks whether plugin installation (initiated by this application only)
+is currently in progress.
+
+
+</description>
+<parameters>
+</parameters>
+<return> TRUE if plugin installation is in progress, otherwise FALSE
+
+Since: 0.10.12
+</return>
+</function>
+
+<function name="gst_rtsp_message_new_data">
+<description>
+Create a new data #GstRTSPMessage with @channel and store the
+result message in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a location for the new #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> the channel
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult. Free with gst_rtsp_message_free().
+</return>
+</function>
+
+<function name="gst_missing_uri_sink_installer_detail_new">
+<description>
+Returns: a newly-allocated detail string, or NULL on error. Free string
+
+</description>
+<parameters>
+<parameter name="protocol">
+<parameter_description> the URI protocol the missing source needs to implement,
+e.g. "http" or "mms"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated detail string, or NULL on error. Free string
+with g_free() when not needed any longer.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_missing_element_installer_detail_new">
+<description>
+Returns: a newly-allocated detail string, or NULL on error. Free string
+
+</description>
+<parameters>
+<parameter name="factory_name">
+<parameter_description> the name of the missing element (element factory),
+e.g. "videoscale" or "cdparanoiasrc"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated detail string, or NULL on error. Free string
+with g_free() when not needed any longer.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_rtsp_connection_send">
+<description>
+Attempt to send @message to the connected @conn, blocking up to
+the specified @timeout. @timeout can be #NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="message">
+<parameter_description> the message to send
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout value or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_version">
+<description>
+Get the version in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_fft_next_fast_length">
+<description>
+Returns: the next fast FFT length.
+
+</description>
+<parameters>
+<parameter name="n">
+<parameter_description> Number for which the next fast length should be returned
+</parameter_description>
+</parameter>
+</parameters>
+<return> the next fast FFT length.
+
+</return>
+</function>
+
+<function name="gst_install_plugins_async">
+<description>
+Requests plugin installation without blocking. Once the plugins have been
+installed or installation has failed, @func will be called with the result
+of the installation and your provided @user_data pointer.
+
+This function requires a running GLib/Gtk main loop. If you are not
+running a GLib/Gtk main loop, make sure to regularly call
+g_main_context_iteration(NULL,FALSE).
+
+The installer strings that make up @detail are typically obtained by
+calling gst_missing_plugin_message_get_installer_detail() on missing-plugin
+messages that have been caught on a pipeline's bus or created by the
+application via the provided API, such as gst_missing_element_message_new().
+
+It is possible to request the installation of multiple missing plugins in
+one go (as might be required if there is a demuxer for a certain format
+installed but no suitable video decoder and no suitable audio decoder).
+
+
+</description>
+<parameters>
+<parameter name="details">
+<parameter_description> NULL-terminated array of installer string details (see below)
+</parameter_description>
+</parameter>
+<parameter name="ctx">
+<parameter_description> a #GstInstallPluginsContext, or NULL
+</parameter_description>
+</parameter>
+<parameter name="func">
+<parameter_description> the function to call when the installer program returns
+</parameter_description>
+</parameter>
+<parameter name="user_data">
+<parameter_description> the user data to pass to @func when called, or NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> result code whether an external installer could be started
+
+Since: 0.10.12
+</return>
+</function>
+
+<function name="gst_rtcp_packet_get_rb">
+<description>
+Parse the values of the @nth report block in @packet and store the result in
+the values.
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SR or RR #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="nth">
+<parameter_description> the nth report block in @packet
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> result for data source being reported
+</parameter_description>
+</parameter>
+<parameter name="fractionlost">
+<parameter_description> result for fraction lost since last SR/RR
+</parameter_description>
+</parameter>
+<parameter name="packetslost">
+<parameter_description> result for the cumululative number of packets lost
+</parameter_description>
+</parameter>
+<parameter name="exthighestseq">
+<parameter_description> result for the extended last sequence number received
+</parameter_description>
+</parameter>
+<parameter name="jitter">
+<parameter_description> result for the interarrival jitter
+</parameter_description>
+</parameter>
+<parameter name="lsr">
+<parameter_description> result for the last SR packet from this source
+</parameter_description>
+</parameter>
+<parameter name="dlsr">
+<parameter_description> result for the delay since last SR packet
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_message_get_bandwidth">
+<description>
+Get the bandwidth at index @idx from @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the bandwidth index
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPBandwidth.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_uri">
+<description>
+Get the URI in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_sdp_media_uninit">
+<description>
+Free all resources allocated in @media. @media should not be used anymore after
+this function. This function should be used when @media was allocated on the
+stack and initialized with gst_sdp_media_init().
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_calc_payload_len">
+<description>
+Calculate the length of the payload of an RTP packet with size @packet_len,
+a padding of @pad_len and a @csrc_count CSRC entries.
+
+
+</description>
+<parameters>
+<parameter name="packet_len">
+<parameter_description> the length of the total RTP packet
+</parameter_description>
+</parameter>
+<parameter name="pad_len">
+<parameter_description> the amount of padding
+</parameter_description>
+</parameter>
+<parameter name="csrc_count">
+<parameter_description> the number of CSRC entries
+</parameter_description>
+</parameter>
+</parameters>
+<return> The length of the payload of an RTP packet with given parameters.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_set_ssrc">
+<description>
+Set the SSRC on the RTP packet in @buffer to @ssrc.
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> the new SSRC
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_ntp_to_unix">
+<description>
+Converts an NTP time to UNIX nanoseconds. @ntptime can typically be
+the NTP time of an SR RTCP message and contains, in the upper 32 bits, the
+number of seconds since 1900 and, in the lower 32 bits, the fractional
+seconds. The resulting value will be the number of nanoseconds since 1970.
+
+
+</description>
+<parameters>
+<parameter name="ntptime">
+<parameter_description> an NTP timestamp
+</parameter_description>
+</parameter>
+</parameters>
+<return> the UNIX time for @ntptime in nanoseconds.
+</return>
+</function>
+
+<function name="gst_basertppayload_push">
+<description>
+Push @buffer to the peer element of the payloader. The SSRC, payload type,
+seqnum and timestamp of the RTP buffer will be updated first.
+
+This function takes ownership of @buffer.
+
+
+</description>
+<parameters>
+<parameter name="payload">
+<parameter_description> a #GstBaseRTPPayload
+</parameter_description>
+</parameter>
+<parameter name="buffer">
+<parameter_description> a #GstBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstFlowReturn.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_attribute_val_n">
+<description>
+Get the @nth attribute with key @key in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> the key
+</parameter_description>
+</parameter>
+<parameter name="nth">
+<parameter_description> the index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the attribute value of the @nth attribute with @key.
+</return>
+</function>
+
+<function name="gst_install_plugins_sync">
+<description>
+Requests plugin installation and block until the plugins have been
+installed or installation has failed.
+
+This function should almost never be used, it only exists for cases where
+a non-GLib main loop is running and the user wants to run it in a separate
+thread and marshal the result back asynchronously into the main thread
+using the other non-GLib main loop. You should almost always use
+gst_install_plugins_async() instead of this function.
+
+
+</description>
+<parameters>
+<parameter name="details">
+<parameter_description> NULL-terminated array of installer string details
+</parameter_description>
+</parameter>
+<parameter name="ctx">
+<parameter_description> a #GstInstallPluginsContext, or NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> the result of the installation.
+
+Since: 0.10.12
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_version">
+<description>
+Get the version number of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The version of @buffer.
+</return>
+</function>
+
+<function name="gst_audio_frame_byte_size">
+<description>
+Calculate byte size of an audio frame.
+
+
+</description>
+<parameters>
+<parameter name="pad">
+<parameter_description> the #GstPad to get the caps from
+</parameter_description>
+</parameter>
+</parameters>
+<return> the byte size, or 0 if there was an error
+</return>
+</function>
+
+<function name="gst_fft_s16_free">
+<description>
+This frees the memory allocated for @self.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS16 instance for this call
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_mixer_mixer_changed">
+<description>
+This function is called by the mixer implementation to produce
+a notification message on the bus indicating that the list of available
+mixer tracks for a given mixer object has changed. Applications should
+rebuild their interface when they receive this message.
+
+This function only works for GstElements that are implementing the
+GstMixer interface, and the element needs to have been provided a bus.
+
+Since: 0.10.18
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) which has changed
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_tuner_get_frequency">
+<description>
+Retrieve the current frequency from the given channel. The same
+applies as for set_frequency (): check the flag.
+
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) that owns the given channel.
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> the #GstTunerChannel to retrieve the frequency from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the current frequency, or 0 on error.
+</return>
+</function>
+
+<function name="gst_basertppayload_set_outcaps">
+<description>
+Configure the output caps with the optional parameters.
+
+Variable arguments should be in the form field name, field type
+(as a GType), value(s). The last variable argument should be NULL.
+
+
+</description>
+<parameters>
+<parameter name="payload">
+<parameter_description> a #GstBaseRTPPayload
+</parameter_description>
+</parameter>
+<parameter name="fieldname">
+<parameter_description> the first field name or %NULL
+</parameter_description>
+</parameter>
+<parameter name="Varargs">
+<parameter_description> field values
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the caps could be set.
+</return>
+</function>
+
+<function name="gst_tag_from_id3_tag">
+<description>
+Looks up the GStreamer tag for a ID3v2 tag.
+
+
+</description>
+<parameters>
+<parameter name="id3_tag">
+<parameter_description> ID3v2 tag to convert to GStreamer tag
+</parameter_description>
+</parameter>
+</parameters>
+<return> The corresponding GStreamer tag or NULL if none exists.
+</return>
+</function>
+
+<function name="gst_mixer_record_toggled">
+<description>
+This function is called by the mixer implementation to produce
+a notification message on the bus indicating that the given track
+has changed recording state.
+
+This function only works for GstElements that are implementing the
+GstMixer interface, and the element needs to have been provided a bus.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> the #GstMixer (a #GstElement) that owns the track
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> the GstMixerTrack that has changed recording state.
+</parameter_description>
+</parameter>
+<parameter name="record">
+<parameter_description> the new state of the record flag on the track
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_message_unset">
+<description>
+Unset the concents of @msg so that it becomes an uninitialized
+#GstRTSPMessage again. This function is mostly used in combination with
+gst_rtsp_message_init_request(), gst_rtsp_message_init_response() and
+gst_rtsp_message_init_data() on stack allocated #GstRTSPMessage structures.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_base_rtp_depayload_push">
+<description>
+Push @out_buf to the peer of @filter. This function takes ownership of
+ out_buf
+
+Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
+any timestamp on the outgoing buffer.
+
+
+</description>
+<parameters>
+<parameter name="filter">
+<parameter_description> a #GstBaseRTPDepayload
+</parameter_description>
+</parameter>
+<parameter name="out_buf">
+<parameter_description> a #GstBuffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstFlowReturn.
+</return>
+</function>
+
+<function name="gst_fft_f32_new">
+<description>
+This returns a new #GstFFTF32 instance with the given parameters. It makes
+sense to keep one instance for several calls for speed reasons.
+
+ len must be even and to get the best performance a product of
+2, 3 and 5. To get the next number with this characteristics use
+gst_fft_next_fast_length().
+
+
+</description>
+<parameters>
+<parameter name="len">
+<parameter_description> Length of the FFT in the time domain
+</parameter_description>
+</parameter>
+<parameter name="inverse">
+<parameter_description> %TRUE if the #GstFFTF32 instance should be used for the inverse FFT
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstFFTF32 instance.
+</return>
+</function>
+
+<function name="gst_pb_utils_get_sink_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="protocol">
+<parameter_description> the protocol the sink element needs to handle, e.g. "http"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_install_plugins_context_free">
+<description>
+Frees a #GstInstallPluginsContext.
+
+Since: 0.10.12
+
+</description>
+<parameters>
+<parameter name="ctx">
+<parameter_description> a #GstInstallPluginsContext
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtp_buffer_validate">
+<description>
+Check if the data pointed to by @buffer is a valid RTP packet using
+gst_rtp_buffer_validate_data().
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer to validate
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @buffer is a valid RTP packet.
+</return>
+</function>
+
+<function name="gst_rtsp_message_get_body">
+<description>
+Get the body of @msg. @data remains valid for as long as @msg is valid and
+unchanged.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> location for the data
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> location for the size of @data
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_mixer_message_parse_option_changed">
+<description>
+Extracts the GstMixerOptions and new value from a option-changed bus notification
+message.
+
+The options and value returned remain valid until the message is freed.
+
+Since: 0.10.14
+
+</description>
+<parameters>
+<parameter name="message">
+<parameter_description> A volume-changed change notification message.
+</parameter_description>
+</parameter>
+<parameter name="options">
+<parameter_description> Pointer to hold a GstMixerOptions object, or NULL.
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> Result location to receive the new options value, or NULL.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_pb_utils_get_decoder_description">
+<description>
+Returns: a newly-allocated description string, or NULL on error. Free
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description> the (fixed) #GstCaps for which an decoder description is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated description string, or NULL on error. Free
+string with g_free() when not needed any longer.
+</return>
+</function>
+
+<function name="gst_sdp_message_set_key">
+<description>
+Adds the encryption information to @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="type">
+<parameter_description> the encryption type
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the encryption data
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtsp_message_free">
+<description>
+Free the memory used by @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_property_probe_needs_probe_name">
+<description>
+Same as gst_property_probe_needs_probe ().
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe object to which the given property belongs.
+</parameter_description>
+</parameter>
+<parameter name="name">
+<parameter_description> the name of the property to check.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the property needs a new probe, FALSE if not.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_media">
+<description>
+Get the media description at index @idx in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> the index
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPMedia.
+</return>
+</function>
+
+<function name="gst_rtsp_base64_encode">
+<description>
+Encode a sequence of binary data into its Base-64 stringified representation.
+
+
+</description>
+<parameters>
+<parameter name="data">
+<parameter_description> the binary data to encode
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the length of @data
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly allocated, zero-terminated Base-64 encoded string
+representing @data.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_get_ssrc">
+<description>
+Get the SSRC of the current SDES item.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> the SSRC of the current item.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_move_to_next">
+<description>
+Move the packet pointer @packet to the next packet in the payload.
+Use gst_rtcp_buffer_get_first_packet() to initialize @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @packet is pointing to a valid packet after calling this
+function.
+</return>
+</function>
+
+<function name="gst_netaddress_set_ip6_address">
+<description>
+Set @naddr with the IPv6 @address and @port pair.
+
+</description>
+<parameters>
+<parameter name="naddr">
+<parameter_description> a network address
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> an IPv6 network address.
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> a port number to set.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_ring_buffer_prepare_read">
+<description>
+Returns: FALSE if the buffer is not started.
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to read from
+</parameter_description>
+</parameter>
+<parameter name="segment">
+<parameter_description> the segment to read
+</parameter_description>
+</parameter>
+<parameter name="readptr">
+<parameter_description> the pointer to the memory where samples can be read
+</parameter_description>
+</parameter>
+<parameter name="len">
+<parameter_description> the number of bytes to read
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE if the buffer is not started.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_sdp_media_as_text">
+<description>
+Convert the contents of @media to a text string.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> A dynamically allocated string representing the media.
+</return>
+</function>
+
+<function name="gst_sdp_message_set_origin">
+<description>
+Configure the SDP origin in @msg with the given parameters.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="username">
+<parameter_description> the user name
+</parameter_description>
+</parameter>
+<parameter name="sess_id">
+<parameter_description> a session id
+</parameter_description>
+</parameter>
+<parameter name="sess_version">
+<parameter_description> a session version
+</parameter_description>
+</parameter>
+<parameter name="nettype">
+<parameter_description> a network type
+</parameter_description>
+</parameter>
+<parameter name="addrtype">
+<parameter_description> an address type
+</parameter_description>
+</parameter>
+<parameter name="addr">
+<parameter_description> an address
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_x_overlay_expose">
+<description>
+Tell an overlay that it has been exposed. This will redraw the current frame
+in the drawable even if the pipeline is PAUSED.
+
+</description>
+<parameters>
+<parameter name="overlay">
+<parameter_description> a #GstXOverlay to expose.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_tag_parse_extended_comment">
+<description>
+Convenience function to parse a GST_TAG_EXTENDED_COMMENT string and
+separate it into its components.
+
+If successful, @key, @lang and/or @value will be set to newly allocated
+strings that you need to free with g_free() when done. @key and @lang
+may also be set to NULL by this function if there is no key or no language
+code in the extended comment string.
+
+
+</description>
+<parameters>
+<parameter name="ext_comment">
+<parameter_description> an extended comment string, see #GST_TAG_EXTENDED_COMMENT
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> return location for the comment description key, or NULL
+</parameter_description>
+</parameter>
+<parameter name="lang">
+<parameter_description> return location for the comment ISO-639 language code, or NULL
+</parameter_description>
+</parameter>
+<parameter name="value">
+<parameter_description> return location for the actual comment string, or NULL
+</parameter_description>
+</parameter>
+<parameter name="fail_if_no_key">
+<parameter_description> whether to fail if strings are not in key=value form
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the string could be parsed, otherwise FALSE
+
+Since: 0.10.10
+</return>
+</function>
+
+<function name="gst_video_format_parse_caps">
+<description>
+Determines the #GstVideoFormat of @caps and places it in the location
+pointed to by @format. Extracts the size of the video and places it
+in the location pointed to by @width and @height. If @caps does not
+represent one of the raw video formats listed in #GstVideoFormat, the
+function will fail and return FALSE.
+
+Since: 0.10.16
+
+
+</description>
+<parameters>
+<parameter name="caps">
+<parameter_description> the #GstCaps to parse
+</parameter_description>
+</parameter>
+<parameter name="format">
+<parameter_description> the #GstVideoFormat of the video represented by @caps (output)
+</parameter_description>
+</parameter>
+<parameter name="width">
+<parameter_description> the width of the video represented by @caps (output)
+</parameter_description>
+</parameter>
+<parameter name="height">
+<parameter_description> the height of the video represented by @caps (output)
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @caps was parsed correctly.
+</return>
+</function>
+
+<function name="gst_base_audio_sink_get_slave_method">
+<description>
+Get the current slave method used by @sink.
+
+
+</description>
+<parameters>
+<parameter name="sink">
+<parameter_description> a #GstBaseAudioSink
+</parameter_description>
+</parameter>
+</parameters>
+<return> The current slave method used by @sink.
+
+Since: 0.10.16
+</return>
+</function>
+
+<function name="gst_mixer_get_option">
+<description>
+Get the current value of a name/value option in the mixer.
+
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> The #GstMixer (a #GstElement) that owns the optionlist.
+</parameter_description>
+</parameter>
+<parameter name="opts">
+<parameter_description> The #GstMixerOptions that we operate on.
+</parameter_description>
+</parameter>
+</parameters>
+<return> current value of the name/value option.
+</return>
+</function>
+
+<function name="gst_is_missing_plugin_message">
+<description>
+Checks whether @msg is a missing plugins message.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if @msg is a missing-plugins message, otherwise %FALSE.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_timestamp">
+<description>
+Get the timestamp of the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> The timestamp in host order.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_write">
+<description>
+Attempt to write @size bytes of @data to the connected @conn, blocking up to
+the specified @timeout. @timeout can be #NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to write
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of @data
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout value or #NULL
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_cdda_base_src_add_track">
+<description>
+CDDA sources use this function from their start vfunc to announce the
+available data and audio tracks to the base source class. The caller
+should allocate @track on the stack, the base source will do a shallow
+copy of the structure (and take ownership of the taglist if there is one).
+
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> a #GstCddaBaseSrc
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> address of #GstCddaBaseSrcTrack to add
+</parameter_description>
+</parameter>
+</parameters>
+<return> FALSE on error, otherwise TRUE.
+</return>
+</function>
+
+<function name="gst_property_probe_get_properties">
+<description>
+Get a list of properties for which probing is supported.
+
+
+</description>
+<parameters>
+<parameter name="probe">
+<parameter_description> the #GstPropertyProbe to get the properties for.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the list of properties for which probing is supported
+by this element.
+</return>
+</function>
+
+<function name="gst_rtsp_url_get_request_uri">
+<description>
+Get a newly allocated string describing the request URI for @url.
+
+
+</description>
+<parameters>
+<parameter name="url">
+<parameter_description> a #GstRTSPUrl
+</parameter_description>
+</parameter>
+</parameters>
+<return> a string with the request URI. g_free() after usage.
+</return>
+</function>
+
+<function name="gst_mixer_set_volume">
+<description>
+Sets the volume on each channel in a track. Short note about
+naming: a track is defined as one separate stream owned by
+the mixer/element, such as 'Line-in' or 'Microphone'. A
+channel is said to be a mono-stream inside this track. A
+stereo track thus contains two channels.
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> The #GstMixer (a #GstElement) that owns the track.
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> The #GstMixerTrack to set the volume on.
+</parameter_description>
+</parameter>
+<parameter name="volumes">
+<parameter_description> an array of integers (of size track-&gt;num_channels)
+that gives the wanted volume for each channel in
+this track.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_sdp_media_get_media">
+<description>
+Get the media description of @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the media description.
+</return>
+</function>
+
+<function name="gst_rtp_buffer_get_marker">
+<description>
+Check if the marker bit is set on the RTP packet in @buffer.
+
+
+</description>
+<parameters>
+<parameter name="buffer">
+<parameter_description> the buffer
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if @buffer has the marker bit set.
+</return>
+</function>
+
+<function name="gst_sdp_message_get_attribute_val">
+<description>
+Get the first attribute with key @key in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="key">
+<parameter_description> the key
+</parameter_description>
+</parameter>
+</parameters>
+<return> the attribute value of the first attribute with @key.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_attribute">
+<description>
+Get the attribute at position @idx in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> an index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstSDPAttribute at position @idx.
+</return>
+</function>
+
+<function name="gst_rtsp_method_as_text">
+<description>
+Convert @method to a string.
+
+
+</description>
+<parameters>
+<parameter name="method">
+<parameter_description> a #GstRTSPMethod
+</parameter_description>
+</parameter>
+</parameters>
+<return> a string representation of @method.
+</return>
+</function>
+
+<function name="gst_sdp_message_new">
+<description>
+Allocate a new GstSDPMessage and store the result in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> pointer to new #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_bye_add_ssrc">
+<description>
+Add @ssrc to the BYE @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="ssrc">
+<parameter_description> an SSRC to add
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE if the ssrc was added. This function can return %FALSE if
+the max MTU is exceeded or the number of sources blocks is greater than
+#GST_RTCP_MAX_BYE_SSRC_COUNT.
+</return>
+</function>
+
+<function name="gst_tuner_set_channel">
+<description>
+Tunes the object to the given channel.
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) that owns the channel.
+</parameter_description>
+</parameter>
+<parameter name="channel">
+<parameter_description> the channel to tune to.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_packet_bye_get_reason_len">
+<description>
+Get the length of the reason string.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> The length of the reason string or 0 when there is no reason string
+present.
+</return>
+</function>
+
+<function name="gst_missing_uri_source_message_new">
+<description>
+Creates a missing-plugin message for @element to notify the application
+that a source element for a particular URI protocol is missing. This
+function is mainly for use in plugins.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> the #GstElement posting the message
+</parameter_description>
+</parameter>
+<parameter name="protocol">
+<parameter_description> the URI protocol the missing source needs to implement,
+e.g. "http" or "mms"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstMessage, or NULL on error
+</return>
+</function>
+
+<function name="gst_netaddress_set_ip4_address">
+<description>
+Set @naddr with the IPv4 @address and @port pair.
+
+</description>
+<parameters>
+<parameter name="naddr">
+<parameter_description> a network address
+</parameter_description>
+</parameter>
+<parameter name="address">
+<parameter_description> an IPv4 network address.
+</parameter_description>
+</parameter>
+<parameter name="port">
+<parameter_description> a port number to set.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_tuner_get_norm">
+<description>
+Get the current video norm from the given tuner object for the
+currently selected channel.
+
+
+</description>
+<parameters>
+<parameter name="tuner">
+<parameter_description> the #GstTuner (a #GstElement) to get the current norm from.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the current norm.
+</return>
+</function>
+
+<function name="gst_video_orientation_get_vflip">
+<description>
+Get the vertical flipping state (%TRUE for flipped) from the given object.
+
+Since: 0.10.11
+
+</description>
+<parameters>
+<parameter name="video_orientation">
+<parameter_description> #GstVideoOrientation interface of a #GstElement
+</parameter_description>
+</parameter>
+<parameter name="flip">
+<parameter_description> return location for the result
+</parameter_description>
+</parameter>
+</parameters>
+<return> %TRUE in case the element supports flipping
+</return>
+</function>
+
+<function name="gst_sdp_media_get_connection">
+<description>
+Get the connection at position @idx in @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="idx">
+<parameter_description> an index
+</parameter_description>
+</parameter>
+</parameters>
+<return> the #GstSDPConnection at position @idx.
+</return>
+</function>
+
+<function name="gst_fft_f64_fft">
+<description>
+This performs the FFT on @timedata and puts the result in @freqdata.
+
+ timedata must have as many samples as specified with the @len parameter while
+allocating the #GstFFTF64 instance with gst_fft_f64_new().
+
+ freqdata must be large enough to hold @len/2 + 1 #GstFFTF64Complex frequency
+domain samples.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF64 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Buffer of the samples in the time domain
+</parameter_description>
+</parameter>
+<parameter name="freqdata">
+<parameter_description> Target buffer for the samples in the frequency domain
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_message_init_request">
+<description>
+Initialize @msg as a request message with @method and @uri. To clear @msg
+again, use gst_rtsp_message_unset().
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="method">
+<parameter_description> the request method to use
+</parameter_description>
+</parameter>
+<parameter name="uri">
+<parameter_description> the uri of the request
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_ring_buffer_pause">
+<description>
+Pause processing samples from the ringbuffer.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to pause
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if the device could be paused, FALSE on error.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_audio_filter_class_add_pad_templates">
+<description>
+Convenience function to add pad templates to this element class, with
+ allowed_caps as the caps that can be handled.
+
+This function is usually used from within a GObject base_init function.
+
+Since: 0.10.12
+
+</description>
+<parameters>
+<parameter name="klass">
+<parameter_description> an #GstAudioFilterClass
+</parameter_description>
+</parameter>
+<parameter name="allowed_caps">
+<parameter_description> what formats the filter can handle, as #GstCaps
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtcp_packet_sdes_next_item">
+<description>
+Move to the next SDES item in @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid SDES #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if there was a next item.
+</return>
+</function>
+
+<function name="gst_fft_s32_window">
+<description>
+This calls the window function @window on the @timedata sample buffer.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTS32 instance for this call
+</parameter_description>
+</parameter>
+<parameter name="timedata">
+<parameter_description> Time domain samples
+</parameter_description>
+</parameter>
+<parameter name="window">
+<parameter_description> Window function to apply
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_missing_element_message_new">
+<description>
+Creates a missing-plugin message for @element to notify the application
+that a certain required element is missing. This function is mainly for
+use in plugins.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> the #GstElement posting the message
+</parameter_description>
+</parameter>
+<parameter name="factory_name">
+<parameter_description> the name of the missing element (element factory),
+e.g. "videoscale" or "cdparanoiasrc"
+</parameter_description>
+</parameter>
+</parameters>
+<return> a new #GstMessage, or NULL on error
+</return>
+</function>
+
+<function name="gst_mixer_message_parse_volume_changed">
+<description>
+Parses a volume-changed notification message and extracts the track object
+it refers to, as well as an array of volumes and the size of the volumes array.
+
+The track object remains valid until the message is freed.
+
+The caller must free the array returned in the volumes parameter using g_free
+when they are done with it.
+
+Since: 0.10.14
+
+</description>
+<parameters>
+<parameter name="message">
+<parameter_description> A volume-changed change notification message.
+</parameter_description>
+</parameter>
+<parameter name="track">
+<parameter_description> Pointer to hold a GstMixerTrack object, or NULL.
+</parameter_description>
+</parameter>
+<parameter name="volumes">
+<parameter_description> A pointer to receive an array of gint values, or NULL.
+</parameter_description>
+</parameter>
+<parameter name="num_channels">
+<parameter_description> Result location to receive the number of channels, or NULL.
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_missing_decoder_installer_detail_new">
+<description>
+Returns: a newly-allocated detail string, or NULL on error. Free string
+
+</description>
+<parameters>
+<parameter name="decode_caps">
+<parameter_description> the (fixed) caps for which a decoder element is needed
+</parameter_description>
+</parameter>
+</parameters>
+<return> a newly-allocated detail string, or NULL on error. Free string
+with g_free() when not needed any longer.
+
+Since: 0.10.15
+</return>
+</function>
+
+<function name="gst_sdp_message_as_text">
+<description>
+Convert the contents of @msg to a text string.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> A dynamically allocated string representing the SDP description.
+</return>
+</function>
+
+<function name="gst_rtsp_message_append_headers">
+<description>
+Append the currently configured headers in @msg to the #GString @str suitable
+for transmission.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstRTSPMessage
+</parameter_description>
+</parameter>
+<parameter name="str">
+<parameter_description> a string
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_sdp_message_add_media">
+<description>
+Adds @media to the array of medias in @msg. This function takes ownership of
+the contents of @media so that @media will have to be reinitialized with
+gst_media_init() before it can be used again.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia to add
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstSDPResult.
+</return>
+</function>
+
+<function name="gst_mixer_get_mixer_flags">
+<description>
+Get the set of supported flags for this mixer implementation.
+
+
+</description>
+<parameters>
+<parameter name="mixer">
+<parameter_description> The #GstMixer implementation
+</parameter_description>
+</parameter>
+</parameters>
+<return> A set of or-ed GstMixerFlags for supported features.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_next_timeout">
+<description>
+Calculate the next timeout for @conn, storing the result in @timeout.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_get_padding">
+<description>
+Get the packet padding of the packet pointed to by @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid #GstRTCPPacket
+</parameter_description>
+</parameter>
+</parameters>
+<return> If the packet has the padding bit set.
+</return>
+</function>
+
+<function name="gst_sdp_media_add_format">
+<description>
+Add the format information to @media.
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+<parameter name="format">
+<parameter_description> the format
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_SDP_OK.
+</return>
+</function>
+
+<function name="gst_rtsp_url_parse">
+<description>
+Parse the RTSP @urlstr into a newly allocated #GstRTSPUrl. Free after usage
+with gst_rtsp_url_free().
+
+
+</description>
+<parameters>
+<parameter name="urlstr">
+<parameter_description> the url string to parse
+</parameter_description>
+</parameter>
+<parameter name="url">
+<parameter_description> location to hold the result.
+</parameter_description>
+</parameter>
+</parameters>
+<return> a #GstRTSPResult.
+</return>
+</function>
+
+<function name="gst_tag_from_vorbis_tag">
+<description>
+Looks up the GStreamer tag for a vorbiscomment tag.
+
+
+</description>
+<parameters>
+<parameter name="vorbis_tag">
+<parameter_description> vorbiscomment tag to convert to GStreamer tag
+</parameter_description>
+</parameter>
+</parameters>
+<return> The corresponding GStreamer tag or NULL if none exists.
+</return>
+</function>
+
+<function name="gst_riff_parse_strh">
+<description>
+Parses a strh structure from input data. Takes ownership of @buf.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging/error).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input data to be used for parsing, stripped from header.
+</parameter_description>
+</parameter>
+<parameter name="strh">
+<parameter_description> a pointer (returned by this function) to a filled-in
+strh structure. Caller should free it.
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if parsing succeeded, otherwise FALSE. The stream
+should be skipped on error, but it is not fatal.
+</return>
+</function>
+
+<function name="gst_ring_buffer_commit_full">
+<description>
+Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
+
+ in_samples and @out_samples define the rate conversion to perform on the the
+samples in @data. For negative rates, @out_samples must be negative and
+ in_samples positive.
+
+When @out_samples is positive, the first sample will be written at position @sample
+in the ringbuffer. When @out_samples is negative, the last sample will be written to
+ sample in reverse order.
+
+ out_samples does not need to be a multiple of the segment size of the ringbuffer
+although it is recommended for optimal performance.
+
+ accum will hold a temporary accumulator used in rate conversion and should be
+set to 0 when this function is first called. In case the commit operation is
+interrupted, one can resume the processing by passing the previously returned
+ accum value back to this function.
+
+
+</description>
+<parameters>
+<parameter name="buf">
+<parameter_description> the #GstRingBuffer to commit
+</parameter_description>
+</parameter>
+<parameter name="sample">
+<parameter_description> the sample position of the data
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to commit
+</parameter_description>
+</parameter>
+<parameter name="in_samples">
+<parameter_description> the number of samples in the data to commit
+</parameter_description>
+</parameter>
+<parameter name="out_samples">
+<parameter_description> the number of samples to write to the ringbuffer
+</parameter_description>
+</parameter>
+<parameter name="accum">
+<parameter_description> accumulator for rate conversion.
+</parameter_description>
+</parameter>
+</parameters>
+<return> The number of samples written to the ringbuffer or -1 on error. The
+number of samples written can be less than @out_samples when @buf was interrupted
+with a flush or stop.
+
+Since: 0.10.11.
+
+MT safe.
+</return>
+</function>
+
+<function name="gst_sdp_message_zones_len">
+<description>
+Get the number of time zone information entries in @msg.
+
+
+</description>
+<parameters>
+<parameter name="msg">
+<parameter_description> a #GstSDPMessage
+</parameter_description>
+</parameter>
+</parameters>
+<return> the number of time zone information entries in @msg.
+</return>
+</function>
+
+<function name="gst_rtsp_connection_read_internal">
+<description>
+Attempt to read @size bytes into @data from the connected @conn, blocking up to
+the specified @timeout. @timeout can be #NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush() only if
+ allow_interrupt is set.
+
+
+</description>
+<parameters>
+<parameter name="conn">
+<parameter_description> a #GstRTSPConnection
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> the data to read
+</parameter_description>
+</parameter>
+<parameter name="size">
+<parameter_description> the size of @data
+</parameter_description>
+</parameter>
+<parameter name="timeout">
+<parameter_description> a timeout value or #NULL
+</parameter_description>
+</parameter>
+<parameter name="allow_interrupt">
+<parameter_description> can the pending read be interrupted
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK on success.
+</return>
+</function>
+
+<function name="gst_base_audio_src_create_ringbuffer">
+<description>
+Create and return the #GstRingBuffer for @src. This function will call the
+::create_ringbuffer vmethod and will set @src as the parent of the returned
+buffer (see gst_object_set_parent()).
+
+
+</description>
+<parameters>
+<parameter name="src">
+<parameter_description> a #GstBaseAudioSrc.
+</parameter_description>
+</parameter>
+</parameters>
+<return> The new ringbuffer of @src.
+</return>
+</function>
+
+<function name="gst_sdp_media_get_proto">
+<description>
+Get the transport protocol of @media
+
+
+</description>
+<parameters>
+<parameter name="media">
+<parameter_description> a #GstSDPMedia
+</parameter_description>
+</parameter>
+</parameters>
+<return> the transport protocol of @media.
+</return>
+</function>
+
+<function name="gst_rtcp_packet_bye_get_nth_ssrc">
+<description>
+Get the @nth SSRC of the BYE @packet.
+
+
+</description>
+<parameters>
+<parameter name="packet">
+<parameter_description> a valid BYE #GstRTCPPacket
+</parameter_description>
+</parameter>
+<parameter name="nth">
+<parameter_description> the nth SSRC to get
+</parameter_description>
+</parameter>
+</parameters>
+<return> The @nth SSRC of @packet.
+</return>
+</function>
+
+<function name="gst_riff_parse_chunk">
+<description>
+Reads a single chunk.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input buffer.
+</parameter_description>
+</parameter>
+<parameter name="offset">
+<parameter_description> offset in the buffer in the caller. Is incremented
+by the read size by this function.
+</parameter_description>
+</parameter>
+<parameter name="fourcc">
+<parameter_description> fourcc (returned by this function0 of the chunk.
+</parameter_description>
+</parameter>
+<parameter name="chunk_data">
+<parameter_description> buffer (returned by the function) containing the
+chunk data.
+</parameter_description>
+</parameter>
+</parameters>
+<return> the fourcc tag of this chunk, or FALSE on error
+</return>
+</function>
+
+<function name="gst_riff_parse_strf_vids">
+<description>
+Parses a video streamÂs strf structure plus optionally some
+extradata from input data. This function takes ownership of @buf.
+
+
+</description>
+<parameters>
+<parameter name="element">
+<parameter_description> caller element (used for debugging/error).
+</parameter_description>
+</parameter>
+<parameter name="buf">
+<parameter_description> input data to be used for parsing, stripped from header.
+</parameter_description>
+</parameter>
+<parameter name="strf">
+<parameter_description> a pointer (returned by this function) to a filled-in
+strf/vids structure. Caller should free it.
+</parameter_description>
+</parameter>
+<parameter name="data">
+<parameter_description> a pointer (returned by this function) to a buffer
+containing extradata for this particular stream (e.g.
+palette, codec initialization data).
+</parameter_description>
+</parameter>
+</parameters>
+<return> TRUE if parsing succeeded, otherwise FALSE. The stream
+should be skipped on error, but it is not fatal.
+</return>
+</function>
+
+<function name="gst_fft_f64_free">
+<description>
+This frees the memory allocated for @self.
+
+
+</description>
+<parameters>
+<parameter name="self">
+<parameter_description> #GstFFTF64 instance for this call
+</parameter_description>
+</parameter>
+</parameters>
+<return></return>
+</function>
+
+<function name="gst_rtsp_transport_get_mime">
+<description>
+Get the mime type of the transport mode @trans. This mime type is typically
+used to generate #GstCaps on buffers.
+
+
+</description>
+<parameters>
+<parameter name="trans">
+<parameter_description> a #GstRTSPTransMode
+</parameter_description>
+</parameter>
+<parameter name="mime">
+<parameter_description> location to hold the result
+</parameter_description>
+</parameter>
+</parameters>
+<return> #GST_RTSP_OK.
+</return>
+</function>
+
</root>
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