If enabled, the payloader will automatically try to enable all the RTP header extensions provided in the src caps, saving the application the need to handle these extensions manually using the GstRTPBasePayload::request-extension: signal.
the time of the clock right before the element is set to
PLAYING. Subtracting base_time
from the current clock time in the PLAYING
state will yield the running_time against the clock.
the bus of the element. This bus is provided to the element by the parent element or the application. A #GstPipeline has a bus of its own.
the clock of the element. This clock is usually provided to the element by the toplevel #GstPipeline.
list of contexts
the current state of an element
flags for this object
the last return value of an element state change
Minimum duration of the packet data in ns (can't go above MTU)
The name of the object
the next state of an element, can be #GST_STATE_VOID_PENDING if the element is in the correct state.
number of pads of the element, includes both source and sink pads.
number of sink pads of the element.
number of source pads of the element.
Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec.
list of pads
updated whenever the a pad is added or removed
this object's parent, weak ref
the final state the element should go to, can be #GST_STATE_VOID_PENDING if the element is in the correct state
Try to use the offset fields to generate perfect RTP timestamps. When this option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of each payloaded buffer. The PTSes of buffers may not necessarily increment with the amount of data in each input buffer, consider e.g. the case where the buffer arrives from a network which means that the PTS is unrelated to the amount of data. Because the RTP timestamps are generated from GST_BUFFER_PTS this can result in RTP timestamps that also don't increment with the amount of data in the payloaded packet. To circumvent this it is possible to set the perfect rtptime option enabled. When this option is enabled the payloader will increment the RTP timestamps based on GST_BUFFER_OFFSET which relates to the amount of data in each packet rather than the GST_BUFFER_PTS of each buffer and therefore the RTP timestamps will more closely correlate with the amount of data in each buffer. Currently GstRTPBasePayload is limited to handling perfect RTP timestamps for audio streams.
Force buffers to be multiples of this duration in ns (0 disables)
Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed).
Example: A server wants to allow streaming a recorded video in double speed but still have the timestamps correspond to the position in the video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled.
list of sink pads
Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's #GstRTPSourceMeta.
list of source pads
the running_time of the last PAUSED state
Used to signal completion of a state change
Used to detect concurrent execution of gst_element_set_state() and gst_element_get_state()
Used to serialize execution of gst_element_set_state()
Various payloader statistics retrieved atomically (and are therefore synchroized with each other), these can be used e.g. to generate an RTP-Info header. This property return a GstStructure named application/x-rtp-payload-stats containing the following fields relating to the last processed buffer and current state of the stream being payloaded:
clock-rate
:#G_TYPE_UINT, clock-rate of the streamrunning-time
:#G_TYPE_UINT64, running timeseqnum
:#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnumtimestamp
:#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestampssrc
:#G_TYPE_UINT, The SSRC in usept
:#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:ptseqnum-offset
:#G_TYPE_UINT, The current offset added to the seqnumtimestamp-offset
:#G_TYPE_UINT, The current offset added to the timestampthe target state of an element as set by the application
Abort the state change of the element. This function is used by elements that do asynchronous state changes and find out something is wrong.
This function should be called with the STATE_LOCK held.
MT safe.
Attach the #GstControlBinding to the object. If there already was a #GstControlBinding for this property it will be replaced.
The object's reference count will be incremented, and any floating reference will be removed (see gst_object_ref_sink())
the #GstControlBinding that should be used
Adds a pad (link point) to element
. pad'
s parent will be set to element;
see gst_object_set_parent() for refcounting information.
Pads are automatically activated when added in the PAUSED or PLAYING state.
The pad and the element should be unlocked when calling this function.
This function will emit the #GstElement::pad-added signal on the element.
Allocate a new #GstBuffer with enough data to hold an RTP packet with
minimum csrc_count
CSRCs, a payload length of payload_len
and padding of
pad_len
. If payload
has #GstRTPBasePayload:source-info %TRUE additional
CSRCs may be allocated and filled with RTP source information.
the length of the payload
the amount of padding
the minimum number of CSRC entries
Creates a binding between source_property
on source
and target_property
on target
.
Whenever the source_property
is changed the target_property
is
updated using the same value. For instance:
g_object_bind_property (action, "active", widget, "sensitive", 0);
Will result in the "sensitive" property of the widget #GObject instance to be updated with the same value of the "active" property of the action #GObject instance.
If flags
contains %G_BINDING_BIDIRECTIONAL then the binding will be mutual:
if target_property
on target
changes then the source_property
on source
will be updated as well.
The binding will automatically be removed when either the source
or the
target
instances are finalized. To remove the binding without affecting the
source
and the target
you can just call g_object_unref() on the returned
#GBinding instance.
Removing the binding by calling g_object_unref() on it must only be done if
the binding, source
and target
are only used from a single thread and it
is clear that both source
and target
outlive the binding. Especially it
is not safe to rely on this if the binding, source
or target
can be
finalized from different threads. Keep another reference to the binding and
use g_binding_unbind() instead to be on the safe side.
A #GObject can have multiple bindings.
the property on source
to bind
the target #GObject
the property on target
to bind
flags to pass to #GBinding
Creates a binding between source_property
on source
and target_property
on target,
allowing you to set the transformation functions to be used by
the binding.
This function is the language bindings friendly version of g_object_bind_property_full(), using #GClosures instead of function pointers.
the property on source
to bind
the target #GObject
the property on target
to bind
flags to pass to #GBinding
a #GClosure wrapping the transformation function from the source
to the target,
or %NULL to use the default
a #GClosure wrapping the transformation function from the target
to the source,
or %NULL to use the default
Calls func
from another thread and passes user_data
to it. This is to be
used for cases when a state change has to be performed from a streaming
thread, directly via gst_element_set_state() or indirectly e.g. via SEEK
events.
Calling those functions directly from the streaming thread will cause deadlocks in many situations, as they might involve waiting for the streaming thread to shut down from this very streaming thread.
MT safe.
Function to call asynchronously from another thread
Perform transition
on element
.
This function must be called with STATE_LOCK held and is mainly used internally.
the requested transition
Commit the state change of the element and proceed to the next pending state if any. This function is used by elements that do asynchronous state changes. The core will normally call this method automatically when an element returned %GST_STATE_CHANGE_SUCCESS from the state change function.
If after calling this method the element still has not reached the pending state, the next state change is performed.
This method is used internally and should normally not be called by plugins or applications.
This function must be called with STATE_LOCK held.
The previous state return value
Creates a pad for each pad template that is always available. This function is only useful during object initialization of subclasses of #GstElement.
Create an RTP buffer and store payload_len
bytes of the adapter as the
payload. Set the timestamp on the new buffer to timestamp
before pushing
the buffer downstream.
If payload_len
is -1, all pending bytes will be flushed. If timestamp
is
-1, the timestamp will be calculated automatically.
length of payload
a #GstClockTime
This function is intended for #GObject implementations to re-enforce a [floating][floating-ref] object reference. Doing this is seldom required: all #GInitiallyUnowneds are created with a floating reference which usually just needs to be sunken by calling g_object_ref_sink().
Call func
with user_data
for each of element'
s pads. func
will be called
exactly once for each pad that exists at the time of this call, unless
one of the calls to func
returns %FALSE in which case we will stop
iterating pads and return early. If new pads are added or pads are removed
while pads are being iterated, this will not be taken into account until
next time this function is used.
function to call for each pad
Call func
with user_data
for each of element'
s sink pads. func
will be
called exactly once for each sink pad that exists at the time of this call,
unless one of the calls to func
returns %FALSE in which case we will stop
iterating pads and return early. If new sink pads are added or sink pads
are removed while the sink pads are being iterated, this will not be taken
into account until next time this function is used.
function to call for each sink pad
Call func
with user_data
for each of element'
s source pads. func
will be
called exactly once for each source pad that exists at the time of this call,
unless one of the calls to func
returns %FALSE in which case we will stop
iterating pads and return early. If new source pads are added or source pads
are removed while the source pads are being iterated, this will not be taken
into account until next time this function is used.
function to call for each source pad
Increases the freeze count on object
. If the freeze count is
non-zero, the emission of "notify" signals on object
is
stopped. The signals are queued until the freeze count is decreased
to zero. Duplicate notifications are squashed so that at most one
#GObject::notify signal is emitted for each property modified while the
object is frozen.
This is necessary for accessors that modify multiple properties to prevent premature notification while the object is still being modified.
Returns the base time of the element. The base time is the absolute time of the clock when this element was last put to PLAYING. Subtracting the base time from the clock time gives the running time of the element.
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.
This function will first attempt to find a compatible unlinked ALWAYS pad,
and if none can be found, it will request a compatible REQUEST pad by looking
at the templates of element
.
the #GstPad to find a compatible one for.
the #GstCaps to use as a filter.
Retrieves a pad template from element
that is compatible with compattempl
.
Pads from compatible templates can be linked together.
the #GstPadTemplate to find a compatible template for
Gets the corresponding #GstControlBinding for the property. This should be unreferenced again after use.
name of the property
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() in between. 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.
Returns the current clock time of the element, as in, the time of the element's clock, or GST_CLOCK_TIME_NONE if there is no clock.
Returns the running time of the element. The running time is the element's clock time minus its base time. Will return GST_CLOCK_TIME_NONE if the element has no clock, or if its base time has not been set.
Gets a named field from the objects table of associations (see g_object_set_data()).
name of the key for that association
Retrieves the factory that was used to create this element.
Gets a number of #GValues for the given controlled property starting at the
requested time. The array values
need to hold enough space for n_values
of
#GValue.
This function is useful if one wants to e.g. draw a graph of the control curve or apply a control curve sample by sample.
the name of the property to get
the time that should be processed
the time spacing between subsequent values
array to put control-values in
Get metadata with key
in klass
.
the key to get
Returns a copy of the name of object
.
Caller should g_free() the return value after usage.
For a nameless object, this returns %NULL, which you can safely g_free()
as well.
Free-function: g_free
Retrieves a padtemplate from element
with the given name.
the name of the #GstPadTemplate to get.
Retrieves a list of the pad templates associated with element
. The
list must not be modified by the calling code.
Generates a string describing the path of object
in
the object hierarchy. Only useful (or used) for debugging.
Free-function: g_free
Gets a property of an object.
The value
can be:
In general, a copy is made of the property contents and the caller is responsible for freeing the memory by calling g_value_unset().
Note that g_object_get_property() is really intended for language bindings, g_object_get() is much more convenient for C programming.
the name of the property to get
return location for the property value
This function gets back user data pointers stored via g_object_set_qdata().
A #GQuark, naming the user data pointer
Count the total number of RTP sources found in the meta of buffer,
which
will be automically added by gst_rtp_base_payload_allocate_output_buffer().
If #GstRTPBasePayload:source-info is %FALSE the count will be 0.
Returns the start time of the element. The start time is the running time of the clock when this element was last put to PAUSED.
Usually the start_time is managed by a toplevel element such as #GstPipeline.
MT safe.
Gets the state of the element.
For elements that performed an ASYNC state change, as reported by gst_element_set_state(), this function will block up to the specified timeout value for the state change to complete. If the element completes the state change or goes into an error, this function returns immediately with a return value of %GST_STATE_CHANGE_SUCCESS or %GST_STATE_CHANGE_FAILURE respectively.
For elements that did not return %GST_STATE_CHANGE_ASYNC, this function returns the current and pending state immediately.
This function returns %GST_STATE_CHANGE_NO_PREROLL if the element successfully changed its state but is not able to provide data yet. This mostly happens for live sources that only produce data in %GST_STATE_PLAYING. While the state change return is equivalent to %GST_STATE_CHANGE_SUCCESS, it is returned to the application to signal that some sink elements might not be able to complete their state change because an element is not producing data to complete the preroll. When setting the element to playing, the preroll will complete and playback will start.
a #GstClockTime to specify the timeout for an async state change or %GST_CLOCK_TIME_NONE for infinite timeout.
Gets the value for the given controlled property at the requested time.
the name of the property to get
the time the control-change should be read from
Gets n_properties
properties for an object
.
Obtained properties will be set to values
. All properties must be valid.
Warnings will be emitted and undefined behaviour may result if invalid
properties are passed in.
the names of each property to get
the values of each property to get
Check if the object
has active controlled properties.
Check if the packet with size
and duration
would exceed the configured
maximum size.
the size of the packet
the duration of the packet
Checks whether object
has a [floating][floating-ref] reference.
Checks if the state of an element is locked. If the state of an element is locked, state changes of the parent don't affect the element. This way you can leave currently unused elements inside bins. Just lock their state before changing the state from #GST_STATE_NULL.
MT safe.
Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from #GstRTPSourceMeta.
Retrieves an iterator of element'
s pads. The iterator should
be freed after usage. Also more specialized iterators exists such as
gst_element_iterate_src_pads() or gst_element_iterate_sink_pads().
The order of pads returned by the iterator will be the order in which the pads were added to the element.
Links src
to dest
. The link must be from source to
destination; the other direction will not be tried. The function looks for
existing pads that aren't linked yet. It will request new pads if necessary.
Such pads need to be released manually when unlinking.
If multiple links are possible, only one is established.
Make sure you have added your elements to a bin or pipeline with gst_bin_add() before trying to link them.
Links src
to dest
using the given caps as filtercaps.
The link must be from source to
destination; the other direction will not be tried. The function looks for
existing pads that aren't linked yet. It will request new pads if necessary.
If multiple links are possible, only one is established.
Make sure you have added your elements to a bin or pipeline with gst_bin_add() before trying to link them.
the #GstElement containing the destination pad.
the #GstCaps to filter the link, or %NULL for no filter.
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.
the name of the #GstPad in source element or %NULL for any pad.
the #GstElement containing the destination pad.
the name of the #GstPad in destination element, or %NULL for any pad.
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
.
the name of the #GstPad in source element or %NULL for any pad.
the #GstElement containing the destination pad.
the name of the #GstPad in destination element or %NULL for any pad.
the #GstCaps to filter the link, or %NULL for no filter.
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.
Calling gst_element_link_pads_full() with flags
== %GST_PAD_LINK_CHECK_DEFAULT
is the same as calling gst_element_link_pads() and the recommended way of
linking pads with safety checks applied.
This is a convenience function for gst_pad_link_full().
the name of the #GstPad in source element or %NULL for any pad.
the #GstElement containing the destination pad.
the name of the #GstPad in destination element, or %NULL for any pad.
the #GstPadLinkCheck to be performed when linking pads.
Brings the element to the lost state. The current state of the element is copied to the pending state so that any call to gst_element_get_state() will return %GST_STATE_CHANGE_ASYNC.
An ASYNC_START message is posted. If the element was PLAYING, it will go to PAUSED. The element will be restored to its PLAYING state by the parent pipeline when it prerolls again.
This is mostly used for elements that lost their preroll buffer in the %GST_STATE_PAUSED or %GST_STATE_PLAYING state after a flush, they will go to their pending state again when a new preroll buffer is queued. This function can only be called when the element is currently not in error or an async state change.
This function is used internally and should normally not be called from plugins or applications.
Post an error, warning or info message on the bus from inside an element.
type
must be of #GST_MESSAGE_ERROR, #GST_MESSAGE_WARNING or
#GST_MESSAGE_INFO.
MT safe.
the #GstMessageType
the GStreamer GError domain this message belongs to
the GError code belonging to the domain
an allocated text string to be used as a replacement for the default message connected to code, or %NULL
an allocated debug message to be used as a replacement for the default debugging information, or %NULL
the source code file where the error was generated
the source code function where the error was generated
the source code line where the error was generated
Post an error, warning or info message on the bus from inside an element.
type
must be of #GST_MESSAGE_ERROR, #GST_MESSAGE_WARNING or
#GST_MESSAGE_INFO.
the #GstMessageType
the GStreamer GError domain this message belongs to
the GError code belonging to the domain
an allocated text string to be used as a replacement for the default message connected to code, or %NULL
an allocated debug message to be used as a replacement for the default debugging information, or %NULL
the source code file where the error was generated
the source code function where the error was generated
the source code line where the error was generated
optional details structure
Use this function to signal that the element does not expect any more pads to show up in the current pipeline. This function should be called whenever pads have been added by the element itself. Elements with #GST_PAD_SOMETIMES pad templates use this in combination with autopluggers to figure out that the element is done initializing its pads.
This function emits the #GstElement::no-more-pads signal.
MT safe.
Emits a "notify" signal for the property property_name
on object
.
When possible, eg. when signaling a property change from within the class that registered the property, you should use g_object_notify_by_pspec() instead.
Note that emission of the notify signal may be blocked with g_object_freeze_notify(). In this case, the signal emissions are queued and will be emitted (in reverse order) when g_object_thaw_notify() is called.
the name of a property installed on the class of object
.
Emits a "notify" signal for the property specified by pspec
on object
.
This function omits the property name lookup, hence it is faster than g_object_notify().
One way to avoid using g_object_notify() from within the class that registered the properties, and using g_object_notify_by_pspec() instead, is to store the GParamSpec used with g_object_class_install_property() inside a static array, e.g.:
enum
{
PROP_0,
PROP_FOO,
PROP_LAST
};
static GParamSpec *properties[PROP_LAST];
static void
my_object_class_init (MyObjectClass *klass)
{
properties[PROP_FOO] = g_param_spec_int ("foo", "Foo", "The foo",
0, 100,
50,
G_PARAM_READWRITE);
g_object_class_install_property (gobject_class,
PROP_FOO,
properties[PROP_FOO]);
}
and then notify a change on the "foo" property with:
g_object_notify_by_pspec (self, properties[PROP_FOO]);
the #GParamSpec of a property installed on the class of object
.
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.
data to set as payload
a #GstClockTime
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
.
Push list
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 list
.
a #GstBufferList
Performs a query on the given element.
For elements that don't implement a query handler, this function forwards the query to a random srcpad or to the peer of a random linked sinkpad of this element.
Please note that some queries might need a running pipeline to work.
Queries an element (usually top-level pipeline or playbin element) for the total stream duration in nanoseconds. This query will only work once the pipeline is prerolled (i.e. reached PAUSED or PLAYING state). The application will receive an ASYNC_DONE message on the pipeline bus when that is the case.
If the duration changes for some reason, you will get a DURATION_CHANGED message on the pipeline bus, in which case you should re-query the duration using this function.
Queries an element (usually top-level pipeline or playbin element) for the stream position in nanoseconds. This will be a value between 0 and the stream duration (if the stream duration is known). This query will usually only work once the pipeline is prerolled (i.e. reached PAUSED or PLAYING state). The application will receive an ASYNC_DONE message on the pipeline bus when that is the case.
If one repeatedly calls this function one can also create a query and reuse it in gst_element_query().
Increments the reference count on object
. This function
does not take the lock on object
because it relies on
atomic refcounting.
This object returns the input parameter to ease writing constructs like : result = gst_object_ref (object->parent);
Increase the reference count of object,
and possibly remove the
[floating][floating-ref] reference, if object
has a floating reference.
In other words, if the object is floating, then this call "assumes ownership" of the floating reference, converting it to a normal reference by clearing the floating flag while leaving the reference count unchanged. If the object is not floating, then this call adds a new normal reference increasing the reference count by one.
Since GLib 2.56, the type of object
will be propagated to the return type
under the same conditions as for g_object_ref().
Makes the element free the previously requested pad as obtained with gst_element_request_pad().
This does not unref the pad. If the pad was created by using
gst_element_request_pad(), gst_element_release_request_pad() needs to be
followed by gst_object_unref() to free the pad
.
MT safe.
Removes the corresponding #GstControlBinding. If it was the last ref of the binding, it will be disposed.
the binding
Removes pad
from element
. pad
will be destroyed if it has not been
referenced elsewhere using gst_object_unparent().
This function is used by plugin developers and should not be used by applications. Pads that were dynamically requested from elements with gst_element_request_pad() should be released with the gst_element_release_request_pad() function instead.
Pads are not automatically deactivated so elements should perform the needed steps to deactivate the pad in case this pad is removed in the PAUSED or PLAYING state. See gst_pad_set_active() for more information about deactivating pads.
The pad and the element should be unlocked when calling this function.
This function will emit the #GstElement::pad-removed signal on the element.
Retrieves a request pad from the element according to the provided template. Pad templates can be looked up using gst_element_factory_get_static_pad_templates().
The pad should be released with gst_element_release_request_pad().
a #GstPadTemplate of which we want a pad of.
the name of the request #GstPad to retrieve. Can be %NULL.
the caps of the pad we want to request. Can be %NULL.
Retrieves a pad from the element by name (e.g. "src_%d"). This version only retrieves request pads. The pad should be released with gst_element_release_request_pad().
This method is slower than manually getting the pad template and calling
gst_element_request_pad() if the pads should have a specific name (e.g.
name
is "src_1" instead of "src_%u").
Note that this function was introduced in GStreamer 1.20 in order to provide a better name to gst_element_get_request_pad(). Prior to 1.20, users should use gst_element_get_request_pad() which provides the same functionality.
the name of the request #GstPad to retrieve.
Releases all references to other objects. This can be used to break reference cycles.
This function should only be called from object system implementations.
Sends a seek event to an element. See gst_event_new_seek() for the details of the parameters. The seek event is sent to the element using gst_element_send_event().
MT safe.
The new playback rate
The format of the seek values
The optional seek flags.
The type and flags for the new start position
The value of the new start position
The type and flags for the new stop position
The value of the new stop position
Simple API to perform a seek on the given element, meaning it just seeks to the given position relative to the start of the stream. For more complex operations like segment seeks (e.g. for looping) or changing the playback rate or seeking relative to the last configured playback segment you should use gst_element_seek().
In a completely prerolled PAUSED or PLAYING pipeline, seeking is always guaranteed to return %TRUE on a seekable media type or %FALSE when the media type is certainly not seekable (such as a live stream).
Some elements allow for seeking in the READY state, in this case they will store the seek event and execute it when they are put to PAUSED. If the element supports seek in READY, it will always return %TRUE when it receives the event in the READY state.
a #GstFormat to execute the seek in, such as #GST_FORMAT_TIME
seek options; playback applications will usually want to use GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT here
position to seek to (relative to the start); if you are doing a seek in #GST_FORMAT_TIME this value is in nanoseconds - multiply with #GST_SECOND to convert seconds to nanoseconds or with #GST_MSECOND to convert milliseconds to nanoseconds.
Sends an event to an element. If the element doesn't implement an event handler, the event will be pushed on a random linked sink pad for downstream events or a random linked source pad for upstream events.
This function takes ownership of the provided event so you should gst_event_ref() it if you want to reuse the event after this call.
MT safe.
Set the base time of an element. See gst_element_get_base_time().
MT safe.
the base time to set.
This function is used to disable the control bindings on a property for some time, i.e. gst_object_sync_values() will do nothing for the property.
property to disable
boolean that specifies whether to disable the controller or not.
This function is used to disable all controlled properties of the object
for
some time, i.e. gst_object_sync_values() will do nothing.
boolean that specifies whether to disable the controller or not.
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() in between. 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.
the new control-rate in nanoseconds.
Each object carries around a table of associations from strings to pointers. This function lets you set an association.
If the object already had an association with that name, the old association will be destroyed.
Internally, the key
is converted to a #GQuark using g_quark_from_string().
This means a copy of key
is kept permanently (even after object
has been
finalized) — so it is recommended to only use a small, bounded set of values
for key
in your program, to avoid the #GQuark storage growing unbounded.
name of the key
data to associate with that key
Tells #GstRTPBaseAudioPayload that the child element is for a frame based audio codec
Sets the options for frame based audio codecs.
The duraction of an audio frame in milliseconds.
The size of an audio frame in bytes.
Locks the state of an element, so state changes of the parent don't affect this element anymore.
Note that this is racy if the state lock of the parent bin is not taken. The parent bin might've just checked the flag in another thread and as the next step proceed to change the child element's state.
MT safe.
%TRUE to lock the element's state
Sets the name of object,
or gives object
a guaranteed unique
name (if name
is %NULL).
This function makes a copy of the provided name, so the caller
retains ownership of the name it sent.
new name of object
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_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps().
the media type (typically "audio" or "video")
if the payload type is dynamic
the encoding name
the clock rate of the media
Sets a property on an object.
the name of the property to set
the value
Tells #GstRTPBaseAudioPayload that the child element is for a sample based audio codec
Sets the options for sample based audio codecs.
Size per sample in bytes.
Sets the options for sample based audio codecs.
Size per sample in bits.
Enable or disable adding contributing sources to RTP packets from #GstRTPSourceMeta.
whether to add contributing sources to RTP packets
Set the start time of an element. The start time of the element is the running time of the element when it last went to the PAUSED state. In READY or after a flushing seek, it is set to 0.
Toplevel elements like #GstPipeline will manage the start_time and base_time on its children. Setting the start_time to #GST_CLOCK_TIME_NONE on such a toplevel element will disable the distribution of the base_time to the children and can be useful if the application manages the base_time itself, for example if you want to synchronize capture from multiple pipelines, and you can also ensure that the pipelines have the same clock.
MT safe.
the base time to set.
Sets the state of the element. This function will try to set the requested state by going through all the intermediary states and calling the class's state change function for each.
This function can return #GST_STATE_CHANGE_ASYNC, in which case the element will perform the remainder of the state change asynchronously in another thread. An application can use gst_element_get_state() to wait for the completion of the state change or it can wait for a %GST_MESSAGE_ASYNC_DONE or %GST_MESSAGE_STATE_CHANGED on the bus.
State changes to %GST_STATE_READY or %GST_STATE_NULL never return #GST_STATE_CHANGE_ASYNC.
Remove a specified datum from the object's data associations, without invoking the association's destroy handler.
name of the key
This function gets back user data pointers stored via
g_object_set_qdata() and removes the data
from object
without invoking its destroy() function (if any was
set).
Usually, calling this function is only required to update
user data pointers with a destroy notifier, for example:
void
object_add_to_user_list (GObject *object,
const gchar *new_string)
{
// the quark, naming the object data
GQuark quark_string_list = g_quark_from_static_string ("my-string-list");
// retrieve the old string list
GList *list = g_object_steal_qdata (object, quark_string_list);
// prepend new string
list = g_list_prepend (list, g_strdup (new_string));
// this changed 'list', so we need to set it again
g_object_set_qdata_full (object, quark_string_list, list, free_string_list);
}
static void
free_string_list (gpointer data)
{
GList *node, *list = data;
for (node = list; node; node = node->next)
g_free (node->data);
g_list_free (list);
}
Using g_object_get_qdata() in the above example, instead of g_object_steal_qdata() would have left the destroy function set, and thus the partial string list would have been freed upon g_object_set_qdata_full().
A #GQuark, naming the user data pointer
Returns a suggestion for timestamps where buffers should be split to get best controller results.
Tries to change the state of the element to the same as its parent. If this function returns %FALSE, the state of element is undefined.
Sets the properties of the object, according to the #GstControlSources that (maybe) handle them and for the given timestamp.
If this function fails, it is most likely the application developers fault. Most probably the control sources are not setup correctly.
the time that should be processed
Reverts the effect of a previous call to
g_object_freeze_notify(). The freeze count is decreased on object
and when it reaches zero, queued "notify" signals are emitted.
Duplicate notifications for each property are squashed so that at most one #GObject::notify signal is emitted for each property, in the reverse order in which they have been queued.
It is an error to call this function when the freeze count is zero.
Unlinks all source pads of the source element with all sink pads of the sink element to which they are linked.
If the link has been made using gst_element_link(), it could have created an requestpad, which has to be released using gst_element_release_request_pad().
Unlinks the two named pads of the source and destination elements.
This is a convenience function for gst_pad_unlink().
the name of the #GstPad in source element.
a #GstElement containing the destination pad.
the name of the #GstPad in destination element.
Clear the parent of object,
removing the associated reference.
This function decreases the refcount of object
.
MT safe. Grabs and releases object'
s lock.
Decrements the reference count on object
. If reference count hits
zero, destroy object
. This function does not take the lock
on object
as it relies on atomic refcounting.
The unref method should never be called with the LOCK held since this might deadlock the dispose function.
Perform transition
on element
.
This function must be called with STATE_LOCK held and is mainly used internally.
the requested transition
Gets the state of the element.
For elements that performed an ASYNC state change, as reported by gst_element_set_state(), this function will block up to the specified timeout value for the state change to complete. If the element completes the state change or goes into an error, this function returns immediately with a return value of %GST_STATE_CHANGE_SUCCESS or %GST_STATE_CHANGE_FAILURE respectively.
For elements that did not return %GST_STATE_CHANGE_ASYNC, this function returns the current and pending state immediately.
This function returns %GST_STATE_CHANGE_NO_PREROLL if the element successfully changed its state but is not able to provide data yet. This mostly happens for live sources that only produce data in %GST_STATE_PLAYING. While the state change return is equivalent to %GST_STATE_CHANGE_SUCCESS, it is returned to the application to signal that some sink elements might not be able to complete their state change because an element is not producing data to complete the preroll. When setting the element to playing, the preroll will complete and playback will start.
a #GstClockTime to specify the timeout for an async state change or %GST_CLOCK_TIME_NONE for infinite timeout.
Use this function to signal that the element does not expect any more pads to show up in the current pipeline. This function should be called whenever pads have been added by the element itself. Elements with #GST_PAD_SOMETIMES pad templates use this in combination with autopluggers to figure out that the element is done initializing its pads.
This function emits the #GstElement::no-more-pads signal.
MT safe.
Emits a "notify" signal for the property property_name
on object
.
When possible, eg. when signaling a property change from within the class that registered the property, you should use g_object_notify_by_pspec() instead.
Note that emission of the notify signal may be blocked with g_object_freeze_notify(). In this case, the signal emissions are queued and will be emitted (in reverse order) when g_object_thaw_notify() is called.
Performs a query on the given element.
For elements that don't implement a query handler, this function forwards the query to a random srcpad or to the peer of a random linked sinkpad of this element.
Please note that some queries might need a running pipeline to work.
Retrieves a request pad from the element according to the provided template. Pad templates can be looked up using gst_element_factory_get_static_pad_templates().
The pad should be released with gst_element_release_request_pad().
a #GstPadTemplate of which we want a pad of.
the name of the request #GstPad to retrieve. Can be %NULL.
the caps of the pad we want to request. Can be %NULL.
Sends an event to an element. If the element doesn't implement an event handler, the event will be pushed on a random linked sink pad for downstream events or a random linked source pad for upstream events.
This function takes ownership of the provided event so you should gst_event_ref() it if you want to reuse the event after this call.
MT safe.
Sets the state of the element. This function will try to set the requested state by going through all the intermediary states and calling the class's state change function for each.
This function can return #GST_STATE_CHANGE_ASYNC, in which case the element will perform the remainder of the state change asynchronously in another thread. An application can use gst_element_get_state() to wait for the completion of the state change or it can wait for a %GST_MESSAGE_ASYNC_DONE or %GST_MESSAGE_STATE_CHANGED on the bus.
State changes to %GST_STATE_READY or %GST_STATE_NULL never return #GST_STATE_CHANGE_ASYNC.
This function essentially limits the life time of the closure
to
the life time of the object. That is, when the object is finalized,
the closure
is invalidated by calling g_closure_invalidate() on
it, in order to prevent invocations of the closure with a finalized
(nonexisting) object. Also, g_object_ref() and g_object_unref() are
added as marshal guards to the closure,
to ensure that an extra
reference count is held on object
during invocation of the
closure
. Usually, this function will be called on closures that
use this object
as closure data.
#GClosure to watch
Checks to see if there is any object named name
in list
. This function
does not do any locking of any kind. You might want to protect the
provided list with the lock of the owner of the list. This function
will lock each #GstObject in the list to compare the name, so be
careful when passing a list with a locked object.
A default deep_notify signal callback for an object. The user data should contain a pointer to an array of strings that should be excluded from the notify. The default handler will print the new value of the property using g_print.
MT safe. This function grabs and releases object'
s LOCK for getting its
path string.
the #GObject that signalled the notify.
a #GstObject that initiated the notify.
a #GParamSpec of the property.
a set of user-specified properties to exclude or %NULL to show all changes.
Find the #GParamSpec with the given name for an
interface. Generally, the interface vtable passed in as g_iface
will be the default vtable from g_type_default_interface_ref(), or,
if you know the interface has already been loaded,
g_type_default_interface_peek().
any interface vtable for the interface, or the default vtable for the interface
name of a property to look up.
Add a property to an interface; this is only useful for interfaces that are added to GObject-derived types. Adding a property to an interface forces all objects classes with that interface to have a compatible property. The compatible property could be a newly created #GParamSpec, but normally g_object_class_override_property() will be used so that the object class only needs to provide an implementation and inherits the property description, default value, bounds, and so forth from the interface property.
This function is meant to be called from the interface's default
vtable initialization function (the class_init
member of
#GTypeInfo.) It must not be called after after class_init
has
been called for any object types implementing this interface.
If pspec
is a floating reference, it will be consumed.
any interface vtable for the interface, or the default vtable for the interface.
the #GParamSpec for the new property
Lists the properties of an interface.Generally, the interface
vtable passed in as g_iface
will be the default vtable from
g_type_default_interface_ref(), or, if you know the interface has
already been loaded, g_type_default_interface_peek().
any interface vtable for the interface, or the default vtable for the interface
Creates a new instance of a #GObject subtype and sets its properties.
Construction parameters (see %G_PARAM_CONSTRUCT, %G_PARAM_CONSTRUCT_ONLY) which are not explicitly specified are set to their default values.
the type id of the #GObject subtype to instantiate
an array of #GParameter
Create a new elementfactory capable of instantiating objects of the
type
and add the factory to plugin
.
#GstPlugin to register the element with, or %NULL for a static element.
name of elements of this type
rank of element (higher rank means more importance when autoplugging)
GType of element to register
Atomically modifies a pointer to point to a new object.
The reference count of oldobj
is decreased and the reference count of
newobj
is increased.
Either newobj
and the value pointed to by oldobj
may be %NULL.
pointer to a place of a #GstObject to replace
a new #GstObject
Gets a string representing the given state change result.
a #GstStateChangeReturn to get the name of.
Marks type
as "documentation should be skipped".
Can be useful for dynamically registered element to be excluded from
plugin documentation system.
Example:
GType my_type;
GTypeInfo my_type_info;
// Fill "my_type_info"
...
my_type = g_type_register_static (GST_TYPE_MY_ELEMENT, "my-type-name",
&my_type_info, 0);
gst_element_type_set_skip_documentation (my_type);
gst_element_register (plugin, "my-plugin-feature-name", rank, my_type);
a #GType of element
Provides a base class for audio RTP payloaders for frame or sample based audio codecs (constant bitrate)
This class derives from GstRTPBasePayload. It can be used for payloading audio codecs. It will only work with constant bitrate codecs. It supports both frame based and sample based codecs. It takes care of packing up the audio data into RTP packets and filling up the headers accordingly. The payloading is done based on the maximum MTU (mtu) and the maximum time per packet (max-ptime). The general idea is to divide large data buffers into smaller RTP packets. The RTP packet size is the minimum of either the MTU, max-ptime (if set) or available data. The RTP packet size is always larger or equal to min-ptime (if set). If min-ptime is not set, any residual data is sent in a last RTP packet. In the case of frame based codecs, the resulting RTP packets always contain full frames.
Usage
To use this base class, your child element needs to call either gst_rtp_base_audio_payload_set_frame_based() or gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the element's
_init()
function. Then, the child element must call either gst_rtp_base_audio_payload_set_frame_options(), gst_rtp_base_audio_payload_set_sample_options() or gst_rtp_base_audio_payload_set_samplebits_options. Since GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element must set any variables or call/override any functions required by that base class. The child element does not need to override any other functions specific to GstRTPBaseAudioPayload.