<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.h">Buffer dropping scheme to avoid the element's internal queue to block when
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Appsink is a sink plugin that supports many different methods for making
the application get a handle on the GStreamer data in a pipeline. Unlike
most GStreamer elements, Appsink provides external API functions.
appsink can be used by linking to the gstappsink.h header file to access the
methods or by using the appsink action signals and properties.
The normal way of retrieving samples from appsink is by using the
gst_app_sink_pull_sample() and gst_app_sink_pull_preroll() methods.
These methods block until a sample becomes available in the sink or when the
sink is shut down or reaches EOS. There are also timed variants of these
methods, gst_app_sink_try_pull_sample() and gst_app_sink_try_pull_preroll(),
which accept a timeout parameter to limit the amount of time to wait.
Appsink will internally use a queue to collect buffers from the streaming
thread. If the application is not pulling samples fast enough, this queue
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state.
This function is typically used when dealing with a pipeline in the PAUSED
state. Calling this function after doing a seek will give the sample right
after the seek position.
Calling this function will clear the internal reference to the preroll
buffer.
Note that the preroll sample will also be returned as the first sample
when calling gst_app_sink_pull_sample().
If an EOS event was received before any buffers, this function returns
%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
This function blocks until a preroll sample or EOS is received or the appsink
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or EOS becomes available or the appsink
element is set to the READY/NULL state.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers will be put in a queue so that the application
can pull samples at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
If an EOS event was received before any buffers, this function returns
%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers and events will be put in a queue so that the application
can pull them at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
Events can be pulled when the appsink is in the READY, PAUSED or PLAYING state.
This function will only pull serialized events, excluding
the EOS event for which this functions returns
%NULL. Use gst_app_sink_is_eos() to check for the EOS condition.
This method is a variant of gst_app_sink_try_pull_sample() that can be used
to handle incoming events events as well as samples.
Note that future releases may extend this API to return other object types
so make sure that your code is checking for the actual type it is handling.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample, or #GstEvent or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state.
This function is typically used when dealing with a pipeline in the PAUSED
state. Calling this function after doing a seek will give the sample right
after the seek position.
Calling this function will clear the internal reference to the preroll
buffer.
Note that the preroll sample will also be returned as the first sample
when calling gst_app_sink_pull_sample().
If an EOS event was received before any buffers or the timeout expires,
this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
condition.
This function blocks until a preroll sample or EOS is received, the appsink
element is set to the READY/NULL state, or the timeout expires.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for the preroll sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers will be put in a queue so that the application
can pull samples at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
If an EOS event was received before any buffers or the timeout expires,
this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the #GstCaps accepted by the sink. gst_caps_unref() after usage.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Check if @appsink will drop old buffers when the maximum amount of queued
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">%TRUE if @appsink is dropping old buffers when the queue is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Check if appsink will emit the "new-preroll" and "new-sample" signals.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">%TRUE if @appsink is emitting the "new-preroll" and "new-sample"
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the maximum amount of buffers that can be queued in @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">The maximum amount of buffers that can be queued.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the maximum total size, in bytes, that can be queued in @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">The maximum amount of bytes that can be queued</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the maximum total duration that can be queued in @appsink.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">The maximum total duration that can be queued.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Check if @appsink will wait for all buffers to be consumed when an EOS is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">%TRUE if @appsink will wait for all buffers to be consumed when an
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Check if @appsink is EOS, which is when no more samples can be pulled because
an EOS event was received.
This function also returns %TRUE when the appsink is not in the PAUSED or
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">%TRUE if no more samples can be pulled and the appsink is EOS.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event becomes available or the appsink
element is set to the READY/NULL state.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers and events will be put in a queue so that the application
can pull them at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
Events can be pulled when the appsink is in the READY, PAUSED or PLAYING state.
This function will only pull serialized events, excluding
the EOS event for which this functions returns
%NULL. Use gst_app_sink_is_eos() to check for the EOS condition.
This method is a variant of gst_app_sink_pull_sample() that can be used
to handle incoming events events as well as samples.
Note that future releases may extend this API to return other object types
so make sure that your code is checking for the actual type it is handling.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample, or a #GstEvent or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state.
This function is typically used when dealing with a pipeline in the PAUSED
state. Calling this function after doing a seek will give the sample right
after the seek position.
Calling this function will clear the internal reference to the preroll
buffer.
Note that the preroll sample will also be returned as the first sample
when calling gst_app_sink_pull_sample().
If an EOS event was received before any buffers, this function returns
%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
This function blocks until a preroll sample or EOS is received or the appsink
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or EOS becomes available or the appsink
element is set to the READY/NULL state.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers will be put in a queue so that the application
can pull samples at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
If an EOS event was received before any buffers, this function returns
%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Instruct @appsink to enable or disable buffer list support.
For backwards-compatibility reasons applications need to opt in
to indicate that they will be able to handle buffer lists.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Set callbacks which will be executed for each new preroll, new sample and eos.
This is an alternative to using the signals, it has lower overhead and is thus
less expensive, but also less flexible.
If callbacks are installed, no signals will be emitted for performance
reasons.
Before 1.16.3 it was not possible to change the callbacks in a thread-safe
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Set the capabilities on the appsink element. This function takes
a copy of the caps structure. After calling this method, the sink will only
accept caps that match @caps. If @caps is non-fixed, or incomplete,
you must check the caps on the samples to get the actual used caps.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Instruct @appsink to drop old buffers when the maximum amount of queued
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Make appsink emit the "new-preroll" and "new-sample" signals. This option is
by default disabled because signal emission is expensive and unneeded when
the application prefers to operate in pull mode.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Set the maximum amount of buffers that can be queued in @appsink. After this
amount of buffers are queued in appsink, any more buffers will block upstream
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Set the maximum total size that can be queued in @appsink. After this
amount of buffers are queued in appsink, any more buffers will block upstream
elements until a sample is pulled from @appsink, unless 'drop' is set, in which
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum total size of buffers to queue, in bytes</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Set the maximum total duration that can be queued in @appsink. After this
amount of buffers are queued in appsink, any more buffers will block upstream
elements until a sample is pulled from @appsink, unless 'drop' is set, in which
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Instruct @appsink to wait for all buffers to be consumed when an EOS is received.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers and events will be put in a queue so that the application
can pull them at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
Events can be pulled when the appsink is in the READY, PAUSED or PLAYING state.
This function will only pull serialized events, excluding
the EOS event for which this functions returns
%NULL. Use gst_app_sink_is_eos() to check for the EOS condition.
This method is a variant of gst_app_sink_try_pull_sample() that can be used
to handle incoming events events as well as samples.
Note that future releases may extend this API to return other object types
so make sure that your code is checking for the actual type it is handling.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample, or #GstEvent or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state.
This function is typically used when dealing with a pipeline in the PAUSED
state. Calling this function after doing a seek will give the sample right
after the seek position.
Calling this function will clear the internal reference to the preroll
buffer.
Note that the preroll sample will also be returned as the first sample
when calling gst_app_sink_pull_sample().
If an EOS event was received before any buffers or the timeout expires,
this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
condition.
This function blocks until a preroll sample or EOS is received, the appsink
element is set to the READY/NULL state, or the timeout expires.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for the preroll sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
This function will only return samples when the appsink is in the PLAYING
state. All rendered buffers will be put in a queue so that the application
can pull samples at its own rate. Note that when the application does not
pull samples fast enough, the queued buffers could consume a lot of memory,
especially when dealing with raw video frames.
If an EOS event was received before any buffers or the timeout expires,
this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Maximum amount of buffers in the queue (0 = unlimited).</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Maximum amount of bytes in the queue (0 = unlimited)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Maximum total duration of data in the queue (0 = unlimited)</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Wait for all buffers to be processed after receiving an EOS.
In cases where it is uncertain if an @appsink will have a consumer for its buffers
when it receives an EOS, set to %FALSE to ensure that the @appsink will not hang.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Signal that the end-of-stream has been reached. This signal is emitted from
the streaming thread.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
</glib:signal>
<glib:signal name="new-preroll" when="last">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Signal that a new preroll sample is available.
This signal is emitted from the streaming thread and only when the
"emit-signals" property is %TRUE.
The new preroll sample can be retrieved with the "pull-preroll" action
signal or gst_app_sink_pull_preroll() either from this signal callback
or from any other thread.
Note that this signal is only emitted when the "emit-signals" property is
set to %TRUE, which it is not by default for performance reasons.</doc>
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn"/>
</return-value>
</glib:signal>
<glib:signal name="new-sample" when="last">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Signal that a new sample is available.
This signal is emitted from the streaming thread and only when the
"emit-signals" property is %TRUE.
The new sample can be retrieved with the "pull-sample" action
signal or gst_app_sink_pull_sample() either from this signal callback
or from any other thread.
Note that this signal is only emitted when the "emit-signals" property is
set to %TRUE, which it is not by default for performance reasons.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Signal that a new downstream serialized event is available.
This signal is emitted from the streaming thread and only when the
"emit-signals" property is %TRUE.
The new event can be retrieved with the "try-pull-object" action
signal or gst_app_sink_pull_object() either from this signal callback
or from any other thread.
EOS will not be notified using this signal, use #GstAppSink::eos instead.
EOS cannot be pulled either, use gst_app_sink_is_eos() to check for it.
Note that this signal is only emitted when the "emit-signals" property is
set to %TRUE, which it is not by default for performance reasons.
The callback should return %TRUE if the event has been handled, which will
skip basesink handling of the event, %FALSE otherwise.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Signal that a new propose_allocation query is available.
This signal is emitted from the streaming thread and only when the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state.
This function is typically used when dealing with a pipeline in the PAUSED
state. Calling this function after doing a seek will give the sample right
after the seek position.
Calling this function will clear the internal reference to the preroll
buffer.
Note that the preroll sample will also be returned as the first sample
when calling gst_app_sink_pull_sample() or the "pull-sample" action signal.
If an EOS event was received before any buffers, this function returns
%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
This function blocks until a preroll sample or EOS is received or the appsink
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or %NULL when the appsink is stopped or EOS.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or EOS becomes available or the appsink
element is set to the READY/NULL state.
This function will only return samples when the appsink is in the PLAYING
state. All rendered samples will be put in a queue so that the application
can pull samples at its own rate.
Note that when the application does not pull samples fast enough, the
queued samples could consume a lot of memory, especially when dealing with
raw video frames. It's possible to control the behaviour of the queue with
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or %NULL when the appsink is stopped or EOS.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or an event becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
This function will only return samples when the appsink is in the PLAYING
state. All rendered samples and events will be put in a queue so that the application
can pull them at its own rate.
Events can be pulled when the appsink is in the READY, PAUSED or PLAYING state.
Note that when the application does not pull samples fast enough, the
queued samples could consume a lot of memory, especially when dealing with
raw video frames. It's possible to control the behaviour of the queue with
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or a #GstEvent or NULL when the appsink is stopped or EOS or the timeout expires.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state.
This function is typically used when dealing with a pipeline in the PAUSED
state. Calling this function after doing a seek will give the sample right
after the seek position.
Calling this function will clear the internal reference to the preroll
buffer.
Note that the preroll sample will also be returned as the first sample
when calling gst_app_sink_pull_sample() or the "pull-sample" action signal.
If an EOS event was received before any buffers or the timeout expires,
this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
condition.
This function blocks until a preroll sample or EOS is received, the appsink
element is set to the READY/NULL state, or the timeout expires.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for the preroll sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">This function blocks until a sample or EOS becomes available or the appsink
element is set to the READY/NULL state or the timeout expires.
This function will only return samples when the appsink is in the PLAYING
state. All rendered samples will be put in a queue so that the application
can pull samples at its own rate.
Note that when the application does not pull samples fast enough, the
queued samples could consume a lot of memory, especially when dealing with
raw video frames. It's possible to control the behaviour of the queue with
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.h">A set of callbacks that can be installed on the appsink with
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for the preroll sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">a #GstSample, or #GstEvent or NULL when the appsink is stopped or EOS or the timeout expires.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsink.c">the maximum amount of time to wait for a sample</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The appsrc element can be used by applications to insert data into a
GStreamer pipeline. Unlike most GStreamer elements, appsrc provides
external API functions.
appsrc can be used by linking with the libgstapp library to access the
methods directly or by using the appsrc action signals.
Before operating appsrc, the caps property must be set to fixed caps
describing the format of the data that will be pushed with appsrc. An
exception to this is when pushing buffers with unknown caps, in which case no
caps should be set. This is typically true of file-like sources that push raw
byte buffers. If you don't want to explicitly set the caps, you can use
gst_app_src_push_sample. This method gets the caps associated with the
sample and sets them on the appsrc replacing any previously set caps (if
different from sample's caps).
The main way of handing data to the appsrc element is by calling the
gst_app_src_push_buffer() method or by emitting the push-buffer action signal.
This will put the buffer onto a queue from which appsrc will read from in its
streaming thread. It is important to note that data transport will not happen
from the thread that performed the push-buffer call.
The "max-bytes", "max-buffers" and "max-time" properties control how much
data can be queued in appsrc before appsrc considers the queue full. A
filled internal queue will always signal the "enough-data" signal, which
signals the application that it should stop pushing data into appsrc. The
"block" property will cause appsrc to block the push-buffer method until
free data becomes available again.
When the internal queue is running out of data, the "need-data" signal is
emitted, which signals the application that it should start pushing more data
into appsrc.
In addition to the "need-data" and "enough-data" signals, appsrc can emit the
"seek-data" signal when the "stream-mode" property is set to "seekable" or
"random-access". The signal argument will contain the new desired position in
the stream expressed in the unit set with the "format" property. After
receiving the seek-data signal, the application should push-buffers from the
new position.
These signals allow the application to operate the appsrc in two different
ways:
The push mode, in which the application repeatedly calls the push-buffer/push-sample
method with a new buffer/sample. Optionally, the queue size in the appsrc
can be controlled with the enough-data and need-data signals by respectively
stopping/starting the push-buffer/push-sample calls. This is a typical
mode of operation for the stream-type "stream" and "seekable". Use this
mode when implementing various network protocols or hardware devices.
The pull mode, in which the need-data signal triggers the next push-buffer call.
This mode is typically used in the "random-access" stream-type. Use this
mode for file access or other randomly accessible sources. In this mode, a
buffer of exactly the amount of bytes given by the need-data signal should be
pushed into appsrc.
In all modes, the size property on appsrc should contain the total stream
size in bytes. Setting this property is mandatory in the random-access mode.
For the stream and seekable modes, setting this property is optional but
recommended.
When the application has finished pushing data into appsrc, it should call
gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
this call, no more buffers can be pushed into appsrc until a flushing seek
occurs or the state of the appsrc has gone through READY.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Indicates to the appsrc element that the last buffer queued in the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Adds a buffer to the queue of buffers that the appsrc element will
push to its source pad. This function takes ownership of the buffer.
When the block property is TRUE, this function can block until free
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Adds a buffer list to the queue of buffers and buffer lists that the
appsrc element will push to its source pad. This function takes ownership
of @buffer_list.
When the block property is TRUE, this function can block until free
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer list was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Extract a buffer from the provided sample and adds it to the queue of
buffers that the appsrc element will push to its source pad. Any
previous caps that were set on appsrc will be replaced by the caps
associated with the sample if not equal.
This function does not take ownership of the
sample so the sample needs to be unreffed after calling this function.
When the block property is TRUE, this function can block until free
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Indicates to the appsrc element that the last buffer queued in the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">the #GstCaps produced by the source. gst_caps_unref() after usage.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the number of currently queued buffers inside @appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the number of currently queued bytes inside @appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the amount of currently queued time inside @appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">the duration of the stream previously set with gst_app_src_set_duration();</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Check if appsrc will emit the "new-preroll" and "new-buffer" signals.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">%TRUE if @appsrc is emitting the "new-preroll" and "new-buffer"
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Retrieve the min and max latencies in @min and @max respectively.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Returns the currently set #GstAppLeakyType. See gst_app_src_set_leaky_type()
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the maximum amount of buffers that can be queued in @appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The maximum amount of buffers that can be queued.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the maximum amount of bytes that can be queued in @appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The maximum amount of bytes that can be queued.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the maximum amount of time that can be queued in @appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the size of the stream in bytes. A value of -1 means that the size is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">the size of the stream previously set with gst_app_src_set_size();</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Get the stream type. Control the stream type of @appsrc
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Adds a buffer to the queue of buffers that the appsrc element will
push to its source pad. This function takes ownership of the buffer.
When the block property is TRUE, this function can block until free
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Adds a buffer list to the queue of buffers and buffer lists that the
appsrc element will push to its source pad. This function takes ownership
of @buffer_list.
When the block property is TRUE, this function can block until free
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer list was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Extract a buffer from the provided sample and adds it to the queue of
buffers that the appsrc element will push to its source pad. Any
previous caps that were set on appsrc will be replaced by the caps
associated with the sample if not equal.
This function does not take ownership of the
sample so the sample needs to be unreffed after calling this function.
When the block property is TRUE, this function can block until free
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set callbacks which will be executed when data is needed, enough data has
been collected or when a seek should be performed.
This is an alternative to using the signals, it has lower overhead and is thus
less expensive, but also less flexible.
If callbacks are installed, no signals will be emitted for performance
reasons.
Before 1.16.3 it was not possible to change the callbacks in a thread-safe
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the capabilities on the appsrc element. This function takes
a copy of the caps structure. After calling this method, the source will
only produce caps that match @caps. @caps must be fixed and the caps on the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Make appsrc emit the "new-preroll" and "new-buffer" signals. This option is
by default disabled because signal emission is expensive and unneeded when
the application prefers to operate in pull mode.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Configure the @min and @max latency in @src. If @min is set to -1, the
default latency calculations for pseudo-live sources will be used.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">When set to any other value than GST_APP_LEAKY_TYPE_NONE then the appsrc
will drop any buffers that are pushed into it once its internal queue is
full. The selected type defines whether to drop the oldest or new
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the maximum amount of buffers that can be queued in @appsrc.
After the maximum amount of buffers are queued, @appsrc will emit the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the maximum amount of bytes that can be queued in @appsrc.
After the maximum amount of bytes are queued, @appsrc will emit the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the maximum amount of time that can be queued in @appsrc.
After the maximum amount of time are queued, @appsrc will emit the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the size of the stream in bytes. A value of -1 means that the size is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Set the stream type on @appsrc. For seekable streams, the "seek" signal must
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">When max-bytes are queued and after the enough-data signal has been emitted,
block any further push-buffer calls until the amount of queued bytes drops
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The GstCaps that will negotiated downstream and will be put
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The number of currently queued buffers inside appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The number of currently queued bytes inside appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The amount of currently queued time inside appsrc.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The total duration in nanoseconds of the data stream. If the total duration is known, it
is recommended to configure it with this property.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The format to use for segment events. When the source is producing
timestamped buffers this property should be set to GST_FORMAT_TIME.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">When enabled, appsrc will check GstSegment in GstSample which was
pushed via gst_app_src_push_sample() or "push-sample" signal action.
If a GstSegment is changed, corresponding segment event will be followed
by next data flow.
FIXME: currently only GST_FORMAT_TIME format is supported and therefore
GstAppSrc::format should be time. However, possibly #GstAppSrc can support
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Instruct the source to behave like a live source. This includes that it
will only push out buffers in the PLAYING state.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">When set to any other value than GST_APP_LEAKY_TYPE_NONE then the appsrc
will drop any buffers that are pushed into it once its internal queue is
full. The selected type defines whether to drop the oldest or new
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The maximum amount of buffers that can be queued internally.
After the maximum amount of buffers are queued, appsrc will emit the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The maximum amount of bytes that can be queued internally.
After the maximum amount of bytes are queued, appsrc will emit the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The maximum amount of time that can be queued internally.
After the maximum amount of time are queued, appsrc will emit the
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The minimum latency of the source. A value of -1 will use the default
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Make appsrc emit the "need-data" signal when the amount of bytes in the
queue drops below this percentage of max-bytes.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The total size in bytes of the data stream. If the total size is known, it
is recommended to configure it with this property.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">The type of stream that this source is producing. For seekable streams the
application should connect to the seek-data signal.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Notify @appsrc that no more buffer are available.</doc>
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn"/>
</return-value>
</glib:signal>
<glib:signal name="enough-data" when="last">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Signal that the source has enough data. It is recommended that the
application stops calling push-buffer until the need-data signal is
emitted again to avoid excessive buffer queueing.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
</glib:signal>
<glib:signal name="need-data" when="last">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Signal that the source needs more data. In the callback or from another
thread you should call push-buffer or end-of-stream.
@length is just a hint and when it is set to -1, any number of bytes can be
pushed into @appsrc.
You can call push-buffer multiple times until the enough-data signal is
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Adds a buffer to the queue of buffers that the appsrc element will
push to its source pad.
This function does not take ownership of the buffer, but it takes a
reference so the buffer can be unreffed at any time after calling this
function.
When the block property is TRUE, this function can block until free space
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Adds a buffer list to the queue of buffers and buffer lists that the
appsrc element will push to its source pad.
This function does not take ownership of the buffer list, but it takes a
reference so the buffer list can be unreffed at any time after calling
this function.
When the block property is TRUE, this function can block until free space
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Extract a buffer from the provided sample and adds the extracted buffer
to the queue of buffers that the appsrc element will
push to its source pad. This function set the appsrc caps based on the caps
in the sample and reset the caps if they change.
Only the caps and the buffer of the provided sample are used and not
for example the segment in the sample.
This function does not take ownership of the sample, but it takes a
reference so the sample can be unreffed at any time after calling this
function.
When the block property is TRUE, this function can block until free space
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">a sample from which extract buffer to push</doc>
<type name="Gst.Sample"/>
</parameter>
</parameters>
</glib:signal>
<glib:signal name="seek-data" when="last">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">Seek to the given offset. The next push-buffer should produce buffers from
the new @offset.
This callback is only called for seekable stream types.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">%TRUE if the seek succeeded.</doc>
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.h">A set of callbacks that can be installed on the appsrc with
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
<doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/app/gstappsrc.c">#GST_FLOW_OK when the buffer list was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
