Commit b25a0711 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵
Browse files

Update gir-files from gstreamer-sys

parent 88a68dff
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......@@ -91,13 +91,15 @@ to avoid polling.</doc>
</virtual-method>
<virtual-method name="pull_preroll" invoker="pull_preroll">
<doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state. This sample can be pulled many times
and remains available to the application even after EOS.
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().
......@@ -146,13 +148,15 @@ If an EOS event was received before any buffers, this function returns
invoker="try_pull_preroll"
version="1.10">
<doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state. This sample can be pulled many times
and remains available to the application even after EOS.
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().
......@@ -316,13 +320,15 @@ PLAYING state.</doc>
</method>
<method name="pull_preroll" c:identifier="gst_app_sink_pull_preroll">
<doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state. This sample can be pulled many times
and remains available to the application even after EOS.
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().
......@@ -517,13 +523,15 @@ elements until a sample is pulled from @appsink.</doc>
c:identifier="gst_app_sink_try_pull_preroll"
version="1.10">
<doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state. This sample can be pulled many times
and remains available to the application even after EOS.
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().
......@@ -653,13 +661,15 @@ set to %TRUE, which it is not by default for performance reasons.</doc>
</glib:signal>
<glib:signal name="pull-preroll" when="last" action="1">
<doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state. This sample can be pulled many times
and remains available to the application even after EOS.
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.
......@@ -698,13 +708,15 @@ If an EOS event was received before any buffers, this function returns
action="1"
version="1.10">
<doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
appsink to preroll in the PAUSED state. This sample can be pulled many times
and remains available to the application even after EOS.
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.
......@@ -1075,6 +1087,32 @@ space becomes available in the queue.</doc>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="push_buffer_list"
invoker="push_buffer_list"
version="1.14">
<doc xml:space="preserve">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
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="appsrc" transfer-ownership="none">
<doc xml:space="preserve">a #GstAppSrc</doc>
<type name="AppSrc" c:type="GstAppSrc*"/>
</instance-parameter>
<parameter name="buffer_list" transfer-ownership="full">
<doc xml:space="preserve">a #GstBufferList to push</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="push_sample" invoker="push_sample" version="1.6">
<doc xml:space="preserve">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
......@@ -1209,7 +1247,7 @@ signals.</doc>
direction="out"
caller-allocates="0"
transfer-ownership="full">
<doc xml:space="preserve">the min latency</doc>
<doc xml:space="preserve">the max latency</doc>
<type name="guint64" c:type="guint64*"/>
</parameter>
</parameters>
......@@ -1279,6 +1317,32 @@ space becomes available in the queue.</doc>
</parameter>
</parameters>
</method>
<method name="push_buffer_list"
c:identifier="gst_app_src_push_buffer_list"
version="1.14">
<doc xml:space="preserve">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
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="appsrc" transfer-ownership="none">
<doc xml:space="preserve">a #GstAppSrc</doc>
<type name="AppSrc" c:type="GstAppSrc*"/>
</instance-parameter>
<parameter name="buffer_list" transfer-ownership="full">
<doc xml:space="preserve">a #GstBufferList to push</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</method>
<method name="push_sample"
c:identifier="gst_app_src_push_sample"
version="1.6">
......@@ -1415,7 +1479,7 @@ default latency calculations for pseudo-live sources will be used.</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
<parameter name="max" transfer-ownership="none">
<doc xml:space="preserve">the min latency</doc>
<doc xml:space="preserve">the max latency</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
</parameters>
......@@ -1575,6 +1639,27 @@ becomes available in the queue.</doc>
</parameter>
</parameters>
</glib:signal>
<glib:signal name="push-buffer-list"
when="last"
action="1"
version="1.14">
<doc xml:space="preserve">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 so the buffer list needs to be unreffed
after calling this function.
When the block property is TRUE, this function can block until free space
becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn"/>
</return-value>
<parameters>
<parameter name="buffer_list" transfer-ownership="none">
<doc xml:space="preserve">a buffer list to push</doc>
<type name="Gst.BufferList"/>
</parameter>
</parameters>
</glib:signal>
<glib:signal name="push-sample" when="last" action="1" version="1.6">
<doc xml:space="preserve">Extract a buffer from the provided sample and adds the extracted buffer
to the queue of buffers that the appsrc element will
......@@ -1791,8 +1876,28 @@ extracted</doc>
</parameters>
</callback>
</field>
<field name="push_buffer_list">
<callback name="push_buffer_list">
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<parameter name="appsrc" transfer-ownership="none">
<doc xml:space="preserve">a #GstAppSrc</doc>
<type name="AppSrc" c:type="GstAppSrc*"/>
</parameter>
<parameter name="buffer_list" transfer-ownership="full">
<doc xml:space="preserve">a #GstBufferList to push</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="3">
<array zero-terminated="0" c:type="gpointer" fixed-size="2">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
......
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......@@ -72,7 +72,7 @@ statistics about clock accuracy and network traffic.</doc>
<doc xml:space="preserve">Create a new #GstNetClientInternalClock that will report the time
provided by the #GstNetTimeProvider on @remote_address and
@remote_port.</doc>
<return-value transfer-ownership="none">
<return-value transfer-ownership="full">
<doc xml:space="preserve">a new #GstClock that receives a time from the remote
clock.</doc>
<type name="Gst.Clock" c:type="GstClock*"/>
......@@ -125,6 +125,9 @@ clock.</doc>
transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="qos-dscp" writable="1" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="round-trip-limit" writable="1" transfer-ownership="none">
<type name="guint64" c:type="guint64"/>
</property>
......@@ -197,7 +200,7 @@ structures.</doc>
caller is responsible for ensuring that @buffer is at least
#GST_NET_TIME_PACKET_SIZE bytes long.
If @buffer is #NULL, the local and remote times will be set to
If @buffer is %NULL, the local and remote times will be set to
#GST_CLOCK_TIME_NONE.
MT safe. Caller owns return value (gst_net_time_packet_free to free).</doc>
......@@ -324,7 +327,7 @@ The #GstNetTimeProvider typically wraps the clock used by a #GstPipeline.</doc>
<implements name="Gio.Initable"/>
<constructor name="new" c:identifier="gst_net_time_provider_new">
<doc xml:space="preserve">Allows network clients to get the current time of @clock.</doc>
<return-value transfer-ownership="none">
<return-value transfer-ownership="full">
<doc xml:space="preserve">the new #GstNetTimeProvider, or NULL on error</doc>
<type name="NetTimeProvider" c:type="GstNetTimeProvider*"/>
</return-value>
......@@ -368,6 +371,9 @@ The #GstNetTimeProvider typically wraps the clock used by a #GstPipeline.</doc>
transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="qos-dscp" writable="1" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<field name="parent">
<type name="Gst.Object" c:type="GstObject"/>
</field>
......@@ -407,7 +413,7 @@ The #GstNetTimeProvider typically wraps the clock used by a #GstPipeline.</doc>
<constructor name="new" c:identifier="gst_ntp_clock_new" version="1.6">
<doc xml:space="preserve">Create a new #GstNtpClock that will report the time provided by
the NTPv4 server on @remote_address and @remote_port.</doc>
<return-value transfer-ownership="none">
<return-value transfer-ownership="full">
<doc xml:space="preserve">a new #GstClock that receives a time from the remote
clock.</doc>
<type name="Gst.Clock" c:type="GstClock*"/>
......@@ -522,7 +528,8 @@ times from the PTP master clock on the network. Once this happens the
GstPtpClock::internal-clock property will become non-NULL. You can
check this with gst_clock_wait_for_sync(), the GstClock::synced signal and
gst_clock_is_synced().</doc>
<return-value transfer-ownership="none">
<return-value transfer-ownership="full">
<doc xml:space="preserve">A new #GstClock</doc>
<type name="Gst.Clock" c:type="GstClock*"/>
</return-value>
<parameters>
......
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......@@ -117,6 +117,32 @@ Since 1.10</doc>
</parameter>
</parameters>
</function>
<function name="config_set_seek_accurate"
c:identifier="gst_player_config_set_seek_accurate"
version="1.12">
<doc xml:space="preserve">Enable or disable accurate seeking. When enabled, elements will try harder
to seek as accurately as possible to the requested seek position. Generally
it will be slower especially for formats that don't have any indexes or
timestamp markers in the stream.
If accurate seeking is disabled, elements will seek as close as the request
position without slowing down seeking too much.
Accurate seeking is disabled by default.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="config" transfer-ownership="none">
<doc xml:space="preserve">a #GstPlayer configuration</doc>
<type name="Gst.Structure" c:type="GstStructure*"/>
</parameter>
<parameter name="accurate" transfer-ownership="none">
<doc xml:space="preserve">accurate seek or not</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</function>
<function name="config_set_user_agent"
c:identifier="gst_player_config_set_user_agent">
<doc xml:space="preserve">Set the user agent to pass to the server if @player needs to connect
......@@ -212,32 +238,6 @@ matching #GstPlayerVideoInfo.</doc>
</array>
</return-value>
</function>
<method name="config_set_seek_accurate"
c:identifier="gst_player_config_set_seek_accurate"
version="1.12">
<doc xml:space="preserve">Enable or disable accurate seeking. When enabled, elements will try harder
to seek as accurately as possible to the requested seek position. Generally
it will be slower especially for formats that don't have any indexes or
timestamp markers in the stream.
If accurate seeking is disabled, elements will seek as close as the request
position without slowing down seeking too much.
Accurate seeking is disabled by default.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="player" transfer-ownership="none">
<doc xml:space="preserve">#GstPlayer instance</doc>
<type name="Player" c:type="GstPlayer*"/>
</instance-parameter>
<parameter name="accurate" transfer-ownership="none">
<doc xml:space="preserve">accurate seek or not</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</method>
<method name="get_audio_video_offset"
c:identifier="gst_player_get_audio_video_offset">
<doc xml:space="preserve">Retrieve the current value of audio-video-offset property</doc>
......@@ -808,7 +808,9 @@ Sets the subtitle strack @stream_index.</doc>
</method>
<method name="set_subtitle_uri"
c:identifier="gst_player_set_subtitle_uri">
<doc xml:space="preserve">Sets the external subtitle URI.</doc>
<doc xml:space="preserve">Sets the external subtitle URI. This should be combined with a call to
gst_player_set_subtitle_track_enabled(@player, TRUE) so the subtitles are actually
rendered.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
......
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