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Update GIR files from 1.12.0 final release

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Sebastian Dröge 2017-05-10 10:52:03 +02:00
parent 0dc9558ea2
commit 385ff00de5
4 changed files with 48 additions and 60 deletions
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100
gir-files/Gst-1.0.gir
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@ -35175,14 +35175,24 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
glib:type-name="GstStateChange" glib:type-name="GstStateChange"
glib:get-type="gst_state_change_get_type" glib:get-type="gst_state_change_get_type"
c:type="GstStateChange"> c:type="GstStateChange">
<doc xml:space="preserve">#GST_STATE_CHANGE_NULL_TO_READY : state change from NULL to READY. <doc xml:space="preserve">These are the different state changes an element goes through.
%GST_STATE_NULL &amp;rArr; %GST_STATE_PLAYING is called an upwards state change
and %GST_STATE_PLAYING &amp;rArr; %GST_STATE_NULL a downwards state change.</doc>
<member name="null_to_ready"
value="10"
c:identifier="GST_STATE_CHANGE_NULL_TO_READY"
glib:nick="null-to-ready">
<doc xml:space="preserve">state change from NULL to READY.
* The element must check if the resources it needs are available. Device * The element must check if the resources it needs are available. Device
sinks and -sources typically try to probe the device to constrain their sinks and -sources typically try to probe the device to constrain their
caps. caps.
* The element opens the device (in case feature need to be probed). * The element opens the device (in case feature need to be probed).</doc>
</member>
#GST_STATE_CHANGE_READY_TO_PAUSED : state change from READY to PAUSED. <member name="ready_to_paused"
value="19"
c:identifier="GST_STATE_CHANGE_READY_TO_PAUSED"
glib:nick="ready-to-paused">
<doc xml:space="preserve">state change from READY to PAUSED.
* The element pads are activated in order to receive data in PAUSED. * The element pads are activated in order to receive data in PAUSED.
Streaming threads are started. Streaming threads are started.
* Some elements might need to return %GST_STATE_CHANGE_ASYNC and complete * Some elements might need to return %GST_STATE_CHANGE_ASYNC and complete
@ -35191,11 +35201,13 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
when they receive the first buffer or %GST_EVENT_EOS (preroll). when they receive the first buffer or %GST_EVENT_EOS (preroll).
Sinks also block the dataflow when in PAUSED. Sinks also block the dataflow when in PAUSED.
* A pipeline resets the running_time to 0. * A pipeline resets the running_time to 0.
* Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data.</doc>
* Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data. </member>
<member name="paused_to_playing"
#GST_STATE_CHANGE_PAUSED_TO_PLAYING: state change from PAUSED to PLAYING. value="28"
c:identifier="GST_STATE_CHANGE_PAUSED_TO_PLAYING"
glib:nick="paused-to-playing">
<doc xml:space="preserve">state change from PAUSED to PLAYING.
* Most elements ignore this state change. * Most elements ignore this state change.
* The pipeline selects a #GstClock and distributes this to all the children * The pipeline selects a #GstClock and distributes this to all the children
before setting them to PLAYING. This means that it is only allowed to before setting them to PLAYING. This means that it is only allowed to
@ -35209,15 +35221,17 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
to post %GST_MESSAGE_EOS when not in the PLAYING state. to post %GST_MESSAGE_EOS when not in the PLAYING state.
* While streaming in PAUSED or PLAYING elements can create and remove * While streaming in PAUSED or PLAYING elements can create and remove
sometimes pads. sometimes pads.
* Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS. * Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS.</doc>
</member>
#GST_STATE_CHANGE_PLAYING_TO_PAUSED: state change from PLAYING to PAUSED. <member name="playing_to_paused"
value="35"
c:identifier="GST_STATE_CHANGE_PLAYING_TO_PAUSED"
glib:nick="playing-to-paused">
<doc xml:space="preserve">state change from PLAYING to PAUSED.
* Most elements ignore this state change. * Most elements ignore this state change.
* The pipeline calculates the running_time based on the last selected * The pipeline calculates the running_time based on the last selected
#GstClock and the base_time. It stores this information to continue #GstClock and the base_time. It stores this information to continue
playback when going back to the PLAYING state. playback when going back to the PLAYING state.
* Sinks unblock any #GstClock wait calls. * Sinks unblock any #GstClock wait calls.
* When a sink does not have a pending buffer to play, it returns * When a sink does not have a pending buffer to play, it returns
#GST_STATE_CHANGE_ASYNC from this state change and completes the state #GST_STATE_CHANGE_ASYNC from this state change and completes the state
@ -35225,56 +35239,28 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
* Any queued %GST_MESSAGE_EOS items are removed since they will be reposted * Any queued %GST_MESSAGE_EOS items are removed since they will be reposted
when going back to the PLAYING state. The EOS messages are queued in when going back to the PLAYING state. The EOS messages are queued in
#GstBin containers. #GstBin containers.
* Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL.</doc>
* Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL. </member>
<member name="paused_to_ready"
#GST_STATE_CHANGE_PAUSED_TO_READY : state change from PAUSED to READY. value="26"
c:identifier="GST_STATE_CHANGE_PAUSED_TO_READY"
glib:nick="paused-to-ready">
<doc xml:space="preserve">state change from PAUSED to READY.
* Sinks unblock any waits in the preroll. * Sinks unblock any waits in the preroll.
* Elements unblock any waits on devices * Elements unblock any waits on devices
* Chain or get_range functions return %GST_FLOW_FLUSHING. * Chain or get_range functions return %GST_FLOW_FLUSHING.
* The element pads are deactivated so that streaming becomes impossible and * The element pads are deactivated so that streaming becomes impossible and
all streaming threads are stopped. all streaming threads are stopped.
* The sink forgets all negotiated formats * The sink forgets all negotiated formats
* Elements remove all sometimes pads * Elements remove all sometimes pads</doc>
#GST_STATE_CHANGE_READY_TO_NULL : state change from READY to NULL.
* Elements close devices
* Elements reset any internal state.
These are the different state changes an element goes through.
%GST_STATE_NULL &amp;rArr; %GST_STATE_PLAYING is called an upwards state change
and %GST_STATE_PLAYING &amp;rArr; %GST_STATE_NULL a downwards state change.</doc>
<member name="null_to_ready"
value="10"
c:identifier="GST_STATE_CHANGE_NULL_TO_READY"
glib:nick="null-to-ready">
</member>
<member name="ready_to_paused"
value="19"
c:identifier="GST_STATE_CHANGE_READY_TO_PAUSED"
glib:nick="ready-to-paused">
</member>
<member name="paused_to_playing"
value="28"
c:identifier="GST_STATE_CHANGE_PAUSED_TO_PLAYING"
glib:nick="paused-to-playing">
</member>
<member name="playing_to_paused"
value="35"
c:identifier="GST_STATE_CHANGE_PLAYING_TO_PAUSED"
glib:nick="playing-to-paused">
</member>
<member name="paused_to_ready"
value="26"
c:identifier="GST_STATE_CHANGE_PAUSED_TO_READY"
glib:nick="paused-to-ready">
</member> </member>
<member name="ready_to_null" <member name="ready_to_null"
value="17" value="17"
c:identifier="GST_STATE_CHANGE_READY_TO_NULL" c:identifier="GST_STATE_CHANGE_READY_TO_NULL"
glib:nick="ready-to-null"> glib:nick="ready-to-null">
<doc xml:space="preserve">state change from READY to NULL.
* Elements close devices
* Elements reset any internal state.</doc>
</member> </member>
</enumeration> </enumeration>
<enumeration name="StateChangeReturn" <enumeration name="StateChangeReturn"
@ -43742,15 +43728,15 @@ determine a order for the two provided values.</doc>
<doc xml:space="preserve">The major version of GStreamer at compile time:</doc> <doc xml:space="preserve">The major version of GStreamer at compile time:</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>
</constant> </constant>
<constant name="VERSION_MICRO" value="2" c:type="GST_VERSION_MICRO"> <constant name="VERSION_MICRO" value="0" c:type="GST_VERSION_MICRO">
<doc xml:space="preserve">The micro version of GStreamer at compile time:</doc> <doc xml:space="preserve">The micro version of GStreamer at compile time:</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>
</constant> </constant>
<constant name="VERSION_MINOR" value="11" c:type="GST_VERSION_MINOR"> <constant name="VERSION_MINOR" value="12" c:type="GST_VERSION_MINOR">
<doc xml:space="preserve">The minor version of GStreamer at compile time:</doc> <doc xml:space="preserve">The minor version of GStreamer at compile time:</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>
</constant> </constant>
<constant name="VERSION_NANO" value="1" c:type="GST_VERSION_NANO"> <constant name="VERSION_NANO" value="0" c:type="GST_VERSION_NANO">
<doc xml:space="preserve">The nano version of GStreamer at compile time: <doc xml:space="preserve">The nano version of GStreamer at compile time:
Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc> Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>

2
gir-files/GstAudio-1.0.gir
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@ -189,6 +189,8 @@ rates.
<constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION" <constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"
value="GstAudioResampler.stop-attenutation" value="GstAudioResampler.stop-attenutation"
c:type="GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"> c:type="GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION">
<doc xml:space="preserve">G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
after the stopband for the kaiser window. 85 dB is the default.</doc>
<type name="utf8" c:type="gchar*"/> <type name="utf8" c:type="gchar*"/>
</constant> </constant>
<constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH" <constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH"

4
gir-files/GstPbutils-1.0.gir
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@ -2683,13 +2683,13 @@ in debugging.</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>
</constant> </constant>
<constant name="PLUGINS_BASE_VERSION_MICRO" <constant name="PLUGINS_BASE_VERSION_MICRO"
value="90" value="0"
c:type="GST_PLUGINS_BASE_VERSION_MICRO"> c:type="GST_PLUGINS_BASE_VERSION_MICRO">
<doc xml:space="preserve">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc> <doc xml:space="preserve">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>
</constant> </constant>
<constant name="PLUGINS_BASE_VERSION_MINOR" <constant name="PLUGINS_BASE_VERSION_MINOR"
value="11" value="12"
c:type="GST_PLUGINS_BASE_VERSION_MINOR"> c:type="GST_PLUGINS_BASE_VERSION_MINOR">
<doc xml:space="preserve">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc> <doc xml:space="preserve">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<type name="gint" c:type="gint"/> <type name="gint" c:type="gint"/>

2
gir-files/GstVideo-1.0.gir
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@ -1804,7 +1804,7 @@ quatization errors.</doc>
</function> </function>
</record> </record>
<record name="VideoAlignment" c:type="GstVideoAlignment"> <record name="VideoAlignment" c:type="GstVideoAlignment">
<doc xml:space="preserve">Extra alignment paramters for the memory of video buffers. This <doc xml:space="preserve">Extra alignment parameters for the memory of video buffers. This
structure is usually used to configure the bufferpool if it supports the structure is usually used to configure the bufferpool if it supports the
#GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT.</doc> #GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT.</doc>
<field name="padding_top" writable="1"> <field name="padding_top" writable="1">