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Update GIR files from 1.12.0 final release
This commit is contained in:
Sebastian Dröge
parent
0dc9558ea2
commit
385ff00de5
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@ -35175,14 +35175,24 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
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glib:type-name="GstStateChange"
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glib:get-type="gst_state_change_get_type"
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c:type="GstStateChange">
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<doc xml:space="preserve">#GST_STATE_CHANGE_NULL_TO_READY : state change from NULL to READY.
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<doc xml:space="preserve">These are the different state changes an element goes through.
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%GST_STATE_NULL &rArr; %GST_STATE_PLAYING is called an upwards state change
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and %GST_STATE_PLAYING &rArr; %GST_STATE_NULL a downwards state change.</doc>
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<member name="null_to_ready"
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value="10"
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c:identifier="GST_STATE_CHANGE_NULL_TO_READY"
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glib:nick="null-to-ready">
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<doc xml:space="preserve">state change from NULL to READY.
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* The element must check if the resources it needs are available. Device
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sinks and -sources typically try to probe the device to constrain their
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caps.
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* The element opens the device (in case feature need to be probed).
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#GST_STATE_CHANGE_READY_TO_PAUSED : state change from READY to PAUSED.
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* The element opens the device (in case feature need to be probed).</doc>
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</member>
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<member name="ready_to_paused"
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value="19"
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c:identifier="GST_STATE_CHANGE_READY_TO_PAUSED"
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glib:nick="ready-to-paused">
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<doc xml:space="preserve">state change from READY to PAUSED.
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* The element pads are activated in order to receive data in PAUSED.
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Streaming threads are started.
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* Some elements might need to return %GST_STATE_CHANGE_ASYNC and complete
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@ -35191,11 +35201,13 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
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when they receive the first buffer or %GST_EVENT_EOS (preroll).
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Sinks also block the dataflow when in PAUSED.
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* A pipeline resets the running_time to 0.
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* Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data.
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#GST_STATE_CHANGE_PAUSED_TO_PLAYING: state change from PAUSED to PLAYING.
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* Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data.</doc>
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</member>
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<member name="paused_to_playing"
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value="28"
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c:identifier="GST_STATE_CHANGE_PAUSED_TO_PLAYING"
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glib:nick="paused-to-playing">
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<doc xml:space="preserve">state change from PAUSED to PLAYING.
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* Most elements ignore this state change.
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* The pipeline selects a #GstClock and distributes this to all the children
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before setting them to PLAYING. This means that it is only allowed to
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@ -35209,15 +35221,17 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
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to post %GST_MESSAGE_EOS when not in the PLAYING state.
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* While streaming in PAUSED or PLAYING elements can create and remove
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sometimes pads.
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* Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS.
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#GST_STATE_CHANGE_PLAYING_TO_PAUSED: state change from PLAYING to PAUSED.
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* Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS.</doc>
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</member>
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<member name="playing_to_paused"
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value="35"
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c:identifier="GST_STATE_CHANGE_PLAYING_TO_PAUSED"
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glib:nick="playing-to-paused">
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<doc xml:space="preserve">state change from PLAYING to PAUSED.
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* Most elements ignore this state change.
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* The pipeline calculates the running_time based on the last selected
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#GstClock and the base_time. It stores this information to continue
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playback when going back to the PLAYING state.
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* Sinks unblock any #GstClock wait calls.
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* When a sink does not have a pending buffer to play, it returns
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#GST_STATE_CHANGE_ASYNC from this state change and completes the state
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@ -35225,56 +35239,28 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
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* Any queued %GST_MESSAGE_EOS items are removed since they will be reposted
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when going back to the PLAYING state. The EOS messages are queued in
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#GstBin containers.
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* Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL.
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#GST_STATE_CHANGE_PAUSED_TO_READY : state change from PAUSED to READY.
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* Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL.</doc>
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</member>
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<member name="paused_to_ready"
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value="26"
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c:identifier="GST_STATE_CHANGE_PAUSED_TO_READY"
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glib:nick="paused-to-ready">
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<doc xml:space="preserve">state change from PAUSED to READY.
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* Sinks unblock any waits in the preroll.
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* Elements unblock any waits on devices
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* Chain or get_range functions return %GST_FLOW_FLUSHING.
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* The element pads are deactivated so that streaming becomes impossible and
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all streaming threads are stopped.
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* The sink forgets all negotiated formats
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* Elements remove all sometimes pads
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#GST_STATE_CHANGE_READY_TO_NULL : state change from READY to NULL.
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* Elements close devices
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* Elements reset any internal state.
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These are the different state changes an element goes through.
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%GST_STATE_NULL &rArr; %GST_STATE_PLAYING is called an upwards state change
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and %GST_STATE_PLAYING &rArr; %GST_STATE_NULL a downwards state change.</doc>
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<member name="null_to_ready"
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value="10"
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c:identifier="GST_STATE_CHANGE_NULL_TO_READY"
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glib:nick="null-to-ready">
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</member>
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<member name="ready_to_paused"
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value="19"
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c:identifier="GST_STATE_CHANGE_READY_TO_PAUSED"
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glib:nick="ready-to-paused">
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</member>
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<member name="paused_to_playing"
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value="28"
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c:identifier="GST_STATE_CHANGE_PAUSED_TO_PLAYING"
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glib:nick="paused-to-playing">
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</member>
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<member name="playing_to_paused"
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value="35"
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c:identifier="GST_STATE_CHANGE_PLAYING_TO_PAUSED"
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glib:nick="playing-to-paused">
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</member>
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<member name="paused_to_ready"
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value="26"
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c:identifier="GST_STATE_CHANGE_PAUSED_TO_READY"
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glib:nick="paused-to-ready">
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* Elements remove all sometimes pads</doc>
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</member>
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<member name="ready_to_null"
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value="17"
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c:identifier="GST_STATE_CHANGE_READY_TO_NULL"
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glib:nick="ready-to-null">
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<doc xml:space="preserve">state change from READY to NULL.
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* Elements close devices
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* Elements reset any internal state.</doc>
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</member>
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</enumeration>
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<enumeration name="StateChangeReturn"
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@ -43742,15 +43728,15 @@ determine a order for the two provided values.</doc>
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<doc xml:space="preserve">The major version of GStreamer at compile time:</doc>
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<type name="gint" c:type="gint"/>
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</constant>
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<constant name="VERSION_MICRO" value="2" c:type="GST_VERSION_MICRO">
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<constant name="VERSION_MICRO" value="0" c:type="GST_VERSION_MICRO">
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<doc xml:space="preserve">The micro version of GStreamer at compile time:</doc>
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<type name="gint" c:type="gint"/>
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</constant>
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<constant name="VERSION_MINOR" value="11" c:type="GST_VERSION_MINOR">
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<constant name="VERSION_MINOR" value="12" c:type="GST_VERSION_MINOR">
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<doc xml:space="preserve">The minor version of GStreamer at compile time:</doc>
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<type name="gint" c:type="gint"/>
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</constant>
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<constant name="VERSION_NANO" value="1" c:type="GST_VERSION_NANO">
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<constant name="VERSION_NANO" value="0" c:type="GST_VERSION_NANO">
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<doc xml:space="preserve">The nano version of GStreamer at compile time:
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Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc>
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<type name="gint" c:type="gint"/>
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@ -189,6 +189,8 @@ rates.
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<constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"
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value="GstAudioResampler.stop-attenutation"
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c:type="GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION">
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<doc xml:space="preserve">G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
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after the stopband for the kaiser window. 85 dB is the default.</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH"
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@ -2683,13 +2683,13 @@ in debugging.</doc>
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<type name="gint" c:type="gint"/>
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</constant>
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<constant name="PLUGINS_BASE_VERSION_MICRO"
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value="90"
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value="0"
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c:type="GST_PLUGINS_BASE_VERSION_MICRO">
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<doc xml:space="preserve">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc>
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<type name="gint" c:type="gint"/>
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</constant>
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<constant name="PLUGINS_BASE_VERSION_MINOR"
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value="11"
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value="12"
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c:type="GST_PLUGINS_BASE_VERSION_MINOR">
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<doc xml:space="preserve">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc>
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<type name="gint" c:type="gint"/>
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@ -1804,7 +1804,7 @@ quatization errors.</doc>
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</function>
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</record>
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<record name="VideoAlignment" c:type="GstVideoAlignment">
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<doc xml:space="preserve">Extra alignment paramters for the memory of video buffers. This
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<doc xml:space="preserve">Extra alignment parameters for the memory of video buffers. This
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structure is usually used to configure the bufferpool if it supports the
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#GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT.</doc>
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<field name="padding_top" writable="1">
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