Commit 385ff00d authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Update GIR files from 1.12.0 final release

parent 0dc9558e
......@@ -35175,14 +35175,24 @@ gst_element_set_state() and checked using gst_element_get_state().</doc>
glib:type-name="GstStateChange"
glib:get-type="gst_state_change_get_type"
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
sinks and -sources typically try to probe the device to constrain their
caps.
* The element opens the device (in case feature need to be probed).
#GST_STATE_CHANGE_READY_TO_PAUSED : state change from READY to PAUSED.
* The element opens the device (in case feature need to be probed).</doc>
</member>
<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.
Streaming threads are started.
* 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).
Sinks also block the dataflow when in PAUSED.
* A pipeline resets the running_time to 0.
* Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data.
#GST_STATE_CHANGE_PAUSED_TO_PLAYING: state change from PAUSED to PLAYING.
* Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data.</doc>
</member>
<member name="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.
* 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
......@@ -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.
* While streaming in PAUSED or PLAYING elements can create and remove
sometimes pads.
* Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS.
#GST_STATE_CHANGE_PLAYING_TO_PAUSED: state change from PLAYING to PAUSED.
* Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS.</doc>
</member>
<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.
* The pipeline calculates the running_time based on the last selected
#GstClock and the base_time. It stores this information to continue
playback when going back to the PLAYING state.
* Sinks unblock any #GstClock wait calls.
* 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
......@@ -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
when going back to the PLAYING state. The EOS messages are queued in
#GstBin containers.
* Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL.
#GST_STATE_CHANGE_PAUSED_TO_READY : state change from PAUSED to READY.
* Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL.</doc>
</member>
<member name="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.
* Elements unblock any waits on devices
* Chain or get_range functions return %GST_FLOW_FLUSHING.
* The element pads are deactivated so that streaming becomes impossible and
all streaming threads are stopped.
* The sink forgets all negotiated formats
* Elements remove all sometimes pads
#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">
* Elements remove all sometimes pads</doc>
</member>
<member name="ready_to_null"
value="17"
c:identifier="GST_STATE_CHANGE_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>
</enumeration>
<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>
<type name="gint" c:type="gint"/>
</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>
<type name="gint" c:type="gint"/>
</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>
<type name="gint" c:type="gint"/>
</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:
Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc>
<type name="gint" c:type="gint"/>
......@@ -189,6 +189,8 @@ rates.
<constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"
value="GstAudioResampler.stop-attenutation"
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*"/>
</constant>
<constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH"
......
......@@ -2683,13 +2683,13 @@ in debugging.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MICRO"
value="90"
value="0"
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>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MINOR"
value="11"
value="12"
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>
<type name="gint" c:type="gint"/>
......
......@@ -1804,7 +1804,7 @@ quatization errors.</doc>
</function>
</record>
<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
#GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT.</doc>
<field name="padding_top" writable="1">
......
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