gstreamer 3.1.6-x64-mingw32 → 3.1.7-x64-mingw32

data/vendor/local/share/gir-1.0/GstAllocators-1.0.gir

@@ -22,6 +22,102 @@ and/or use gtk-doc annotations.  -->
     <constant name="ALLOCATOR_FD" value="fd" c:type="GST_ALLOCATOR_FD">
       <type name="utf8" c:type="gchar*"/>
     </constant>
+    <constant name="CAPS_FEATURE_MEMORY_DMABUF"
+              value="memory:DMABuf"
+              c:type="GST_CAPS_FEATURE_MEMORY_DMABUF"
+              version="1.11">
+      <doc xml:space="preserve">Constant that defines the caps feature name for DMA buffer sharing.
+
+It has to be used for non-mappable dma-buf only, i.e. when the underlying
+memory is not mappable to user space. Or when the mapped memory contains
+non meaningful data. It can be the case for protected content or when the
+user wants explicitly avoid any software post processing.
+
+In these cases all elements between the exported and the importer has to work
+in passthrough mode. This is done by adding this caps feature.
+
+When the memory is mappable for read and write requests then it is assumes
+to be a fast path and so this caps feature should not be used. Though
+according to the dma-buf protocal, while it is mapped it prevents the
+exporter to migrate the buffer.
+
+This caps feature should not serve at all the purpose of selecting the
+@GST_ALLOCATOR_DMABUF allocator during caps negotiation.
+When the exporter is the upstream element from the importer point of view,
+the exporter should try to map the dma buffer at runtime (preferrably during
+decide_allocation phase). When it succeeds for #GST_MAP_READWRITE this caps
+feature should not be used. This allows scalers, color converts and any image
+processing filters to work directly on the dma buffer.
+In this case the importer element should check all incomming memory using
+gst_is_dmabuf_memory().</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <class name="DmaBufAllocator"
+           c:symbol-prefix="dmabuf_allocator"
+           c:type="GstDmaBufAllocator"
+           version="1.12"
+           parent="FdAllocator"
+           glib:type-name="GstDmaBufAllocator"
+           glib:get-type="gst_dmabuf_allocator_get_type"
+           glib:type-struct="DmaBufAllocatorClass">
+      <doc xml:space="preserve">Base class for allocators with dmabuf-backed memory</doc>
+      <constructor name="new"
+                   c:identifier="gst_dmabuf_allocator_new"
+                   version="1.2">
+        <doc xml:space="preserve">Return a new dmabuf allocator.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a new dmabuf allocator, or NULL if the allocator
+   isn't available. Use gst_object_unref() to release the allocator after
+   usage</doc>
+          <type name="Gst.Allocator" c:type="GstAllocator*"/>
+        </return-value>
+      </constructor>
+      <function name="alloc"
+                c:identifier="gst_dmabuf_allocator_alloc"
+                version="1.2">
+        <doc xml:space="preserve">Return a %GstMemory that wraps a dmabuf file descriptor.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a GstMemory based on @allocator.
+When the buffer will be released dmabuf allocator will close the @fd.
+The memory is only mmapped on gst_buffer_mmap() request.</doc>
+          <type name="Gst.Memory" c:type="GstMemory*"/>
+        </return-value>
+        <parameters>
+          <parameter name="allocator" transfer-ownership="none">
+            <doc xml:space="preserve">allocator to be used for this memory</doc>
+            <type name="Gst.Allocator" c:type="GstAllocator*"/>
+          </parameter>
+          <parameter name="fd" transfer-ownership="none">
+            <doc xml:space="preserve">dmabuf file descriptor</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="size" transfer-ownership="none">
+            <doc xml:space="preserve">memory size</doc>
+            <type name="gsize" c:type="gsize"/>
+          </parameter>
+        </parameters>
+      </function>
+      <field name="parent">
+        <type name="FdAllocator" c:type="GstFdAllocator"/>
+      </field>
+      <field name="_gst_reserved" readable="0" private="1">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </class>
+    <record name="DmaBufAllocatorClass"
+            c:type="GstDmaBufAllocatorClass"
+            glib:is-gtype-struct-for="DmaBufAllocator">
+      <field name="parent_class">
+        <type name="FdAllocatorClass" c:type="GstFdAllocatorClass"/>
+      </field>
+      <field name="_gst_reserved" readable="0" private="1">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+          <type name="gpointer" c:type="gpointer"/>
+        </array>
+      </field>
+    </record>
     <class name="FdAllocator"
            c:symbol-prefix="fd_allocator"
            c:type="GstFdAllocator"
@@ -107,42 +203,6 @@ The memory is only mmapped on gst_buffer_mmap() request.</doc>
        the memory is freed. Since: 1.10.</doc>
       </member>
     </bitfield>
-    <function name="dmabuf_allocator_alloc"
-              c:identifier="gst_dmabuf_allocator_alloc"
-              version="1.2">
-      <doc xml:space="preserve">Return a %GstMemory that wraps a dmabuf file descriptor.</doc>
-      <return-value transfer-ownership="full">
-        <doc xml:space="preserve">a GstMemory based on @allocator.
-When the buffer will be released dmabuf allocator will close the @fd.
-The memory is only mmapped on gst_buffer_mmap() request.</doc>
-        <type name="Gst.Memory" c:type="GstMemory*"/>
-      </return-value>
-      <parameters>
-        <parameter name="allocator" transfer-ownership="none">
-          <doc xml:space="preserve">allocator to be used for this memory</doc>
-          <type name="Gst.Allocator" c:type="GstAllocator*"/>
-        </parameter>
-        <parameter name="fd" transfer-ownership="none">
-          <doc xml:space="preserve">dmabuf file descriptor</doc>
-          <type name="gint" c:type="gint"/>
-        </parameter>
-        <parameter name="size" transfer-ownership="none">
-          <doc xml:space="preserve">memory size</doc>
-          <type name="gsize" c:type="gsize"/>
-        </parameter>
-      </parameters>
-    </function>
-    <function name="dmabuf_allocator_new"
-              c:identifier="gst_dmabuf_allocator_new"
-              version="1.2">
-      <doc xml:space="preserve">Return a new dmabuf allocator.</doc>
-      <return-value transfer-ownership="full">
-        <doc xml:space="preserve">a new dmabuf allocator, or NULL if the allocator
-   isn't available. Use gst_object_unref() to release the allocator after
-   usage</doc>
-        <type name="Gst.Allocator" c:type="GstAllocator*"/>
-      </return-value>
-    </function>
     <function name="dmabuf_memory_get_fd"
               c:identifier="gst_dmabuf_memory_get_fd"
               version="1.2">

data/vendor/local/share/gir-1.0/GstApp-1.0.gir

@@ -209,6 +209,21 @@ Call gst_sample_unref() after usage.</doc>
           </parameter>
         </parameters>
       </virtual-method>
+      <method name="get_buffer_list_support"
+              c:identifier="gst_app_sink_get_buffer_list_support"
+              version="1.12">
+        <doc xml:space="preserve">Check if @appsink supports buffer lists.</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">%TRUE if @appsink supports buffer lists.</doc>
+          <type name="gboolean" c:type="gboolean"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="appsink" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAppSink</doc>
+            <type name="AppSink" c:type="GstAppSink*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
       <method name="get_caps" c:identifier="gst_app_sink_get_caps">
         <doc xml:space="preserve">Get the configured caps on @appsink.</doc>
         <return-value transfer-ownership="full">
@@ -352,6 +367,27 @@ If an EOS event was received before any buffers, this function returns
           </instance-parameter>
         </parameters>
       </method>
+      <method name="set_buffer_list_support"
+              c:identifier="gst_app_sink_set_buffer_list_support"
+              version="1.12">
+        <doc xml:space="preserve">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>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="appsink" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAppSink</doc>
+            <type name="AppSink" c:type="GstAppSink*"/>
+          </instance-parameter>
+          <parameter name="enable_lists" transfer-ownership="none">
+            <doc xml:space="preserve">enable or disable buffer list support</doc>
+            <type name="gboolean" c:type="gboolean"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_callbacks"
               c:identifier="gst_app_sink_set_callbacks"
               introspectable="0">
@@ -544,6 +580,9 @@ Call gst_sample_unref() after usage.</doc>
           </parameter>
         </parameters>
       </method>
+      <property name="buffer-list" writable="1" transfer-ownership="none">
+        <type name="gboolean" c:type="gboolean"/>
+      </property>
       <property name="caps" writable="1" transfer-ownership="none">
         <type name="Gst.Caps"/>
       </property>

data/vendor/local/share/gir-1.0/GstAudio-1.0.gir

@@ -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"
@@ -656,37 +658,45 @@ discontinuity happens.</doc>
     </callback>
     <enumeration name="AudioBaseSinkDiscontReason"
                  version="1.6"
+                 glib:type-name="GstAudioBaseSinkDiscontReason"
+                 glib:get-type="gst_audio_base_sink_discont_reason_get_type"
                  c:type="GstAudioBaseSinkDiscontReason">
       <doc xml:space="preserve">Different possible reasons for discontinuities. This enum is useful for the custom
 slave method.</doc>
       <member name="no_discont"
               value="0"
-              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT">
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT"
+              glib:nick="no-discont">
         <doc xml:space="preserve">No discontinuity occurred</doc>
       </member>
       <member name="new_caps"
               value="1"
-              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS">
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS"
+              glib:nick="new-caps">
         <doc xml:space="preserve">New caps are set, causing renegotiotion</doc>
       </member>
       <member name="flush"
               value="2"
-              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH">
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH"
+              glib:nick="flush">
         <doc xml:space="preserve">Samples have been flushed</doc>
       </member>
       <member name="sync_latency"
               value="3"
-              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY">
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY"
+              glib:nick="sync-latency">
         <doc xml:space="preserve">Sink was synchronized to the estimated latency (occurs during initialization)</doc>
       </member>
       <member name="alignment"
               value="4"
-              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT">
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT"
+              glib:nick="alignment">
         <doc xml:space="preserve">Aligning buffers failed because the timestamps are too discontinuous</doc>
       </member>
       <member name="device_failure"
               value="5"
-              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE">
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE"
+              glib:nick="device-failure">
         <doc xml:space="preserve">Audio output device experienced and recovered from an error but introduced latency in the process (see also @gst_audio_base_sink_report_device_failure())</doc>
       </member>
     </enumeration>
@@ -928,7 +938,7 @@ not selected as the pipeline clock.</doc>
       </member>
       <member name="re_timestamp"
               value="1"
-              c:identifier="GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP"
+              c:identifier="GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP"
               glib:nick="re-timestamp">
         <doc xml:space="preserve">Retimestamp output buffers with master
 clock time.</doc>
@@ -954,60 +964,49 @@ drifts too much.</doc>
            glib:type-name="GstAudioCdSrc"
            glib:get-type="gst_audio_cd_src_get_type"
            glib:type-struct="AudioCdSrcClass">
-      <doc xml:space="preserve">&lt;para&gt;
-Provides a base class for CD digital audio (CDDA) sources, which handles
+      <doc xml:space="preserve">Provides a base class for CD digital audio (CDDA) sources, which handles
 things like seeking, querying, discid calculation, tags, and buffer
 timestamping.
-&lt;/para&gt;
-&lt;refsect2&gt;
-&lt;title&gt;Using GstAudioCdSrc-based elements in applications&lt;/title&gt;
-&lt;para&gt;
+
+## Using GstAudioCdSrc-based elements in applications
+
 GstAudioCdSrc registers two #GstFormat&lt;!-- --&gt;s of its own, namely
 the "track" format and the "sector" format. Applications will usually
 only find the "track" format interesting. You can retrieve that #GstFormat
 for use in seek events or queries with gst_format_get_by_nick("track").
-&lt;/para&gt;
-&lt;para&gt;
+
 In order to query the number of tracks, for example, an application would
 set the CDDA source element to READY or PAUSED state and then query the
 the number of tracks via gst_element_query_duration() using the track
 format acquired above. Applications can query the currently playing track
 in the same way.
-&lt;/para&gt;
-&lt;para&gt;
+
 Alternatively, applications may retrieve the currently playing track and
 the total number of tracks from the taglist that will posted on the bus
 whenever the CD is opened or the currently playing track changes. The
 taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags.
-&lt;/para&gt;
-&lt;para&gt;
+
 Applications playing back CD audio using playbin and cdda://n URIs should
 issue a seek command in track format to change between tracks, rather than
 setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin
 involves closing and re-opening the CD device, which is much much slower).
-&lt;/para&gt;
-&lt;refsect2&gt;
-&lt;/refsect2&gt;
-&lt;title&gt;Tags and meta-information&lt;/title&gt;
-&lt;para&gt;
+
+## Tags and meta-information
+
 CDDA sources will automatically emit a number of tags, details about which
 can be found in the libgsttag documentation. Those tags are:
 #GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL,
 #GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL,
 among others.
-&lt;/para&gt;
-&lt;/refsect2&gt;
-&lt;refsect2&gt;
-&lt;title&gt;Tracks and Table of Contents (TOC)&lt;/title&gt;
-&lt;para&gt;
+
+## Tracks and Table of Contents (TOC)
+
 Applications will be informed of the available tracks via a TOC message
 on the pipeline's #GstBus. The #GstToc will contain a #GstTocEntry for
 each track, with information about each track. The duration for each
 track can be retrieved via the #GST_TAG_DURATION tag from each entry's
 tag list, or calculated via gst_toc_entry_get_start_stop_times().
-The track entries in the TOC will be sorted by track number.
-&lt;/para&gt;
-&lt;/refsect2&gt;</doc>
+The track entries in the TOC will be sorted by track number.</doc>
       <implements name="Gst.URIHandler"/>
       <virtual-method name="close">
         <return-value transfer-ownership="none">
@@ -1158,13 +1157,15 @@ copy of the structure (and take ownership of the taglist if there is one).</doc>
 EOS handling and seeking.</doc>
       <member name="normal"
               value="0"
-              c:identifier="Stream consists of a single track"
+              c:identifier="GST_AUDIO_CD_SRC_MODE_NORMAL"
               glib:nick="normal">
+        <doc xml:space="preserve">each single track is a stream</doc>
       </member>
       <member name="continuous"
               value="1"
-              c:identifier="Stream consists of the whole disc"
+              c:identifier="GST_AUDIO_CD_SRC_MODE_CONTINUOUS"
               glib:nick="continuous">
+        <doc xml:space="preserve">the entire disc is a single stream</doc>
       </member>
     </enumeration>
     <record name="AudioCdSrcPrivate"
@@ -1654,24 +1655,24 @@ calculated it will call @func with @user_data. When @func returns
           </parameter>
         </parameters>
       </constructor>
-      <function name="adjust" c:identifier="gst_audio_clock_adjust">
+      <method name="adjust" c:identifier="gst_audio_clock_adjust">
         <doc xml:space="preserve">Adjust @time with the internal offset of the audio clock.</doc>
         <return-value transfer-ownership="none">
           <doc xml:space="preserve">@time adjusted with the internal offset.</doc>
           <type name="Gst.ClockTime" c:type="GstClockTime"/>
         </return-value>
         <parameters>
-          <parameter name="clock" transfer-ownership="none">
+          <instance-parameter name="clock" transfer-ownership="none">
             <doc xml:space="preserve">a #GstAudioClock</doc>
-            <type name="Gst.Clock" c:type="GstClock*"/>
-          </parameter>
+            <type name="AudioClock" c:type="GstAudioClock*"/>
+          </instance-parameter>
           <parameter name="time" transfer-ownership="none">
             <doc xml:space="preserve">a #GstClockTime</doc>
             <type name="Gst.ClockTime" c:type="GstClockTime"/>
           </parameter>
         </parameters>
-      </function>
-      <function name="get_time" c:identifier="gst_audio_clock_get_time">
+      </method>
+      <method name="get_time" c:identifier="gst_audio_clock_get_time">
         <doc xml:space="preserve">Report the time as returned by the #GstAudioClockGetTimeFunc without applying
 any offsets.</doc>
         <return-value transfer-ownership="none">
@@ -1679,13 +1680,13 @@ any offsets.</doc>
           <type name="Gst.ClockTime" c:type="GstClockTime"/>
         </return-value>
         <parameters>
-          <parameter name="clock" transfer-ownership="none">
+          <instance-parameter name="clock" transfer-ownership="none">
             <doc xml:space="preserve">a #GstAudioClock</doc>
-            <type name="Gst.Clock" c:type="GstClock*"/>
-          </parameter>
+            <type name="AudioClock" c:type="GstAudioClock*"/>
+          </instance-parameter>
         </parameters>
-      </function>
-      <function name="invalidate" c:identifier="gst_audio_clock_invalidate">
+      </method>
+      <method name="invalidate" c:identifier="gst_audio_clock_invalidate">
         <doc xml:space="preserve">Invalidate the clock function. Call this function when the provided
 #GstAudioClockGetTimeFunc cannot be called anymore, for example, when the
 user_data becomes invalid.
@@ -1696,12 +1697,12 @@ the rest of its lifetime.</doc>
           <type name="none" c:type="void"/>
         </return-value>
         <parameters>
-          <parameter name="clock" transfer-ownership="none">
+          <instance-parameter name="clock" transfer-ownership="none">
             <doc xml:space="preserve">a #GstAudioClock</doc>
-            <type name="Gst.Clock" c:type="GstClock*"/>
-          </parameter>
+            <type name="AudioClock" c:type="GstAudioClock*"/>
+          </instance-parameter>
         </parameters>
-      </function>
+      </method>
       <method name="reset" c:identifier="gst_audio_clock_reset">
         <doc xml:space="preserve">Inform @clock that future calls to #GstAudioClockGetTimeFunc will return values
 starting from @time. The clock will update an internal offset to make sure that
@@ -1937,6 +1938,21 @@ are matching and @in and @out point to enough memory.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="supports_inplace"
+              c:identifier="gst_audio_converter_supports_inplace">
+        <doc xml:space="preserve">Returns whether the audio converter can perform the conversion in-place.
+The return value would be typically input to gst_base_transform_set_in_place()</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">%TRUE when the conversion can be done in place.</doc>
+          <type name="gboolean" c:type="gboolean"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="convert" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioConverter</doc>
+            <type name="AudioConverter" c:type="GstAudioConverter*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
       <method name="update_config"
               c:identifier="gst_audio_converter_update_config">
         <doc xml:space="preserve">Set @in_rate, @out_rate and @config as extra configuration for @convert.
@@ -2049,72 +2065,48 @@ parameters for details about the options and values.</doc>
 raw audio samples.
 
 GstAudioDecoder and subclass should cooperate as follows.
-&lt;orderedlist&gt;
-&lt;listitem&gt;
-  &lt;itemizedlist&gt;&lt;title&gt;Configuration&lt;/title&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    Initially, GstAudioDecoder calls @start when the decoder element
+
+## Configuration
+
+  * Initially, GstAudioDecoder calls @start when the decoder element
     is activated, which allows subclass to perform any global setup.
     Base class (context) parameters can already be set according to subclass
     capabilities (or possibly upon receive more information in subsequent
     @set_format).
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    GstAudioDecoder calls @set_format to inform subclass of the format
+  * GstAudioDecoder calls @set_format to inform subclass of the format
     of input audio data that it is about to receive.
     While unlikely, it might be called more than once, if changing input
     parameters require reconfiguration.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    GstAudioDecoder calls @stop at end of all processing.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/itemizedlist&gt;
-&lt;/listitem&gt;
+  * GstAudioDecoder calls @stop at end of all processing.
+
 As of configuration stage, and throughout processing, GstAudioDecoder
 provides various (context) parameters, e.g. describing the format of
 output audio data (valid when output caps have been set) or current parsing state.
 Conversely, subclass can and should configure context to inform
 base class of its expectation w.r.t. buffer handling.
-&lt;listitem&gt;
-  &lt;itemizedlist&gt;
-  &lt;title&gt;Data processing&lt;/title&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      Base class gathers input data, and optionally allows subclass
+
+## Data processing
+    * Base class gathers input data, and optionally allows subclass
       to parse this into subsequently manageable (as defined by subclass)
       chunks.  Such chunks are subsequently referred to as 'frames',
       though they may or may not correspond to 1 (or more) audio format frame.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      Input frame is provided to subclass' @handle_frame.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      If codec processing results in decoded data, subclass should call
+    * Input frame is provided to subclass' @handle_frame.
+    * If codec processing results in decoded data, subclass should call
       @gst_audio_decoder_finish_frame to have decoded data pushed
       downstream.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      Just prior to actually pushing a buffer downstream,
+    * Just prior to actually pushing a buffer downstream,
       it is passed to @pre_push.  Subclass should either use this callback
       to arrange for additional downstream pushing or otherwise ensure such
       custom pushing occurs after at least a method call has finished since
       setting src pad caps.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      During the parsing process GstAudioDecoderClass will handle both
+    * During the parsing process GstAudioDecoderClass will handle both
       srcpad and sinkpad events. Sink events will be passed to subclass
       if @event callback has been provided.
-    &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/itemizedlist&gt;
-&lt;/listitem&gt;
-&lt;listitem&gt;
-  &lt;itemizedlist&gt;&lt;title&gt;Shutdown phase&lt;/title&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    GstAudioDecoder class calls @stop to inform the subclass that data
+
+## Shutdown phase
+
+  * GstAudioDecoder class calls @stop to inform the subclass that data
     parsing will be stopped.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/itemizedlist&gt;
-&lt;/listitem&gt;
-&lt;/orderedlist&gt;
 
 Subclass is responsible for providing pad template caps for
 source and sink pads. The pads need to be named "sink" and "src". It also
@@ -2144,23 +2136,17 @@ approach caters for seeking and duration reporting using estimated input
 bitrates.
 
 Things that subclass need to take care of:
-&lt;itemizedlist&gt;
-  &lt;listitem&gt;&lt;para&gt;Provide pad templates&lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Set source pad caps when appropriate
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Set user-configurable properties to sane defaults for format and
+
+  * Provide pad templates
+  * Set source pad caps when appropriate
+  * Set user-configurable properties to sane defaults for format and
      implementing codec at hand, and convey some subclass capabilities and
      expectations in context.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Accept data in @handle_frame and provide encoded results to
+
+  * Accept data in @handle_frame and provide encoded results to
      @gst_audio_decoder_finish_frame.  If it is prepared to perform
      PLC, it should also accept NULL data in @handle_frame and provide for
-     data for indicated duration.
-  &lt;/para&gt;&lt;/listitem&gt;
-&lt;/itemizedlist&gt;</doc>
+     data for indicated duration.</doc>
       <virtual-method name="close">
         <return-value transfer-ownership="none">
           <type name="gboolean" c:type="gboolean"/>
@@ -3400,65 +3386,46 @@ of the results.</doc>
 encoded audio data.
 
 GstAudioEncoder and subclass should cooperate as follows.
-&lt;orderedlist&gt;
-&lt;listitem&gt;
-  &lt;itemizedlist&gt;&lt;title&gt;Configuration&lt;/title&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    Initially, GstAudioEncoder calls @start when the encoder element
+
+## Configuration
+
+  * Initially, GstAudioEncoder calls @start when the encoder element
     is activated, which allows subclass to perform any global setup.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    GstAudioEncoder calls @set_format to inform subclass of the format
+
+  * GstAudioEncoder calls @set_format to inform subclass of the format
     of input audio data that it is about to receive.  Subclass should
     setup for encoding and configure various base class parameters
     appropriately, notably those directing desired input data handling.
     While unlikely, it might be called more than once, if changing input
     parameters require reconfiguration.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    GstAudioEncoder calls @stop at end of all processing.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/itemizedlist&gt;
-&lt;/listitem&gt;
+
+  * GstAudioEncoder calls @stop at end of all processing.
+
 As of configuration stage, and throughout processing, GstAudioEncoder
 maintains various parameters that provide required context,
 e.g. describing the format of input audio data.
 Conversely, subclass can and should configure these context parameters
 to inform base class of its expectation w.r.t. buffer handling.
-&lt;listitem&gt;
-  &lt;itemizedlist&gt;
-  &lt;title&gt;Data processing&lt;/title&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      Base class gathers input sample data (as directed by the context's
+
+## Data processing
+
+    * Base class gathers input sample data (as directed by the context's
       frame_samples and frame_max) and provides this to subclass' @handle_frame.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      If codec processing results in encoded data, subclass should call
+    * If codec processing results in encoded data, subclass should call
       gst_audio_encoder_finish_frame() to have encoded data pushed
       downstream. Alternatively, it might also call
       gst_audio_encoder_finish_frame() (with a NULL buffer and some number of
       dropped samples) to indicate dropped (non-encoded) samples.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      Just prior to actually pushing a buffer downstream,
+    * Just prior to actually pushing a buffer downstream,
       it is passed to @pre_push.
-    &lt;/para&gt;&lt;/listitem&gt;
-    &lt;listitem&gt;&lt;para&gt;
-      During the parsing process GstAudioEncoderClass will handle both
+    * During the parsing process GstAudioEncoderClass will handle both
       srcpad and sinkpad events. Sink events will be passed to subclass
       if @event callback has been provided.
-    &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/itemizedlist&gt;
-&lt;/listitem&gt;
-&lt;listitem&gt;
-  &lt;itemizedlist&gt;&lt;title&gt;Shutdown phase&lt;/title&gt;
-  &lt;listitem&gt;&lt;para&gt;
-    GstAudioEncoder class calls @stop to inform the subclass that data
+
+## Shutdown phase
+
+  * GstAudioEncoder class calls @stop to inform the subclass that data
     parsing will be stopped.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/itemizedlist&gt;
-&lt;/listitem&gt;
-&lt;/orderedlist&gt;
 
 Subclass is responsible for providing pad template caps for
 source and sink pads. The pads need to be named "sink" and "src". It also
@@ -3497,25 +3464,16 @@ including buffer) and OFFSET_END to corresponding timestamp (as determined
 by same sample count and sample rate).
 
 Things that subclass need to take care of:
-&lt;itemizedlist&gt;
-  &lt;listitem&gt;&lt;para&gt;Provide pad templates&lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Set source pad caps when appropriate
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Inform base class of buffer processing needs using context's
+
+  * Provide pad templates
+  * Set source pad caps when appropriate
+  * Inform base class of buffer processing needs using context's
      frame_samples and frame_bytes.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Set user-configurable properties to sane defaults for format and
+  * Set user-configurable properties to sane defaults for format and
      implementing codec at hand, e.g. those controlling timestamp behaviour
      and discontinuity processing.
-  &lt;/para&gt;&lt;/listitem&gt;
-  &lt;listitem&gt;&lt;para&gt;
-     Accept data in @handle_frame and provide encoded results to
-     gst_audio_encoder_finish_frame().
-  &lt;/para&gt;&lt;/listitem&gt;
-&lt;/itemizedlist&gt;</doc>
+  * Accept data in @handle_frame and provide encoded results to
+     gst_audio_encoder_finish_frame().</doc>
       <implements name="Gst.Preset"/>
       <virtual-method name="close">
         <return-value transfer-ownership="none">
@@ -6094,19 +6052,14 @@ for @quality in @options.</doc>
            glib:type-name="GstAudioRingBuffer"
            glib:get-type="gst_audio_ring_buffer_get_type"
            glib:type-struct="AudioRingBufferClass">
-      <doc xml:space="preserve">&lt;refsect2&gt;
-&lt;para&gt;
-This object is the base class for audio ringbuffers used by the base
+      <doc xml:space="preserve">This object is the base class for audio ringbuffers used by the base
 audio source and sink classes.
-&lt;/para&gt;
-&lt;para&gt;
+
 The ringbuffer abstracts a circular buffer of data. One reader and
 one writer can operate on the data from different threads in a lockfree
 manner. The base class is sufficiently flexible to be used as an
 abstraction for DMA based ringbuffers as well as a pure software
-implementations.
-&lt;/para&gt;
-&lt;/refsect2&gt;</doc>
+implementations.</doc>
       <function name="debug_spec_buff"
                 c:identifier="gst_audio_ring_buffer_debug_spec_buff">
         <doc xml:space="preserve">Print debug info about the buffer sized in @spec to the debug log.</doc>
@@ -6821,7 +6774,9 @@ MT safe.</doc>
         </parameters>
       </method>
       <method name="set_callback"
-              c:identifier="gst_audio_ring_buffer_set_callback">
+              c:identifier="gst_audio_ring_buffer_set_callback"
+              shadowed-by="set_callback_full"
+              introspectable="0">
         <doc xml:space="preserve">Sets the given callback function on the buffer. This function
 will be called every time a segment has been written to a device.
 
@@ -6836,7 +6791,8 @@ MT safe.</doc>
           </instance-parameter>
           <parameter name="cb"
                      transfer-ownership="none"
-                     scope="async"
+                     nullable="1"
+                     allow-none="1"
                      closure="1">
             <doc xml:space="preserve">the callback to set</doc>
             <type name="AudioRingBufferCallback"
@@ -6851,6 +6807,46 @@ MT safe.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="set_callback_full"
+              c:identifier="gst_audio_ring_buffer_set_callback_full"
+              shadows="set_callback"
+              version="1.12">
+        <doc xml:space="preserve">Sets the given callback function on the buffer. This function
+will be called every time a segment has been written to a device.
+
+MT safe.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="buf" transfer-ownership="none">
+            <doc xml:space="preserve">the #GstAudioRingBuffer to set the callback on</doc>
+            <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
+          </instance-parameter>
+          <parameter name="cb"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1"
+                     scope="notified"
+                     closure="1"
+                     destroy="2">
+            <doc xml:space="preserve">the callback to set</doc>
+            <type name="AudioRingBufferCallback"
+                  c:type="GstAudioRingBufferCallback"/>
+          </parameter>
+          <parameter name="user_data"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">user data passed to the callback</doc>
+            <type name="gpointer" c:type="gpointer"/>
+          </parameter>
+          <parameter name="notify" transfer-ownership="none" scope="async">
+            <doc xml:space="preserve">function to be called when @user_data is no longer needed</doc>
+            <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_channel_positions"
               c:identifier="gst_audio_ring_buffer_set_channel_positions">
         <doc xml:space="preserve">Tell the ringbuffer about the device's channel positions. This must
@@ -7038,8 +7034,11 @@ MT safe.</doc>
       <field name="active" readable="0" private="1">
         <type name="gboolean" c:type="gboolean"/>
       </field>
+      <field name="cb_data_notify" readable="0" private="1">
+        <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
+      </field>
       <field name="_gst_reserved" readable="0" private="1">
-        <array zero-terminated="0" c:type="gpointer" fixed-size="4">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="3">
           <type name="gpointer" c:type="gpointer"/>
         </array>
       </field>
@@ -7372,13 +7371,31 @@ with a flush or stop.</doc>
               value="10"
               c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC"
               glib:nick="mpeg2-aac">
-        <doc xml:space="preserve">samples in MPEG-2 AAC format</doc>
+        <doc xml:space="preserve">samples in MPEG-2 AAC ADTS format</doc>
       </member>
       <member name="mpeg4_aac"
               value="11"
               c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC"
               glib:nick="mpeg4-aac">
-        <doc xml:space="preserve">samples in MPEG-4 AAC format</doc>
+        <doc xml:space="preserve">samples in MPEG-4 AAC ADTS format</doc>
+      </member>
+      <member name="mpeg2_aac_raw"
+              value="12"
+              c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW"
+              glib:nick="mpeg2-aac-raw">
+        <doc xml:space="preserve">samples in MPEG-2 AAC raw format (Since 1.12)</doc>
+      </member>
+      <member name="mpeg4_aac_raw"
+              value="13"
+              c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW"
+              glib:nick="mpeg4-aac-raw">
+        <doc xml:space="preserve">samples in MPEG-4 AAC raw format (Since 1.12)</doc>
+      </member>
+      <member name="flac"
+              value="14"
+              c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC"
+              glib:nick="flac">
+        <doc xml:space="preserve">samples in FLAC format (Since 1.12)</doc>
       </member>
     </enumeration>
     <record name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec">
@@ -7465,37 +7482,20 @@ with a flush or stop.</doc>
       <doc xml:space="preserve">This is the most simple base class for audio sinks that only requires
 subclasses to implement a set of simple functions:
 
-&lt;variablelist&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;open()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Open the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;prepare()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Configure the device with the specified format.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;write()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Write samples to the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;reset()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Unblock writes and flush the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;delay()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Get the number of samples written but not yet played
-    by the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;unprepare()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Undo operations done by prepare.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;close()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Close the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-&lt;/variablelist&gt;
+* `open()` :Open the device.
+
+* `prepare()` :Configure the device with the specified format.
+
+* `write()` :Write samples to the device.
+
+* `reset()` :Unblock writes and flush the device.
+
+* `delay()` :Get the number of samples written but not yet played
+by the device.
+
+* `unprepare()` :Undo operations done by prepare.
+
+* `close()` :Close the device.
 
 All scheduling of samples and timestamps is done in this base class
 together with #GstAudioBaseSink using a default implementation of a
@@ -7715,37 +7715,13 @@ together with #GstAudioBaseSink using a default implementation of a
       <doc xml:space="preserve">This is the most simple base class for audio sources that only requires
 subclasses to implement a set of simple functions:
 
-&lt;variablelist&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;open()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Open the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;prepare()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Configure the device with the specified format.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;read()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Read samples from the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;reset()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Unblock reads and flush the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;delay()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Get the number of samples in the device but not yet read.
-    &lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;unprepare()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Undo operations done by prepare.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-  &lt;varlistentry&gt;
-    &lt;term&gt;close()&lt;/term&gt;
-    &lt;listitem&gt;&lt;para&gt;Close the device.&lt;/para&gt;&lt;/listitem&gt;
-  &lt;/varlistentry&gt;
-&lt;/variablelist&gt;
+* `open()` :Open the device.
+* `prepare()` :Configure the device with the specified format.
+* `read()` :Read samples from the device.
+* `reset()` :Unblock reads and flush the device.
+* `delay()` :Get the number of samples in the device but not yet read.
+* `unprepare()` :Undo operations done by prepare.
+* `close()` :Close the device.
 
 All scheduling of samples and timestamps is done in this base class
 together with #GstAudioBaseSrc using a default implementation of a
@@ -7989,12 +7965,9 @@ functionality.</doc>
                glib:type-name="GstStreamVolume"
                glib:get-type="gst_stream_volume_get_type"
                glib:type-struct="StreamVolumeInterface">
-      <doc xml:space="preserve">&lt;refsect2&gt;
-&lt;para&gt;
-This interface is implemented by elements that provide a stream volume. Examples for
+      <doc xml:space="preserve">This interface is implemented by elements that provide a stream volume. Examples for
 such elements are #volume and #playbin.
-&lt;/para&gt;
-&lt;para&gt;
+
 Applications can use this interface to get or set the current stream volume. For this
 the "volume" #GObject property can be used or the helper functions gst_stream_volume_set_volume()
 and gst_stream_volume_get_volume(). This volume is always a linear factor, i.e. 0.0 is muted
@@ -8004,13 +7977,10 @@ use a cubic volume.
 
 Separate from the volume the stream can also be muted by the "mute" #GObject property or
 gst_stream_volume_set_mute() and gst_stream_volume_get_mute().
-&lt;/para&gt;
-&lt;para&gt;
+
 Elements that provide some kind of stream volume should implement the "volume" and
 "mute" #GObject properties and handle setting and getting of them properly.
-The volume property is defined to be a linear volume factor.
-&lt;/para&gt;
-&lt;/refsect2&gt;</doc>
+The volume property is defined to be a linear volume factor.</doc>
       <function name="convert_volume"
                 c:identifier="gst_stream_volume_convert_volume">
         <return-value transfer-ownership="none">
@@ -8318,11 +8288,12 @@ in the order required by GStreamer.</doc>
           <doc xml:space="preserve">Only consider the GStreamer channel order.</doc>
           <type name="gboolean" c:type="gboolean"/>
         </parameter>
-        <parameter name="channel_mask" transfer-ownership="none">
+        <parameter name="channel_mask"
+                   direction="out"
+                   caller-allocates="0"
+                   transfer-ownership="full">
           <doc xml:space="preserve">the output channel mask</doc>
-          <array zero-terminated="0" c:type="guint64*">
-            <type name="guint64" c:type="guint64"/>
-          </array>
+          <type name="guint64" c:type="guint64*"/>
         </parameter>
       </parameters>
     </function>