gstreamer 2.2.5-x86-mingw32 → 3.0.0-x86-mingw32

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

@@ -19,6 +19,90 @@ and/or use gtk-doc annotations.  -->
               c:type="GST_ALLOCATOR_DMABUF">
       <type name="utf8" c:type="gchar*"/>
     </constant>
+    <constant name="ALLOCATOR_FD" value="fd" c:type="GST_ALLOCATOR_FD">
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <class name="FdAllocator"
+           c:symbol-prefix="fd_allocator"
+           c:type="GstFdAllocator"
+           version="1.6"
+           parent="Gst.Allocator"
+           glib:type-name="GstFdAllocator"
+           glib:get-type="gst_fd_allocator_get_type"
+           glib:type-struct="FdAllocatorClass">
+      <doc xml:space="preserve">Base class for allocators with fd-backed memory</doc>
+      <constructor name="new"
+                   c:identifier="gst_fd_allocator_new"
+                   version="1.6">
+        <doc xml:space="preserve">Return a new fd allocator.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a new fd 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_fd_allocator_alloc"
+                version="1.6">
+        <doc xml:space="preserve">Return a %GstMemory that wraps a generic file descriptor.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a GstMemory based on @allocator.
+When the buffer will be released the 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"
+                     nullable="1"
+                     allow-none="1">
+            <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">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>
+          <parameter name="flags" transfer-ownership="none">
+            <doc xml:space="preserve">extra #GstFdMemoryFlags</doc>
+            <type name="FdMemoryFlags" c:type="GstFdMemoryFlags"/>
+          </parameter>
+        </parameters>
+      </function>
+      <field name="parent">
+        <type name="Gst.Allocator" c:type="GstAllocator"/>
+      </field>
+    </class>
+    <record name="FdAllocatorClass"
+            c:type="GstFdAllocatorClass"
+            glib:is-gtype-struct-for="FdAllocator">
+      <field name="parent_class">
+        <type name="Gst.AllocatorClass" c:type="GstAllocatorClass"/>
+      </field>
+    </record>
+    <bitfield name="FdMemoryFlags" version="1.6" c:type="GstFdMemoryFlags">
+      <doc xml:space="preserve">Various flags to control the operation of the fd backed memory.</doc>
+      <member name="none" value="0" c:identifier="GST_FD_MEMORY_FLAG_NONE">
+        <doc xml:space="preserve">no flag</doc>
+      </member>
+      <member name="keep_mapped"
+              value="1"
+              c:identifier="GST_FD_MEMORY_FLAG_KEEP_MAPPED">
+        <doc xml:space="preserve">once the memory is mapped,
+       keep it mapped until the memory is destroyed.</doc>
+      </member>
+      <member name="map_private"
+              value="2"
+              c:identifier="GST_FD_MEMORY_FLAG_MAP_PRIVATE">
+        <doc xml:space="preserve">do a private mapping instead of
+       the default shared mapping.</doc>
+      </member>
+    </bitfield>
     <function name="dmabuf_allocator_alloc"
               c:identifier="gst_dmabuf_allocator_alloc"
               version="1.2">
@@ -63,7 +147,9 @@ The memory is only mmapped on gst_buffer_mmap() request.</doc>
               version="1.2">
       <doc xml:space="preserve">Return the file descriptor associated with @mem.</doc>
       <return-value transfer-ownership="none">
-        <doc xml:space="preserve">the file descriptor associated with the memory, or -1</doc>
+        <doc xml:space="preserve">the file descriptor associated with the memory, or -1.  The file
+    descriptor is still owned by the GstMemory.  Use dup to take a copy
+    if you intend to use it beyond the lifetime of this GstMemory.</doc>
         <type name="gint" c:type="gint"/>
       </return-value>
       <parameters>
@@ -73,6 +159,22 @@ The memory is only mmapped on gst_buffer_mmap() request.</doc>
         </parameter>
       </parameters>
     </function>
+    <function name="fd_memory_get_fd"
+              c:identifier="gst_fd_memory_get_fd"
+              version="1.6">
+      <doc xml:space="preserve">Get the fd from @mem. Call gst_is_fd_memory() to check if @mem has
+an fd.</doc>
+      <return-value transfer-ownership="none">
+        <doc xml:space="preserve">the fd of @mem or -1 when there is no fd on @mem</doc>
+        <type name="gint" c:type="gint"/>
+      </return-value>
+      <parameters>
+        <parameter name="mem" transfer-ownership="none">
+          <doc xml:space="preserve">#GstMemory</doc>
+          <type name="Gst.Memory" c:type="GstMemory*"/>
+        </parameter>
+      </parameters>
+    </function>
     <function name="is_dmabuf_memory"
               c:identifier="gst_is_dmabuf_memory"
               version="1.2">
@@ -88,5 +190,21 @@ The memory is only mmapped on gst_buffer_mmap() request.</doc>
         </parameter>
       </parameters>
     </function>
+    <function name="is_fd_memory"
+              c:identifier="gst_is_fd_memory"
+              version="1.6">
+      <doc xml:space="preserve">Check if @mem is memory backed by an fd</doc>
+      <return-value transfer-ownership="none">
+        <doc xml:space="preserve">%TRUE when @mem has an fd that can be retrieved with
+gst_fd_memory_get_fd().</doc>
+        <type name="gboolean" c:type="gboolean"/>
+      </return-value>
+      <parameters>
+        <parameter name="mem" transfer-ownership="none">
+          <doc xml:space="preserve">#GstMemory</doc>
+          <type name="Gst.Memory" c:type="GstMemory*"/>
+        </parameter>
+      </parameters>
+    </function>
   </namespace>
 </repository>

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

@@ -105,7 +105,8 @@ If an EOS event was received before any buffers, this function returns
 This function blocks until a preroll sample or EOS is received or the appsink
 element is set to the READY/NULL state.</doc>
         <return-value transfer-ownership="full">
-          <doc xml:space="preserve">a #GstBuffer or NULL when the appsink is stopped or EOS.</doc>
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS.
+         Call gst_sample_unref() after usage.</doc>
           <type name="Gst.Sample" c:type="GstSample*"/>
         </return-value>
         <parameters>
@@ -128,7 +129,8 @@ especially when dealing with raw video frames.
 If an EOS event was received before any buffers, this function returns
 %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.</doc>
         <return-value transfer-ownership="full">
-          <doc xml:space="preserve">a #GstBuffer or NULL when the appsink is stopped or EOS.</doc>
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS.
+         Call gst_sample_unref() after usage.</doc>
           <type name="Gst.Sample" c:type="GstSample*"/>
         </return-value>
         <parameters>
@@ -230,7 +232,8 @@ If an EOS event was received before any buffers, this function returns
 This function blocks until a preroll sample or EOS is received or the appsink
 element is set to the READY/NULL state.</doc>
         <return-value transfer-ownership="full">
-          <doc xml:space="preserve">a #GstBuffer or NULL when the appsink is stopped or EOS.</doc>
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS.
+         Call gst_sample_unref() after usage.</doc>
           <type name="Gst.Sample" c:type="GstSample*"/>
         </return-value>
         <parameters>
@@ -253,7 +256,8 @@ especially when dealing with raw video frames.
 If an EOS event was received before any buffers, this function returns
 %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.</doc>
         <return-value transfer-ownership="full">
-          <doc xml:space="preserve">a #GstBuffer or NULL when the appsink is stopped or EOS.</doc>
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS.
+         Call gst_sample_unref() after usage.</doc>
           <type name="Gst.Sample" c:type="GstSample*"/>
         </return-value>
         <parameters>
@@ -577,7 +581,8 @@ gst_app_sink_set_callbacks().</doc>
       <field name="pull_preroll">
         <callback name="pull_preroll">
           <return-value transfer-ownership="full">
-            <doc xml:space="preserve">a #GstBuffer or NULL when the appsink is stopped or EOS.</doc>
+            <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS.
+         Call gst_sample_unref() after usage.</doc>
             <type name="Gst.Sample" c:type="GstSample*"/>
           </return-value>
           <parameters>
@@ -591,7 +596,8 @@ gst_app_sink_set_callbacks().</doc>
       <field name="pull_sample">
         <callback name="pull_sample">
           <return-value transfer-ownership="full">
-            <doc xml:space="preserve">a #GstBuffer or NULL when the appsink is stopped or EOS.</doc>
+            <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS.
+         Call gst_sample_unref() after usage.</doc>
             <type name="Gst.Sample" c:type="GstSample*"/>
           </return-value>
           <parameters>
@@ -618,17 +624,20 @@ gst_app_sink_set_callbacks().</doc>
            glib:get-type="gst_app_src_get_type"
            glib:type-struct="AppSrcClass">
       <doc xml:space="preserve">The appsrc element can be used by applications to insert data into a
-GStreamer pipeline. Unlike most GStreamer elements, Appsrc provides
+GStreamer pipeline. Unlike most GStreamer elements, appsrc provides
 external API functions.
 
 appsrc can be used by linking with the libgstapp library to access the
 methods directly or by using the appsrc action signals.
 
-Before operating appsrc, the caps property must be set to a fixed caps
+Before operating appsrc, the caps property must be set to fixed caps
 describing the format of the data that will be pushed with appsrc. An
 exception to this is when pushing buffers with unknown caps, in which case no
 caps should be set. This is typically true of file-like sources that push raw
-byte buffers.
+byte buffers. If you don't want to explicitly set the caps, you can use
+gst_app_src_push_sample. This method gets the caps associated with the
+sample and sets them on the appsrc replacing any previously set caps (if
+different from sample's caps).
 
 The main way of handing data to the appsrc element is by calling the
 gst_app_src_push_buffer() method or by emitting the push-buffer action signal.
@@ -656,16 +665,16 @@ new position.
 These signals allow the application to operate the appsrc in two different
 ways:
 
-The push model, in which the application repeatedly calls the push-buffer method
-with a new buffer. Optionally, the queue size in the appsrc can be controlled
-with the enough-data and need-data signals by respectively stopping/starting
-the push-buffer calls. This is a typical mode of operation for the
-stream-type "stream" and "seekable". Use this model when implementing various
-network protocols or hardware devices.
+The push mode, in which the application repeatedly calls the push-buffer/push-sample
+method with a new buffer/sample. Optionally, the queue size in the appsrc
+can be controlled with the enough-data and need-data signals by respectively
+stopping/starting the push-buffer/push-sample calls. This is a typical
+mode of operation for the stream-type "stream" and "seekable". Use this
+mode when implementing various network protocols or hardware devices.
 
-The pull model where the need-data signal triggers the next push-buffer call.
+The pull mode, in which the need-data signal triggers the next push-buffer call.
 This mode is typically used in the "random-access" stream-type. Use this
-model for file access or other randomly accessable sources. In this mode, a
+mode for file access or other randomly accessable sources. In this mode, a
 buffer of exactly the amount of bytes given by the need-data signal should be
 pushed into appsrc.
 
@@ -674,10 +683,10 @@ size in bytes. Setting this property is mandatory in the random-access mode.
 For the stream and seekable modes, setting this property is optional but
 recommended.
 
-When the application is finished pushing data into appsrc, it should call
+When the application has finished pushing data into appsrc, it should call
 gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
 this call, no more buffers can be pushed into appsrc until a flushing seek
-happened or the state of the appsrc has gone through READY.</doc>
+occurs or the state of the appsrc has gone through READY.</doc>
       <implements name="Gst.URIHandler"/>
       <virtual-method name="end_of_stream" invoker="end_of_stream">
         <doc xml:space="preserve">Indicates to the appsrc element that the last buffer queued in the
@@ -740,6 +749,32 @@ space becomes available in the queue.</doc>
           </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
+previous caps setted on appsrc will be replaced by the caps associated
+with the sample if not equal.
+
+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 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="sample" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstSample from wich extract buffer to
+push and caps to set</doc>
+            <type name="Gst.Sample" c:type="GstSample*"/>
+          </parameter>
+        </parameters>
+      </virtual-method>
       <virtual-method name="seek_data">
         <return-value transfer-ownership="none">
           <type name="gboolean" c:type="gboolean"/>
@@ -896,6 +931,34 @@ space becomes available in the queue.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="push_sample"
+              c:identifier="gst_app_src_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
+previous caps setted on appsrc will be replaced by the caps associated
+with the sample if not equal.
+
+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 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="sample" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstSample from wich extract buffer to
+push and caps to set</doc>
+            <type name="Gst.Sample" c:type="GstSample*"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_callbacks"
               c:identifier="gst_app_src_set_callbacks"
               introspectable="0">
@@ -1139,6 +1202,28 @@ becomes available in the queue.</doc>
           </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
+push to its source pad. This function set the appsrc caps based on the caps
+in the sample and reset the caps if they change.
+Only the caps and the buffer of the provided sample are used and not
+for example the segment in the sample.
+This function does not take ownership of the
+sample so the sample 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="sample" transfer-ownership="none">
+            <doc xml:space="preserve">a sample from which extract buffer to push</doc>
+            <type name="Gst.Sample"/>
+          </parameter>
+        </parameters>
+      </glib:signal>
       <glib:signal name="seek-data" when="last">
         <doc xml:space="preserve">Seek to the given offset. The next push-buffer should produce buffers from
 the new @offset.
@@ -1300,8 +1385,29 @@ gst_app_src_set_callbacks().</doc>
           </parameters>
         </callback>
       </field>
+      <field name="push_sample">
+        <callback name="push_sample">
+          <return-value transfer-ownership="none">
+            <doc xml:space="preserve">#GST_FLOW_OK when the buffer 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="sample" transfer-ownership="none">
+              <doc xml:space="preserve">a #GstSample from wich extract buffer to
+push and caps to set</doc>
+              <type name="Gst.Sample" c:type="GstSample*"/>
+            </parameter>
+          </parameters>
+        </callback>
+      </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>

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

@@ -206,6 +206,24 @@ gst_audio_base_sink_set_provide_clock.</doc>
           </instance-parameter>
         </parameters>
       </method>
+      <method name="report_device_failure"
+              c:identifier="gst_audio_base_sink_report_device_failure"
+              version="1.6">
+        <doc xml:space="preserve">Informs this base class that the audio output device has failed for
+some reason, causing a discontinuity (for example, because the device
+recovered from the error, but lost all contents of its ring buffer).
+This function is typically called by derived classes, and is useful
+for the custom slave method.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="sink" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioBaseSink</doc>
+            <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
       <method name="set_alignment_threshold"
               c:identifier="gst_audio_base_sink_set_alignment_threshold">
         <doc xml:space="preserve">Controls the sink's alignment threshold.</doc>
@@ -223,6 +241,44 @@ gst_audio_base_sink_set_provide_clock.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="set_custom_slaving_callback"
+              c:identifier="gst_audio_base_sink_set_custom_slaving_callback"
+              version="1.6">
+        <doc xml:space="preserve">Sets the custom slaving callback. This callback will
+be invoked if the slave-method property is set to
+GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink
+receives and plays samples.
+
+Setting the callback to NULL causes the sink to
+behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE
+method were used.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="sink" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioBaseSink</doc>
+            <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
+          </instance-parameter>
+          <parameter name="callback"
+                     transfer-ownership="none"
+                     scope="notified"
+                     closure="1"
+                     destroy="2">
+            <doc xml:space="preserve">a #GstAudioBaseSinkCustomSlavingCallback</doc>
+            <type name="AudioBaseSinkCustomSlavingCallback"
+                  c:type="GstAudioBaseSinkCustomSlavingCallback"/>
+          </parameter>
+          <parameter name="user_data" transfer-ownership="none">
+            <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">called when user_data becomes unused</doc>
+            <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_discont_wait"
               c:identifier="gst_audio_base_sink_set_discont_wait">
         <doc xml:space="preserve">Controls how long the sink will wait before creating a discontinuity.</doc>
@@ -401,6 +457,97 @@ functionality.</doc>
         </array>
       </field>
     </record>
+    <callback name="AudioBaseSinkCustomSlavingCallback"
+              c:type="GstAudioBaseSinkCustomSlavingCallback"
+              version="1.6">
+      <doc xml:space="preserve">This function is set with gst_audio_base_sink_set_custom_slaving_callback()
+and is called during playback. It receives the current time of external and
+internal clocks, which the callback can then use to apply any custom
+slaving/synchronization schemes. The external clock is the sink's element clock,
+the internal one is the internal audio clock. The internal audio clock's
+calibration is applied to the timestamps before they are passed to the
+callback. The difference between etime and itime is the skew; how much
+internal and external clock lie apart from each other. A skew of 0 means
+both clocks are perfectly in sync. itime &gt; etime means the external clock
+is going slower, while itime &lt; etime means it is going faster than the
+internal clock. etime and itime are always valid timestamps, except for when
+discont is set to TRUE.
+requested_skew is an output value the callback can write to. It informs the sink
+of whether or not it should move the playout pointer, and if so, by how much.
+This pointer is only NULL if discont is true; otherwise, it is safe to write
+to *requested_skew. The default skew is 0.
+
+The sink may experience discontinuities. If one happens, discont is TRUE,
+itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
+This makes it possible to reset custom clock slaving algorithms when a
+discontinuity happens.</doc>
+      <return-value transfer-ownership="none">
+        <type name="none" c:type="void"/>
+      </return-value>
+      <parameters>
+        <parameter name="sink" transfer-ownership="none">
+          <doc xml:space="preserve">a #GstAudioBaseSink</doc>
+          <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
+        </parameter>
+        <parameter name="etime" transfer-ownership="none">
+          <doc xml:space="preserve">external clock time</doc>
+          <type name="Gst.ClockTime" c:type="GstClockTime"/>
+        </parameter>
+        <parameter name="itime" transfer-ownership="none">
+          <doc xml:space="preserve">internal clock time</doc>
+          <type name="Gst.ClockTime" c:type="GstClockTime"/>
+        </parameter>
+        <parameter name="requested_skew" transfer-ownership="none">
+          <doc xml:space="preserve">skew amount requested by the callback</doc>
+          <type name="Gst.ClockTimeDiff" c:type="GstClockTimeDiff*"/>
+        </parameter>
+        <parameter name="discont_reason" transfer-ownership="none">
+          <doc xml:space="preserve">TRUE if there was a discontinuity in the average skew</doc>
+          <type name="AudioBaseSinkDiscontReason"
+                c:type="GstAudioBaseSinkDiscontReason"/>
+        </parameter>
+        <parameter name="user_data" transfer-ownership="none" closure="5">
+          <doc xml:space="preserve">user data</doc>
+          <type name="gpointer" c:type="gpointer"/>
+        </parameter>
+      </parameters>
+    </callback>
+    <enumeration name="AudioBaseSinkDiscontReason"
+                 version="1.6"
+                 c:type="GstAudioBaseSinkDiscontReason">
+      <doc xml:space="preserve">GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT: No discontinuity occurred
+GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS: New caps are set, causing renegotiotion
+GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH: Samples have been flushed
+GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY: Sink was synchronized to the estimated latency (occurs during initialization)
+GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT: Aligning buffers failed because the timestamps are too discontinuous
+GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE: Audio output device experienced and recovered from an error but introduced latency in the process (see also @gst_audio_base_sink_report_device_failure)
+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">
+      </member>
+      <member name="new_caps"
+              value="1"
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS">
+      </member>
+      <member name="flush"
+              value="2"
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH">
+      </member>
+      <member name="sync_latency"
+              value="3"
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY">
+      </member>
+      <member name="alignment"
+              value="4"
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT">
+      </member>
+      <member name="device_failure"
+              value="5"
+              c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE">
+      </member>
+    </enumeration>
     <record name="AudioBaseSinkPrivate"
             c:type="GstAudioBaseSinkPrivate"
             disguised="1">
@@ -430,6 +577,12 @@ drifts too much.</doc>
               glib:nick="none">
         <doc xml:space="preserve">No adjustment is done.</doc>
       </member>
+      <member name="custom"
+              value="3"
+              c:identifier="GST_AUDIO_BASE_SINK_SLAVE_CUSTOM"
+              glib:nick="custom">
+        <doc xml:space="preserve">Use custom clock slaving algorithm (Since: 1.6)</doc>
+      </member>
     </enumeration>
     <class name="AudioBaseSrc"
            c:symbol-prefix="audio_base_src"
@@ -1461,6 +1614,19 @@ Things that subclass need to take care of:
           </parameter>
         </parameters>
       </virtual-method>
+      <virtual-method name="getcaps">
+        <return-value transfer-ownership="full">
+          <type name="Gst.Caps" c:type="GstCaps*"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="dec" transfer-ownership="none">
+            <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+          </instance-parameter>
+          <parameter name="filter" transfer-ownership="none">
+            <type name="Gst.Caps" c:type="GstCaps*"/>
+          </parameter>
+        </parameters>
+      </virtual-method>
       <virtual-method name="handle_frame">
         <return-value transfer-ownership="none">
           <type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
@@ -1570,6 +1736,19 @@ negotiate fails.</doc>
           </parameter>
         </parameters>
       </virtual-method>
+      <virtual-method name="sink_query">
+        <return-value transfer-ownership="none">
+          <type name="gboolean" c:type="gboolean"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="dec" transfer-ownership="none">
+            <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+          </instance-parameter>
+          <parameter name="query" transfer-ownership="none">
+            <type name="Gst.Query" c:type="GstQuery*"/>
+          </parameter>
+        </parameters>
+      </virtual-method>
       <virtual-method name="src_event">
         <return-value transfer-ownership="none">
           <type name="gboolean" c:type="gboolean"/>
@@ -1583,6 +1762,19 @@ negotiate fails.</doc>
           </parameter>
         </parameters>
       </virtual-method>
+      <virtual-method name="src_query">
+        <return-value transfer-ownership="none">
+          <type name="gboolean" c:type="gboolean"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="dec" transfer-ownership="none">
+            <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+          </instance-parameter>
+          <parameter name="query" transfer-ownership="none">
+            <type name="Gst.Query" c:type="GstQuery*"/>
+          </parameter>
+        </parameters>
+      </virtual-method>
       <virtual-method name="start">
         <return-value transfer-ownership="none">
           <type name="gboolean" c:type="gboolean"/>
@@ -1927,6 +2119,37 @@ negotiate fails.</doc>
           </instance-parameter>
         </parameters>
       </method>
+      <method name="proxy_getcaps"
+              c:identifier="gst_audio_decoder_proxy_getcaps"
+              version="1.6">
+        <doc xml:space="preserve">Returns caps that express @caps (or sink template caps if @caps == NULL)
+restricted to rate/channels/... combinations supported by downstream
+elements.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a #GstCaps owned by caller</doc>
+          <type name="Gst.Caps" c:type="GstCaps*"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="decoder" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioDecoder</doc>
+            <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+          </instance-parameter>
+          <parameter name="caps"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">initial caps</doc>
+            <type name="Gst.Caps" c:type="GstCaps*"/>
+          </parameter>
+          <parameter name="filter"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">filter caps</doc>
+            <type name="Gst.Caps" c:type="GstCaps*"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_drainable"
               c:identifier="gst_audio_decoder_set_drainable">
         <doc xml:space="preserve">Configures decoder drain handling.  If drainable, subclass might
@@ -2371,8 +2594,53 @@ overridden.</doc>
           </parameters>
         </callback>
       </field>
+      <field name="sink_query">
+        <callback name="sink_query">
+          <return-value transfer-ownership="none">
+            <type name="gboolean" c:type="gboolean"/>
+          </return-value>
+          <parameters>
+            <parameter name="dec" transfer-ownership="none">
+              <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+            </parameter>
+            <parameter name="query" transfer-ownership="none">
+              <type name="Gst.Query" c:type="GstQuery*"/>
+            </parameter>
+          </parameters>
+        </callback>
+      </field>
+      <field name="src_query">
+        <callback name="src_query">
+          <return-value transfer-ownership="none">
+            <type name="gboolean" c:type="gboolean"/>
+          </return-value>
+          <parameters>
+            <parameter name="dec" transfer-ownership="none">
+              <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+            </parameter>
+            <parameter name="query" transfer-ownership="none">
+              <type name="Gst.Query" c:type="GstQuery*"/>
+            </parameter>
+          </parameters>
+        </callback>
+      </field>
+      <field name="getcaps">
+        <callback name="getcaps">
+          <return-value transfer-ownership="full">
+            <type name="Gst.Caps" c:type="GstCaps*"/>
+          </return-value>
+          <parameters>
+            <parameter name="dec" transfer-ownership="none">
+              <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+            </parameter>
+            <parameter name="filter" transfer-ownership="none">
+              <type name="Gst.Caps" c:type="GstCaps*"/>
+            </parameter>
+          </parameters>
+        </callback>
+      </field>
       <field name="_gst_reserved" readable="0" private="1">
-        <array zero-terminated="0" c:type="gpointer" fixed-size="20">
+        <array zero-terminated="0" c:type="gpointer" fixed-size="17">
           <type name="gpointer" c:type="gpointer"/>
         </array>
       </field>
@@ -3048,11 +3316,17 @@ restricted to channel/rate combinations supported by downstream elements
             <doc xml:space="preserve">a #GstAudioEncoder</doc>
             <type name="AudioEncoder" c:type="GstAudioEncoder*"/>
           </instance-parameter>
-          <parameter name="caps" transfer-ownership="none">
+          <parameter name="caps"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
             <doc xml:space="preserve">initial caps</doc>
             <type name="Gst.Caps" c:type="GstCaps*"/>
           </parameter>
-          <parameter name="filter" transfer-ownership="none">
+          <parameter name="filter"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
             <doc xml:space="preserve">filter caps</doc>
             <type name="Gst.Caps" c:type="GstCaps*"/>
           </parameter>
@@ -6534,7 +6808,9 @@ positions and the same positions, only in a different order.
       <doc xml:space="preserve">Convert the @channels present in @channel_mask to a @position array
 (which should have at least @channels entries ensured by caller).
 If @channel_mask is set to 0, it is considered as 'not present' for purpose
-of conversion.</doc>
+of conversion.
+A partially valid @channel_mask with less bits set than the number
+of channels is considered valid.</doc>
       <return-value transfer-ownership="none">
         <doc xml:space="preserve">%TRUE if channel and channel mask are valid and could be converted</doc>
         <type name="gboolean" c:type="gboolean"/>
@@ -6915,15 +7191,14 @@ positions and the same positions, only in a different order.</doc>
       </parameters>
     </function>
     <function name="buffer_add_audio_downmix_meta"
-              c:identifier="gst_buffer_add_audio_downmix_meta"
-              introspectable="0">
+              c:identifier="gst_buffer_add_audio_downmix_meta">
       <doc xml:space="preserve">Attaches #GstAudioDownmixMeta metadata to @buffer with the given parameters.
 
 @matrix is an two-dimensional array of @to_channels times @from_channels
 coefficients, i.e. the i-th output channels is constructed by multiplicating
 the input channels with the coefficients in @matrix[i] and taking the sum
 of the results.</doc>
-      <return-value>
+      <return-value transfer-ownership="none">
         <doc xml:space="preserve">the #GstAudioDownmixMeta on @buffer.</doc>
         <type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/>
       </return-value>
@@ -6967,11 +7242,10 @@ of the results.</doc>
       </parameters>
     </function>
     <function name="buffer_get_audio_downmix_meta_for_channels"
-              c:identifier="gst_buffer_get_audio_downmix_meta_for_channels"
-              introspectable="0">
+              c:identifier="gst_buffer_get_audio_downmix_meta_for_channels">
       <doc xml:space="preserve">Find the #GstAudioDownmixMeta on @buffer for the given destination
 channel positions.</doc>
-      <return-value>
+      <return-value transfer-ownership="none">
         <doc xml:space="preserve">the #GstAudioDownmixMeta on @buffer.</doc>
         <type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/>
       </return-value>