gstreamer 3.0.9-x64-mingw32 → 3.1.0-x64-mingw32

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

@@ -48,7 +48,8 @@ and/or use gtk-doc annotations.  -->
         <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.
+When the buffer will be released the allocator will close the @fd unless
+the %GST_FD_MEMORY_FLAG_DONT_CLOSE flag is specified.
 The memory is only mmapped on gst_buffer_mmap() request.</doc>
           <type name="Gst.Memory" c:type="GstMemory*"/>
         </return-value>
@@ -99,6 +100,12 @@ The memory is only mmapped on gst_buffer_mmap() request.</doc>
         <doc xml:space="preserve">do a private mapping instead of
        the default shared mapping.</doc>
       </member>
+      <member name="dont_close"
+              value="4"
+              c:identifier="GST_FD_MEMORY_FLAG_DONT_CLOSE">
+        <doc xml:space="preserve">don't close the file descriptor when
+       the memory is freed. Since: 1.10.</doc>
+      </member>
     </bitfield>
     <function name="dmabuf_allocator_alloc"
               c:identifier="gst_dmabuf_allocator_alloc"

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

@@ -32,7 +32,9 @@ methods or by using the appsink action signals and properties.
 The normal way of retrieving samples from appsink is by using the
 gst_app_sink_pull_sample() and gst_app_sink_pull_preroll() methods.
 These methods block until a sample becomes available in the sink or when the
-sink is shut down or reaches EOS.
+sink is shut down or reaches EOS. There are also timed variants of these
+methods, gst_app_sink_try_pull_sample() and gst_app_sink_try_pull_preroll(),
+which accept a timeout parameter to limit the amount of time to wait.
 
 Appsink will internally use a queue to collect buffers from the streaming
 thread. If the application is not pulling samples fast enough, this queue
@@ -140,6 +142,73 @@ If an EOS event was received before any buffers, this function returns
           </instance-parameter>
         </parameters>
       </virtual-method>
+      <virtual-method name="try_pull_preroll"
+                      invoker="try_pull_preroll"
+                      version="1.10">
+        <doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state. This sample can be pulled many times
+and remains available to the application even after EOS.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample().
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+This function blocks until a preroll sample or EOS is received, the appsink
+element is set to the READY/NULL state, or the timeout expires.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+         Call gst_sample_unref() after usage.</doc>
+          <type name="Gst.Sample" c:type="GstSample*"/>
+        </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="timeout" transfer-ownership="none">
+            <doc xml:space="preserve">the maximum amount of time to wait for the preroll sample</doc>
+            <type name="Gst.ClockTime" c:type="GstClockTime"/>
+          </parameter>
+        </parameters>
+      </virtual-method>
+      <virtual-method name="try_pull_sample"
+                      invoker="try_pull_sample"
+                      version="1.10">
+        <doc xml:space="preserve">This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state or the timeout expires.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered buffers will be put in a queue so that the application
+can pull samples at its own rate. Note that when the application does not
+pull samples fast enough, the queued buffers could consume a lot of memory,
+especially when dealing with raw video frames.
+
+If an EOS event was received before any buffers or the timeout expires,
+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 #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+Call gst_sample_unref() after usage.</doc>
+          <type name="Gst.Sample" c:type="GstSample*"/>
+        </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="timeout" transfer-ownership="none">
+            <doc xml:space="preserve">the maximum amount of time to wait for a sample</doc>
+            <type name="Gst.ClockTime" c:type="GstClockTime"/>
+          </parameter>
+        </parameters>
+      </virtual-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">
@@ -408,6 +477,73 @@ elements until a sample is pulled from @appsink.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="try_pull_preroll"
+              c:identifier="gst_app_sink_try_pull_preroll"
+              version="1.10">
+        <doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state. This sample can be pulled many times
+and remains available to the application even after EOS.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample().
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+This function blocks until a preroll sample or EOS is received, the appsink
+element is set to the READY/NULL state, or the timeout expires.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+         Call gst_sample_unref() after usage.</doc>
+          <type name="Gst.Sample" c:type="GstSample*"/>
+        </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="timeout" transfer-ownership="none">
+            <doc xml:space="preserve">the maximum amount of time to wait for the preroll sample</doc>
+            <type name="Gst.ClockTime" c:type="GstClockTime"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="try_pull_sample"
+              c:identifier="gst_app_sink_try_pull_sample"
+              version="1.10">
+        <doc xml:space="preserve">This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state or the timeout expires.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered buffers will be put in a queue so that the application
+can pull samples at its own rate. Note that when the application does not
+pull samples fast enough, the queued buffers could consume a lot of memory,
+especially when dealing with raw video frames.
+
+If an EOS event was received before any buffers or the timeout expires,
+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 #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+Call gst_sample_unref() after usage.</doc>
+          <type name="Gst.Sample" c:type="GstSample*"/>
+        </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="timeout" transfer-ownership="none">
+            <doc xml:space="preserve">the maximum amount of time to wait for a sample</doc>
+            <type name="Gst.ClockTime" c:type="GstClockTime"/>
+          </parameter>
+        </parameters>
+      </method>
       <property name="caps" writable="1" transfer-ownership="none">
         <type name="Gst.Caps"/>
       </property>
@@ -518,6 +654,68 @@ If an EOS event was received before any buffers, this function returns
           <type name="Gst.Sample"/>
         </return-value>
       </glib:signal>
+      <glib:signal name="try-pull-preroll"
+                   when="last"
+                   action="1"
+                   version="1.10">
+        <doc xml:space="preserve">Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state. This sample can be pulled many times
+and remains available to the application even after EOS.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample() or the "pull-sample" action signal.
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+This function blocks until a preroll sample or EOS is received, the appsink
+element is set to the READY/NULL state, or the timeout expires.</doc>
+        <return-value transfer-ownership="full">
+          <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.</doc>
+          <type name="Gst.Sample"/>
+        </return-value>
+        <parameters>
+          <parameter name="timeout" transfer-ownership="none">
+            <doc xml:space="preserve">the maximum amount of time to wait for the preroll sample</doc>
+            <type name="guint64" c:type="guint64"/>
+          </parameter>
+        </parameters>
+      </glib:signal>
+      <glib:signal name="try-pull-sample"
+                   when="last"
+                   action="1"
+                   version="1.10">
+        <doc xml:space="preserve">This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state or the timeout expires.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered samples will be put in a queue so that the application
+can pull samples at its own rate.
+
+Note that when the application does not pull samples fast enough, the
+queued samples could consume a lot of memory, especially when dealing with
+raw video frames. It's possible to control the behaviour of the queue with
+the "drop" and "max-buffers" properties.
+
+If an EOS event was received before any buffers or the timeout expires,
+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 #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.</doc>
+          <type name="Gst.Sample"/>
+        </return-value>
+        <parameters>
+          <parameter name="timeout" transfer-ownership="none">
+            <doc xml:space="preserve">the maximum amount of time to wait for a sample</doc>
+            <type name="guint64" c:type="guint64"/>
+          </parameter>
+        </parameters>
+      </glib:signal>
     </class>
     <record name="AppSinkCallbacks"
             c:type="GstAppSinkCallbacks"
@@ -659,8 +857,46 @@ gst_app_sink_set_callbacks().</doc>
           </parameters>
         </callback>
       </field>
+      <field name="try_pull_preroll">
+        <callback name="try_pull_preroll">
+          <return-value transfer-ownership="full">
+            <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+         Call gst_sample_unref() after usage.</doc>
+            <type name="Gst.Sample" c:type="GstSample*"/>
+          </return-value>
+          <parameters>
+            <parameter name="appsink" transfer-ownership="none">
+              <doc xml:space="preserve">a #GstAppSink</doc>
+              <type name="AppSink" c:type="GstAppSink*"/>
+            </parameter>
+            <parameter name="timeout" transfer-ownership="none">
+              <doc xml:space="preserve">the maximum amount of time to wait for the preroll sample</doc>
+              <type name="Gst.ClockTime" c:type="GstClockTime"/>
+            </parameter>
+          </parameters>
+        </callback>
+      </field>
+      <field name="try_pull_sample">
+        <callback name="try_pull_sample">
+          <return-value transfer-ownership="full">
+            <doc xml:space="preserve">a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+Call gst_sample_unref() after usage.</doc>
+            <type name="Gst.Sample" c:type="GstSample*"/>
+          </return-value>
+          <parameters>
+            <parameter name="appsink" transfer-ownership="none">
+              <doc xml:space="preserve">a #GstAppSink</doc>
+              <type name="AppSink" c:type="GstAppSink*"/>
+            </parameter>
+            <parameter name="timeout" transfer-ownership="none">
+              <doc xml:space="preserve">the maximum amount of time to wait for a sample</doc>
+              <type name="Gst.ClockTime" c:type="GstClockTime"/>
+            </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="2">
           <type name="gpointer" c:type="gpointer"/>
         </array>
       </field>
@@ -882,6 +1118,22 @@ element is the last buffer of the stream.</doc>
           </instance-parameter>
         </parameters>
       </method>
+      <method name="get_duration"
+              c:identifier="gst_app_src_get_duration"
+              version="1.10">
+        <doc xml:space="preserve">Get the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
+not known.</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">the duration of the stream previously set with gst_app_src_set_duration();</doc>
+          <type name="Gst.ClockTime" c:type="GstClockTime"/>
+        </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>
+        </parameters>
+      </method>
       <method name="get_emit_signals"
               c:identifier="gst_app_src_get_emit_signals">
         <doc xml:space="preserve">Check if appsrc will emit the "new-preroll" and "new-buffer" signals.</doc>
@@ -1064,6 +1316,25 @@ buffers must match the caps or left NULL.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="set_duration"
+              c:identifier="gst_app_src_set_duration"
+              version="1.10">
+        <doc xml:space="preserve">Set the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
+not known.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </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="duration" transfer-ownership="none">
+            <doc xml:space="preserve">the duration to set</doc>
+            <type name="Gst.ClockTime" c:type="GstClockTime"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_emit_signals"
               c:identifier="gst_app_src_set_emit_signals">
         <doc xml:space="preserve">Make appsrc emit the "new-preroll" and "new-buffer" signals. This option is
@@ -1168,6 +1439,9 @@ A stream_type stream</doc>
       <property name="current-level-bytes" transfer-ownership="none">
         <type name="guint64" c:type="guint64"/>
       </property>
+      <property name="duration" writable="1" transfer-ownership="none">
+        <type name="guint64" c:type="guint64"/>
+      </property>
       <property name="emit-signals" writable="1" transfer-ownership="none">
         <type name="gboolean" c:type="gboolean"/>
       </property>

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

@@ -46,6 +46,14 @@ quantized to multiples of this value.
 Default is 1</doc>
       <type name="utf8" c:type="gchar*"/>
     </constant>
+    <constant name="AUDIO_CONVERTER_OPT_RESAMPLER_METHOD"
+              value="GstAudioConverter.resampler-method"
+              c:type="GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD">
+      <doc xml:space="preserve">#GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
+changing sample rates.
+Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
     <constant name="AUDIO_DECODER_MAX_ERRORS"
               value="10"
               c:type="GST_AUDIO_DECODER_MAX_ERRORS">
@@ -104,6 +112,107 @@ Default is 1</doc>
       <doc xml:space="preserve">Maximum range of allowed sample rates, for use in template caps strings.</doc>
       <type name="utf8" c:type="gchar*"/>
     </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_CUBIC_B"
+              value="GstAudioResampler.cubic-b"
+              c:type="GST_AUDIO_RESAMPLER_OPT_CUBIC_B">
+      <doc xml:space="preserve">G_TYPE_DOUBLE, B parameter of the cubic filter.
+Values between 0.0 and 2.0 are accepted. 1.0 is the default.
+
+Below are some values of popular filters:
+                   B       C
+Hermite           0.0     0.0
+Spline            1.0     0.0
+Catmull-Rom       0.0     1/2</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_CUBIC_C"
+              value="GstAudioResampler.cubic-c"
+              c:type="GST_AUDIO_RESAMPLER_OPT_CUBIC_C">
+      <doc xml:space="preserve">G_TYPE_DOUBLE, C parameter of the cubic filter.
+Values between 0.0 and 2.0 are accepted. 0.0 is the default.
+
+See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_CUTOFF"
+              value="GstAudioResampler.cutoff"
+              c:type="GST_AUDIO_RESAMPLER_OPT_CUTOFF">
+      <doc xml:space="preserve">G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION"
+              value="GstAudioResampler.filter-interpolation"
+              c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION">
+      <doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
+   interpolated.
+GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_FILTER_MODE"
+              value="GstAudioResampler.filter-mode"
+              c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_MODE">
+      <doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
+constructed.
+GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD"
+              value="GstAudioResampler.filter-mode-threshold"
+              c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD">
+      <doc xml:space="preserve">G_TYPE_UINT: the amount of memory to use for full filter tables before
+switching to interpolated filter tables.
+1048576 is the default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE"
+              value="GstAudioResampler.filter-oversample"
+              c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE">
+      <doc xml:space="preserve">G_TYPE_UINT, oversampling to use when interpolating filters
+8 is the default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR"
+              value="GstAudioResampler.max-phase-error"
+              c:type="GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR">
+      <doc xml:space="preserve">G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
+rates.
+0.1 is the default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_N_TAPS"
+              value="GstAudioResampler.n-taps"
+              c:type="GST_AUDIO_RESAMPLER_OPT_N_TAPS">
+      <doc xml:space="preserve">G_TYPE_INT: the number of taps to use for the filter.
+0 is the default and selects the taps automatically.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"
+              value="GstAudioResampler.stop-attenutation"
+              c:type="GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION">
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH"
+              value="GstAudioResampler.transition-bandwidth"
+              c:type="GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH">
+      <doc xml:space="preserve">G_TYPE_DOUBLE, transition bandwidth. The width of the
+transition band for the kaiser window. 0.087 is the default.</doc>
+      <type name="utf8" c:type="gchar*"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_QUALITY_DEFAULT"
+              value="4"
+              c:type="GST_AUDIO_RESAMPLER_QUALITY_DEFAULT">
+      <type name="gint" c:type="gint"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_QUALITY_MAX"
+              value="10"
+              c:type="GST_AUDIO_RESAMPLER_QUALITY_MAX">
+      <type name="gint" c:type="gint"/>
+    </constant>
+    <constant name="AUDIO_RESAMPLER_QUALITY_MIN"
+              value="0"
+              c:type="GST_AUDIO_RESAMPLER_QUALITY_MIN">
+      <type name="gint" c:type="gint"/>
+    </constant>
     <class name="AudioBaseSink"
            c:symbol-prefix="audio_base_sink"
            c:type="GstAudioBaseSink"
@@ -1832,7 +1941,7 @@ are matching and @in and @out point to enough memory.</doc>
               c:identifier="gst_audio_converter_update_config">
         <doc xml:space="preserve">Set @in_rate, @out_rate and @config as extra configuration for @convert.
 
-in_rate and @out_rate specify the new sample rates of input and output
+@in_rate and @out_rate specify the new sample rates of input and output
 formats. A value of 0 leaves the sample rate unchanged.
 
 @config can be %NULL, in which case, the current configuration is not
@@ -1884,7 +1993,7 @@ parameters for details about the options and values.</doc>
         </return-value>
         <parameters>
           <parameter name="flags" transfer-ownership="none">
-            <doc xml:space="preserve">#GstAudioConverterFlags</doc>
+            <doc xml:space="preserve">extra #GstAudioConverterFlags</doc>
             <type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
           </parameter>
           <parameter name="in_info" transfer-ownership="none">
@@ -2643,6 +2752,30 @@ elements.</doc>
           </parameter>
         </parameters>
       </method>
+      <method name="set_allocation_caps"
+              c:identifier="gst_audio_decoder_set_allocation_caps"
+              version="1.10">
+        <doc xml:space="preserve">Sets a caps in allocation query which are different from the set
+pad's caps. Use this function before calling
+gst_audio_decoder_negotiate(). Setting to %NULL the allocation
+query will use the caps from the pad.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="dec" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioDecoder</doc>
+            <type name="AudioDecoder" c:type="GstAudioDecoder*"/>
+          </instance-parameter>
+          <parameter name="allocation_caps"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">a #GstCaps or %NULL</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
@@ -3947,6 +4080,30 @@ restricted to channel/rate combinations supported by downstream elements
           </parameter>
         </parameters>
       </method>
+      <method name="set_allocation_caps"
+              c:identifier="gst_audio_encoder_set_allocation_caps"
+              version="1.10">
+        <doc xml:space="preserve">Sets a caps in allocation query which are different from the set
+pad's caps. Use this function before calling
+gst_audio_encoder_negotiate(). Setting to %NULL the allocation
+query will use the caps from the pad.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="enc" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioEncoder</doc>
+            <type name="AudioEncoder" c:type="GstAudioEncoder*"/>
+          </instance-parameter>
+          <parameter name="allocation_caps"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">a #GstCaps or %NULL</doc>
+            <type name="Gst.Caps" c:type="GstCaps*"/>
+          </parameter>
+        </parameters>
+      </method>
       <method name="set_drainable"
               c:identifier="gst_audio_encoder_set_drainable">
         <doc xml:space="preserve">Configures encoder drain handling.  If drainable, subclass might
@@ -4154,14 +4311,18 @@ MT safe.</doc>
       </method>
       <method name="set_output_format"
               c:identifier="gst_audio_encoder_set_output_format">
+        <doc xml:space="preserve">Configure output caps on the srcpad of @enc.</doc>
         <return-value transfer-ownership="none">
+          <doc xml:space="preserve">%TRUE on success.</doc>
           <type name="gboolean" c:type="gboolean"/>
         </return-value>
         <parameters>
           <instance-parameter name="enc" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioEncoder</doc>
             <type name="AudioEncoder" c:type="GstAudioEncoder*"/>
           </instance-parameter>
           <parameter name="caps" transfer-ownership="none">
+            <doc xml:space="preserve">#GstCaps</doc>
             <type name="Gst.Caps" c:type="GstCaps*"/>
           </parameter>
         </parameters>
@@ -5564,6 +5725,367 @@ the @dither and @ns parameters.</doc>
         <doc xml:space="preserve">samples are non-interleaved</doc>
       </member>
     </bitfield>
+    <record name="AudioResampler" c:type="GstAudioResampler" disguised="1">
+      <doc xml:space="preserve">#GstAudioResampler is a structure which holds the information
+required to perform various kinds of resampling filtering.</doc>
+      <method name="free"
+              c:identifier="gst_audio_resampler_free"
+              version="1.6">
+        <doc xml:space="preserve">Free a previously allocated #GstAudioResampler @resampler.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
+      <method name="get_in_frames"
+              c:identifier="gst_audio_resampler_get_in_frames">
+        <doc xml:space="preserve">Get the number of input frames that would currently be needed
+to produce @out_frames from @resampler.</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">The number of input frames needed for producing
+@out_frames of data from @resampler.</doc>
+          <type name="gsize" c:type="gsize"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+          <parameter name="out_frames" transfer-ownership="none">
+            <doc xml:space="preserve">number of input frames</doc>
+            <type name="gsize" c:type="gsize"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="get_max_latency"
+              c:identifier="gst_audio_resampler_get_max_latency">
+        <doc xml:space="preserve">Get the maximum number of input samples that the resampler would
+need before producing output.</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">the latency of @resampler as expressed in the number of
+frames.</doc>
+          <type name="gsize" c:type="gsize"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
+      <method name="get_out_frames"
+              c:identifier="gst_audio_resampler_get_out_frames">
+        <doc xml:space="preserve">Get the number of output frames that would be currently available when
+@in_frames are given to @resampler.</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">The number of frames that would be availabe after giving
+@in_frames as input to @resampler.</doc>
+          <type name="gsize" c:type="gsize"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+          <parameter name="in_frames" transfer-ownership="none">
+            <doc xml:space="preserve">number of input frames</doc>
+            <type name="gsize" c:type="gsize"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="resample" c:identifier="gst_audio_resampler_resample">
+        <doc xml:space="preserve">Perform resampling on @in_frames frames in @in and write @out_frames to @out.
+
+In case the samples are interleaved, @in and @out must point to an
+array with a single element pointing to a block of interleaved samples.
+
+If non-interleaved samples are used, @in and @out must point to an
+array with pointers to memory blocks, one for each channel.
+
+@in may be %NULL, in which case @in_frames of silence samples are pushed
+into the resampler.
+
+This function always produces @out_frames of output and consumes @in_frames of
+input. Use gst_audio_resampler_get_out_frames() and
+gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames
+are matching and @in and @out point to enough memory.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+          <parameter name="in"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">input samples</doc>
+            <type name="gpointer" c:type="gpointer"/>
+          </parameter>
+          <parameter name="in_frames" transfer-ownership="none">
+            <doc xml:space="preserve">number of input frames</doc>
+            <type name="gsize" c:type="gsize"/>
+          </parameter>
+          <parameter name="out"
+                     transfer-ownership="none"
+                     nullable="1"
+                     allow-none="1">
+            <doc xml:space="preserve">output samples</doc>
+            <type name="gpointer" c:type="gpointer"/>
+          </parameter>
+          <parameter name="out_frames" transfer-ownership="none">
+            <doc xml:space="preserve">number of output frames</doc>
+            <type name="gsize" c:type="gsize"/>
+          </parameter>
+        </parameters>
+      </method>
+      <method name="reset" c:identifier="gst_audio_resampler_reset">
+        <doc xml:space="preserve">Reset @resampler to the state it was when it was first created, discarding
+all sample history.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+        </parameters>
+      </method>
+      <method name="update" c:identifier="gst_audio_resampler_update">
+        <doc xml:space="preserve">Update the resampler parameters for @resampler. This function should
+not be called concurrently with any other function on @resampler.
+
+When @in_rate or @out_rate is 0, its value is unchanged.
+
+When @options is %NULL, the previously configured options are reused.</doc>
+        <return-value transfer-ownership="none">
+          <doc xml:space="preserve">%TRUE if the new parameters could be set</doc>
+          <type name="gboolean" c:type="gboolean"/>
+        </return-value>
+        <parameters>
+          <instance-parameter name="resampler" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResampler</doc>
+            <type name="AudioResampler" c:type="GstAudioResampler*"/>
+          </instance-parameter>
+          <parameter name="in_rate" transfer-ownership="none">
+            <doc xml:space="preserve">new input rate</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="out_rate" transfer-ownership="none">
+            <doc xml:space="preserve">new output rate</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="options" transfer-ownership="none">
+            <doc xml:space="preserve">new options or %NULL</doc>
+            <type name="Gst.Structure" c:type="GstStructure*"/>
+          </parameter>
+        </parameters>
+      </method>
+      <function name="new" c:identifier="gst_audio_resampler_new">
+        <doc xml:space="preserve">Make a new resampler.</doc>
+        <return-value transfer-ownership="full" skip="1">
+          <doc xml:space="preserve">%TRUE on success</doc>
+          <type name="AudioResampler" c:type="GstAudioResampler*"/>
+        </return-value>
+        <parameters>
+          <parameter name="method" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResamplerMethod</doc>
+            <type name="AudioResamplerMethod"
+                  c:type="GstAudioResamplerMethod"/>
+          </parameter>
+          <parameter name="flags" transfer-ownership="none">
+            <doc xml:space="preserve">#GstAudioResamplerFlags</doc>
+            <type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
+          </parameter>
+          <parameter name="format" transfer-ownership="none">
+            <type name="AudioFormat" c:type="GstAudioFormat"/>
+          </parameter>
+          <parameter name="channels" transfer-ownership="none">
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="in_rate" transfer-ownership="none">
+            <doc xml:space="preserve">input rate</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="out_rate" transfer-ownership="none">
+            <doc xml:space="preserve">output rate</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="options" transfer-ownership="none">
+            <doc xml:space="preserve">extra options</doc>
+            <type name="Gst.Structure" c:type="GstStructure*"/>
+          </parameter>
+        </parameters>
+      </function>
+      <function name="options_set_quality"
+                c:identifier="gst_audio_resampler_options_set_quality">
+        <doc xml:space="preserve">Set the parameters for resampling from @in_rate to @out_rate using @method
+for @quality in @options.</doc>
+        <return-value transfer-ownership="none">
+          <type name="none" c:type="void"/>
+        </return-value>
+        <parameters>
+          <parameter name="method" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstAudioResamplerMethod</doc>
+            <type name="AudioResamplerMethod"
+                  c:type="GstAudioResamplerMethod"/>
+          </parameter>
+          <parameter name="quality" transfer-ownership="none">
+            <doc xml:space="preserve">the quality</doc>
+            <type name="guint" c:type="guint"/>
+          </parameter>
+          <parameter name="in_rate" transfer-ownership="none">
+            <doc xml:space="preserve">the input rate</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="out_rate" transfer-ownership="none">
+            <doc xml:space="preserve">the output rate</doc>
+            <type name="gint" c:type="gint"/>
+          </parameter>
+          <parameter name="options" transfer-ownership="none">
+            <doc xml:space="preserve">a #GstStructure</doc>
+            <type name="Gst.Structure" c:type="GstStructure*"/>
+          </parameter>
+        </parameters>
+      </function>
+    </record>
+    <enumeration name="AudioResamplerFilterInterpolation"
+                 glib:type-name="GstAudioResamplerFilterInterpolation"
+                 glib:get-type="gst_audio_resampler_filter_interpolation_get_type"
+                 c:type="GstAudioResamplerFilterInterpolation">
+      <doc xml:space="preserve">The different filter interpolation methods.</doc>
+      <member name="none"
+              value="0"
+              c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE"
+              glib:nick="none">
+        <doc xml:space="preserve">no interpolation</doc>
+      </member>
+      <member name="linear"
+              value="1"
+              c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR"
+              glib:nick="linear">
+        <doc xml:space="preserve">linear interpolation of the
+  filter coeficients.</doc>
+      </member>
+      <member name="cubic"
+              value="2"
+              c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC"
+              glib:nick="cubic">
+        <doc xml:space="preserve">cubic interpolation of the
+  filter coeficients.</doc>
+      </member>
+    </enumeration>
+    <enumeration name="AudioResamplerFilterMode"
+                 glib:type-name="GstAudioResamplerFilterMode"
+                 glib:get-type="gst_audio_resampler_filter_mode_get_type"
+                 c:type="GstAudioResamplerFilterMode">
+      <doc xml:space="preserve">Select for the filter tables should be set up.</doc>
+      <member name="interpolated"
+              value="0"
+              c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED"
+              glib:nick="interpolated">
+        <doc xml:space="preserve">Use interpolated filter tables. This
+    uses less memory but more CPU and is slightly less accurate but it allows for more
+    efficient variable rate resampling with gst_audio_resampler_update().</doc>
+      </member>
+      <member name="full"
+              value="1"
+              c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_FULL"
+              glib:nick="full">
+        <doc xml:space="preserve">Use full filter table. This uses more memory
+    but less CPU.</doc>
+      </member>
+      <member name="auto"
+              value="2"
+              c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO"
+              glib:nick="auto">
+        <doc xml:space="preserve">Automatically choose between interpolated
+    and full filter tables.</doc>
+      </member>
+    </enumeration>
+    <bitfield name="AudioResamplerFlags"
+              glib:type-name="GstAudioResamplerFlags"
+              glib:get-type="gst_audio_resampler_flags_get_type"
+              c:type="GstAudioResamplerFlags">
+      <doc xml:space="preserve">Different resampler flags.</doc>
+      <member name="none"
+              value="0"
+              c:identifier="GST_AUDIO_RESAMPLER_FLAG_NONE"
+              glib:nick="none">
+        <doc xml:space="preserve">no flags</doc>
+      </member>
+      <member name="non_interleaved_in"
+              value="1"
+              c:identifier="GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN"
+              glib:nick="non-interleaved-in">
+        <doc xml:space="preserve">input samples are non-interleaved.
+   an array of blocks of samples, one for each channel, should be passed to the
+   resample function.</doc>
+      </member>
+      <member name="non_interleaved_out"
+              value="2"
+              c:identifier="GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT"
+              glib:nick="non-interleaved-out">
+        <doc xml:space="preserve">output samples are non-interleaved.
+   an array of blocks of samples, one for each channel, should be passed to the
+   resample function.</doc>
+      </member>
+      <member name="variable_rate"
+              value="4"
+              c:identifier="GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE"
+              glib:nick="variable-rate">
+        <doc xml:space="preserve">optimize for dynamic updates of the sample
+   rates with gst_audio_resampler_update(). This will select an interpolating filter
+   when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.</doc>
+      </member>
+    </bitfield>
+    <enumeration name="AudioResamplerMethod"
+                 version="1.6"
+                 glib:type-name="GstAudioResamplerMethod"
+                 glib:get-type="gst_audio_resampler_method_get_type"
+                 c:type="GstAudioResamplerMethod">
+      <doc xml:space="preserve">Different subsampling and upsampling methods</doc>
+      <member name="nearest"
+              value="0"
+              c:identifier="GST_AUDIO_RESAMPLER_METHOD_NEAREST"
+              glib:nick="nearest">
+        <doc xml:space="preserve">Duplicates the samples when
+   upsampling and drops when downsampling</doc>
+      </member>
+      <member name="linear"
+              value="1"
+              c:identifier="GST_AUDIO_RESAMPLER_METHOD_LINEAR"
+              glib:nick="linear">
+        <doc xml:space="preserve">Uses linear interpolation to reconstruct
+   missing samples and averaging to downsample</doc>
+      </member>
+      <member name="cubic"
+              value="2"
+              c:identifier="GST_AUDIO_RESAMPLER_METHOD_CUBIC"
+              glib:nick="cubic">
+        <doc xml:space="preserve">Uses cubic interpolation</doc>
+      </member>
+      <member name="blackman_nuttall"
+              value="3"
+              c:identifier="GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL"
+              glib:nick="blackman-nuttall">
+        <doc xml:space="preserve">Uses Blackman-Nuttall windowed sinc interpolation</doc>
+      </member>
+      <member name="kaiser"
+              value="4"
+              c:identifier="GST_AUDIO_RESAMPLER_METHOD_KAISER"
+              glib:nick="kaiser">
+        <doc xml:space="preserve">Uses Kaiser windowed sinc interpolation</doc>
+      </member>
+    </enumeration>
     <class name="AudioRingBuffer"
            c:symbol-prefix="audio_ring_buffer"
            c:type="GstAudioRingBuffer"
@@ -7627,15 +8149,15 @@ is not clipped</doc>
         <parameter name="segment" transfer-ownership="none">
           <doc xml:space="preserve">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
           the buffer should be clipped.</doc>
-          <type name="Gst.Segment" c:type="GstSegment*"/>
+          <type name="Gst.Segment" c:type="const GstSegment*"/>
         </parameter>
         <parameter name="rate" transfer-ownership="none">
           <doc xml:space="preserve">sample rate.</doc>
           <type name="gint" c:type="gint"/>
         </parameter>
         <parameter name="bpf" transfer-ownership="none">
-          <doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample
-* channels.</doc>
+          <doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
+number of channels.</doc>
           <type name="gint" c:type="gint"/>
         </parameter>
       </parameters>
@@ -7688,7 +8210,7 @@ This function returns a reasonable fallback channel-mask and should be
 called as a last resort when the specific channel map is unknown.</doc>
       <return-value transfer-ownership="none">
         <doc xml:space="preserve">a fallback channel-mask for @channels or 0 when there is no
-mask.</doc>
+mask and mono.</doc>
         <type name="guint64" c:type="guint64"/>
       </return-value>
       <parameters>
@@ -7804,6 +8326,34 @@ in the order required by GStreamer.</doc>
         </parameter>
       </parameters>
     </function>
+    <function name="audio_channel_positions_to_string"
+              c:identifier="gst_audio_channel_positions_to_string">
+      <doc xml:space="preserve">Converts @position to a human-readable string representation for
+debugging purposes.</doc>
+      <return-value transfer-ownership="full">
+        <doc xml:space="preserve">a newly allocated string representing
+@position
+
+Since 1.10</doc>
+        <type name="utf8" c:type="gchar*"/>
+      </return-value>
+      <parameters>
+        <parameter name="position" transfer-ownership="none">
+          <doc xml:space="preserve">The %GstAudioChannelPositions
+  to convert.</doc>
+          <array length="1"
+                 zero-terminated="0"
+                 c:type="GstAudioChannelPosition*">
+            <type name="AudioChannelPosition"
+                  c:type="GstAudioChannelPosition"/>
+          </array>
+        </parameter>
+        <parameter name="channels" transfer-ownership="none">
+          <doc xml:space="preserve">The number of channels.</doc>
+          <type name="gint" c:type="gint"/>
+        </parameter>
+      </parameters>
+    </function>
     <function name="audio_channel_positions_to_valid_order"
               c:identifier="gst_audio_channel_positions_to_valid_order">
       <doc xml:space="preserve">Reorders the channel positions in @position from any order to
@@ -7888,7 +8438,7 @@ parameters for details about the options and values.</doc>
       </return-value>
       <parameters>
         <parameter name="flags" transfer-ownership="none">
-          <doc xml:space="preserve">#GstAudioConverterFlags</doc>
+          <doc xml:space="preserve">extra #GstAudioConverterFlags</doc>
           <type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
         </parameter>
         <parameter name="in_info" transfer-ownership="none">
@@ -8210,6 +8760,74 @@ positions and the same positions, only in a different order.</doc>
         </parameter>
       </parameters>
     </function>
+    <function name="audio_resampler_new"
+              c:identifier="gst_audio_resampler_new"
+              moved-to="AudioResampler.new">
+      <doc xml:space="preserve">Make a new resampler.</doc>
+      <return-value transfer-ownership="full" skip="1">
+        <doc xml:space="preserve">%TRUE on success</doc>
+        <type name="AudioResampler" c:type="GstAudioResampler*"/>
+      </return-value>
+      <parameters>
+        <parameter name="method" transfer-ownership="none">
+          <doc xml:space="preserve">a #GstAudioResamplerMethod</doc>
+          <type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
+        </parameter>
+        <parameter name="flags" transfer-ownership="none">
+          <doc xml:space="preserve">#GstAudioResamplerFlags</doc>
+          <type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
+        </parameter>
+        <parameter name="format" transfer-ownership="none">
+          <type name="AudioFormat" c:type="GstAudioFormat"/>
+        </parameter>
+        <parameter name="channels" transfer-ownership="none">
+          <type name="gint" c:type="gint"/>
+        </parameter>
+        <parameter name="in_rate" transfer-ownership="none">
+          <doc xml:space="preserve">input rate</doc>
+          <type name="gint" c:type="gint"/>
+        </parameter>
+        <parameter name="out_rate" transfer-ownership="none">
+          <doc xml:space="preserve">output rate</doc>
+          <type name="gint" c:type="gint"/>
+        </parameter>
+        <parameter name="options" transfer-ownership="none">
+          <doc xml:space="preserve">extra options</doc>
+          <type name="Gst.Structure" c:type="GstStructure*"/>
+        </parameter>
+      </parameters>
+    </function>
+    <function name="audio_resampler_options_set_quality"
+              c:identifier="gst_audio_resampler_options_set_quality"
+              moved-to="AudioResampler.options_set_quality">
+      <doc xml:space="preserve">Set the parameters for resampling from @in_rate to @out_rate using @method
+for @quality in @options.</doc>
+      <return-value transfer-ownership="none">
+        <type name="none" c:type="void"/>
+      </return-value>
+      <parameters>
+        <parameter name="method" transfer-ownership="none">
+          <doc xml:space="preserve">a #GstAudioResamplerMethod</doc>
+          <type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
+        </parameter>
+        <parameter name="quality" transfer-ownership="none">
+          <doc xml:space="preserve">the quality</doc>
+          <type name="guint" c:type="guint"/>
+        </parameter>
+        <parameter name="in_rate" transfer-ownership="none">
+          <doc xml:space="preserve">the input rate</doc>
+          <type name="gint" c:type="gint"/>
+        </parameter>
+        <parameter name="out_rate" transfer-ownership="none">
+          <doc xml:space="preserve">the output rate</doc>
+          <type name="gint" c:type="gint"/>
+        </parameter>
+        <parameter name="options" transfer-ownership="none">
+          <doc xml:space="preserve">a #GstStructure</doc>
+          <type name="Gst.Structure" c:type="GstStructure*"/>
+        </parameter>
+      </parameters>
+    </function>
     <function name="buffer_add_audio_clipping_meta"
               c:identifier="gst_buffer_add_audio_clipping_meta"
               version="1.8">