audioaggregator: refactor conversion API

For the rationale, see:

https://bugzilla.gnome.org/show_bug.cgi?id=793917

Also test audiomixer conversion of current output buffer

gst-libs/gst/audio/gstaudioaggregator.c

@@ -28,22 +28,24 @@
  * @title: GstAudioAggregator
  * @short_description: Base class that manages a set of audio input pads
  * with the purpose of aggregating or mixing their raw audio input buffers
- * @see_also: #GstAggregator
+ * @see_also: #GstAggregator, #GstAudioMixer
  *
- * #GstAudioAggregator will perform conversion on the data arriving
+ * Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
- * on its sink pads, based on the format expected downstream.
+ * their source and sink pads,
+ * gst_element_class_add_static_pad_template_with_gtype() is a convenient
+ * helper.
  *
- * Subclasses can opt out of the conversion behaviour by setting
+ * #GstAudioAggregator can perform conversion on the data arriving
- * #GstAudioAggregatorClass.convert_buffer() to %NULL.
+ * on its sink pads, based on the format expected downstream: in order
+ * to enable that behaviour, the GType of the sink pads must either be
+ * a (subclass of) #GstAudioAggregatorConvertPad to use the default
+ * #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
+ * implementing #GstAudioAggregatorPad.convert_buffer.
  *
- * Subclasses that wish to use the default conversion implementation
+ * To allow for the output caps to change, the mechanism is the same as
- * should use a (subclass of) #GstAudioAggregatorConvertPad as their
+ * above, with the GType of the source pad.
- * #GstAggregatorClass.sinkpads_type, as it will cache the created
- * #GstAudioConverter and install a property allowing to configure it,
- * #GstAudioAggregatorPad:converter-config.
  *
- * Subclasses that wish to perform custom conversion should override
+ * See #GstAudioMixer for an example.
- * #GstAudioAggregatorClass.convert_buffer().
  *
  * When conversion is enabled, #GstAudioAggregator will accept
  * any type of raw audio caps and perform conversion
@@ -54,10 +56,6 @@
  * the first configured sink pad to finish fixating its source pad
  * caps.
  *
- * Additionally, handling audio conversion directly in the element
- * means that this base class supports safely reconfiguring its
- * source pad.
- *
  * A notable exception for now is the sample rate, sink pads must
  * have the same sample rate as either the downstream requirement,
  * or the first configured pad, or a combination of both (when
@@ -223,12 +221,22 @@ gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
   aaggcpad->priv->converter_config_changed = FALSE;
 }
 
+static void
+gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad *
+    aaggpad)
+{
+  GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed =
+      TRUE;
+}
+
 static GstBuffer *
-gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorConvertPad *
+gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad *
-    aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
+    aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
     GstBuffer * input_buffer)
 {
   GstBuffer *res;
+  GstAudioAggregatorConvertPad *aaggcpad =
+      GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad);
 
   gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info,
       out_info);
@@ -327,6 +335,8 @@ gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
     klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstAudioAggregatorPadClass *aaggpad_class =
+      (GstAudioAggregatorPadClass *) klass;
   g_type_class_add_private (klass,
       sizeof (GstAudioAggregatorConvertPadPrivate));
 
@@ -339,6 +349,12 @@ gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
           "when converting this pad's audio buffers",
           GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  aaggpad_class->convert_buffer =
+      gst_audio_aggregator_convert_pad_convert_buffer;
+
+  aaggpad_class->update_conversion_info =
+      gst_audio_aggregator_pad_update_conversion_info;
+
   gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize;
 }
 
@@ -449,64 +465,16 @@ gst_audio_aggregator_get_next_time (GstAggregator * agg)
   return next_time;
 }
 
-static GstBuffer *
-gst_audio_aggregator_convert_once (GstAudioAggregator * aagg, GstPad * pad,
-    GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
-{
-  GstAudioConverter *converter =
-      gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE,
-      in_info, out_info, NULL);
-  gint insize = gst_buffer_get_size (buffer);
-  gsize insamples = insize / in_info->bpf;
-  gsize outsamples = gst_audio_converter_get_out_frames (converter,
-      insamples);
-  gint outsize = outsamples * out_info->bpf;
-  GstMapInfo inmap, outmap;
-  GstBuffer *converted = gst_buffer_new_allocate (NULL, outsize, NULL);
-
-  gst_buffer_copy_into (converted, buffer,
-      GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
-      GST_BUFFER_COPY_META, 0, -1);
-
-  gst_buffer_map (buffer, &inmap, GST_MAP_READ);
-  gst_buffer_map (converted, &outmap, GST_MAP_WRITE);
-
-  gst_audio_converter_samples (converter,
-      GST_AUDIO_CONVERTER_FLAG_NONE,
-      (gpointer *) & inmap.data, insamples,
-      (gpointer *) & outmap.data, outsamples);
-
-  gst_buffer_unmap (buffer, &inmap);
-  gst_buffer_unmap (converted, &outmap);
-  gst_audio_converter_free (converter);
-
-  return converted;
-}
-
-static GstBuffer *
-gst_audio_aggregator_default_convert_buffer (GstAudioAggregator * aagg,
-    GstPad * pad, GstAudioInfo * in_info, GstAudioInfo * out_info,
-    GstBuffer * buffer)
-{
-  if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
-    return
-        gst_audio_aggregator_convert_pad_convert_buffer
-        (GST_AUDIO_AGGREGATOR_CONVERT_PAD (pad),
-        &GST_AUDIO_AGGREGATOR_PAD (pad)->info, out_info, buffer);
-  else
-    return gst_audio_aggregator_convert_once (aagg, pad, in_info, out_info,
-        buffer);
-}
-
 static GstBuffer *
 gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
     GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
 {
-  GstAudioAggregatorClass *klass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
+  GstAudioAggregatorPadClass *klass = GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad);
+  GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad);
 
   g_assert (klass->convert_buffer);
 
-  return klass->convert_buffer (aagg, pad, in_info, out_info, buffer);
+  return klass->convert_buffer (aaggpad, in_info, out_info, buffer);
 }
 
 static void
@@ -542,7 +510,6 @@ gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
       gst_audio_aggregator_negotiated_src_caps;
 
   klass->create_output_buffer = gst_audio_aggregator_create_output_buffer;
-  klass->convert_buffer = gst_audio_aggregator_default_convert_buffer;
 
   GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator",
       GST_DEBUG_FG_MAGENTA, "GstAudioAggregator");
@@ -746,11 +713,12 @@ gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad,
     gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
     ret = FALSE;
   } else {
+    GstAudioAggregatorPadClass *klass =
+        GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
     GST_OBJECT_LOCK (aaggpad);
     gst_audio_info_from_caps (&aaggpad->info, caps);
-    if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
+    if (klass->update_conversion_info)
-      GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
+      klass->update_conversion_info (aaggpad);
-          priv->converter_config_changed = TRUE;
     GST_OBJECT_UNLOCK (aaggpad);
   }
 
@@ -792,10 +760,9 @@ gst_audio_aggregator_update_src_caps (GstAggregator * agg,
 static GstCaps *
 gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps)
 {
-  GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
   GstAudioAggregatorPad *first_configured_pad;
 
-  if (!aaggclass->convert_buffer)
+  if (!GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer)
     return
         GST_AGGREGATOR_CLASS
         (gst_audio_aggregator_parent_class)->fixate_src_caps (agg, caps);
@@ -844,10 +811,11 @@ gst_audio_aggregator_update_converters (GstAudioAggregator * aagg,
 
   for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
     GstAudioAggregatorPad *aaggpad = l->data;
+    GstAudioAggregatorPadClass *klass =
+        GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
 
-    if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad))
+    if (klass->update_conversion_info)
-      GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->
+      klass->update_conversion_info (aaggpad);
-          priv->converter_config_changed = TRUE;
 
     /* If we currently were mixing a buffer, we need to convert it to the new
      * format */
@@ -865,7 +833,6 @@ static gboolean
 gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
 {
   GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
-  GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg);
   GstAudioInfo info;
 
   GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps);
@@ -878,12 +845,19 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps)
   GST_AUDIO_AGGREGATOR_LOCK (aagg);
   GST_OBJECT_LOCK (aagg);
 
-  if (aaggclass->convert_buffer) {
+  if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer) {
     gst_audio_aggregator_update_converters (aagg, &info);
 
     if (aagg->priv->current_buffer
         && !gst_audio_info_is_equal (&aagg->info, &info)) {
-      GstBuffer *converted =
+      GstBuffer *converted;
+      GstAudioAggregatorPadClass *klass =
+          GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);
+
+      if (klass->update_conversion_info)
+        klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad));
+
+      converted =
           gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info,
           &info, aagg->priv->current_buffer);
       gst_buffer_unref (aagg->priv->current_buffer);
@@ -1324,7 +1298,6 @@ static gboolean
 gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
     GstAudioAggregatorPad * pad)
 {
-  GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg);
   GstClockTime start_time, end_time;
   gboolean discont = FALSE;
   guint64 start_offset, end_offset;
@@ -1333,7 +1306,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
   GstAggregator *agg = GST_AGGREGATOR (aagg);
   GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
 
-  if (aaggclass->convert_buffer) {
+  if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
     rate = GST_AUDIO_INFO_RATE (&aagg->info);
     bpf = GST_AUDIO_INFO_BPF (&aagg->info);
   } else {
@@ -1802,7 +1775,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
 
     /* New buffer? */
     if (!pad->priv->buffer) {
-      if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad))
+      if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
         pad->priv->buffer =
             gst_audio_aggregator_convert_buffer
             (aagg, GST_PAD (pad), &pad->info, &aagg->info,

gst-libs/gst/audio/gstaudioaggregator.h

@@ -79,12 +79,21 @@ struct _GstAudioAggregatorPad
 
 /**
  * GstAudioAggregatorPadClass:
- *
+ * @convert_buffer: Convert a buffer from one format to another.
+ * @update_conversion_info: Called when either the input or output
+ *  formats have changed.
  */
 struct _GstAudioAggregatorPadClass
   {
   GstAggregatorPadClass   parent_class;
 
+  GstBuffer * (* convert_buffer) (GstAudioAggregatorPad * pad,
+                                  GstAudioInfo *in_info,
+                                  GstAudioInfo *out_info,
+                                  GstBuffer * buffer);
+
+  void        (* update_conversion_info) (GstAudioAggregatorPad *pad);
+
   /*< private >*/
   gpointer      _gst_reserved[GST_PADDING_LARGE];
 };
@@ -181,10 +190,6 @@ struct _GstAudioAggregator
  *  buffer.  The in_offset and out_offset are in "frames", which is
  *  the size of a sample times the number of channels. Returns TRUE if
  *  any non-silence was added to the buffer
- * @convert_buffer: Convert a buffer from one format to another. The pad
- *  is either a sinkpad, when converting an input buffer, or the source pad,
- *  when converting the output buffer after a downstream format change is
- *  requested.
  */
 struct _GstAudioAggregatorClass {
   GstAggregatorClass   parent_class;
@@ -194,11 +199,6 @@ struct _GstAudioAggregatorClass {
   gboolean (* aggregate_one_buffer) (GstAudioAggregator * aagg,
       GstAudioAggregatorPad * pad, GstBuffer * inbuf, guint in_offset,
       GstBuffer * outbuf, guint out_offset, guint num_frames);
-  GstBuffer * (* convert_buffer) (GstAudioAggregator *aagg,
-                                  GstPad * pad,
-                                  GstAudioInfo *in_info,
-                                  GstAudioInfo *out_info,
-                                  GstBuffer * buffer);
 
   /*< private >*/
   gpointer          _gst_reserved[GST_PADDING_LARGE];

gst/audiomixer/gstaudiointerleave.c

@@ -560,8 +560,8 @@ gst_audio_interleave_class_init (GstAudioInterleaveClass * klass)
   gobject_class->get_property = gst_audio_interleave_get_property;
   gobject_class->finalize = gst_audio_interleave_finalize;
 
-  gst_element_class_add_static_pad_template (gstelement_class,
+  gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
-      &gst_audio_interleave_src_template);
+      &gst_audio_interleave_src_template, GST_TYPE_AUDIO_AGGREGATOR_PAD);
   gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
       &gst_audio_interleave_sink_template, GST_TYPE_AUDIO_INTERLEAVE_PAD);
   gst_element_class_set_static_metadata (gstelement_class, "AudioInterleave",
@@ -580,7 +580,6 @@ gst_audio_interleave_class_init (GstAudioInterleaveClass * klass)
   agg_class->negotiated_src_caps = gst_audio_interleave_negotiated_src_caps;
 
   aagg_class->aggregate_one_buffer = gst_audio_interleave_aggregate_one_buffer;
-  aagg_class->convert_buffer = NULL;
 
   /**
    * GstInterleave:channel-positions

gst/audiomixer/gstaudiomixer.c

@@ -224,8 +224,8 @@ gst_audiomixer_class_init (GstAudioMixerClass * klass)
   GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass;
 
-  gst_element_class_add_static_pad_template (gstelement_class,
+  gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
-      &gst_audiomixer_src_template);
+      &gst_audiomixer_src_template, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD);
   gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
       &gst_audiomixer_sink_template, GST_TYPE_AUDIO_MIXER_PAD);
   gst_element_class_set_static_metadata (gstelement_class, "AudioMixer",

tests/check/elements/audiomixer.c

@@ -1849,6 +1849,141 @@ GST_START_TEST (test_change_output_caps)
 
 GST_END_TEST;
 
+/* In this test, we create two input buffers with a duration of 1 second,
+ * and require the audiomixer to output 1.5 second long buffers.
+ *
+ * After we have input two buffers, we change the output format
+ * from S8 to S32, then push a last buffer.
+ *
+ * This makes audioaggregator convert its "half-mixed" current_buffer,
+ * we can then ensure that the second output buffer is as expected.
+ */
+GST_START_TEST (test_change_output_caps_mid_output_buffer)
+{
+  GstSegment segment;
+  GstElement *bin, *audiomixer, *capsfilter, *sink;
+  GstBus *bus;
+  GstPad *sinkpad;
+  gboolean res;
+  GstStateChangeReturn state_res;
+  GstFlowReturn ret;
+  GstEvent *event;
+  GstBuffer *buffer;
+  GstCaps *caps;
+  GstQuery *drain;
+  GstMapInfo inmap;
+  GstMapInfo outmap;
+  guint i;
+
+  bin = gst_pipeline_new ("pipeline");
+  bus = gst_element_get_bus (bin);
+  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
+  g_object_set (audiomixer, "output-buffer-duration", 1500 * GST_MSECOND, NULL);
+  capsfilter = gst_element_factory_make ("capsfilter", NULL);
+  sink = gst_element_factory_make ("fakesink", "sink");
+  gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
+
+  res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
+  fail_unless (res == TRUE, NULL);
+
+  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
+  fail_if (sinkpad == NULL, NULL);
+
+  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, "S8",
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
+
+  gst_pad_set_caps (sinkpad, caps);
+  g_object_set (capsfilter, "caps", caps, NULL);
+  gst_caps_unref (caps);
+
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  segment.start = 0;
+  segment.stop = 3 * GST_SECOND;
+  segment.time = 0;
+  event = gst_event_new_segment (&segment);
+  gst_pad_send_event (sinkpad, event);
+
+  buffer = new_buffer (10, 0, 0, 1 * GST_SECOND, 0);
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  buffer = new_buffer (10, 0, 1 * GST_SECOND, 1 * GST_SECOND, 0);
+  gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
+  memset (inmap.data, 1, 10);
+  gst_buffer_unmap (buffer, &inmap);
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  drain = gst_query_new_drain ();
+  gst_pad_query (sinkpad, drain);
+  gst_query_unref (drain);
+
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
+  g_object_set (capsfilter, "caps", caps, NULL);
+  gst_caps_unref (caps);
+
+  gst_buffer_replace (&handoff_buffer, NULL);
+  g_object_set (sink, "signal-handoffs", TRUE, NULL);
+  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
+
+  buffer = new_buffer (10, 0, 2 * GST_SECOND, 1 * GST_SECOND, 0);
+  gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
+  memset (inmap.data, 0, 10);
+  gst_buffer_unmap (buffer, &inmap);
+  ret = gst_pad_chain (sinkpad, buffer);
+  ck_assert_int_eq (ret, GST_FLOW_OK);
+
+  drain = gst_query_new_drain ();
+  gst_pad_query (sinkpad, drain);
+  gst_query_unref (drain);
+
+  fail_unless (handoff_buffer);
+  fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 60);
+
+  gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
+  for (i = 0; i < 15; i++) {
+    guint32 sample;
+
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+    sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
+#else
+    sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
+#endif
+
+    if (i < 5) {
+      fail_unless_equals_int (sample, 1 << 24);
+    } else {
+      fail_unless_equals_int (sample, 0);
+    }
+  }
+
+  gst_buffer_unmap (handoff_buffer, &outmap);
+
+  gst_element_release_request_pad (audiomixer, sinkpad);
+  gst_object_unref (sinkpad);
+  gst_element_set_state (bin, GST_STATE_NULL);
+  gst_bus_remove_signal_watch (bus);
+  gst_object_unref (bus);
+  gst_object_unref (bin);
+}
+
+GST_END_TEST;
 static Suite *
 audiomixer_suite (void)
 {
@@ -1876,6 +2011,7 @@ audiomixer_suite (void)
   tcase_add_test (tc_chain, test_sinkpad_property_controller);
   tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
   tcase_add_test (tc_chain, test_change_output_caps);
+  tcase_add_test (tc_chain, test_change_output_caps_mid_output_buffer);
 
   /* Use a longer timeout */
 #ifdef HAVE_VALGRIND