diff --git a/gst-libs/gst/audio/gstaudioringbuffer.c b/gst-libs/gst/audio/gstaudioringbuffer.c
index 2e9af680a0..203472347f 100644
--- a/gst-libs/gst/audio/gstaudioringbuffer.c
+++ b/gst-libs/gst/audio/gstaudioringbuffer.c
@@ -1291,6 +1291,12 @@ gst_audio_ring_buffer_set_sample (GstAudioRingBuffer * buf, guint64 sample)
       sample, buf->segbase);
 }
 
+/**
+ * default_clear_all:
+ * @buf: the #GstAudioRingBuffer to clear
+ *
+ * Fill the ringbuffer with silence.
+ */
 static void
 default_clear_all (GstAudioRingBuffer * buf)
 {
@@ -1311,7 +1317,7 @@ default_clear_all (GstAudioRingBuffer * buf)
  * gst_audio_ring_buffer_clear_all:
  * @buf: the #GstAudioRingBuffer to clear
  *
- * Fill the ringbuffer with silence.
+ * Clear all samples from the ringbuffer.
  *
  * MT safe.
  */
diff --git a/gst-libs/gst/audio/gstaudioringbuffer.h b/gst-libs/gst/audio/gstaudioringbuffer.h
index 9dee681b8f..615041f71d 100644
--- a/gst-libs/gst/audio/gstaudioringbuffer.h
+++ b/gst-libs/gst/audio/gstaudioringbuffer.h
@@ -234,7 +234,10 @@ struct _GstAudioRingBuffer {
  * @activate: activate the thread that starts pulling and monitoring the
  * consumed segments in the device.
  * @commit: write samples into the ringbuffer
- * @clear_all: clear the entire ringbuffer.
+ * @clear_all: Optional.
+ *             Clear the entire ringbuffer.
+ *             Subclasses should chain up to the parent implementation to
+ *             invoke the default handler.
  *
  * The vmethods that subclasses can override to implement the ringbuffer.
  */
diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c
index 3892432bfd..5eb6039baf 100644
--- a/gst-libs/gst/audio/gstaudiosink.c
+++ b/gst-libs/gst/audio/gstaudiosink.c
@@ -117,10 +117,12 @@ static gboolean gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf,
 static gboolean gst_audio_sink_ring_buffer_release (GstAudioRingBuffer * buf);
 static gboolean gst_audio_sink_ring_buffer_start (GstAudioRingBuffer * buf);
 static gboolean gst_audio_sink_ring_buffer_pause (GstAudioRingBuffer * buf);
+static gboolean gst_audio_sink_ring_buffer_resume (GstAudioRingBuffer * buf);
 static gboolean gst_audio_sink_ring_buffer_stop (GstAudioRingBuffer * buf);
 static guint gst_audio_sink_ring_buffer_delay (GstAudioRingBuffer * buf);
 static gboolean gst_audio_sink_ring_buffer_activate (GstAudioRingBuffer * buf,
     gboolean active);
+static void gst_audio_sink_ring_buffer_clear_all (GstAudioRingBuffer * buf);
 
 /* ringbuffer abstract base class */
 static GType
@@ -176,14 +178,15 @@ gst_audio_sink_ring_buffer_class_init (GstAudioSinkRingBufferClass * klass)
   gstringbuffer_class->pause =
       GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_pause);
   gstringbuffer_class->resume =
-      GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_start);
+      GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_resume);
   gstringbuffer_class->stop =
       GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_stop);
-
   gstringbuffer_class->delay =
       GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_delay);
   gstringbuffer_class->activate =
       GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_activate);
+  gstringbuffer_class->clear_all =
+      GST_DEBUG_FUNCPTR (gst_audio_sink_ring_buffer_clear_all);
 }
 
 typedef gint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
@@ -531,12 +534,36 @@ gst_audio_sink_ring_buffer_pause (GstAudioRingBuffer * buf)
   csink = GST_AUDIO_SINK_GET_CLASS (sink);
 
   /* unblock any pending writes to the audio device */
-  if (csink->reset) {
+  if (csink->pause) {
+    GST_DEBUG_OBJECT (sink, "pause...");
+    csink->pause (sink);
+    GST_DEBUG_OBJECT (sink, "pause done");
+  } else if (csink->reset) {
+    /* fallback to reset for audio sinks that don't provide pause */
     GST_DEBUG_OBJECT (sink, "reset...");
     csink->reset (sink);
     GST_DEBUG_OBJECT (sink, "reset done");
   }
+  return TRUE;
+}
 
+static gboolean
+gst_audio_sink_ring_buffer_resume (GstAudioRingBuffer * buf)
+{
+  GstAudioSink *sink;
+  GstAudioSinkClass *csink;
+
+  sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+  csink = GST_AUDIO_SINK_GET_CLASS (sink);
+
+  if (csink->resume) {
+    GST_DEBUG_OBJECT (sink, "resume...");
+    csink->resume (sink);
+    GST_DEBUG_OBJECT (sink, "resume done");
+  } else {
+    /* fallback to start for audio sinks that don't provide resume */
+    gst_audio_sink_ring_buffer_start (buf);
+  }
   return TRUE;
 }
 
@@ -550,7 +577,12 @@ gst_audio_sink_ring_buffer_stop (GstAudioRingBuffer * buf)
   csink = GST_AUDIO_SINK_GET_CLASS (sink);
 
   /* unblock any pending writes to the audio device */
-  if (csink->reset) {
+  if (csink->stop) {
+    GST_DEBUG_OBJECT (sink, "stop...");
+    csink->stop (sink);
+    GST_DEBUG_OBJECT (sink, "stop done");
+  } else if (csink->reset) {
+    /* fallback to reset for audio sinks that don't provide stop */
     GST_DEBUG_OBJECT (sink, "reset...");
     csink->reset (sink);
     GST_DEBUG_OBJECT (sink, "reset done");
@@ -582,6 +614,24 @@ gst_audio_sink_ring_buffer_delay (GstAudioRingBuffer * buf)
   return res;
 }
 
+static void
+gst_audio_sink_ring_buffer_clear_all (GstAudioRingBuffer * buf)
+{
+  GstAudioSink *sink;
+  GstAudioSinkClass *csink;
+
+  sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+  csink = GST_AUDIO_SINK_GET_CLASS (sink);
+
+  if (csink->clear_all) {
+    GST_DEBUG_OBJECT (sink, "clear all");
+    csink->clear_all (sink);
+  }
+
+  /* chain up to the parent implementation */
+  ring_parent_class->clear_all (buf);
+}
+
 /* AudioSink signals and args */
 enum
 {
diff --git a/gst-libs/gst/audio/gstaudiosink.h b/gst-libs/gst/audio/gstaudiosink.h
index f29df97d82..409cf6b017 100644
--- a/gst-libs/gst/audio/gstaudiosink.h
+++ b/gst-libs/gst/audio/gstaudiosink.h
@@ -66,12 +66,25 @@ struct _GstAudioSink {
  * @unprepare: Undo operations done in prepare.
  * @close: Close the device.
  * @write: Write data to the device.
- * @delay: Return how many frames are still in the device. This is used to
- *         drive the synchronisation.
+ *         This vmethod is allowed to block until all the data is written.
+ *         If such is the case then it is expected that pause, stop and
+ *         reset will unblock the write when called.
+ * @delay: Return how many frames are still in the device. Participates in
+ *         computing the time for audio clocks and drives the synchronisation.
  * @reset: Returns as quickly as possible from a write and flush any pending
  *         samples from the device.
- *
- * #GstAudioSink class. Override the vmethods to implement functionality.
+ *         This vmethod is deprecated. Please provide pause and stop instead.
+ * @pause: Pause the device and unblock write as fast as possible.
+ *         For retro compatibility, the audio sink will fallback
+ *         to calling reset if this vmethod is not provided.
+ * @resume: Resume the device.
+ *          For retro compatibility, the audio sink will fallback
+ *          to calling start if this vmethod is not provided.
+ * @stop: Stop the device and unblock write as fast as possible.
+ *        Pending samples are flushed from the device.
+ *        For retro compatibility, the audio sink will fallback
+ *        to calling reset if this vmethod is not provided.
+ * @clear-all: Clear the device.
  */
 struct _GstAudioSinkClass {
   GstAudioBaseSinkClass parent_class;
@@ -90,11 +103,19 @@ struct _GstAudioSinkClass {
   gint     (*write)     (GstAudioSink *sink, gpointer data, guint length);
   /* get number of frames queued in the device */
   guint    (*delay)     (GstAudioSink *sink);
-  /* reset the audio device, unblock from a write */
+  /* deprecated: reset the audio device, unblock from a write */
   void     (*reset)     (GstAudioSink *sink);
+  /* pause the audio device, unblock from a write */
+  void     (*pause)     (GstAudioSink *sink);
+  /* resume the audio device */
+  void     (*resume)    (GstAudioSink *sink);
+  /* stop the audio device, unblock from a write */
+  void     (*stop)      (GstAudioSink *sink);
+  /* clear the audio device */
+  void     (*clear_all) (GstAudioSink *sink);
 
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING];
+  gpointer _gst_reserved[GST_PADDING - 4];
 };
 
 GST_AUDIO_API