diff --git a/sys/decklink/gstdecklinkaudiosink.cpp b/sys/decklink/gstdecklinkaudiosink.cpp
index 2e6d871cdb..e4a0ec8092 100644
--- a/sys/decklink/gstdecklinkaudiosink.cpp
+++ b/sys/decklink/gstdecklinkaudiosink.cpp
@@ -54,6 +54,9 @@ struct _GstDecklinkAudioSinkRingBuffer
 
   GstDecklinkOutput *output;
   GstDecklinkAudioSink *sink;
+
+  GMutex clock_id_lock;
+  GstClockID clock_id;
 };
 
 struct _GstDecklinkAudioSinkRingBufferClass
@@ -122,6 +125,7 @@ static void
 static void
 gst_decklink_audio_sink_ringbuffer_init (GstDecklinkAudioSinkRingBuffer * self)
 {
+  g_mutex_init (&self->clock_id_lock);
 }
 
 static void
@@ -132,6 +136,7 @@ gst_decklink_audio_sink_ringbuffer_finalize (GObject * object)
 
   gst_object_unref (self->sink);
   self->sink = NULL;
+  g_mutex_clear (&self->clock_id_lock);
 
   G_OBJECT_CLASS (ringbuffer_parent_class)->finalize (object);
 }
@@ -193,10 +198,60 @@ public:
     gint bpf;
     guint written, written_sum;
     HRESULT res;
+    const GstAudioRingBufferSpec *spec =
+        &GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->spec;
+    guint delay, max_delay;
 
     GST_LOG_OBJECT (m_ringbuffer->sink, "Writing audio samples (preroll: %d)",
         preroll);
 
+    delay =
+        gst_audio_ring_buffer_delay (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer));
+    max_delay = MAX ((spec->segtotal * spec->segsize) / 2, spec->segsize);
+    max_delay /= GST_AUDIO_INFO_BPF (&spec->info);
+    if (delay > max_delay) {
+      GstClock *clock =
+          gst_element_get_clock (GST_ELEMENT_CAST (m_ringbuffer->sink));
+      GstClockTime wait_time;
+      GstClockID clock_id;
+      GstClockReturn clock_ret;
+
+      GST_DEBUG_OBJECT (m_ringbuffer->sink, "Delay %u > max delay %u", delay,
+          max_delay);
+
+      wait_time =
+          gst_util_uint64_scale (delay - max_delay, GST_SECOND,
+          GST_AUDIO_INFO_RATE (&spec->info));
+      GST_DEBUG_OBJECT (m_ringbuffer->sink, "Waiting for %" GST_TIME_FORMAT,
+          GST_TIME_ARGS (wait_time));
+      wait_time += gst_clock_get_time (clock);
+
+      g_mutex_lock (&m_ringbuffer->clock_id_lock);
+      if (!GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->acquired) {
+        GST_DEBUG_OBJECT (m_ringbuffer->sink,
+            "Ringbuffer not acquired anymore");
+        g_mutex_unlock (&m_ringbuffer->clock_id_lock);
+        gst_object_unref (clock);
+        return S_OK;
+      }
+      clock_id = gst_clock_new_single_shot_id (clock, wait_time);
+      m_ringbuffer->clock_id = clock_id;
+      g_mutex_unlock (&m_ringbuffer->clock_id_lock);
+      gst_object_unref (clock);
+
+      clock_ret = gst_clock_id_wait (clock_id, NULL);
+
+      g_mutex_lock (&m_ringbuffer->clock_id_lock);
+      gst_clock_id_unref (clock_id);
+      m_ringbuffer->clock_id = NULL;
+      g_mutex_unlock (&m_ringbuffer->clock_id_lock);
+
+      if (clock_ret == GST_CLOCK_UNSCHEDULED) {
+        GST_DEBUG_OBJECT (m_ringbuffer->sink, "Flushing");
+        return S_OK;
+      }
+    }
+
     if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER_CAST
             (m_ringbuffer), &seg, &ptr, &len)) {
       GST_WARNING_OBJECT (m_ringbuffer->sink, "No segment available");
@@ -392,6 +447,11 @@ gst_decklink_audio_sink_ringbuffer_release (GstAudioRingBuffer * rb)
   GST_DEBUG_OBJECT (self->sink, "Release");
 
   if (self->output) {
+    g_mutex_lock (&self->clock_id_lock);
+    if (self->clock_id)
+      gst_clock_id_unschedule (self->clock_id);
+    g_mutex_unlock (&self->clock_id_lock);
+
     g_mutex_lock (&self->output->lock);
     self->output->audio_enabled = FALSE;
     if (self->output->start_scheduled_playback)