diff --git a/gst-libs/gst/audio/gstaudiobasesink.c b/gst-libs/gst/audio/gstaudiobasesink.c
index acd8c98a4a..3510754de2 100644
--- a/gst-libs/gst/audio/gstaudiobasesink.c
+++ b/gst-libs/gst/audio/gstaudiobasesink.c
@@ -2187,14 +2187,22 @@ gst_audio_base_sink_change_state (GstElement * element,
   GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element);
 
   switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      if (sink->ringbuffer == NULL) {
-        gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
-        sink->ringbuffer = gst_audio_base_sink_create_ringbuffer (sink);
-      }
+    case GST_STATE_CHANGE_NULL_TO_READY:{
+      GstAudioRingBuffer *rb;
+
+      gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
+      rb = gst_audio_base_sink_create_ringbuffer (sink);
+      if (rb == NULL)
+        goto create_failed;
+
+      GST_OBJECT_LOCK (sink);
+      sink->ringbuffer = rb;
+      GST_OBJECT_UNLOCK (sink);
+
       if (!gst_audio_ring_buffer_open_device (sink->ringbuffer))
         goto open_failed;
       break;
+    }
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       gst_audio_base_sink_reset_sync (sink);
       gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
@@ -2289,6 +2297,12 @@ gst_audio_base_sink_change_state (GstElement * element,
   return ret;
 
   /* ERRORS */
+create_failed:
+  {
+    /* subclass must post a meaningful error message */
+    GST_DEBUG_OBJECT (sink, "create failed");
+    return GST_STATE_CHANGE_FAILURE;
+  }
 open_failed:
   {
     /* subclass must post a meaningful error message */
diff --git a/gst-libs/gst/audio/gstaudiobasesrc.c b/gst-libs/gst/audio/gstaudiobasesrc.c
index 38b22b54c9..7517cd88f1 100644
--- a/gst-libs/gst/audio/gstaudiobasesrc.c
+++ b/gst-libs/gst/audio/gstaudiobasesrc.c
@@ -1124,17 +1124,23 @@ gst_audio_base_src_change_state (GstElement * element,
   GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element);
 
   switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
+    case GST_STATE_CHANGE_NULL_TO_READY:{
+      GstAudioRingBuffer *rb;
+
       GST_DEBUG_OBJECT (src, "NULL->READY");
+      gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
+      rb = gst_audio_base_src_create_ringbuffer (src);
+      if (rb == NULL)
+        goto create_failed;
+
       GST_OBJECT_LOCK (src);
-      if (src->ringbuffer == NULL) {
-        gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
-        src->ringbuffer = gst_audio_base_src_create_ringbuffer (src);
-      }
+      src->ringbuffer = rb;
       GST_OBJECT_UNLOCK (src);
+
       if (!gst_audio_ring_buffer_open_device (src->ringbuffer))
         goto open_failed;
       break;
+    }
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       GST_DEBUG_OBJECT (src, "READY->PAUSED");
       src->next_sample = -1;
@@ -1197,6 +1203,12 @@ gst_audio_base_src_change_state (GstElement * element,
   return ret;
 
   /* ERRORS */
+create_failed:
+  {
+    /* subclass must post a meaningful error message */
+    GST_DEBUG_OBJECT (src, "create failed");
+    return GST_STATE_CHANGE_FAILURE;
+  }
 open_failed:
   {
     /* subclass must post a meaningful error message */