diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c
index b56a3887e5..85299ff25b 100644
--- a/tests/check/elements/audiomixer.c
+++ b/tests/check/elements/audiomixer.c
@@ -891,36 +891,17 @@ GST_END_TEST;
 
 
 static GstBuffer *handoff_buffer = NULL;
-static gboolean
-_quit (GMainLoop * ml)
-{
-  g_main_loop_quit (ml);
-
-  return G_SOURCE_REMOVE;
-}
 
 static void
 handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
-    GstClockTime * wanted_end)
+    gpointer user_data)
 {
   GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT
-      " -- %p DURATION is %" GST_TIME_FORMAT " -- WANTED END %" GST_TIME_FORMAT,
+      " -- %p DURATION is %" GST_TIME_FORMAT,
       gst_buffer_get_size (buffer), buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)),
-      GST_TIME_ARGS (*wanted_end));
+      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
 
   gst_buffer_replace (&handoff_buffer, buffer);
-
-
-  /* Buffers we push in will be 'cut' into different smaller buffers,
-   * we make sure that the last chunck was pushes before we concider the buffer
-   * we pushed as being done */
-  if (main_loop && *wanted_end
-      && *wanted_end <=
-      GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)) {
-    *wanted_end = 0;
-    g_idle_add ((GSourceFunc) _quit, main_loop);
-  }
 }
 
 /* check if clipping works as expected */
@@ -936,14 +917,10 @@ GST_START_TEST (test_clip)
   GstEvent *event;
   GstBuffer *buffer;
   GstCaps *caps;
-  GMainLoop *local_mainloop;
-  GstClockTime wanted_end = 0;
+  GstQuery *drain = gst_query_new_drain ();
 
   GST_INFO ("preparing test");
 
-  local_mainloop = g_main_loop_new (NULL, FALSE);
-  main_loop = NULL;
-
   /* build pipeline */
   bin = gst_pipeline_new ("pipeline");
   bus = gst_element_get_bus (bin);
@@ -957,8 +934,7 @@ GST_START_TEST (test_clip)
   audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
   sink = gst_element_factory_make ("fakesink", "sink");
   g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb,
-      &wanted_end);
+  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
   gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
 
   res = gst_element_link (audiomixer, sink);
@@ -999,16 +975,13 @@ GST_START_TEST (test_clip)
       buffer,
       GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
   ret = gst_pad_chain (sinkpad, buffer);
-  main_loop = local_mainloop;
+  ck_assert_int_eq (ret, GST_FLOW_OK);
 
   /* The aggregation is done in a dedicated thread, so we can't
-   * not know when it is actually going to happen, so we just add\
-   * a 100 ms timeout to be able to then check that the aggregation
-   * did happen as we do not have much other choice.
+   * not know when it is actually going to happen, so we use a DRAIN query
+   * to wait for it to complete.
    */
-  g_timeout_add (100, (GSourceFunc) _quit, main_loop);
-  g_main_loop_run (main_loop);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
+  gst_pad_query (sinkpad, drain);
   fail_unless (handoff_buffer == NULL);
 
   /* should be partially clipped */
@@ -1016,15 +989,14 @@ GST_START_TEST (test_clip)
   GST_BUFFER_TIMESTAMP (buffer) = 900 * GST_MSECOND;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
 
-  wanted_end = 135 * GST_MSECOND;
   GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
       GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
 
-  main_loop = local_mainloop;
   ret = gst_pad_chain (sinkpad, buffer);
   ck_assert_int_eq (ret, GST_FLOW_OK);
-  g_main_loop_run (main_loop);
+  gst_pad_query (sinkpad, drain);
+
   fail_unless (handoff_buffer != NULL);
   gst_buffer_replace (&handoff_buffer, NULL);
 
@@ -1033,18 +1005,18 @@ GST_START_TEST (test_clip)
   GST_BUFFER_TIMESTAMP (buffer) = 1 * GST_SECOND;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
 
-  wanted_end = 390 * GST_MSECOND;
   GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
       buffer,
       GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
   ret = gst_pad_chain (sinkpad, buffer);
   ck_assert_int_eq (ret, GST_FLOW_OK);
-  g_main_loop_run (main_loop);
+  gst_pad_query (sinkpad, drain);
   fail_unless (handoff_buffer != NULL);
   gst_buffer_replace (&handoff_buffer, NULL);
   fail_unless (handoff_buffer == NULL);
 
   /* should be clipped and ok */
+
   buffer = gst_buffer_new_and_alloc (44100);
   GST_BUFFER_TIMESTAMP (buffer) = 2 * GST_SECOND;
   GST_BUFFER_DURATION (buffer) = 250 * GST_MSECOND;
@@ -1052,9 +1024,8 @@ GST_START_TEST (test_clip)
       buffer,
       GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
   ret = gst_pad_chain (sinkpad, buffer);
-  g_timeout_add (100, (GSourceFunc) _quit, main_loop);
-  g_main_loop_run (main_loop);
   ck_assert_int_eq (ret, GST_FLOW_OK);
+  gst_pad_query (sinkpad, drain);
   fail_unless (handoff_buffer == NULL);
 
   gst_element_release_request_pad (audiomixer, sinkpad);
@@ -1063,6 +1034,7 @@ GST_START_TEST (test_clip)
   gst_bus_remove_signal_watch (bus);
   gst_object_unref (bus);
   gst_object_unref (bin);
+  gst_query_unref (drain);
 }
 
 GST_END_TEST;