diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c
index 2acbc55784..1ce7364671 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/gst/audioresample/gstaudioresample.c
@@ -295,14 +295,34 @@ static GstCaps *
 gst_audio_resample_transform_caps (GstBaseTransform * base,
     GstPadDirection direction, GstCaps * caps)
 {
+  const GValue *val;
+  GstStructure *s;
   GstCaps *res;
-  GstStructure *structure;
 
-  /* transform caps gives one single caps so we can just replace
-   * the rate property with our range. */
+  /* transform single caps into input_caps + input_caps with the rate
+   * field set to our supported range. This ensures that upstream knows
+   * about downstream's prefered rate(s) and can negotiate accordingly. */
   res = gst_caps_copy (caps);
-  structure = gst_caps_get_structure (res, 0);
-  gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+
+  /* first, however, check if the caps contain a range for the rate field, in
+   * which case that side isn't going to care much about the exact sample rate
+   * chosen and we should just assume things will get fixated to something sane
+   * and we may just as well offer our full range instead of the range in the
+   * caps. If the rate is not an int range value, it's likely to express a
+   * real preference or limitation and we should maintain that structure as
+   * preference by putting it first into the transformed caps, and only add
+   * our full rate range as second option  */
+  s = gst_caps_get_structure (res, 0);
+  val = gst_structure_get_value (s, "rate");
+  if (val == NULL || GST_VALUE_HOLDS_INT_RANGE (val)) {
+    /* overwrite existing range, or add field if it doesn't exist yet */
+    gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+  } else {
+    /* append caps with full range to existing caps with non-range rate field */
+    s = gst_structure_copy (s);
+    gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+    gst_caps_append_structure (res, s);
+  }
 
   return res;
 }
diff --git a/tests/check/elements/audioresample.c b/tests/check/elements/audioresample.c
index 7b8408d410..24f77a1400 100644
--- a/tests/check/elements/audioresample.c
+++ b/tests/check/elements/audioresample.c
@@ -674,6 +674,59 @@ GST_START_TEST (test_pipelines)
 }
 
 GST_END_TEST;
+
+GST_START_TEST (test_preference_passthrough)
+{
+  GstStateChangeReturn ret;
+  GstElement *pipeline, *src;
+  GstStructure *s;
+  GstMessage *msg;
+  GstCaps *caps;
+  GstPad *pad;
+  GstBus *bus;
+  GError *error = NULL;
+  gint rate = 0;
+
+  pipeline = gst_parse_launch ("audiotestsrc num-buffers=1 name=src ! "
+      "audioresample ! audio/x-raw-int,channels=1,width=16,depth=16,"
+      "endianness=1234,signed=true,rate=8000 ! "
+      "fakesink can-activate-pull=false", &error);
+  fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
+      error ? error->message : "(invalid error)");
+
+  ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
+  fail_unless_equals_int (ret, GST_STATE_CHANGE_ASYNC);
+
+  /* run until we receive EOS */
+  bus = gst_element_get_bus (pipeline);
+  fail_if (bus == NULL);
+  msg = gst_bus_timed_pop_filtered (bus, -1, GST_MESSAGE_EOS);
+  gst_message_unref (msg);
+  gst_object_unref (bus);
+
+  src = gst_bin_get_by_name (GST_BIN (pipeline), "src");
+  fail_unless (src != NULL);
+  pad = gst_element_get_static_pad (src, "src");
+  fail_unless (pad != NULL);
+  caps = gst_pad_get_negotiated_caps (pad);
+  GST_LOG ("negotiated audiotestsrc caps: %" GST_PTR_FORMAT, caps);
+  fail_unless (caps != NULL);
+  s = gst_caps_get_structure (caps, 0);
+  fail_unless (gst_structure_get_int (s, "rate", &rate));
+  /* there's no need to resample, audiotestsrc supports any rate, so make
+   * sure audioresample provided upstream with the right caps to negotiate
+   * this correctly */
+  fail_unless_equals_int (rate, 8000);
+  gst_caps_unref (caps);
+  gst_object_unref (pad);
+  gst_object_unref (src);
+
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+GST_END_TEST;
+
 #endif
 
 static Suite *
@@ -692,6 +745,7 @@ audioresample_suite (void)
 #ifndef GST_DISABLE_PARSE
   tcase_set_timeout (tc_chain, 360);
   tcase_add_test (tc_chain, test_pipelines);
+  tcase_add_test (tc_chain, test_preference_passthrough);
 #endif
 
   return s;