diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c
index f5ecd4e810..37e62a1b4d 100644
--- a/gst/audioparsers/gstaacparse.c
+++ b/gst/audioparsers/gstaacparse.c
@@ -243,7 +243,6 @@ gst_aacparse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
 
   GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
 
-  gst_pad_use_fixed_caps (GST_BASE_PARSE (aacparse)->srcpad);
   res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
   gst_caps_unref (src_caps);
   return res;
@@ -265,6 +264,7 @@ gst_aacparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
   GstAacParse *aacparse;
   GstStructure *structure;
   gchar *caps_str;
+  const GValue *value;
 
   aacparse = GST_AACPARSE (parse);
   structure = gst_caps_get_structure (caps, 0);
@@ -276,12 +276,11 @@ gst_aacparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
   /* This is needed at least in case of RTP
    * Parses the codec_data information to get ObjectType,
    * number of channels and samplerate */
-  if (gst_structure_has_field (structure, "codec_data")) {
+  value = gst_structure_get_value (structure, "codec_data");
+  if (value) {
+    GstBuffer *buf = gst_value_get_buffer (value);
 
-    const GValue *value = gst_structure_get_value (structure, "codec_data");
-
-    if (value) {
-      GstBuffer *buf = gst_value_get_buffer (value);
+    if (buf) {
       const guint8 *buffer = GST_BUFFER_DATA (buf);
       guint sr_idx;