diff --git a/ChangeLog b/ChangeLog
index 9f81bf7430..a6249870eb 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,11 +1,3 @@
-2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
-
-	* gst-libs/gst/rtp/gstrtpbuffer.h:
-	Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
-	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
-	* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
-	New RTP audio base payloader class. Supports frame or sample based codecs
-
 2006-04-12  Thomas Vander Stichele  <thomas at apestaart dot org>
 
 	* configure.ac:
diff --git a/gst-libs/gst/rtp/Makefile.am b/gst-libs/gst/rtp/Makefile.am
index d1a7f21e80..ea02606c58 100644
--- a/gst-libs/gst/rtp/Makefile.am
+++ b/gst-libs/gst/rtp/Makefile.am
@@ -2,14 +2,12 @@ libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
 
 libgstrtpinclude_HEADERS = gstrtpbuffer.h \
 			gstbasertppayload.h \
-			gstbasertpaudiopayload.h \
 			gstbasertpdepayload.h
 
 lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
 
 libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
   				gstbasertppayload.c \
-  				gstbasertpaudiopayload.c \
   				gstbasertpdepayload.c
 
 libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_CFLAGS)
diff --git a/gst-libs/gst/rtp/gstrtpbuffer.h b/gst-libs/gst/rtp/gstrtpbuffer.h
index 32ed24a670..9fcfbcd626 100644
--- a/gst-libs/gst/rtp/gstrtpbuffer.h
+++ b/gst-libs/gst/rtp/gstrtpbuffer.h
@@ -71,8 +71,6 @@ typedef enum
 #define GST_RTP_PAYLOAD_MPV_STRING "32"
 #define GST_RTP_PAYLOAD_H263_STRING "34"
 
-#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]"
-
 /* creating buffers */
 GstBuffer*      gst_rtp_buffer_new              (void);
 void            gst_rtp_buffer_allocate_data    (GstBuffer *buffer, guint payload_len,