reverting rtp patches to fix freeze break on -base as explained on the list
authorThomas Vander Stichele <thomas@apestaart.org>
Thu, 13 Apr 2006 09:26:27 +0000 (09:26 +0000)
committerThomas Vander Stichele <thomas@apestaart.org>
Thu, 13 Apr 2006 09:26:27 +0000 (09:26 +0000)
Original commit message from CVS:
reverting rtp patches to fix freeze break on -base as explained on the list

ChangeLog
gst-libs/gst/rtp/Makefile.am
gst-libs/gst/rtp/gstrtpbuffer.h

index 9f81bf7..a624987 100644 (file)
--- 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:
index d1a7f21..ea02606 100644 (file)
@@ -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)
index 32ed24a..9fcfbcd 100644 (file)
@@ -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,