+ checking in plugin category changes

Original commit message from CVS:
+ checking in plugin category changes

8 files changed, 10 insertions, 10 deletions

diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c
index 2aa3207c..3a18e295 100644
--- a/gst/rtp/gstrtpL16depay.c
+++ b/gst/rtp/gstrtpL16depay.c
@@ -9,7 +9,7 @@
  * This library is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more 
+ * Library General Public License for more
  */
 
 #ifdef HAVE_CONFIG_H
@@ -22,7 +22,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtp_L16parse_details = {
   "RTP packet parser",
-  "Codec/Network",
+  "Codec/Parser/Network",
   "Extracts raw audio from RTP packets",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpL16enc.c b/gst/rtp/gstrtpL16enc.c
index 0ec42a29..610ef051 100644
--- a/gst/rtp/gstrtpL16enc.c
+++ b/gst/rtp/gstrtpL16enc.c
@@ -26,7 +26,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtpL16enc_details = {
   "RTP RAW Audio Encoder",
-  "Codec/Network",
+  "Codec/Encoder/Network",
   "Encodes Raw Audio into an RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpL16parse.c b/gst/rtp/gstrtpL16parse.c
index 2aa3207c..3a18e295 100644
--- a/gst/rtp/gstrtpL16parse.c
+++ b/gst/rtp/gstrtpL16parse.c
@@ -9,7 +9,7 @@
  * This library is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more 
+ * Library General Public License for more
  */
 
 #ifdef HAVE_CONFIG_H
@@ -22,7 +22,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtp_L16parse_details = {
   "RTP packet parser",
-  "Codec/Network",
+  "Codec/Parser/Network",
   "Extracts raw audio from RTP packets",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index 0ec42a29..610ef051 100644
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -26,7 +26,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtpL16enc_details = {
   "RTP RAW Audio Encoder",
-  "Codec/Network",
+  "Codec/Encoder/Network",
   "Encodes Raw Audio into an RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpgsmdepay.c b/gst/rtp/gstrtpgsmdepay.c
index ee009417..edc027b1 100644
--- a/gst/rtp/gstrtpgsmdepay.c
+++ b/gst/rtp/gstrtpgsmdepay.c
@@ -23,7 +23,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtp_gsmparse_details = {
   "RTP packet parser",
-  "Codec/Network",
+  "Codec/Parser/Network",
   "Extracts GSM audio from RTP packets",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpgsmenc.c b/gst/rtp/gstrtpgsmenc.c
index 47a622b7..c32e32f8 100644
--- a/gst/rtp/gstrtpgsmenc.c
+++ b/gst/rtp/gstrtpgsmenc.c
@@ -27,7 +27,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtpgsmenc_details = {
   "RTP GSM Audio Encoder",
-  "Codec/Network",
+  "Codec/Encoder/Network",
   "Encodes GSM audio into an RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpgsmparse.c b/gst/rtp/gstrtpgsmparse.c
index ee009417..edc027b1 100644
--- a/gst/rtp/gstrtpgsmparse.c
+++ b/gst/rtp/gstrtpgsmparse.c
@@ -23,7 +23,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtp_gsmparse_details = {
   "RTP packet parser",
-  "Codec/Network",
+  "Codec/Parser/Network",
   "Extracts GSM audio from RTP packets",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c
index 47a622b7..c32e32f8 100644
--- a/gst/rtp/gstrtpgsmpay.c
+++ b/gst/rtp/gstrtpgsmpay.c
@@ -27,7 +27,7 @@
 /* elementfactory information */
 static GstElementDetails gst_rtpgsmenc_details = {
   "RTP GSM Audio Encoder",
-  "Codec/Network",
+  "Codec/Encoder/Network",
   "Encodes GSM audio into an RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };