From 306ab03865ff6b074699c61daee6c368a351bd04 Mon Sep 17 00:00:00 2001
From: Philippe Kalaf <philippe.kalaf@collabora.co.uk>
Date: Sat, 30 Sep 2006 00:14:20 +0000
Subject: [PATCH] gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty *
 between paragraphs

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Removed empty * between paragraphs
---
 ChangeLog                                 | 5 +++++
 gst-libs/gst/rtp/gstbasertpaudiopayload.c | 3 ---
 2 files changed, 5 insertions(+), 3 deletions(-)

diff --git a/ChangeLog b/ChangeLog
index 567057e6bc..d28425a603 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,8 @@
+2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
+
+	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+	Removed empty * between paragraphs
+
 2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
 
 	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
index 1f700a011f..975787844b 100644
--- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c
+++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
@@ -26,7 +26,6 @@
  * Provides a base class for audio RTP payloaders for frame or sample based
  * audio codecs (constant bitrate)
  * </para>
- *
  * <para>
  * This class derives from GstBaseRTPPayload. It can be used for payloading
  * audio codecs. It will only work with constant bitrate codecs. It supports
@@ -40,7 +39,6 @@
  * added in future versions if the need arises. In the case of frame
  * based codecs, the resulting RTP packets always contain full frames.
  * </para>
- *
  * <title>Usage</title>
  * <para>
  * To use this base class, your child element needs to call either
@@ -55,7 +53,6 @@
  * GstBaseRTPAudioPayload.
  * </para>
  * </refsect2>
- *
  */
 
 #ifdef HAVE_CONFIG_H