Commit 306ab038 authored by Philippe Kalaf's avatar Philippe Kalaf
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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
parent 5ba46c08
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:
......@@ -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
......
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