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gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs

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Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Removed empty * between paragraphs
This commit is contained in:
Philippe Kalaf 2006-09-30 00:14:20 +00:00
parent 5ba46c0866
commit 306ab03865
2 changed files with 5 additions and 3 deletions
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ChangeLog
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@ -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:

3
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -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