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