tizen 2.0 init

[framework/multimedia/gst-plugins-base0.10.git] / gst-libs / gst / rtp / gstbasertpaudiopayload.c

diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
index d8ba21b..c8d0b57 100644 (file)
--- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c
+++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
@@ -21,12 +21,9 @@
  * SECTION:gstbasertpaudiopayload
  * @short_description: Base class for audio RTP payloader
  *
- * <refsect2>
- * <para>
  * 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
  * both frame based and sample based codecs. It takes care of packing up the
@@ -38,7 +35,8 @@
  * equal to min-ptime (if set). If min-ptime is not set, any residual data is
  * sent in a last RTP packet. In the case of frame based codecs, the resulting
  * RTP packets always contain full frames.
- * </para>
+ *
+ * <refsect2>
  * <title>Usage</title>
  * <para>
  * To use this base class, your child element needs to call either
@@ -841,7 +839,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
         GstClockTime diff;
         guint64 bytes;
         /* we're only going to apply a positive gap, otherwise we let the marker
-         * bit do its thing. simply convert to bytes and add the the current
+         * bit do its thing. simply convert to bytes and add the current
          * offset */
         diff = timestamp - priv->last_timestamp;
         bytes = priv->time_to_bytes (payload, diff);