gst-libs/gst/rtp/: Moved some documentation into .c file

Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/README:
Moved some documentation into .c file

diff --git a/ChangeLog b/ChangeLog
index b3ab21a..567057e 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,9 @@
+2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>      
+
+       * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
+       * gst-libs/gst/rtp/README:
+       Moved some documentation into .c file
+
 2006-09-29  Wim Taymans  <wim@fluendo.com>
 
        * gst/playback/gstdecodebin.c: (no_more_pads):

diff --git a/gst-libs/gst/rtp/README b/gst-libs/gst/rtp/README
index 4507325..77253b7 100644 (file)
--- a/gst-libs/gst/rtp/README
+++ b/gst-libs/gst/rtp/README
@@ -39,29 +39,6 @@ The RTP libraries
   RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
   -------------------------------------------------------
 
-  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 audio data into RTP packets and filling up the headers
-  accordingly. The payloading is done based on the maximum MTU (mtu) and the
-  maximum time per packet (max-ptime). The general idea is to divide large data
-  buffers into smaller RTP packets. The RTP packet size is the minimum of either
-  the MTU, max-ptime (if set) or available data. Any residual data is always
-  sent in a last RTP packet (no minimum RTP packet size). The idea is that since
-  this is a real time protocol, data should never be delayed. In the case of
-  frame based codecs, the resulting RTP packets always contain full frames.
-
-  To use this base class, your child element needs to call either
-  gst_basertpaudiopayload_set_frame_based() or
-  gst_basertpaudiopayload_set_sample_based(). This is usually done in the
-  element's _init() function. Then, the child element must call either
-  gst_basertpaudiopayload_set_frame_options() or
-  gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
-  derives from GstBaseRTPPayload, the child element must set any variables or
-  call/override any functions required by that base class. The child element
-  does not need to override any other functions specific to
-  GstBaseRTPAudioPayload.
-
   This base class can be tested through it's children classes. Here is an
   example using the iLBC payloader (frame based).
 

diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
index 7727c48..1f700a0 100644 (file)
--- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c
+++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
@@ -17,6 +17,47 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/**
+ * 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
+ * audio data into RTP packets and filling up the headers accordingly. The
+ * payloading is done based on the maximum MTU (mtu) and the maximum time per
+ * packet (max-ptime). The general idea is to divide large data buffers into
+ * smaller RTP packets. The RTP packet size is the minimum of either the MTU,
+ * max-ptime (if set) or available data. Any residual data is always sent in a
+ * last RTP packet (no minimum RTP packet size). A minimum packet size might be
+ * 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
+ * gst_basertpaudiopayload_set_frame_based() or
+ * gst_basertpaudiopayload_set_sample_based(). This is usually done in the
+ * element's _init() function. Then, the child element must call either
+ * gst_basertpaudiopayload_set_frame_options() or
+ * gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
+ * derives from GstBaseRTPPayload, the child element must set any variables or
+ * call/override any functions required by that base class. The child element
+ * does not need to override any other functions specific to
+ * GstBaseRTPAudioPayload.
+ * </para>
+ * </refsect2>
+ *
+ */
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif