From 269205b1ad5cd37a3c04b7110314a0eca44b40bd Mon Sep 17 00:00:00 2001
From: Stefan Kost <ensonic@users.sf.net>
Date: Mon, 23 May 2011 23:12:50 +0300
Subject: [PATCH] docs: rtp library docs update

---
 gst-libs/gst/rtp/gstbasertpaudiopayload.c |  8 +++-----
 gst-libs/gst/rtp/gstbasertpaudiopayload.h |  7 +++++++
 gst-libs/gst/rtp/gstbasertpdepayload.c    |  6 +-----
 gst-libs/gst/rtp/gstbasertpdepayload.h    | 14 +++++++++++++-
 gst-libs/gst/rtp/gstbasertppayload.c      |  4 ----
 gst-libs/gst/rtp/gstbasertppayload.h      | 10 ++++++++++
 6 files changed, 34 insertions(+), 15 deletions(-)

diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c
index d8ba21bf3f..d1a43a90c7 100644
--- 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
diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.h b/gst-libs/gst/rtp/gstbasertpaudiopayload.h
index 3fdb488a61..13b93667b9 100644
--- a/gst-libs/gst/rtp/gstbasertpaudiopayload.h
+++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.h
@@ -61,10 +61,17 @@ struct _GstBaseRTPAudioPayload
   gpointer _gst_reserved[GST_PADDING];
 };
 
+/**
+ * GstBaseRTPAudioPayloadClass:
+ * @parent_class: the parent class
+ *
+ * Base class for audio RTP payloader.
+ */
 struct _GstBaseRTPAudioPayloadClass
 {
   GstBaseRTPPayloadClass parent_class;
 
+  /*< private >*/
   gpointer _gst_reserved[GST_PADDING];
 };
 
diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.c b/gst-libs/gst/rtp/gstbasertpdepayload.c
index 978a26252e..9c334135b8 100644
--- a/gst-libs/gst/rtp/gstbasertpdepayload.c
+++ b/gst-libs/gst/rtp/gstbasertpdepayload.c
@@ -1,5 +1,5 @@
 /* GStreamer
- * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org> 
+ * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
  * Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.com>
  *
  * This library is free software; you can redistribute it and/or
@@ -22,11 +22,7 @@
  * SECTION:gstbasertpdepayload
  * @short_description: Base class for RTP depayloader
  *
- * <refsect2>
- * <para>
  * Provides a base class for RTP depayloaders
- * </para>
- * </refsect2>
  */
 
 #include "gstbasertpdepayload.h"
diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.h b/gst-libs/gst/rtp/gstbasertpdepayload.h
index 000c1167ff..d0326a48b0 100644
--- a/gst-libs/gst/rtp/gstbasertpdepayload.h
+++ b/gst-libs/gst/rtp/gstbasertpdepayload.h
@@ -1,5 +1,5 @@
 /* GStreamer
- * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org> 
+ * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -92,6 +92,18 @@ struct _GstBaseRTPDepayload
   gpointer _gst_reserved[GST_PADDING-1];
 };
 
+/**
+ * GstBaseRTPDepayloadClass:
+ * @parent_class: the parent class
+ * @set_caps: configure the depayloader
+ * @add_to_queue: (deprecated)
+ * @process: process incoming rtp packets
+ * @set_gst_timestamp: convert from RTP timestamp to GST timestamp
+ * @packet_lost: signal the depayloader about packet loss
+ * @handle_event: custom event handling
+ *
+ * Base class for audio RTP payloader.
+ */
 struct _GstBaseRTPDepayloadClass
 {
   GstElementClass parent_class;
diff --git a/gst-libs/gst/rtp/gstbasertppayload.c b/gst-libs/gst/rtp/gstbasertppayload.c
index 060e4e0188..e27d97aa56 100644
--- a/gst-libs/gst/rtp/gstbasertppayload.c
+++ b/gst-libs/gst/rtp/gstbasertppayload.c
@@ -16,11 +16,7 @@
  * SECTION:gstbasertppayload
  * @short_description: Base class for RTP payloader
  *
- * <refsect2>
- * <para>
  * Provides a base class for RTP payloaders
- * </para>
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst-libs/gst/rtp/gstbasertppayload.h b/gst-libs/gst/rtp/gstbasertppayload.h
index 9707be06e1..86a5721855 100644
--- a/gst-libs/gst/rtp/gstbasertppayload.h
+++ b/gst-libs/gst/rtp/gstbasertppayload.h
@@ -120,6 +120,16 @@ struct _GstBaseRTPPayload
   } abidata;
 };
 
+/**
+ * GstBaseRTPPayloadClass:
+ * @parent_class: the parent class
+ * @set_caps: configure the payloader
+ * @handle_buffer: process data
+ * @handle_event: custom event handling
+ * @get_caps: get desired caps
+ *
+ * Base class for audio RTP payloader.
+ */
 struct _GstBaseRTPPayloadClass
 {
   GstElementClass parent_class;