* 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
* 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
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];
};
/* 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
* SECTION:gstbasertpdepayload
* @short_description: Base class for RTP depayloader
*
- * <refsect2>
- * <para>
* Provides a base class for RTP depayloaders
- * </para>
- * </refsect2>
*/
#include "gstbasertpdepayload.h"
/* 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
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;
* SECTION:gstbasertppayload
* @short_description: Base class for RTP payloader
*
- * <refsect2>
- * <para>
* Provides a base class for RTP payloaders
- * </para>
- * </refsect2>
*/
#ifdef HAVE_CONFIG_H
} 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;