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[framework/multimedia/gst-plugins-base0.10.git] / docs / libs / html / gst-plugins-base-libs-gstbasertpaudiopayload.html
diff --git a/docs/libs/html/gst-plugins-base-libs-gstbasertpaudiopayload.html b/docs/libs/html/gst-plugins-base-libs-gstbasertpaudiopayload.html
index 15b6dc2..72dc4e0 100644 (file)
--- a/docs/libs/html/gst-plugins-base-libs-gstbasertpaudiopayload.html
+++ b/docs/libs/html/gst-plugins-base-libs-gstbasertpaudiopayload.html
@@ -3,12 +3,12 @@
 <head>
 <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
 <title>gstbasertpaudiopayload</title>
-<meta name="generator" content="DocBook XSL Stylesheets V1.75.2">
+<meta name="generator" content="DocBook XSL Stylesheets V1.76.1">
 <link rel="home" href="index.html" title="GStreamer Base Plugins 0.10 Library Reference Manual">
 <link rel="up" href="gstreamer-rtp.html" title="RTP Library">
 <link rel="prev" href="gstreamer-rtp.html" title="RTP Library">
 <link rel="next" href="gst-plugins-base-libs-gstbasertpdepayload.html" title="gstbasertpdepayload">
-<meta name="generator" content="GTK-Doc V1.17 (XML mode)">
+<meta name="generator" content="GTK-Doc V1.18 (XML mode)">
 <link rel="stylesheet" href="style.css" type="text/css">
 </head>
 <body bgcolor="white" text="black" link="#0000FF" vlink="#840084" alink="#0000FF">
@@ -60,7 +60,7 @@ struct              <a class="link" href="gst-plugins-base-libs-gstbasertpaudiop
 <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html"><span class="returnvalue">GstAdapter</span></a> *        <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-get-adapter" title="gst_base_rtp_audio_payload_get_adapter ()">gst_base_rtp_audio_payload_get_adapter</a>
                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);
 <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>       <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-push" title="gst_base_rtp_audio_payload_push ()">gst_base_rtp_audio_payload_push</a>     (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *baseaudiopayload</code></em>,
-                                                         <em class="parameter"><code>const <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint8"><span class="type">guint8</span></a> *data</code></em>,
+                                                         <em class="parameter"><code>const <span class="type">guint8</span> *data</code></em>,
                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> payload_len</code></em>,
                                                          <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> timestamp</code></em>);
 <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>       <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-flush" title="gst_base_rtp_audio_payload_flush ()">gst_base_rtp_audio_payload_flush</a>    (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *baseaudiopayload</code></em>,
@@ -90,10 +90,6 @@ struct              <a class="link" href="gst-plugins-base-libs-gstbasertpaudiop
 <div class="refsect1">
 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.description"></a><h2>Description</h2>
 <p>
-</p>
-<div class="refsect2">
-<a name="idp16015920"></a><h3>Usage</h3>
-<p>
 Provides a base class for audio RTP payloaders for frame or sample based
 audio codecs (constant bitrate)
 </p>
@@ -111,6 +107,10 @@ sent in a last RTP packet. In the case of frame based codecs, the resulting
 RTP packets always contain full frames.
 </p>
 <p>
+</p>
+<div class="refsect2">
+<a name="idp18656560"></a><h3>Usage</h3>
+<p>
 To use this base class, your child element needs to call either
 <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-based" title="gst_base_rtp_audio_payload_set_frame_based ()"><code class="function">gst_base_rtp_audio_payload_set_frame_based()</code></a> or
 <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-sample-based" title="gst_base_rtp_audio_payload_set_sample_based ()"><code class="function">gst_base_rtp_audio_payload_set_sample_based()</code></a>. This is usually done in the
@@ -138,10 +138,18 @@ specific to GstBaseRTPAudioPayload.
 <a name="GstBaseRTPAudioPayloadClass"></a><h3>struct GstBaseRTPAudioPayloadClass</h3>
 <pre class="programlisting">struct GstBaseRTPAudioPayloadClass {
   GstBaseRTPPayloadClass parent_class;
-
-  gpointer _gst_reserved[GST_PADDING];
 };
 </pre>
+<p>
+Base class for audio RTP payloader.
+</p>
+<div class="variablelist"><table border="0">
+<col align="left" valign="top">
+<tbody><tr>
+<td><p><span class="term"><a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayloadClass" title="struct GstBaseRTPPayloadClass"><span class="type">GstBaseRTPPayloadClass</span></a> <em class="structfield"><code><a name="GstBaseRTPAudioPayloadClass.parent-class"></a>parent_class</code></em>;</span></p></td>
+<td>the parent class</td>
+</tr></tbody>
+</table></div>
 </div>
 <hr>
 <div class="refsect2">
@@ -256,7 +264,7 @@ Gets the internal adapter used by the depayloader.
 <div class="refsect2">
 <a name="gst-base-rtp-audio-payload-push"></a><h3>gst_base_rtp_audio_payload_push ()</h3>
 <pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>       gst_base_rtp_audio_payload_push     (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *baseaudiopayload</code></em>,
-                                                         <em class="parameter"><code>const <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint8"><span class="type">guint8</span></a> *data</code></em>,
+                                                         <em class="parameter"><code>const <span class="type">guint8</span> *data</code></em>,
                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> payload_len</code></em>,
                                                          <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> timestamp</code></em>);</pre>
 <p>
@@ -372,6 +380,6 @@ Sets the options for sample based audio codecs.
 </div>
 <div class="footer">
 <hr>
-          Generated by GTK-Doc V1.17</div>
+          Generated by GTK-Doc V1.18</div>
 </body>
 </html>
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Domain: Multimedia / Media Playback;