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<div class="refentry">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload"></a><div class="titlepage"></div>
<div class="refnamediv"><table width="100%"><tr>
<td valign="top">
<h2><span class="refentrytitle"><a name="gst-plugins-base-libs-gstbasertpaudiopayload.top_of_page"></a>gstbasertpaudiopayload</span></h2>
<p>gstbasertpaudiopayload — Base class for audio RTP payloader</p>
</td>
<td valign="top" align="right"></td>
</tr></table></div>
<div class="refsynopsisdiv">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload.synopsis"></a><h2>Synopsis</h2>
<a name="GstBaseRTPAudioPayload"></a><pre class="synopsis">
#include <gst/rtp/gstbasertpaudiopayload.h>
struct <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload-struct" title="struct GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a>;
struct <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayloadClass" title="struct GstBaseRTPAudioPayloadClass">GstBaseRTPAudioPayloadClass</a>;
<span class="returnvalue">void</span> <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 ()">gst_base_rtp_audio_payload_set_frame_based</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>);
<span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-options" title="gst_base_rtp_audio_payload_set_frame_options ()">gst_base_rtp_audio_payload_set_frame_options</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>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_duration</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_size</code></em>);
<span class="returnvalue">void</span> <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 ()">gst_base_rtp_audio_payload_set_sample_based</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>);
<span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-sample-options" title="gst_base_rtp_audio_payload_set_sample_options ()">gst_base_rtp_audio_payload_set_sample_options</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>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);
<a href="/usr/share/gtk-doc/html/gstreamer-libs-0.10/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="/usr/share/gtk-doc/html/gstreamer-0.10/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 <span class="type">guint8</span> *data</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a> payload_len</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> timestamp</code></em>);
<a href="/usr/share/gtk-doc/html/gstreamer-0.10/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>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a> payload_len</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> timestamp</code></em>);
<span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-samplebits-options" title="gst_base_rtp_audio_payload_set_samplebits_options ()">gst_base_rtp_audio_payload_set_samplebits_options</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>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);
</pre>
</div>
<div class="refsect1">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload.object-hierarchy"></a><h2>Object Hierarchy</h2>
<pre class="synopsis">
<a href="/usr/share/gtk-doc/html/gobject/gobject-The-Base-Object-Type.html#GObject">GObject</a>
+----<a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstObject.html">GstObject</a>
+----<a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstElement.html">GstElement</a>
+----<a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload">GstBaseRTPPayload</a>
+----GstBaseRTPAudioPayload
</pre>
</div>
<div class="refsect1">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload.properties"></a><h2>Properties</h2>
<pre class="synopsis">
"<a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload--buffer-list" title='The "buffer-list" property'>buffer-list</a>" <a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> : Read / Write
</pre>
</div>
<div class="refsect1">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload.description"></a><h2>Description</h2>
<p>
Provides a base class for audio RTP payloaders for frame or sample based
audio codecs (constant bitrate)
</p>
<p>
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. The RTP packet size is always larger or
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.
</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
element's <code class="function">_init()</code> function. Then, the child element must call either
<a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-options" title="gst_base_rtp_audio_payload_set_frame_options ()"><code class="function">gst_base_rtp_audio_payload_set_frame_options()</code></a>,
<a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-sample-options" title="gst_base_rtp_audio_payload_set_sample_options ()"><code class="function">gst_base_rtp_audio_payload_set_sample_options()</code></a> or
gst_base_rtp_audio_payload_set_samplebits_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.
</p>
</div>
<p>
</p>
</div>
<div class="refsect1">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload.details"></a><h2>Details</h2>
<div class="refsect2">
<a name="GstBaseRTPAudioPayload-struct"></a><h3>struct GstBaseRTPAudioPayload</h3>
<pre class="programlisting">struct GstBaseRTPAudioPayload;</pre>
</div>
<hr>
<div class="refsect2">
<a name="GstBaseRTPAudioPayloadClass"></a><h3>struct GstBaseRTPAudioPayloadClass</h3>
<pre class="programlisting">struct GstBaseRTPAudioPayloadClass {
GstBaseRTPPayloadClass parent_class;
};
</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">
<a name="gst-base-rtp-audio-payload-set-frame-based"></a><h3>gst_base_rtp_audio_payload_set_frame_based ()</h3>
<pre class="programlisting"><span class="returnvalue">void</span> gst_base_rtp_audio_payload_set_frame_based
(<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);</pre>
<p>
Tells <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> that the child element is for a frame based
audio codec
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody><tr>
<td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
<td>a pointer to the element.</td>
</tr></tbody>
</table></div>
</div>
<hr>
<div class="refsect2">
<a name="gst-base-rtp-audio-payload-set-frame-options"></a><h3>gst_base_rtp_audio_payload_set_frame_options ()</h3>
<pre class="programlisting"><span class="returnvalue">void</span> gst_base_rtp_audio_payload_set_frame_options
(<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_duration</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_size</code></em>);</pre>
<p>
Sets the options for frame based audio codecs.
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody>
<tr>
<td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
<td>a pointer to the element.</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>frame_duration</code></em> :</span></p></td>
<td>The duraction of an audio frame in milliseconds.</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>frame_size</code></em> :</span></p></td>
<td>The size of an audio frame in bytes.</td>
</tr>
</tbody>
</table></div>
</div>
<hr>
<div class="refsect2">
<a name="gst-base-rtp-audio-payload-set-sample-based"></a><h3>gst_base_rtp_audio_payload_set_sample_based ()</h3>
<pre class="programlisting"><span class="returnvalue">void</span> gst_base_rtp_audio_payload_set_sample_based
(<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);</pre>
<p>
Tells <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> that the child element is for a sample based
audio codec
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody><tr>
<td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
<td>a pointer to the element.</td>
</tr></tbody>
</table></div>
</div>
<hr>
<div class="refsect2">
<a name="gst-base-rtp-audio-payload-set-sample-options"></a><h3>gst_base_rtp_audio_payload_set_sample_options ()</h3>
<pre class="programlisting"><span class="returnvalue">void</span> gst_base_rtp_audio_payload_set_sample_options
(<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);</pre>
<p>
Sets the options for sample based audio codecs.
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody>
<tr>
<td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
<td>a pointer to the element.</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>sample_size</code></em> :</span></p></td>
<td>Size per sample in bytes.</td>
</tr>
</tbody>
</table></div>
</div>
<hr>
<div class="refsect2">
<a name="gst-base-rtp-audio-payload-get-adapter"></a><h3>gst_base_rtp_audio_payload_get_adapter ()</h3>
<pre class="programlisting"><a href="/usr/share/gtk-doc/html/gstreamer-libs-0.10/GstAdapter.html"><span class="returnvalue">GstAdapter</span></a> * gst_base_rtp_audio_payload_get_adapter
(<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);</pre>
<p>
Gets the internal adapter used by the depayloader.
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody>
<tr>
<td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
<td>a <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a>
</td>
</tr>
<tr>
<td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td>
<td>a <a href="/usr/share/gtk-doc/html/gstreamer-libs-0.10/GstAdapter.html"><span class="type">GstAdapter</span></a>.</td>
</tr>
</tbody>
</table></div>
<p class="since">Since 0.10.13</p>
</div>
<hr>
<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="/usr/share/gtk-doc/html/gstreamer-0.10/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 <span class="type">guint8</span> *data</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a> payload_len</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> timestamp</code></em>);</pre>
<p>
Create an RTP buffer and store <em class="parameter"><code>payload_len</code></em> bytes of <em class="parameter"><code>data</code></em> as the
payload. Set the timestamp on the new buffer to <em class="parameter"><code>timestamp</code></em> before pushing
the buffer downstream.
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody>
<tr>
<td><p><span class="term"><em class="parameter"><code>baseaudiopayload</code></em> :</span></p></td>
<td>a <a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload"><span class="type">GstBaseRTPPayload</span></a>
</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>data</code></em> :</span></p></td>
<td>data to set as payload</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>payload_len</code></em> :</span></p></td>
<td>length of payload</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>timestamp</code></em> :</span></p></td>
<td>a <a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a>
</td>
</tr>
<tr>
<td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td>
<td>a <a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstPad.html#GstFlowReturn"><span class="type">GstFlowReturn</span></a>
</td>
</tr>
</tbody>
</table></div>
<p class="since">Since 0.10.13</p>
</div>
<hr>
<div class="refsect2">
<a name="gst-base-rtp-audio-payload-flush"></a><h3>gst_base_rtp_audio_payload_flush ()</h3>
<pre class="programlisting"><a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a> gst_base_rtp_audio_payload_flush (<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><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#guint"><span class="type">guint</span></a> payload_len</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> timestamp</code></em>);</pre>
<p>
Create an RTP buffer and store <em class="parameter"><code>payload_len</code></em> bytes of the adapter as the
payload. Set the timestamp on the new buffer to <em class="parameter"><code>timestamp</code></em> before pushing
the buffer downstream.
</p>
<p>
If <em class="parameter"><code>payload_len</code></em> is -1, all pending bytes will be flushed. If <em class="parameter"><code>timestamp</code></em> is
-1, the timestamp will be calculated automatically.
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody>
<tr>
<td><p><span class="term"><em class="parameter"><code>baseaudiopayload</code></em> :</span></p></td>
<td>a <a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload"><span class="type">GstBaseRTPPayload</span></a>
</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>payload_len</code></em> :</span></p></td>
<td>length of payload</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>timestamp</code></em> :</span></p></td>
<td>a <a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a>
</td>
</tr>
<tr>
<td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td>
<td>a <a href="/usr/share/gtk-doc/html/gstreamer-0.10/GstPad.html#GstFlowReturn"><span class="type">GstFlowReturn</span></a>
</td>
</tr>
</tbody>
</table></div>
<p class="since">Since 0.10.25</p>
</div>
<hr>
<div class="refsect2">
<a name="gst-base-rtp-audio-payload-set-samplebits-options"></a><h3>gst_base_rtp_audio_payload_set_samplebits_options ()</h3>
<pre class="programlisting"><span class="returnvalue">void</span> gst_base_rtp_audio_payload_set_samplebits_options
(<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
<em class="parameter"><code><a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);</pre>
<p>
Sets the options for sample based audio codecs.
</p>
<div class="variablelist"><table border="0">
<col align="left" valign="top">
<tbody>
<tr>
<td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
<td>a pointer to the element.</td>
</tr>
<tr>
<td><p><span class="term"><em class="parameter"><code>sample_size</code></em> :</span></p></td>
<td>Size per sample in bits.</td>
</tr>
</tbody>
</table></div>
<p class="since">Since 0.10.18</p>
</div>
</div>
<div class="refsect1">
<a name="gst-plugins-base-libs-gstbasertpaudiopayload.property-details"></a><h2>Property Details</h2>
<div class="refsect2">
<a name="GstBaseRTPAudioPayload--buffer-list"></a><h3>The <code class="literal">"buffer-list"</code> property</h3>
<pre class="programlisting"> "buffer-list" <a href="/usr/share/gtk-doc/html/glib/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> : Read / Write</pre>
<p>Use Buffer Lists.</p>
<p>Default value: FALSE</p>
</div>
</div>
</div>
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