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  24 <a href="#gst-plugins-base-libs-gstbasertpaudiopayload.synopsis" class="shortcut">Top</a>
  25                    |
  26                   <a href="#gst-plugins-base-libs-gstbasertpaudiopayload.description" class="shortcut">Description</a>
  27                    |
  28                   <a href="#gst-plugins-base-libs-gstbasertpaudiopayload.object-hierarchy" class="shortcut">Object Hierarchy</a>
  29                    |
  30                   <a href="#gst-plugins-base-libs-gstbasertpaudiopayload.properties" class="shortcut">Properties</a>
  31 </td></tr>
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  33 <div class="refentry">
  34 <a name="gst-plugins-base-libs-gstbasertpaudiopayload"></a><div class="titlepage"></div>
  35 <div class="refnamediv"><table width="100%"><tr>
  36 <td valign="top">
  37 <h2><span class="refentrytitle"><a name="gst-plugins-base-libs-gstbasertpaudiopayload.top_of_page"></a>gstbasertpaudiopayload</span></h2>
  38 <p>gstbasertpaudiopayload — Base class for audio RTP payloader</p>
  39 </td>
  40 <td valign="top" align="right"></td>
  41 </tr></table></div>
  42 <div class="refsynopsisdiv">
  43 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.synopsis"></a><h2>Synopsis</h2>
  44 <a name="GstBaseRTPAudioPayload"></a><pre class="synopsis">
  45 #include &lt;gst/rtp/gstbasertpaudiopayload.h&gt;
  46
  47 struct              <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload-struct" title="struct GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a>;
  48 struct              <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayloadClass" title="struct GstBaseRTPAudioPayloadClass">GstBaseRTPAudioPayloadClass</a>;
  49 <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>
  50                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);
  51 <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>
  52                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
  53                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_duration</code></em>,
  54                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_size</code></em>);
  55 <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>
  56                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);
  57 <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>
  58                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
  59                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);
  60 <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>
  61                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>);
  62 <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>,
  63                                                          <em class="parameter"><code>const <span class="type">guint8</span> *data</code></em>,
  64                                                          <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>,
  65                                                          <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>);
  66 <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>,
  67                                                          <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>,
  68                                                          <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>);
  69 <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>
  70                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
  71                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);
  72 </pre>
  73 </div>
  74 <div class="refsect1">
  75 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.object-hierarchy"></a><h2>Object Hierarchy</h2>
  76 <pre class="synopsis">
  77   <a href="http://library.gnome.org/devel/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
  78    +----<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstObject.html">GstObject</a>
  79          +----<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html">GstElement</a>
  80                +----<a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload">GstBaseRTPPayload</a>
  81                      +----GstBaseRTPAudioPayload
  82 </pre>
  83 </div>
  84 <div class="refsect1">
  85 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.properties"></a><h2>Properties</h2>
  86 <pre class="synopsis">
  87   "<a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload--buffer-list" title='The "buffer-list" property'>buffer-list</a>"              <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a>              : Read / Write
  88 </pre>
  89 </div>
  90 <div class="refsect1">
  91 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.description"></a><h2>Description</h2>
  92 <p>
  93 Provides a base class for audio RTP payloaders for frame or sample based
  94 audio codecs (constant bitrate)
  95 </p>
  96 <p>
  97 This class derives from GstBaseRTPPayload. It can be used for payloading
  98 audio codecs. It will only work with constant bitrate codecs. It supports
  99 both frame based and sample based codecs. It takes care of packing up the
 100 audio data into RTP packets and filling up the headers accordingly. The
 101 payloading is done based on the maximum MTU (mtu) and the maximum time per
 102 packet (max-ptime). The general idea is to divide large data buffers into
 103 smaller RTP packets. The RTP packet size is the minimum of either the MTU,
 104 max-ptime (if set) or available data. The RTP packet size is always larger or
 105 equal to min-ptime (if set). If min-ptime is not set, any residual data is
 106 sent in a last RTP packet. In the case of frame based codecs, the resulting
 107 RTP packets always contain full frames.
 108 </p>
 109 <p>
 110 </p>
 111 <div class="refsect2">
 112 <a name="idp18656560"></a><h3>Usage</h3>
 113 <p>
 114 To use this base class, your child element needs to call either
 115 <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
 116 <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
 117 element's <code class="function">_init()</code> function. Then, the child element must call either
 118 <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>,
 119 <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
 120 gst_base_rtp_audio_payload_set_samplebits_options. Since
 121 GstBaseRTPAudioPayload derives from GstBaseRTPPayload, the child element
 122 must set any variables or call/override any functions required by that base
 123 class. The child element does not need to override any other functions
 124 specific to GstBaseRTPAudioPayload.
 125 </p>
 126 </div>
 127 <p>
 128 </p>
 129 </div>
 130 <div class="refsect1">
 131 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.details"></a><h2>Details</h2>
 132 <div class="refsect2">
 133 <a name="GstBaseRTPAudioPayload-struct"></a><h3>struct GstBaseRTPAudioPayload</h3>
 134 <pre class="programlisting">struct GstBaseRTPAudioPayload;</pre>
 135 </div>
 136 <hr>
 137 <div class="refsect2">
 138 <a name="GstBaseRTPAudioPayloadClass"></a><h3>struct GstBaseRTPAudioPayloadClass</h3>
 139 <pre class="programlisting">struct GstBaseRTPAudioPayloadClass {
 140   GstBaseRTPPayloadClass parent_class;
 141 };
 142 </pre>
 143 <p>
 144 Base class for audio RTP payloader.
 145 </p>
 146 <div class="variablelist"><table border="0">
 147 <col align="left" valign="top">
 148 <tbody><tr>
 149 <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>
 150 <td>the parent class</td>
 151 </tr></tbody>
 152 </table></div>
 153 </div>
 154 <hr>
 155 <div class="refsect2">
 156 <a name="gst-base-rtp-audio-payload-set-frame-based"></a><h3>gst_base_rtp_audio_payload_set_frame_based ()</h3>
 157 <pre class="programlisting"><span class="returnvalue">void</span>                gst_base_rtp_audio_payload_set_frame_based
 158                                                         (<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>
 159 <p>
 160 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
 161 audio codec
 162 </p>
 163 <div class="variablelist"><table border="0">
 164 <col align="left" valign="top">
 165 <tbody><tr>
 166 <td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
 167 <td>a pointer to the element.</td>
 168 </tr></tbody>
 169 </table></div>
 170 </div>
 171 <hr>
 172 <div class="refsect2">
 173 <a name="gst-base-rtp-audio-payload-set-frame-options"></a><h3>gst_base_rtp_audio_payload_set_frame_options ()</h3>
 174 <pre class="programlisting"><span class="returnvalue">void</span>                gst_base_rtp_audio_payload_set_frame_options
 175                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
 176                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_duration</code></em>,
 177                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> frame_size</code></em>);</pre>
 178 <p>
 179 Sets the options for frame based audio codecs.
 180 </p>
 181 <div class="variablelist"><table border="0">
 182 <col align="left" valign="top">
 183 <tbody>
 184 <tr>
 185 <td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
 186 <td>a pointer to the element.</td>
 187 </tr>
 188 <tr>
 189 <td><p><span class="term"><em class="parameter"><code>frame_duration</code></em> :</span></p></td>
 190 <td>The duraction of an audio frame in milliseconds.</td>
 191 </tr>
 192 <tr>
 193 <td><p><span class="term"><em class="parameter"><code>frame_size</code></em> :</span></p></td>
 194 <td>The size of an audio frame in bytes.</td>
 195 </tr>
 196 </tbody>
 197 </table></div>
 198 </div>
 199 <hr>
 200 <div class="refsect2">
 201 <a name="gst-base-rtp-audio-payload-set-sample-based"></a><h3>gst_base_rtp_audio_payload_set_sample_based ()</h3>
 202 <pre class="programlisting"><span class="returnvalue">void</span>                gst_base_rtp_audio_payload_set_sample_based
 203                                                         (<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>
 204 <p>
 205 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
 206 audio codec
 207 </p>
 208 <div class="variablelist"><table border="0">
 209 <col align="left" valign="top">
 210 <tbody><tr>
 211 <td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
 212 <td>a pointer to the element.</td>
 213 </tr></tbody>
 214 </table></div>
 215 </div>
 216 <hr>
 217 <div class="refsect2">
 218 <a name="gst-base-rtp-audio-payload-set-sample-options"></a><h3>gst_base_rtp_audio_payload_set_sample_options ()</h3>
 219 <pre class="programlisting"><span class="returnvalue">void</span>                gst_base_rtp_audio_payload_set_sample_options
 220                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
 221                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);</pre>
 222 <p>
 223 Sets the options for sample based audio codecs.
 224 </p>
 225 <div class="variablelist"><table border="0">
 226 <col align="left" valign="top">
 227 <tbody>
 228 <tr>
 229 <td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
 230 <td>a pointer to the element.</td>
 231 </tr>
 232 <tr>
 233 <td><p><span class="term"><em class="parameter"><code>sample_size</code></em> :</span></p></td>
 234 <td>Size per sample in bytes.</td>
 235 </tr>
 236 </tbody>
 237 </table></div>
 238 </div>
 239 <hr>
 240 <div class="refsect2">
 241 <a name="gst-base-rtp-audio-payload-get-adapter"></a><h3>gst_base_rtp_audio_payload_get_adapter ()</h3>
 242 <pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html"><span class="returnvalue">GstAdapter</span></a> *        gst_base_rtp_audio_payload_get_adapter
 243                                                         (<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>
 244 <p>
 245 Gets the internal adapter used by the depayloader.
 246 </p>
 247 <div class="variablelist"><table border="0">
 248 <col align="left" valign="top">
 249 <tbody>
 250 <tr>
 251 <td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
 252 <td>a <a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a>
 253 </td>
 254 </tr>
 255 <tr>
 256 <td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td>
 257 <td>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html"><span class="type">GstAdapter</span></a>.</td>
 258 </tr>
 259 </tbody>
 260 </table></div>
 261 <p class="since">Since 0.10.13</p>
 262 </div>
 263 <hr>
 264 <div class="refsect2">
 265 <a name="gst-base-rtp-audio-payload-push"></a><h3>gst_base_rtp_audio_payload_push ()</h3>
 266 <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>,
 267                                                          <em class="parameter"><code>const <span class="type">guint8</span> *data</code></em>,
 268                                                          <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>,
 269                                                          <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>
 270 <p>
 271 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
 272 payload. Set the timestamp on the new buffer to <em class="parameter"><code>timestamp</code></em> before pushing
 273 the buffer downstream.
 274 </p>
 275 <div class="variablelist"><table border="0">
 276 <col align="left" valign="top">
 277 <tbody>
 278 <tr>
 279 <td><p><span class="term"><em class="parameter"><code>baseaudiopayload</code></em> :</span></p></td>
 280 <td>a <a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload"><span class="type">GstBaseRTPPayload</span></a>
 281 </td>
 282 </tr>
 283 <tr>
 284 <td><p><span class="term"><em class="parameter"><code>data</code></em> :</span></p></td>
 285 <td>data to set as payload</td>
 286 </tr>
 287 <tr>
 288 <td><p><span class="term"><em class="parameter"><code>payload_len</code></em> :</span></p></td>
 289 <td>length of payload</td>
 290 </tr>
 291 <tr>
 292 <td><p><span class="term"><em class="parameter"><code>timestamp</code></em> :</span></p></td>
 293 <td>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a>
 294 </td>
 295 </tr>
 296 <tr>
 297 <td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td>
 298 <td>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="type">GstFlowReturn</span></a>
 299 </td>
 300 </tr>
 301 </tbody>
 302 </table></div>
 303 <p class="since">Since 0.10.13</p>
 304 </div>
 305 <hr>
 306 <div class="refsect2">
 307 <a name="gst-base-rtp-audio-payload-flush"></a><h3>gst_base_rtp_audio_payload_flush ()</h3>
 308 <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_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>,
 309                                                          <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>,
 310                                                          <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>
 311 <p>
 312 Create an RTP buffer and store <em class="parameter"><code>payload_len</code></em> bytes of the adapter as the
 313 payload. Set the timestamp on the new buffer to <em class="parameter"><code>timestamp</code></em> before pushing
 314 the buffer downstream.
 315 </p>
 316 <p>
 317 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
 318 -1, the timestamp will be calculated automatically.
 319 </p>
 320 <div class="variablelist"><table border="0">
 321 <col align="left" valign="top">
 322 <tbody>
 323 <tr>
 324 <td><p><span class="term"><em class="parameter"><code>baseaudiopayload</code></em> :</span></p></td>
 325 <td>a <a class="link" href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload"><span class="type">GstBaseRTPPayload</span></a>
 326 </td>
 327 </tr>
 328 <tr>
 329 <td><p><span class="term"><em class="parameter"><code>payload_len</code></em> :</span></p></td>
 330 <td>length of payload</td>
 331 </tr>
 332 <tr>
 333 <td><p><span class="term"><em class="parameter"><code>timestamp</code></em> :</span></p></td>
 334 <td>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a>
 335 </td>
 336 </tr>
 337 <tr>
 338 <td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td>
 339 <td>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="type">GstFlowReturn</span></a>
 340 </td>
 341 </tr>
 342 </tbody>
 343 </table></div>
 344 <p class="since">Since 0.10.25</p>
 345 </div>
 346 <hr>
 347 <div class="refsect2">
 348 <a name="gst-base-rtp-audio-payload-set-samplebits-options"></a><h3>gst_base_rtp_audio_payload_set_samplebits_options ()</h3>
 349 <pre class="programlisting"><span class="returnvalue">void</span>                gst_base_rtp_audio_payload_set_samplebits_options
 350                                                         (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> *basertpaudiopayload</code></em>,
 351                                                          <em class="parameter"><code><a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> sample_size</code></em>);</pre>
 352 <p>
 353 Sets the options for sample based audio codecs.
 354 </p>
 355 <div class="variablelist"><table border="0">
 356 <col align="left" valign="top">
 357 <tbody>
 358 <tr>
 359 <td><p><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></p></td>
 360 <td>a pointer to the element.</td>
 361 </tr>
 362 <tr>
 363 <td><p><span class="term"><em class="parameter"><code>sample_size</code></em> :</span></p></td>
 364 <td>Size per sample in bits.</td>
 365 </tr>
 366 </tbody>
 367 </table></div>
 368 <p class="since">Since 0.10.18</p>
 369 </div>
 370 </div>
 371 <div class="refsect1">
 372 <a name="gst-plugins-base-libs-gstbasertpaudiopayload.property-details"></a><h2>Property Details</h2>
 373 <div class="refsect2">
 374 <a name="GstBaseRTPAudioPayload--buffer-list"></a><h3>The <code class="literal">"buffer-list"</code> property</h3>
 375 <pre class="programlisting">  "buffer-list"              <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a>              : Read / Write</pre>
 376 <p>Use Buffer Lists.</p>
 377 <p>Default value: FALSE</p>
 378 </div>
 379 </div>
 380 </div>
 381 <div class="footer">
 382 <hr>
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