<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="Content-Type" content="text/html; charset=US-ASCII"> <title>gstbasertpaudiopayload</title> <meta name="generator" content="DocBook XSL Stylesheets V1.72.0"> <link rel="start" 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 V1.8 (XML mode)"> <link rel="stylesheet" href="style.css" type="text/css"> <link rel="part" href="gstreamer-plugins-base.html" title="Part I. GStreamer Base Plugins Libraries"> <link rel="chapter" href="gstreamer-audio.html" title="Audio Library"> <link rel="chapter" href="gstreamer-cdda.html" title="CDDA Library"> <link rel="chapter" href="gstreamer-ffft.html" title="FFT Library"> <link rel="chapter" href="gstreamer-floatcast.html" title="FloatCast Library"> <link rel="chapter" href="gstreamer-interfaces.html" title="Interfaces Library"> <link rel="chapter" href="gstreamer-netbuffer.html" title="Network Buffer Library"> <link rel="chapter" href="gstreamer-riff.html" title="Riff Media Library"> <link rel="chapter" href="gstreamer-rtp.html" title="RTP Library"> <link rel="chapter" href="gstreamer-rtsp.html" title="RTSP Library"> <link rel="chapter" href="gstreamer-sdp.html" title="SDP Library"> <link rel="chapter" href="gstreamer-tag.html" title="Tag Support Library"> <link rel="chapter" href="gstreamer-base-utils.html" title="Base Utils Library"> <link rel="chapter" href="gstreamer-video.html" title="Video Library"> <link rel="part" href="gstreamer-libs-hierarchy.html" title="Part II. Object Hierarchy"> <link rel="index" href="api-index.html" title="Index"> </head> <body bgcolor="white" text="black" link="#0000FF" vlink="#840084" alink="#0000FF"> <table class="navigation" id="top" width="100%" summary="Navigation header" cellpadding="2" cellspacing="2"> <tr valign="middle"> <td><a accesskey="p" href="gstreamer-rtp.html"><img src="left.png" width="24" height="24" border="0" alt="Prev"></a></td> <td><a accesskey="u" href="gstreamer-rtp.html"><img src="up.png" width="24" height="24" border="0" alt="Up"></a></td> <td><a accesskey="h" href="index.html"><img src="home.png" width="24" height="24" border="0" alt="Home"></a></td> <th width="100%" align="center">GStreamer Base Plugins 0.10 Library Reference Manual</th> <td><a accesskey="n" href="gst-plugins-base-libs-gstbasertpdepayload.html"><img src="right.png" width="24" height="24" border="0" alt="Next"></a></td> </tr> <tr><td colspan="5" class="shortcuts"><nobr><a href="#id2646228" class="shortcut">Top</a>  |  <a href="#id2646518" class="shortcut">Description</a>  |  <a href="#id2646470" class="shortcut">Object Hierarchy</a></nobr></td></tr> </table> <div class="refentry" lang="en"> <a name="gst-plugins-base-libs-gstbasertpaudiopayload"></a><div class="titlepage"></div> <div class="refnamediv"><table width="100%"><tr> <td valign="top"> <h2> <a name="id2646228"></a><span class="refentrytitle">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"> <h2>Synopsis</h2> <a name="GstBaseRTPAudioPayload"></a><pre class="synopsis"> #include <gst/rtp/gstbasertpaudiopayload.h> <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload-struct">GstBaseRTPAudioPayload</a>; <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayloadClass">GstBaseRTPAudioPayloadClass</a>; void <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-based">gst_base_rtp_audio_payload_set_frame_based</a> (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload); void <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-options">gst_base_rtp_audio_payload_set_frame_options</a> (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload, gint frame_duration, gint frame_size); void <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-sample-based">gst_base_rtp_audio_payload_set_sample_based</a> (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload); void <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-sample-options">gst_base_rtp_audio_payload_set_sample_options</a> (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload, gint sample_size); GstAdapter* <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-get-adapter">gst_base_rtp_audio_payload_get_adapter</a> (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload); GstFlowReturn <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-push">gst_base_rtp_audio_payload_push</a> (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *baseaudiopayload, const guint8 *data, guint payload_len, GstClockTime timestamp); </pre> </div> <div class="refsect1" lang="en"> <a name="id2646470"></a><h2>Object Hierarchy</h2> <pre class="synopsis"> GObject +----GstObject +----GstElement +----<a href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload">GstBaseRTPPayload</a> +----GstBaseRTPAudioPayload </pre> </div> <div class="refsect1" lang="en"> <a name="id2646518"></a><h2>Description</h2> <p> </p> <div class="refsect2" lang="en"> <a name="id2646530"></a><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> <h3>Usage</h3> <p> To use this base class, your child element needs to call either <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-based"><code class="function">gst_base_rtp_audio_payload_set_frame_based()</code></a> or <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#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 href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-frame-options"><code class="function">gst_base_rtp_audio_payload_set_frame_options()</code></a> or <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#gst-base-rtp-audio-payload-set-sample-options"><code class="function">gst_base_rtp_audio_payload_set_sample_options()</code></a>. 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" lang="en"> <a name="id2646633"></a><h2>Details</h2> <div class="refsect2" lang="en"> <a name="id2646643"></a><h3> <a name="GstBaseRTPAudioPayload-struct"></a>GstBaseRTPAudioPayload</h3> <a class="indexterm" name="id2646658"></a><pre class="programlisting">typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;</pre> <p> </p> </div> <hr> <div class="refsect2" lang="en"> <a name="id2646674"></a><h3> <a name="GstBaseRTPAudioPayloadClass"></a>GstBaseRTPAudioPayloadClass</h3> <a class="indexterm" name="id2646687"></a><pre class="programlisting">typedef struct { GstBaseRTPPayloadClass parent_class; gpointer _gst_reserved[GST_PADDING]; } GstBaseRTPAudioPayloadClass; </pre> <p> </p> </div> <hr> <div class="refsect2" lang="en"> <a name="id2646704"></a><h3> <a name="gst-base-rtp-audio-payload-set-frame-based"></a>gst_base_rtp_audio_payload_set_frame_based ()</h3> <a class="indexterm" name="id2646719"></a><pre class="programlisting">void gst_base_rtp_audio_payload_set_frame_based (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload);</pre> <p> Tells <a 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> <p> </p> <div class="variablelist"><table border="0"> <col align="left" valign="top"> <tbody><tr> <td><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></td> <td> a pointer to the element. </td> </tr></tbody> </table></div> </div> <hr> <div class="refsect2" lang="en"> <a name="id2646780"></a><h3> <a name="gst-base-rtp-audio-payload-set-frame-options"></a>gst_base_rtp_audio_payload_set_frame_options ()</h3> <a class="indexterm" name="id2646795"></a><pre class="programlisting">void gst_base_rtp_audio_payload_set_frame_options (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload, gint frame_duration, gint frame_size);</pre> <p> Sets the options for frame based audio codecs.</p> <p> </p> <div class="variablelist"><table border="0"> <col align="left" valign="top"> <tbody> <tr> <td><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></td> <td> a pointer to the element. </td> </tr> <tr> <td><span class="term"><em class="parameter"><code>frame_duration</code></em> :</span></td> <td> The duraction of an audio frame in milliseconds. </td> </tr> <tr> <td><span class="term"><em class="parameter"><code>frame_size</code></em> :</span></td> <td> The size of an audio frame in bytes. </td> </tr> </tbody> </table></div> </div> <hr> <div class="refsect2" lang="en"> <a name="id2646893"></a><h3> <a name="gst-base-rtp-audio-payload-set-sample-based"></a>gst_base_rtp_audio_payload_set_sample_based ()</h3> <a class="indexterm" name="id2646908"></a><pre class="programlisting">void gst_base_rtp_audio_payload_set_sample_based (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload);</pre> <p> Tells <a 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> <p> </p> <div class="variablelist"><table border="0"> <col align="left" valign="top"> <tbody><tr> <td><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></td> <td> a pointer to the element. </td> </tr></tbody> </table></div> </div> <hr> <div class="refsect2" lang="en"> <a name="id2646970"></a><h3> <a name="gst-base-rtp-audio-payload-set-sample-options"></a>gst_base_rtp_audio_payload_set_sample_options ()</h3> <a class="indexterm" name="id2646984"></a><pre class="programlisting">void gst_base_rtp_audio_payload_set_sample_options (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload, gint sample_size);</pre> <p> Sets the options for sample based audio codecs.</p> <p> </p> <div class="variablelist"><table border="0"> <col align="left" valign="top"> <tbody> <tr> <td><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></td> <td> a pointer to the element. </td> </tr> <tr> <td><span class="term"><em class="parameter"><code>sample_size</code></em> :</span></td> <td> Size per sample in bytes. </td> </tr> </tbody> </table></div> </div> <hr> <div class="refsect2" lang="en"> <a name="id2647060"></a><h3> <a name="gst-base-rtp-audio-payload-get-adapter"></a>gst_base_rtp_audio_payload_get_adapter ()</h3> <a class="indexterm" name="id2647076"></a><pre class="programlisting">GstAdapter* gst_base_rtp_audio_payload_get_adapter (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *basertpaudiopayload);</pre> <p> Gets the internal adapter used by the depayloader.</p> <p> </p> <div class="variablelist"><table border="0"> <col align="left" valign="top"> <tbody> <tr> <td><span class="term"><em class="parameter"><code>basertpaudiopayload</code></em> :</span></td> <td> a <a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload"><span class="type">GstBaseRTPAudioPayload</span></a> </td> </tr> <tr> <td><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></td> <td> a <span class="type">GstAdapter</span>. </td> </tr> </tbody> </table></div> <p class="since">Since 0.10.13 </p> </div> <hr> <div class="refsect2" lang="en"> <a name="id2647163"></a><h3> <a name="gst-base-rtp-audio-payload-push"></a>gst_base_rtp_audio_payload_push ()</h3> <a class="indexterm" name="id2647179"></a><pre class="programlisting">GstFlowReturn gst_base_rtp_audio_payload_push (<a href="gst-plugins-base-libs-gstbasertpaudiopayload.html#GstBaseRTPAudioPayload">GstBaseRTPAudioPayload</a> *baseaudiopayload, const guint8 *data, guint payload_len, GstClockTime timestamp);</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> <p> </p> <div class="variablelist"><table border="0"> <col align="left" valign="top"> <tbody> <tr> <td><span class="term"><em class="parameter"><code>baseaudiopayload</code></em> :</span></td> <td> a <a href="gst-plugins-base-libs-gstbasertppayload.html#GstBaseRTPPayload"><span class="type">GstBaseRTPPayload</span></a> </td> </tr> <tr> <td><span class="term"><em class="parameter"><code>data</code></em> :</span></td> <td> data to set as payload </td> </tr> <tr> <td><span class="term"><em class="parameter"><code>payload_len</code></em> :</span></td> <td> length of payload </td> </tr> <tr> <td><span class="term"><em class="parameter"><code>timestamp</code></em> :</span></td> <td> a <span class="type">GstClockTime</span> </td> </tr> <tr> <td><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></td> <td> a <span class="type">GstFlowReturn</span> </td> </tr> </tbody> </table></div> <p class="since">Since 0.10.13 </p> </div> </div> </div> </body> </html>