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| <a href="#gst-plugins-base-libs-gstrtpbaseaudiopayload.synopsis" class="shortcut">Top</a> |
| | |
| <a href="#gst-plugins-base-libs-gstrtpbaseaudiopayload.description" class="shortcut">Description</a> |
| | |
| <a href="#gst-plugins-base-libs-gstrtpbaseaudiopayload.object-hierarchy" class="shortcut">Object Hierarchy</a> |
| | |
| <a href="#gst-plugins-base-libs-gstrtpbaseaudiopayload.properties" class="shortcut">Properties</a> |
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| <div class="refentry"> |
| <a name="gst-plugins-base-libs-gstrtpbaseaudiopayload"></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-gstrtpbaseaudiopayload.top_of_page"></a>gstrtpbaseaudiopayload</span></h2> |
| <p>gstrtpbaseaudiopayload — 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-gstrtpbaseaudiopayload.synopsis"></a><h2>Synopsis</h2> |
| <a name="GstRTPBaseAudioPayload"></a><pre class="synopsis"> |
| #include <gst/rtp/gstrtpbaseaudiopayload.h> |
| |
| struct <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload-struct" title="struct GstRTPBaseAudioPayload">GstRTPBaseAudioPayload</a>; |
| struct <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayloadClass" title="struct GstRTPBaseAudioPayloadClass">GstRTPBaseAudioPayloadClass</a>; |
| <span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-frame-based" title="gst_rtp_base_audio_payload_set_frame_based ()">gst_rtp_base_audio_payload_set_frame_based</a> |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>); |
| <span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-frame-options" title="gst_rtp_base_audio_payload_set_frame_options ()">gst_rtp_base_audio_payload_set_frame_options</a> |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>, |
| <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>, |
| <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>); |
| <span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-sample-based" title="gst_rtp_base_audio_payload_set_sample_based ()">gst_rtp_base_audio_payload_set_sample_based</a> |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>); |
| <span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-sample-options" title="gst_rtp_base_audio_payload_set_sample_options ()">gst_rtp_base_audio_payload_set_sample_options</a> |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>, |
| <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>); |
| <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-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-get-adapter" title="gst_rtp_base_audio_payload_get_adapter ()">gst_rtp_base_audio_payload_get_adapter</a> |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</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-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-push" title="gst_rtp_base_audio_payload_push ()">gst_rtp_base_audio_payload_push</a> (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *baseaudiopayload</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-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-flush" title="gst_rtp_base_audio_payload_flush ()">gst_rtp_base_audio_payload_flush</a> (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *baseaudiopayload</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>); |
| <span class="returnvalue">void</span> <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-samplebits-options" title="gst_rtp_base_audio_payload_set_samplebits_options ()">gst_rtp_base_audio_payload_set_samplebits_options</a> |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>, |
| <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> |
| </div> |
| <div class="refsect1"> |
| <a name="gst-plugins-base-libs-gstrtpbaseaudiopayload.object-hierarchy"></a><h2>Object Hierarchy</h2> |
| <pre class="synopsis"> |
| <a href="http://library.gnome.org/devel/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a> |
| +----<a href="http://library.gnome.org/devel/gobject/unstable/gobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a> |
| +----<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstObject.html">GstObject</a> |
| +----<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html">GstElement</a> |
| +----<a class="link" href="gst-plugins-base-libs-gstrtpbasepayload.html#GstRTPBasePayload">GstRTPBasePayload</a> |
| +----GstRTPBaseAudioPayload |
| </pre> |
| </div> |
| <div class="refsect1"> |
| <a name="gst-plugins-base-libs-gstrtpbaseaudiopayload.properties"></a><h2>Properties</h2> |
| <pre class="synopsis"> |
| "<a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload--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 |
| </pre> |
| </div> |
| <div class="refsect1"> |
| <a name="gst-plugins-base-libs-gstrtpbaseaudiopayload.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 GstRTPBasePayload. 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="idp14680368"></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-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-frame-based" title="gst_rtp_base_audio_payload_set_frame_based ()"><code class="function">gst_rtp_base_audio_payload_set_frame_based()</code></a> or |
| <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-sample-based" title="gst_rtp_base_audio_payload_set_sample_based ()"><code class="function">gst_rtp_base_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-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-frame-options" title="gst_rtp_base_audio_payload_set_frame_options ()"><code class="function">gst_rtp_base_audio_payload_set_frame_options()</code></a>, |
| <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#gst-rtp-base-audio-payload-set-sample-options" title="gst_rtp_base_audio_payload_set_sample_options ()"><code class="function">gst_rtp_base_audio_payload_set_sample_options()</code></a> or |
| gst_rtp_base_audio_payload_set_samplebits_options. Since |
| GstRTPBaseAudioPayload derives from GstRTPBasePayload, 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 GstRTPBaseAudioPayload. |
| </p> |
| </div> |
| <p> |
| </p> |
| </div> |
| <div class="refsect1"> |
| <a name="gst-plugins-base-libs-gstrtpbaseaudiopayload.details"></a><h2>Details</h2> |
| <div class="refsect2"> |
| <a name="GstRTPBaseAudioPayload-struct"></a><h3>struct GstRTPBaseAudioPayload</h3> |
| <pre class="programlisting">struct GstRTPBaseAudioPayload;</pre> |
| </div> |
| <hr> |
| <div class="refsect2"> |
| <a name="GstRTPBaseAudioPayloadClass"></a><h3>struct GstRTPBaseAudioPayloadClass</h3> |
| <pre class="programlisting">struct GstRTPBaseAudioPayloadClass { |
| GstRTPBasePayloadClass 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-gstrtpbasepayload.html#GstRTPBasePayloadClass" title="struct GstRTPBasePayloadClass"><span class="type">GstRTPBasePayloadClass</span></a> <em class="structfield"><code><a name="GstRTPBaseAudioPayloadClass.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-rtp-base-audio-payload-set-frame-based"></a><h3>gst_rtp_base_audio_payload_set_frame_based ()</h3> |
| <pre class="programlisting"><span class="returnvalue">void</span> gst_rtp_base_audio_payload_set_frame_based |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>);</pre> |
| <p> |
| Tells <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</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>rtpbaseaudiopayload</code></em> :</span></p></td> |
| <td>a pointer to the element.</td> |
| </tr></tbody> |
| </table></div> |
| </div> |
| <hr> |
| <div class="refsect2"> |
| <a name="gst-rtp-base-audio-payload-set-frame-options"></a><h3>gst_rtp_base_audio_payload_set_frame_options ()</h3> |
| <pre class="programlisting"><span class="returnvalue">void</span> gst_rtp_base_audio_payload_set_frame_options |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>, |
| <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>, |
| <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> |
| <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>rtpbaseaudiopayload</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-rtp-base-audio-payload-set-sample-based"></a><h3>gst_rtp_base_audio_payload_set_sample_based ()</h3> |
| <pre class="programlisting"><span class="returnvalue">void</span> gst_rtp_base_audio_payload_set_sample_based |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>);</pre> |
| <p> |
| Tells <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</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>rtpbaseaudiopayload</code></em> :</span></p></td> |
| <td>a pointer to the element.</td> |
| </tr></tbody> |
| </table></div> |
| </div> |
| <hr> |
| <div class="refsect2"> |
| <a name="gst-rtp-base-audio-payload-set-sample-options"></a><h3>gst_rtp_base_audio_payload_set_sample_options ()</h3> |
| <pre class="programlisting"><span class="returnvalue">void</span> gst_rtp_base_audio_payload_set_sample_options |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>, |
| <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> |
| <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>rtpbaseaudiopayload</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-rtp-base-audio-payload-get-adapter"></a><h3>gst_rtp_base_audio_payload_get_adapter ()</h3> |
| <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_rtp_base_audio_payload_get_adapter |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</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>rtpbaseaudiopayload</code></em> :</span></p></td> |
| <td>a <a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> |
| </td> |
| </tr> |
| <tr> |
| <td><p><span class="term"><span class="emphasis"><em>Returns</em></span> :</span></p></td> |
| <td>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html"><span class="type">GstAdapter</span></a>. <span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span> |
| </td> |
| </tr> |
| </tbody> |
| </table></div> |
| </div> |
| <hr> |
| <div class="refsect2"> |
| <a name="gst-rtp-base-audio-payload-push"></a><h3>gst_rtp_base_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_rtp_base_audio_payload_push (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *baseaudiopayload</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> |
| 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-gstrtpbasepayload.html#GstRTPBasePayload"><span class="type">GstRTPBasePayload</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="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/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="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="type">GstFlowReturn</span></a> |
| </td> |
| </tr> |
| </tbody> |
| </table></div> |
| </div> |
| <hr> |
| <div class="refsect2"> |
| <a name="gst-rtp-base-audio-payload-flush"></a><h3>gst_rtp_base_audio_payload_flush ()</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_rtp_base_audio_payload_flush (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *baseaudiopayload</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> |
| 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-gstrtpbasepayload.html#GstRTPBasePayload"><span class="type">GstRTPBasePayload</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="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/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="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="type">GstFlowReturn</span></a> |
| </td> |
| </tr> |
| </tbody> |
| </table></div> |
| </div> |
| <hr> |
| <div class="refsect2"> |
| <a name="gst-rtp-base-audio-payload-set-samplebits-options"></a><h3>gst_rtp_base_audio_payload_set_samplebits_options ()</h3> |
| <pre class="programlisting"><span class="returnvalue">void</span> gst_rtp_base_audio_payload_set_samplebits_options |
| (<em class="parameter"><code><a class="link" href="gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload"><span class="type">GstRTPBaseAudioPayload</span></a> *rtpbaseaudiopayload</code></em>, |
| <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> |
| <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>rtpbaseaudiopayload</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> |
| </div> |
| </div> |
| <div class="refsect1"> |
| <a name="gst-plugins-base-libs-gstrtpbaseaudiopayload.property-details"></a><h2>Property Details</h2> |
| <div class="refsect2"> |
| <a name="GstRTPBaseAudioPayload--buffer-list"></a><h3>The <code class="literal">"buffer-list"</code> property</h3> |
| <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> |
| <p>Use Buffer Lists.</p> |
| <p>Default value: FALSE</p> |
| </div> |
| </div> |
| </div> |
| <div class="footer"> |
| <hr> |
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