RubyGems - gstreamer - Versions diffs - 3.1.6-x64-mingw32 → 3.1.7-x64-mingw32 - Mend

gstreamer 3.1.6-x64-mingw32 → 3.1.7-x64-mingw32

This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.

Files changed (1658) hide show

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtppcmudepay.html CHANGED Viewed

@@ -51,7 +51,7 @@
     <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gobjectgobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a>
         <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstObject.html#GstObject-struct">GstObject</a>
             <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstElement.html#GstElement-struct">GstElement</a>
-                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0gst-plugins-base-libs-gstrtpbasedepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
+                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0GstRTPBaseDepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
                     <span class="lineart">╰──</span> GstRtpPcmuDepay
 </pre>
 </div>
@@ -75,7 +75,7 @@
 </tr>
 <tr>
 <td><p><span class="term">author</span></p></td>
-<td>Edgard Lima &lt;edgard.lima@indt.org.br&gt;, Zeeshan Ali &lt;zeenix@gmail.com&gt;</td>
+<td>Edgard Lima &lt;edgard.lima@gmail.com&gt;, Zeeshan Ali &lt;zeenix@gmail.com&gt;</td>
 </tr>
 <tr>
 <td><p><span class="term">class</span></p></td>

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtppcmupay.html CHANGED Viewed

@@ -51,8 +51,8 @@
     <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gobjectgobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a>
         <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstObject.html#GstObject-struct">GstObject</a>
             <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstElement.html#GstElement-struct">GstElement</a>
-                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0gst-plugins-base-libs-gstrtpbasepayload.html#GstRTPBasePayload-struct">GstRTPBasePayload</a>
-                    <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0gst-plugins-base-libs-gstrtpbaseaudiopayload.html#GstRTPBaseAudioPayload-struct">GstRTPBaseAudioPayload</a>
+                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0GstRTPBasePayload.html#GstRTPBasePayload-struct">GstRTPBasePayload</a>
+                    <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0GstRTPBaseAudioPayload.html#GstRTPBaseAudioPayload-struct">GstRTPBaseAudioPayload</a>
                         <span class="lineart">╰──</span> GstRtpPcmuPay
 </pre>
 </div>
@@ -76,7 +76,7 @@
 </tr>
 <tr>
 <td><p><span class="term">author</span></p></td>
-<td>Edgard Lima &lt;edgard.lima@indt.org.br&gt;</td>
+<td>Edgard Lima &lt;edgard.lima@gmail.com&gt;</td>
 </tr>
 <tr>
 <td><p><span class="term">class</span></p></td>

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpqcelpdepay.html CHANGED Viewed

@@ -51,7 +51,7 @@
     <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gobjectgobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a>
         <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstObject.html#GstObject-struct">GstObject</a>
             <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstElement.html#GstElement-struct">GstElement</a>
-                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0gst-plugins-base-libs-gstrtpbasedepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
+                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0GstRTPBaseDepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
                     <span class="lineart">╰──</span> GstRtpQCELPDepay
 </pre>
 </div>

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpqdm2depay.html CHANGED Viewed

@@ -51,7 +51,7 @@
     <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gobjectgobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a>
         <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstObject.html#GstObject-struct">GstObject</a>
             <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstElement.html#GstElement-struct">GstElement</a>
-                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0gst-plugins-base-libs-gstrtpbasedepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
+                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0GstRTPBaseDepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
                     <span class="lineart">╰──</span> GstRtpQDM2Depay
 </pre>
 </div>

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtprtxqueue.html CHANGED Viewed

@@ -52,6 +52,16 @@
 <td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxqueue.html#GstRTPRtxQueue--max-size-time" title="The “max-size-time” property">max-size-time</a></td>
 <td class="property_flags">Read / Write</td>
 </tr>
+<tr>
+<td class="property_type"><a href="/usr/share/gtk-doc/html/glibglib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
+<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxqueue.html#GstRTPRtxQueue--fulfilled-requests" title="The “fulfilled-requests” property">fulfilled-requests</a></td>
+<td class="property_flags">Read</td>
+</tr>
+<tr>
+<td class="property_type"><a href="/usr/share/gtk-doc/html/glibglib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
+<td class="property_name"><a class="link" href="gst-plugins-good-plugins-rtprtxqueue.html#GstRTPRtxQueue--requests" title="The “requests” property">requests</a></td>
+<td class="property_flags">Read</td>
+</tr>
 </tbody>
 </table></div>
 </div>
@@ -79,10 +89,75 @@
 </div>
 <div class="refsect1">
 <a name="gst-plugins-good-plugins-rtprtxqueue.description"></a><h2>Description</h2>
+<p>rtprtxqueue maintains a queue of transmitted RTP packets, up to a
+configurable limit (see <span class="type">“max-size-time”</span>,
+<span class="type">“max-size-packets”</span>), and retransmits them upon request
+from the downstream rtpsession (GstRTPRetransmissionRequest event).</p>
+<p>This element is similar to rtprtxsend, but it has differences:</p>
+<div class="itemizedlist"><ul class="itemizedlist" style="list-style-type: disc; ">
+<li class="listitem"><p>Retransmission from rtprtxqueue is not RFC 4588 compliant. The
+retransmitted packets have the same ssrc and payload type as the original
+stream.</p></li>
+<li class="listitem"><p>As a side-effect of the above, rtprtxqueue does not require the use of
+rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to
+reconstruct the stream.</p></li>
+<li class="listitem"><p>Retransmission from rtprtxqueue happens as soon as the next regular flow
+packet is chained, while rtprtxsend retransmits as soon as the retransmission
+event is received, using a helper thread.</p></li>
+<li class="listitem"><p>rtprtxqueue can be used with rtpbin without the need of hooking to its
+<a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-request-aux-sender" title="The “request-aux-sender” signal"><span class="type">“request-aux-sender”</span></a> signal, which means it can be used with
+rtpbin using gst-launch.</p></li>
+</ul></div>
+<p>See also <a class="link" href="gst-plugins-good-plugins-rtprtxsend.html#GstRtpRtxSend"><span class="type">GstRtpRtxSend</span></a>, <a class="link" href="gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive"><span class="type">GstRtpRtxReceive</span></a></p>
+<div class="refsect2">
+<a name="id-1.2.200.7.6"></a><h3>Example pipelines</h3>
+<div class="informalexample">
+  <table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
+    <tbody>
+      <tr>
+        <td class="listing_lines" align="right"><pre>1
+2
+3
+4
+5</pre></td>
+        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpbin name<span class="gtkdoc opt">=</span>b rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
+   audiotestsrc is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">96</span> <span class="gtkdoc opt">!</span> rtprtxqueue <span class="gtkdoc opt">!</span> b<span class="gtkdoc opt">.</span>send_rtp_sink_0 \
+   b<span class="gtkdoc opt">.</span>send_rtp_src_0 <span class="gtkdoc opt">!</span> identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.01</span> <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
+   udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> b<span class="gtkdoc opt">.</span>recv_rtcp_sink_0 \
+   b<span class="gtkdoc opt">.</span>send_rtcp_src_0 <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
+      </tr>
+    </tbody>
+  </table>
+</div>
+<p> Sender pipeline</p>
+<div class="informalexample">
+  <table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
+    <tbody>
+      <tr>
+        <td class="listing_lines" align="right"><pre>1
+2
+3
+4
+5
+6</pre></td>
+        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpbin name<span class="gtkdoc opt">=</span>b rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> \
+   udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96&quot;</span> <span class="gtkdoc opt">!</span> \
+       b<span class="gtkdoc opt">.</span>recv_rtp_sink_0 \
+   b<span class="gtkdoc opt">. !</span> rtpopusdepay <span class="gtkdoc opt">!</span> opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioresample <span class="gtkdoc opt">!</span> autoaudiosink \
+   udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> b<span class="gtkdoc opt">.</span>recv_rtcp_sink_0 \
+   b<span class="gtkdoc opt">.</span>send_rtcp_src_0 <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
+      </tr>
+    </tbody>
+  </table>
+</div>
+<p> Receiver pipeline</p>
+</div>
 <div class="refsynopsisdiv">
 <h2>Synopsis</h2>
 <div class="refsect2">
-<a name="id-1.2.200.7.2.1"></a><h3>Element Information</h3>
+<a name="id-1.2.200.7.7.1"></a><h3>Element Information</h3>
 <div class="variablelist"><table border="0" class="variablelist">
 <colgroup>
 <col align="left" valign="top">
@@ -108,7 +183,7 @@
 </div>
 <hr>
 <div class="refsect2">
-<a name="id-1.2.200.7.2.2"></a><h3>Element Pads</h3>
+<a name="id-1.2.200.7.7.2"></a><h3>Element Pads</h3>
 <div class="variablelist"><table border="0" class="variablelist">
 <colgroup>
 <col align="left" valign="top">
@@ -189,6 +264,22 @@
 <p>Flags: Read / Write</p>
 <p>Default value: 0</p>
 </div>
+<hr>
+<div class="refsect2">
+<a name="GstRTPRtxQueue--fulfilled-requests"></a><h3>The <code class="literal">“fulfilled-requests”</code> property</h3>
+<pre class="programlisting">  “fulfilled-requests”       <a href="/usr/share/gtk-doc/html/glibglib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
+<p>Number of fulfilled retransmission requests.</p>
+<p>Flags: Read</p>
+<p>Default value: 0</p>
+</div>
+<hr>
+<div class="refsect2">
+<a name="GstRTPRtxQueue--requests"></a><h3>The <code class="literal">“requests”</code> property</h3>
+<pre class="programlisting">  “requests”                 <a href="/usr/share/gtk-doc/html/glibglib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
+<p>Total number of retransmission requests.</p>
+<p>Flags: Read</p>
+<p>Default value: 0</p>
+</div>
 </div>
 </div>
 <div class="footer">

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtprtxreceive.html CHANGED Viewed

@@ -90,21 +90,24 @@
 </div>
 <div class="refsect1">
 <a name="gst-plugins-good-plugins-rtprtxreceive.description"></a><h2>Description</h2>
-<p>The receiver will listen to the custom retransmission events from the
-downstream jitterbuffer and will remember the SSRC1 of the stream and
-seqnum that was requested. When it sees a packet with one of the stored
-seqnum, it associates the SSRC2 of the stream with the SSRC1 of the
-master stream. From then it knows that SSRC2 is the retransmission
-stream of SSRC1. This algorithm is stated in RFC 4588. For this
-algorithm to work, RFC4588 also states that no two pending retransmission
-requests can exist for the same seqnum and different SSRCs or else it
-would be impossible to associate the retransmission with the original
-requester SSRC.
-When the RTX receiver has associated the retransmission packets,
-it can depayload and forward them to the source pad of the element.
-RTX is SSRC-multiplexed. See <a class="link" href="gst-plugins-good-plugins-rtprtxsend.html#GstRtpRtxSend"><span class="type">GstRtpRtxSend</span></a></p>
+<p>rtprtxreceive listens to the retransmission events from the
+downstream rtpjitterbuffer and remembers the SSRC (ssrc1) of the stream and
+the sequence number that was requested. When it receives a packet with
+a sequence number equal to one of the ones stored and with a different SSRC,
+it identifies the new SSRC (ssrc2) as the retransmission stream of ssrc1.
+From this point on, it replaces ssrc2 with ssrc1 in all packets of the
+ssrc2 stream and flags them as retransmissions, so that rtpjitterbuffer
+can reconstruct the original stream.</p>
+<p>This algorithm is implemented as specified in RFC 4588.</p>
+<p>This element is meant to be used with rtprtxsend on the sender side.
+See <a class="link" href="gst-plugins-good-plugins-rtprtxsend.html#GstRtpRtxSend"><span class="type">GstRtpRtxSend</span></a></p>
+<p>Below you can see some examples that illustrate how rtprtxreceive and
+rtprtxsend fit among the other rtp elements and how they work internally.
+Normally, hoewever, you should avoid using such pipelines and use
+rtpbin instead, with its <a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-request-aux-sender" title="The “request-aux-sender” signal"><span class="type">“request-aux-sender”</span></a> and
+<a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin-request-aux-receiver" title="The “request-aux-receiver” signal"><span class="type">“request-aux-receiver”</span></a> signals. See <a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin"><span class="type">GstRtpBin</span></a>.</p>
 <div class="refsect2">
-<a name="id-1.2.201.7.3"></a><h3>Example pipelines</h3>
+<a name="id-1.2.201.7.6"></a><h3>Example pipelines</h3>
 <div class="informalexample">
   <table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
     <tbody>
@@ -114,18 +117,26 @@ RTX is SSRC-multiplexed. See <a class="link" href="gst-plugins-good-plugins-rtpr
 3
 4
 5
-6</pre></td>
-        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession \
-        audiotestsrc <span class="gtkdoc opt">!</span> speexenc <span class="gtkdoc opt">!</span> rtpspeexpay pt<span class="gtkdoc opt">=</span><span class="number">97</span> <span class="gtkdoc opt">!</span> rtprtxsend rtx<span class="gtkdoc opt">-</span>payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">=</span><span class="number">99</span> <span class="gtkdoc opt">!</span> \
-            identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.1</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>send_rtp_sink \
-            rtpsession<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
-        rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
+6
+7
+8
+9</pre></td>
+        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
+    audiotestsrc is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">96</span> <span class="gtkdoc opt">!</span> \
+        rtprtxsend payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp-pt-map,96=(uint)97&quot;</span> <span class="gtkdoc opt">!</span> \
+        rtpsession<span class="gtkdoc opt">.</span>send_rtp_sink \
+    rtpsession<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.01</span> <span class="gtkdoc opt">!</span> \
+        udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
+    udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
+    rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> \
+        sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
       </tr>
     </tbody>
   </table>
 </div>
- Send audio stream through port 5000. (5001 and 5002 are just the rtcp link with the receiver)
+<p> Send audio stream through port 5000 (5001 and 5002 are just the rtcp
+link with the receiver)</p>
 <div class="informalexample">
   <table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
     <tbody>
@@ -136,36 +147,41 @@ RTX is SSRC-multiplexed. See <a class="link" href="gst-plugins-good-plugins-rtpr
 4
 5
 6
-7</pre></td>
-        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession \
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1&quot;</span> <span class="gtkdoc opt">!</span> \
-            rtpsession<span class="gtkdoc opt">.</span>recv_rtp_sink \
-            rtpsession<span class="gtkdoc opt">.</span>recv_rtp_src <span class="gtkdoc opt">!</span> rtprtxreceive rtx<span class="gtkdoc opt">-</span>payload<span class="gtkdoc opt">-</span>types<span class="gtkdoc opt">=</span><span class="string">&quot;99&quot;</span> <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpspeexdepay <span class="gtkdoc opt">!</span> \
-            speexdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
-        rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> \
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink sync<span class="gtkdoc opt">=</span>fakse async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
+7
+8
+9
+10</pre></td>
+        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
+    udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96&quot;</span> <span class="gtkdoc opt">!</span> \
+        rtpsession<span class="gtkdoc opt">.</span>recv_rtp_sink \
+    rtpsession<span class="gtkdoc opt">.</span>recv_rtp_src <span class="gtkdoc opt">!</span> \
+        rtprtxreceive payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp-pt-map,96=(uint)97&quot;</span> <span class="gtkdoc opt">!</span> \
+        rtpssrcdemux <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> \
+        rtpopusdepay <span class="gtkdoc opt">!</span> opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> audioresample <span class="gtkdoc opt">!</span> autoaudiosink \
+    rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> \
+        udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span> \
+    udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink</pre></td>
       </tr>
     </tbody>
   </table>
 </div>
- Receive audio stream from port 5000. (5001 and 5002 are just the rtcp link with the sender)
-On sender side make sure to use a different payload type for the stream and
-its associated retransmission stream (see <a class="link" href="gst-plugins-good-plugins-rtprtxsend.html#GstRtpRtxSend"><span class="type">GstRtpRtxSend</span></a>). Note that several retransmission streams can
-have the same payload type so this is not deterministic. Actually the
-rtprtxreceiver element does the association using seqnum values.
-On receiver side set all the retransmission payload types (Those informations are retrieve
-through SDP).
-You should still hear a clear sound when setting drop-probability to something greater than 0.
-The rtpjitterbuffer will generate a custom upstream event GstRTPRetransmissionRequest when
-it assumes that one packet is missing. Then this request is translated to a FB NACK in the rtcp link
-Finally the rtpsession of the sender side re-convert it in a GstRTPRetransmissionRequest that will
-be handle by rtprtxsend.
-When increasing this value it may be possible that even the retransmission stream would be dropped
-so the receiver will ask to resend the packets again and again until it actually receive them.
-If the value is too high the rtprtxsend will not be able to retrieve the packet in its list of
-stored packets. For learning purpose you could try to increase the max-size-packets or max-size-time
-rtprtxsender's properties.
-Also note that you should use rtprtxsend through rtpbin and its set-aux-send property. See <a class="link" href="gst-plugins-good-plugins-rtpbin.html#GstRtpBin"><span class="type">GstRtpBin</span></a>.
+<p> Receive audio stream from port 5000 (5001 and 5002 are just the rtcp
+link with the sender)</p>
+<p>In this example we can see a simple streaming of an OPUS stream with some
+of the packets being artificially dropped by the identity element.
+Thanks to retransmission, you should still hear a clear sound when setting
+drop-probability to something greater than 0.</p>
+<p>Internally, the rtpjitterbuffer will generate a custom upstream event,
+GstRTPRetransmissionRequest, when it detects that one packet is missing.
+Then this request is translated to a FB NACK in the rtcp link by rtpsession.
+Finally the rtpsession of the sender side will re-convert it in a
+GstRTPRetransmissionRequest that will be handled by rtprtxsend. rtprtxsend
+will then re-send the missing packet with a new srrc and a different payload
+type (here, 97), but with the same original sequence number. On the receiver
+side, rtprtxreceive will associate this new stream with the original and
+forward the retransmission packets to rtpjitterbuffer with the original
+ssrc and payload type.</p>
 <div class="informalexample">
   <table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
     <tbody>
@@ -181,24 +197,27 @@ Also note that you should use rtprtxsend through rtpbin and its set-aux-send pro
 9
 10
 11
-12</pre></td>
-        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession0 \
-        audiotestsrc wave<span class="gtkdoc opt">=</span><span class="number">0</span> <span class="gtkdoc opt">!</span> speexenc <span class="gtkdoc opt">!</span> rtpspeexpay pt<span class="gtkdoc opt">=</span><span class="number">97</span> <span class="gtkdoc opt">!</span> rtprtxsend rtx<span class="gtkdoc opt">-</span>payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">=</span><span class="number">99</span> seqnum<span class="gtkdoc opt">-</span>offset<span class="gtkdoc opt">=</span><span class="number">1</span> <span class="gtkdoc opt">!</span> \
-            identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.1</span> <span class="gtkdoc opt">!</span> rtpsession0<span class="gtkdoc opt">.</span>send_rtp_sink \
-            rtpsession0<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> rtpsession0<span class="gtkdoc opt">.</span>recv_rtcp_sink \
-        rtpsession0<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span> \
-               rtpsession name<span class="gtkdoc opt">=</span>rtpsession1 \
-        audiotestsrc wave<span class="gtkdoc opt">=</span><span class="number">0</span> <span class="gtkdoc opt">!</span> speexenc <span class="gtkdoc opt">!</span> rtpspeexpay pt<span class="gtkdoc opt">=</span><span class="number">97</span> <span class="gtkdoc opt">!</span> rtprtxsend rtx<span class="gtkdoc opt">-</span>payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">=</span><span class="number">99</span> seqnum<span class="gtkdoc opt">-</span>offset<span class="gtkdoc opt">=</span><span class="number">10</span> <span class="gtkdoc opt">!</span> \
-            identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.1</span> <span class="gtkdoc opt">!</span> rtpsession1<span class="gtkdoc opt">.</span>send_rtp_sink \
-            rtpsession1<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5004</span> <span class="gtkdoc opt">!</span> rtpsession1<span class="gtkdoc opt">.</span>recv_rtcp_sink \
-        rtpsession1<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
+12
+13</pre></td>
+        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
+    audiotestsrc is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">97</span> seqnum<span class="gtkdoc opt">-</span>offset<span class="gtkdoc opt">=</span><span class="number">1</span> <span class="gtkdoc opt">!</span> \
+        rtprtxsend payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp-pt-map,97=(uint)99&quot;</span> <span class="gtkdoc opt">!</span> \
+        funnel name<span class="gtkdoc opt">=</span>f <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>send_rtp_sink \
+    audiotestsrc freq<span class="gtkdoc opt">=</span><span class="number">660.0</span> is<span class="gtkdoc opt">-</span>live<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> opusenc <span class="gtkdoc opt">!</span> \
+        rtpopuspay pt<span class="gtkdoc opt">=</span><span class="number">97</span> seqnum<span class="gtkdoc opt">-</span>offset<span class="gtkdoc opt">=</span><span class="number">100</span> <span class="gtkdoc opt">!</span> \
+        rtprtxsend payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp-pt-map,97=(uint)99&quot;</span> <span class="gtkdoc opt">!</span> \
+        f<span class="gtkdoc opt">.</span> \
+    rtpsession<span class="gtkdoc opt">.</span>send_rtp_src <span class="gtkdoc opt">!</span> identity drop<span class="gtkdoc opt">-</span>probability<span class="gtkdoc opt">=</span><span class="number">0.01</span> <span class="gtkdoc opt">!</span> \
+        udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5000</span> \
+    udpsrc port<span class="gtkdoc opt">=</span><span class="number">5001</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
+    rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5002</span> \
+        sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
       </tr>
     </tbody>
   </table>
 </div>
- Send two audio streams to port 5000.
+<p> Send two audio streams to port 5000.</p>
 <div class="informalexample">
   <table class="listing_frame" border="0" cellpadding="0" cellspacing="0">
     <tbody>
@@ -210,34 +229,48 @@ Also note that you should use rtprtxsend through rtpbin and its set-aux-send pro
 5
 6
 7
-8</pre></td>
-        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1&quot;</span> <span class="gtkdoc opt">!</span> \
-            rtpsession<span class="gtkdoc opt">.</span>recv_rtp_sink \
-            rtpsession<span class="gtkdoc opt">.</span>recv_rtp_src <span class="gtkdoc opt">!</span> rtprtxreceive rtx<span class="gtkdoc opt">-</span>payload<span class="gtkdoc opt">-</span>types<span class="gtkdoc opt">=</span><span class="string">&quot;99&quot;</span> <span class="gtkdoc opt">!</span> rtpssrcdemux name<span class="gtkdoc opt">=</span>demux \
-            demux<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpspeexdepay <span class="gtkdoc opt">!</span> speexdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
-            demux<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpspeexdepay <span class="gtkdoc opt">!</span> speexdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
-        rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">! !</span> tee name<span class="gtkdoc opt">=</span>t <span class="gtkdoc opt">!</span> queue <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> t<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5004</span> \
-        udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink sync<span class="gtkdoc opt">=</span>fakse async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
+8
+9
+10
+11
+12
+13</pre></td>
+        <td class="listing_code"><pre class="programlisting">gst<span class="gtkdoc opt">-</span>launch<span class="gtkdoc opt">-</span><span class="number">1.0</span> rtpsession name<span class="gtkdoc opt">=</span>rtpsession rtp<span class="gtkdoc opt">-</span>profile<span class="gtkdoc opt">=</span>avpf \
+    udpsrc port<span class="gtkdoc opt">=</span><span class="number">5000</span> caps<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97&quot;</span> <span class="gtkdoc opt">!</span> \
+        rtpsession<span class="gtkdoc opt">.</span>recv_rtp_sink \
+    rtpsession<span class="gtkdoc opt">.</span>recv_rtp_src <span class="gtkdoc opt">!</span> \
+        rtprtxreceive payload<span class="gtkdoc opt">-</span>type<span class="gtkdoc opt">-</span>map<span class="gtkdoc opt">=</span><span class="string">&quot;application/x-rtp-pt-map,97=(uint)99&quot;</span> <span class="gtkdoc opt">!</span> \
+        rtpssrcdemux name<span class="gtkdoc opt">=</span>demux \
+    demux<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpopusdepay <span class="gtkdoc opt">!</span> \
+        opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
+    demux<span class="gtkdoc opt">. !</span> queue <span class="gtkdoc opt">!</span> rtpjitterbuffer <span class="keyword">do</span><span class="gtkdoc opt">-</span>retransmission<span class="gtkdoc opt">=</span><span class="keyword">true</span> <span class="gtkdoc opt">!</span> rtpopusdepay <span class="gtkdoc opt">!</span> \
+        opusdec <span class="gtkdoc opt">!</span> audioconvert <span class="gtkdoc opt">!</span> autoaudiosink \
+    udpsrc port<span class="gtkdoc opt">=</span><span class="number">5002</span> <span class="gtkdoc opt">!</span> rtpsession<span class="gtkdoc opt">.</span>recv_rtcp_sink \
+    rtpsession<span class="gtkdoc opt">.</span>send_rtcp_src <span class="gtkdoc opt">!</span> udpsink host<span class="gtkdoc opt">=</span><span class="string">&quot;127.0.0.1&quot;</span> port<span class="gtkdoc opt">=</span><span class="number">5001</span> \
+        sync<span class="gtkdoc opt">=</span><span class="keyword">false</span> async<span class="gtkdoc opt">=</span><span class="keyword">false</span></pre></td>
       </tr>
     </tbody>
   </table>
 </div>
- Receive audio stream from port 5000.
-On sender side the two streams have the same payload type for master streams, Same about retransmission streams.
-The streams are sent to the network through two distincts sessions.
-But we need to set a different seqnum-offset to make sure their seqnum navigate at a different rate like in concrete cases.
-We could also choose the same seqnum offset but we would require to set a different initial seqnum value.
-This is also why the rtprtxreceive can succeed to do the association between master and retransmission stream.
-On receiver side the same session is used to receive the two streams. So the rtpssrcdemux is here to demultiplex
-those two streams. The rtprtxreceive is responsible for reconstructing the original packets from the two retransmission streams.
-You can play with the drop-probability value for one or both streams.
-You should hear a clear sound. (after a few seconds the two streams wave feel synchronized)
+<p> Receive two audio streams from port 5000.</p>
+<p>In this example we are streaming two streams of the same type through the
+same port. They, however, are using a different SSRC (ssrc is randomly
+generated on each payloader - rtpopuspay in this example), so they can be
+identified and demultiplexed by rtpssrcdemux on the receiver side. This is
+an example of SSRC-multiplexing.</p>
+<p>It is important here to use a different starting sequence number
+(seqnum-offset), since this is the only means of identification that
+rtprtxreceive uses the very first time to identify retransmission streams.
+It is an error, according to RFC4588 to have two retransmission requests for
+packets belonging to two different streams but with the same sequence number.
+Note that the default seqnum-offset value (-1, which means random) would
+work just fine, but it is overriden here for illustration purposes.</p>
 </div>
 <div class="refsynopsisdiv">
 <h2>Synopsis</h2>
 <div class="refsect2">
-<a name="id-1.2.201.7.4.1"></a><h3>Element Information</h3>
+<a name="id-1.2.201.7.7.1"></a><h3>Element Information</h3>
 <div class="variablelist"><table border="0" class="variablelist">
 <colgroup>
 <col align="left" valign="top">
@@ -263,7 +296,7 @@ You should hear a clear sound. (after a few seconds the two streams wave feel sy
 </div>
 <hr>
 <div class="refsect2">
-<a name="id-1.2.201.7.4.2"></a><h3>Element Pads</h3>
+<a name="id-1.2.201.7.7.2"></a><h3>Element Pads</h3>
 <div class="variablelist"><table border="0" class="variablelist">
 <colgroup>
 <col align="left" valign="top">

data/vendor/local/share/gtk-doc/html/gst-plugins-good-plugins-1.0/gst-plugins-good-plugins-rtpsbcdepay.html CHANGED Viewed

@@ -51,7 +51,7 @@
     <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gobjectgobject-The-Base-Object-Type.html#GInitiallyUnowned">GInitiallyUnowned</a>
         <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstObject.html#GstObject-struct">GstObject</a>
             <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gstreamer-1.0GstElement.html#GstElement-struct">GstElement</a>
-                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0gst-plugins-base-libs-gstrtpbasedepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
+                <span class="lineart">╰──</span> <a href="/usr/share/gtk-doc/html/gst-plugins-base-libs-1.0GstRTPBaseDepayload.html#GstRTPBaseDepayload-struct">GstRTPBaseDepayload</a>
                     <span class="lineart">╰──</span> GstRtpSbcDepay
 </pre>
 </div>