<div dir="ltr"><div dir="ltr"><div dir="ltr">Thanks Matthew.</div><div dir="ltr"><br></div><div dir="ltr">Just to understand clearly, I have the following in a pipeline,<br><div><br></div><div>"wasapisrc role=comms use-audioclient3=true low-latency=true ! queue ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! queue name=audio_queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=111 ! webrtcbin ".</div><div><br></div><div>To mute, the queue (named audio_queue) src pad and the webrtcbin sink_pad (sink_%u) have to be unlinked and</div><div><br></div><div>to unmute, both pads must be linked back together.</div><div><br></div><div>Is my understanding correct?</div><div><br></div><div>Thanks.</div><div><br><div><br></div><div><br></div></div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Feb 18, 2022 at 2:53 AM Matthew Waters <<a href="mailto:ystreet00@gmail.com">ystreet00@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
Setting the sendrecv/recvonly/sendonly does not change what data you
push into webrtcbin. If you want 'mute' functionality, stop pushing
audio/video into webrtcbin.<br>
<br>
Cheers<br>
-Matt<br>
<br>
<div>On 18/2/22 05:15, Busayo Famutimi via
gstreamer-devel wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div dir="ltr">Hello.
<div><br>
</div>
<div>Please I have a working webrtc peer connection using.
However, I am unable to set the transceivers (both audio and
video) dynamically.</div>
<div><br>
</div>
<div>I have the following code:</div>
<div><br>
</div>
<div>
<div style="color:rgb(212,212,212);background-color:rgb(30,30,30);font-family:Consolas,"Courier New",monospace;font-size:14px;line-height:19px;white-space:pre-wrap"><div><span style="color:rgb(78,201,176)">GArray</span> *<span style="color:rgb(156,220,254)">transceivers</span>;</div><div> <span style="color:rgb(220,220,170)">g_signal_emit_by_name</span>(<span style="color:rgb(156,220,254)">voip_data</span>-><span style="color:rgb(156,220,254)">webrtc</span>, <span style="color:rgb(206,145,120)">"get-transceivers"</span>, &<span style="color:rgb(156,220,254)">transceivers</span>);</div><div> <span style="color:rgb(86,156,214)">g_assert</span>(<span style="color:rgb(156,220,254)">transceivers</span> != <span style="color:rgb(86,156,214)">NULL</span> && <span style="color:rgb(156,220,254)">transceivers</span>-><span style="color:rgb(156,220,254)">len</span> > <span style="color:rgb(181,206,168)">1</span>);</div><div> <span style="color:rgb(197,134,192)">for</span> (<span style="color:rgb(78,201,176)">guint</span> <span style="color:rgb(156,220,254)">i</span> = <span style="color:rgb(181,206,168)">0</span>; <span style="color:rgb(156,220,254)">i</span> < <span style="color:rgb(156,220,254)">transceivers</span>-><span style="color:rgb(156,220,254)">len</span>; <span style="color:rgb(156,220,254)">i</span>++) </div><div> {</div><div> <span style="color:rgb(78,201,176)">GstWebRTCRTPTransceiver</span> *<span style="color:rgb(156,220,254)">transceiver</span> = <span style="color:rgb(86,156,214)">g_array_index</span>(<span style="color:rgb(156,220,254)">transceivers</span>, <span style="color:rgb(78,201,176)">GstWebRTCRTPTransceiver</span> *, <span style="color:rgb(156,220,254)">i</span>);</div><div> <span style="color:rgb(78,201,176)">GstWebRTCKind</span> <span style="color:rgb(156,220,254)">kind</span>;</div><div> <span style="color:rgb(220,220,170)">g_object_get</span>(<span style="color:rgb(156,220,254)">transceiver</span>, <span style="color:rgb(206,145,120)">"kind"</span>, &<span style="color:rgb(156,220,254)">kind</span>, <span style="color:rgb(86,156,214)">NULL</span>);</div><div> <span style="color:rgb(197,134,192)">if</span> (<span style="color:rgb(156,220,254)">kind</span> == <span style="color:rgb(79,193,255)">GST_WEBRTC_KIND_AUDIO</span>)</div><div> {</div><div> <span style="color:rgb(197,134,192)">if</span> (<span style="color:rgb(156,220,254)">mic_action</span> == <span style="color:rgb(79,193,255)">MUTE_MIC</span>)
</div><div> <span style="color:rgb(220,220,170)">g_object_set</span>(<span style="color:rgb(156,220,254)">transceiver</span>, <span style="color:rgb(206,145,120)">"direction"</span>, <span style="color:rgb(79,193,255)">GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY</span>, <span style="color:rgb(86,156,214)">NULL</span>);</div><div> <span style="color:rgb(197,134,192)">else</span> <span style="color:rgb(197,134,192)">if</span> (<span style="color:rgb(156,220,254)">mic_action</span> == <span style="color:rgb(79,193,255)">UNMUTE_MIC</span>)</div><div> <span style="color:rgb(220,220,170)">g_object_set</span>(<span style="color:rgb(156,220,254)">transceiver</span>, <span style="color:rgb(206,145,120)">"direction"</span>, <span style="color:rgb(79,193,255)">GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV</span>, <span style="color:rgb(86,156,214)">NULL</span>);</div><div> <span style="color:rgb(197,134,192)">else</span></div><div> <span style="color:rgb(86,156,214)">g_assert_not_reached</span>();</div><div> }</div><div> }</div><div> <span style="color:rgb(220,220,170)">g_array_unref</span>(<span style="color:rgb(156,220,254)">transceivers</span>);</div></div>
</div>
<div><br>
</div>
<div>but the other peer still receives audio.</div>
<div><br>
</div>
<div>I am trying to create the mute/unmute microphone
functionality.</div>
<div><br>
</div>
<div>How can this be achieved?</div>
<div><br>
</div>
<div>Thanks.</div>
</div>
</div>
</blockquote>
<br>
</div>
</blockquote></div>