<html> <head> <style></style></head> <body class='hmmessage'><div dir='ltr'> <span style="color: rgb(34, 34, 34); font-family: arial, sans-serif; background-color: rgb(255, 255, 255); ">Thank you Stefan, this worked with acceptable </span><span class="il" style="background-color: rgb(255, 255, 204); color: rgb(34, 34, 34); font-family: arial, sans-serif; ">audio</span><span style="color: rgb(34, 34, 34); font-family: arial, sans-serif; background-color: rgb(255, 255, 255); "> quality.</span><div style="color: rgb(34, 34, 34); font-family: arial, sans-serif; background-color: rgb(255, 255, 255); ">B.<br><div><br><div class="gmail_quote">On Thu, Sep 6, 2012 at 1:05 AM, Ștefan *** wrote:<div class="im adL" style="color: rgb(80, 0, 80); "><br><blockquote class="gmail_quote" style="margin-right: 0px; margin-left: 0.8ex; border-left-width: 1px; border-left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; ">Hello,<br><br>As far as I'm aware, you will need to use null and combine sinks as<br>well as module-loopback. I'm thinking of something along those lines:<br><br>1. Set up two null sinks (module-null-sink), NULL1 and NULL2.<br>2. Set up two combine sinks (module-combine-sink) that combine each of<br>the null sinks with your built-in sink, COMBINE1 and COMBINE2<br>3. Set up two loopbacks (module-loopback) from your built-in source<br>(microphone) to each of the null sinks<br>4. Have Skype play to COMBINE1 and record from the monitor of NULL2<br>5. Have Google Talk play to COMBINE2 and record from the monitor of NULL1<br><br>You can automate steps 1–3 by appending the commands you would type in<br>pacmd to ~/.pulse/<a href="http://default.pa/" target="_blank" style="color: rgb(17, 85, 204); ">default.pa</a> (assuming you are recording from a card<br>that doesn't get plugged in after pulseaudio has started).<br><br>I don't know if this is the best choice, it might have a high latency,<br>but VoIP has quite high latency anyway.<br><br>Best,<br>Ștefan</blockquote></div></div></div></div><br><div><div id="SkyDrivePlaceholder"></div><hr id="stopSpelling">From: berteh@<br>To: pulseaudio-discuss@lists.freedesktop.org<br>Subject: help with audio pipe / monitor<br>Date: Wed, 5 Sep 2012 13:48:30 +0200<br><br> <style></style> <div dir="ltr"> Hello,<br><br>I'd like to configure my pulse audio (on ubuntu precise)to bridge over different conferencing software. I've seen various approaches online (pactl null sink, monitors...) and wonder what would be the best way to do it.<br><br>The audio sources are Google Hangout (WEBRTC playStream), Skype (Input), and my internal (analog) microphone.<br>The audio outputs are similarly Hangout (WEBRTC recStream), Skype (Output), and my internal analog headphones.<br><br>I want, in the end, something I'd call a full duplex setup, where <b>each audio out gets the two other audio ins relayed</b>, e.a. <br><ol><li>Skype Output would get mic+hangout playStream,</li><li>Hangout recStream would get mic+Skype out, and <br></li><li>my headphones would get Skype input + Hangout playStream.</li></ol><br>I'm not very familiar yet with pulseaudio, but any suggestion would be really appreciated. I'm comfortable with the pavucontrol, paprefs, paman tools, don't know much about the pulse command lines yet.<br><br>Have a nice day,<br>Berteh.<br> </div></div> </div></body> </html>