Thanks Anthony. Question: what if in the web RTC client SDP you have it set so that you are sending mono and receiving stereo. What happens then? Does the core allow for that kind of asymmetry? It must, right, because, for instance, I was sending a mono stream (or perhaps dual mono) and hearing stereo in the conference on Friday. <div>Thanks, chris<span></span><br><br>On Tuesday, June 23, 2015, Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>If you call in and negotiate stereo, it uses stereo, if you try to play a mono audio stream it will resample and or mux it appropriately.</div><div>If you call in with mono to mod_conference in stereo mode it will up-sample the input and the core will down-sample the output.</div><div><br></div><div>If you are not using our verto demo with the stereo checked and calling with opus, there are not many other ways to turn stereo on.</div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Jun 23, 2015 at 5:46 PM, Chris Mandra <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','mandra@gmail.com');" target="_blank">mandra@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Anthony - thanks for the info. <div>So, "channels" is already set to 2 in conf/autoload_configs/conference.conf.xml <div>
<p> <profile name="video-mcu-stereo"></p>
<p> <param name="domain" value="$${domain}"/></p>
<p> <param name="rate" value="48000"/></p>
<p> <param name="channels" value="2"/></p>
<p> <param name="interval" value="20"/></p>
<p> <param name="energy-level" value="200"/></p><p><br></p><p>Does freeswitch test for actual stereo (as opposed to two channel mono) and if it sees 2 chan mono does it downsample to mono?</p><p>thanks!</p><p>chris</p></div></div></div><div class="gmail_extra"><div><div><br><div class="gmail_quote">On Tue, Jun 23, 2015 at 6:26 PM, Anthony Minessale <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','anthony.minessale@gmail.com');" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Yes its interleaved.</div></div><div><br></div><div>If you are calling into a conference you will have to run the conference in stereo or the channel will be downsampled to mono to match.</div><div>Set channels to 2 in the mod_conference profile.</div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Jun 23, 2015 at 5:21 PM, Chris Mandra <span dir="ltr"><<a href="javascript:_e(%7B%7D,'cvml','mandra@gmail.com');" target="_blank">mandra@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi guys, I have another question about freeswitch, conferences and stereo.<div><br></div><div>When I make a webrtc call into 3500 on my freeswitch server, mod_ladspa is reporting the number of samples as 1920 (20 ms stereo interleaved I assume) fine for a few seconds, and then the number of samples drops to 960 (20 ms mono) Any idea why this might be happening?</div><div><br></div><div>Also, I have been assuming that stereo in freeswitch is interleaved - is that the case? Is that assumption incorrect? If not interleaved, how does fs handle stereo. <br clear="all"><div><br></div><div><br></div><div>thanks!</div><span><font color="#888888"><div>chris</div><br>
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