<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"> <meta name="Generator" content="Microsoft Exchange Server">  <style>.EmailQuote { margin-left: 1pt; padding-left: 4pt; border-left: #800000 2px solid; }</style></head> <body> <div class="PlainText">Did you get this to work? I am facing a perhaps similar issue with a webrtc pipeline on Windows, where it seems like echo cancellation simply has no effect at all. In my pipeline I use directsoundsrc ! webrtcdsp ! ... ultimately going to webrtcbin and on the webrtc receiving side ... ! webrtcechoprobe ! autoaudiosink buffer-time=40000 low-latency=true. I have tried different values for buffer-time and low-latency, but it seems that regardless of what I do there is no audible echo cancellation at the other end of the webrtc connection. I would appreciate any further advice. Best regards Thorsten -----Original Message----- From: gstreamer-devel <gstreamer-devel-bounces@lists.freedesktop.org> On Behalf Of Maksim Danilov Sent: 24. marts 2021 07:49 To: gstreamer-devel@lists.freedesktop.org Subject: [EXT] Re: How to configure webrtc echo cancellation in Windows 10? EXTERNAL MAIL: Thanks for the answer. I gave up on configuring AEC over rtp and came up with simple example that look like: wasapisrc ! audioconvert ! audio/x-raw, format=S16LE, rate=48000 ! audioconvert ! webrtcdsp ! webrtcechoprobe ! audioconvert ! wasapisink. It gives no result (in linux everything work as expected). I tried you suggestion with low-latency mode, however got messages with 'invalid latency add some queues to the pipeline'. Even if I add queue to pipeline. I can't hear my self at all. That pipeline gives the same result: wasapisrc low-latency=true ! <queue> ! wasapisink low-latency=true. So I think the problem is in plugin. Probably it add some additional latency in implementation, cause it doesn't act like real time. I inspected the code of plugin and it looks like a very fast implementation of api under Windows. Moreover it can't even capture specific caps of src/sink and uses shared format F32LE (maybe it is resampling and converting audio in core and we get some latency through). -- Sent from: <a href="https://urldefense.com/v3/__http://gstreamer-devel.966125.n4.nabble.com/__;!!B0kNdk3R__Efj5F6ERhD!tlRfLYhjrcUeKRO_m1dp9p_NlMnuA49CZbs_9QlxDV1lxxbTB63mpcy-t0zv4zmqxf37OPehiA$">https://urldefense.com/v3/__http://gstreamer-devel.966125.n4.nabble.com/__;!!B0kNdk3R__Efj5F6ERhD!tlRfLYhjrcUeKRO_m1dp9p_NlMnuA49CZbs_9QlxDV1lxxbTB63mpcy-t0zv4zmqxf37OPehiA$</a> _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.freedesktop.org <a href="https://urldefense.com/v3/__https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel__;!!B0kNdk3R__Efj5F6ERhD!tlRfLYhjrcUeKRO_m1dp9p_NlMnuA49CZbs_9QlxDV1lxxbTB63mpcy-t0zv4zmqxf32AanurA$">https://urldefense.com/v3/__https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel__;!!B0kNdk3R__Efj5F6ERhD!tlRfLYhjrcUeKRO_m1dp9p_NlMnuA49CZbs_9QlxDV1lxxbTB63mpcy-t0zv4zmqxf32AanurA$</a> ? </div> </body> </html>