AES16e-50 sync problem

Mixer version is 2.00 build 23, firmware version is 14.0 but I have already tried version 13.2 with no success.

I installed my Motu 896mk3 to the same computer and programs and plugins run nice with same samplerates and buffer lengths so after all it really seems to be a hardware issue.

I have been having similar problems with the lynx AES16e-50 hooked up to a Midas DL32 stage box running RML labs Software Audio Console ( SAC ) as the software. I don't think it is a clocking issue. For me It seems to be a dropped buffer issue between the DAW software and the Lynx Mixer? I could not get clean audio out at 1 buffer 64 samples, 1 buffer 128 samples or even 1 buffer 264 samples, but when I tried 3 buffers at 32 samples it worked. I then tried 2 buffers at 32 samples and it remained stable with no audible glitches. SAC has a window in the software Where it reports slipped buffers and while I was trying all this it never reported any slipped buffers yet something was definitely going on.

Phase 2:

AS I started to test further with more audio source input and screen changes, plugin window opening etc..

I've found that the SAC program ( which has several "settings" to choose from such as Realtime Priority, High Priority, Single CPU etc.. ) needed to be set to High Priority NOT Realtime Priority in order to have mostly clean audio. I say mostly, because with this setting I got rid of the constant "dropped Buffer" sounds but SAC Itself will now drop the occasional buffer and register the event in its window.

This is what has lead me to the possibility that there could be some conflict in the transfer between the DAW and The Lynx mixer? i.e. both cannot have "Real time priority" just a theory as the lynx mixer does not report slipped buffers as far as I can tell.

What DAW type software was used in testing with the Lynx AES16e-50?

Is it even possible for the card/lynx mixer to run fine at 2x32 samples but NOT at 1x higher buffer sizes? this would seem strange to me.

Thoughts?

Hal

edit: I tested both the stagebox and the aes16e-50 cards with both desks.

Because of all the christmas-hassle I didn't get chance to get that RMA-quote so I ran some more tests. When running Reaper or Waves TracksLive I can get audio through clean with 64 sample buffers, with plugins in Reaper I need 128 samples for somewhat reliable operation. Most of the problems seem to come along between software output and aes50 input of host device as my last plugin in chain is fft spectrogram that shows almost no noise at all but the audio from aes50 has clearly problems. I've also tested recording audio and then playing it back several times and the noise is not in the recorded audio as it shows up in different times.

With 256 sample buffers I get acceptable reliability but I'd like to have shorter buffers. The strange thing is that even with very long buffers the Waves Multirack still refuses to start audio. I still get the same message every 10 seconds or so stating that MultiRack has lost sync and audio will be stopped. Is this still likely to be hardware issue or is there something I could test or try? If it is not likely to be hardware problem then I will contact Waves support and ask their opinion.

Is the card installed directly or are you using an expansion chassis? 

Do you have a different computer you can test the card in?

 

First off, I would suggesting running the DPC latency checker to see if you are getting spikes. If so, this will give us an objective way of measuring if changes made to the system will help or not. 

Look into your Bios. If you have Speedstep or EIST, disable it. Same with C States (C1, C3. C6, etc). 

If the card does not get sync with aes50 the whole connection is dead. Also when using only lynx mixer to route audio from stagebox inputs back to stagebox outputs everything is fine so the aes50 side and lynx mixer works ok. Only when trying to record, play or loop back audio through any software everything becomes glitchy. I have tried every possible combination of clock sources and SR selection sources but so far nothing has worked.

Firmware was version 14, I downgraded it and didn't notice any difference. My educated guess is that the ASIO-driver somehow is unable to communicate the sync between software and hardware as both ends report all-ok. If the software is given permission to decide samplerate and the card is on internal clock it is possible to change hardware samplerate from software. (that naturally drops sync between stagebox and the card)

I have tried to re-install the drivers but it haven't helped, I'll try tomorrow a complete clean install of windows, drivers and software. Is it possible to somehow extract software clock and compare it to the cards clock?