AES16e-50 sync problem

I have a AES16e-50 and it seems like all of my software is unable to sync with the card. The card syncs nice in both internal and aes50 clocks but there is clicks, pops and noises with all buffer lengths and samplerates. Strange thing is that with card syncing from aes50 (96k) it is possible to set programs to run with other rates. With long buffers (4096 or 8192 samples) the problem does not significantly get better or worse even if programs are running on 88.2k rate while card is running 96k. It is possible to set software even to 44,1k rate while card is running 96k and still get somewhat recognizable audio through. When routing through internal mixer audio is ok as long as card and aes50 source are properly set up and they manage to lock sync.

Both software and firmware are latest available and I have tried to re-install the drivers and software. I'm running win7 64bit, CPU is AMD A10-5800K and I have 8gb of ram installed so that should be enough of power. I plan on running vst-plugins in live environment so low latency is necessary. Is there any way to test where the sync is lost or do you have any ideas what to try?

Thanks in advance!

 -Ossi Miikkulainen

Lynx mixer never shows that the sync is lost as it would mean complete loss of aes50 stream also. If I force another rate or select wrong clock master/slave settings it does not find sync and all audio through aes50 drops. 

My source is locked to 96k samplerate, if I let it be master and tell card to take sync from aes50 I can choose any samplerate in software but the card is locked to 96k and audio is glitchy. If I tell card to be master at 96k and my source to slave same thing happens, card shows sync lock to 96k, software can be set to any rate and audio is broken. If card is in any other rate than 96k it does not find sync and there is no audio through aes50, also if there are two masters or two slaves at 96k it doesn't find sync and no aes50 audio is present.

No, the mixer lock is not enabled.

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?

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.

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). 

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

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

 

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.