asus p8z68 v pro gen 3 pcie slots

**TheJackAttack** · 03-11-2012, 10:46 AM

Basically. I would run it this way while I bought the coax to hookup the BNC wordclock. It would be better all around and it really doesn't cost that much I promise. Still use your RayDat as the master and then run to ADA8000 #1 with a T connector and then run on to ADA8000 #2. If you add 3 and 4 you just add a T onto the #2 unit and continue down the chain. The last in the line still has a T and then you put a termination cap on the unused connection to finish.

**djmukilteo** · 03-11-2012, 11:20 AM

You can try the WC, but I still wouldn't be using the Berhingers as master...
I think you really need to run both ADA's like you have it setup right now and track the 16 channels simultaneously.
That is the only time you will have both fully packed data streams coming into the RayDAT and being recorded to Sonar (at least 5 minutes). That will be the proof that it's stable and everything stays in sync and if the ADAT glitches or looses sync or acts strange then that's when it will happen. Then I would look into doing the WC setup.

Just out of curiosity....are both the ADA's Sync Lock LED's lit on their front panels?
Does the RayDAT indicate ADAT is sync locked?

**offcenter2005** · 03-11-2012, 12:12 PM

Both the Behringer locked lights are on and in the hdsp settings the input status says adat1 sync 44.1 kHZ and adat2 sync 44.1 kHZ. Is there any other settings that i should mess with in sonar to optimize the raydat. I set it to ASIO and the at 44.1 kHZ and didnt mess with anything else.

**TheJackAttack** · 03-11-2012, 12:36 PM

No.

**offcenter2005** · 03-11-2012, 12:59 PM

I set the buffer size to 32 samples and opened a 36 track song that i did for someone and it is loaded with plugins and it ran smooth without any pops or glitchs.

**TheJackAttack** · 03-11-2012, 01:02 PM

Great, but that isn't recording. Do test dj suggested.

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**djmukilteo** · 03-11-2012, 01:02 PM

No that is perfect!
The bullet proof test is the 16 track test....full blown max all ADAT channels going.
If it passes that stress test for at least 5 minutes of recording time and everything is accurately recorded with no dropouts and the playback is glitch free.....I would confidently say you can expect it will work flawlessly 24/7.
Make a note of the latency figures when you do that.
Even though you may never do 16....you are heavily dependent on the ADAT working properly as that is your only interface method and it will verify that everything stays together...

**offcenter2005** · 03-11-2012, 01:06 PM

Originally Posted by djmukilteo

No that is perfect!
The bullet proof test is the 16 track test....full blown max all ADAT channels going.
If it passes that stress test for at least 5 minutes of recording time and everything is accurately recorded with no dropouts and the playback is glitch free.....I would confidently say you can expect it will work flawlessly 24/7.
Make a note of the latency figures when you do that.
Even though you may never do 16....you are heavily dependent on the ADAT working properly as that is your only interface method and it will verify that everything stays together...

If i add more inputs would the stability at that sample rate be affected?

**offcenter2005** · 03-11-2012, 01:09 PM

Also on that track my cpu meter is reading barley anything. Is that a good sign that it will stay stable with that many tracks and plugins?

**djmukilteo** · 03-11-2012, 02:11 PM

Originally Posted by offcenter2005

If i add more inputs would the stability at that sample rate be affected?

The sample rate (44.1khz) won't but the sample buffer setting will. That's why it's important to do that test.
The ability to stream 16 channel into your system and "print" those to your internal HDD (which is your maxed out setup) is all dependent on the buffer size which is a temporarily space storing the incoming bits before transferring them onto the HDD. It's how the incoming stream keeps up with the CPU, RAM and HDD. If it's too small and it fills up with data coming in and the HDD isn't ready to accept the buffer data it will crap out....dropouts, clicks, pops will start to occur. The bigger the buffer the less likely there will be a problem with that throughput. The bigger it is the more time it takes to fill before it transfers, the smaller the quicker...but if it's too quick somebody is not going to be ready and they have a problem.....they keep going but they will skip or drop some of the data. Somewhere in the middle is where you will find the perfect match of size and speed to keep up with each other. And as you know the latency increases with increasing the sample buffer size.
A 32 sample buffer in terms of size is small and fast and might work for one track or maybe even two or three, but 16 data packets coming in is a whole other story and may not work. Setting the buffer and finding that happy medium is all part of optimizing "your" system. The buffer can be changed depending on what your doing too...if your just playing back some tracks and recording one overdub and you want low latency you might find that 128 works best or 64 or 32, but when you set up 12 mics and 4 instruments and recording all that live at once...it will need to go up..
You'd have to find someone who has the "exact" same system you have in front of you (highly unlikely) to be able to figure out what works correctly!...so the only way for you to know is do it....there is no fixed setting that works on every machine...

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