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Hello,
...but here goes...and please excuse my neophyte-ness...
As I understand it, the quality of the audio available in SAC - and SAW for the at matter, is superb.
As a result, would the quality or the cost of an audio interface have a positive or a negative effect on the quality of this output?
Note: I hope that this all makes sense?
Anything that you record through whatever interface and converters is certainly affected by the quality of both, although past a certain point price rises a lot faster than quality. Of course, the quality of what you hear on playback is also affected by the quality of the interface and converters you use.
But... suppose you are mixing files from a session recorded by someone else. Your own interface certainly does not affect the quality of those received files. And if you do a "build mix" operation, thus generating a file, the quality of the file generated is not at all affected by what interface you are using.
So, the short answer to your question is "It depends..."
Hello,
Thanks for the comments, Carey.
Again, I beg your indulgence, when using the RMD HDSP 9652 as Master and the ADA-82000 as slave combo, are we hearing the quality of both devices or of just one of the devices?
For all practical purposes, your only hearing the ADA-8200.Quote:
Originally Posted by mr_es335
The ADA-8200 is the component that deals with amplifying the input signal and converting it to a digital form (ADAT).
If used as an output, it is also the component that is responsible for converting the digital signal (ADAT) back to an analog form.
The RME card is only the transport mechanism for moving the digital content into and out of the computer in a manner that can be processed by the software. (ADAT to computer data, and computer data to ADAT). It is strictly a transport layer and aside from very small artifacts that might occur due to jitter, it has no impact or affect on the sound quality going in or out of the system.
With the interface part of the setup, the two primary concerns are the driver and the stability of the clock. A driver must be well written to enable low latency operation (even an ASIO driver, if poorly written, will not get you really low latency. Clock stability does have a subtle effect on sound quality. What you want is a clock with low jitter to give you the best clarity of sound.Quote:
Originally Posted by mr_es335
The good news here for you is that the RME interfaces have excellent clock stability, and even with your "older" model, you have nothing to worry about from it in the quality department. Particularly if your computer and your converters are mounted together in the same rack, I would say definitely keep the RME HDSP 9652. When I had one, I had noting but good from it. It never failed me (I use something else now, but not because of any quality issue with the RME).
The primary effect on sound quality will generally be in the converters. When I used the ADA8000 units, I did have some problems in that department, and I later switched over to Presonus Digimax FS, which have worked better for me. I have no direct experience with the ADA8200, but my guess is that these are better than the earlier model, and if so, particularly in live PA use, you have nothing to worry about.
Another thing about clocking: In many cases, you will get the most stable clocking by using the internal clock source of the converter and slaving the interface to that. However, if you have more than one converter box, you need to make the interface the master clock source (as you say you have done). Every time clock has to be carried from one place to another, you can introduce a certain (usually very small) amount of jitter. The most stable way to transmit clock is by coaxial cable. Optical cables generally introduce more jitter than copper. The PLL (circuit that is used to control the internal clock from an external master) in the slave device may introduce a little more jitter. For my own setup, I never noticed any audible problem from clock carried over ADAT, but it is technically not the best way.
You should also know that clock jitter only affects sound quality at the converter itself. In the rest of the system, it is completely ignored.
In my studio, I let the interface clock be the master. It is more convenient that way, because then any clock rate I set in SAW or SAC is automatically followed by the converters.
In my remote recording rig, I make the internal clock of my Digimax FS converter box the master and the interface follows it. That gets me the best clock stability for the first 8 channels. The downside is, I then have to set clock rate manually on the Digimax FS as well as select the same clock rate in SAW or SAC and in the control application for my Focusrite Scarlett 18i8. This means more hoops to jump through during setup before recording.
Anyway, my take is that with your setup, and in your live performance application, I probably wouldn't worry too much.
Go thou out and play, and have fun doing it!