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Trying to test audiocards...the wrong way?
post Feb 9 2013, 22:35
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Hi there.

I was trying to make some tests to audiocards that I own (Roland UA-1G, M-Audio 2494, M-Audio Delta 1010LT), in order to do some other testing after that with several CD, DVD players, Media Centers, etc.

My first thought was something like having the outputs of the soundcard connected to the inputs, play one song and record it on another track (I'm using Nuendo).
That's the part of the process where all things worked out as planned, except...
...I was thinking that by doing this, if I phase reversed one of the stereo tracks, I would end up only with the differences between the two (or nothing, on a perfect scenario).

I had to compensate for the delay and any volume difference, but on the best 'phase cancellation' spot, I still hear (mostly) the very high frequencies.

That didn't surprised me on the Roland UA-1G, as it's a USB card with volume control and dedicated to record instruments on the go, so I would find fair enough that it wasn't very flat on high frequencies. That's something that even REW confirms in the 'soundcard calibration' part of the setup.

But having the same behaviour with both the M-Audio made me wonder if there's something wrong with my test basics.

I'm playing and recording at 44.1 and 16 bit (trying to keep it from the original source to the end Wave file), so any input on this theory would be appreciated. smile.gif


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post Feb 9 2013, 22:37
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The standard recommendation is a loopback test using RightMark Audio Analyzer.

This page offers some warnings, best practices, and criticisms.

This post has been edited by db1989: Feb 9 2013, 22:41
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