Why I never got that perfect null in difference testing part one
Entry posted by esldude
I set out to see what the difference between expensive analog interconnects and cheap ones were. My method was simple in essence. I would send a signal to a DAC, send the analog output over an interconnect, record that digitally with an ADC. Then repeat with a different cable. Take the two digital files and subtract one from the other using software. There was a long thread posted in the general forum about my attempts. I learned some things along the way and eventually could get consistent results. But not perfection.
An absolutely perfect result would indicate no differences whatsoever. A perfect difference file would be an infinitely deep null with nothing left. While possible in the digital realm that actually isn't possible in the analog world. At a minimum you have thermal noise. So one step from total perfection in one sense. However if one differenced two files and nothing was left above the thermal noise floor that would be practical perfection. Mainly the thermal noise limit is defined by the impedance of your electronics. It is simply calculated like here at this online calculator:
http://www.sengpielaudio.com/calculator-noise.htm
Notice bandwidth and impedance are the main factors with temperature being a minor effect.
Along with thermal noise most of the time you run into flicker noise. Thermal noise is white in character while flicker noise is pink. Your real world noise floors typically will be a combination of those two. In my case, recording silence with a digital stream going I got this with my best equipment. You may notice a couple idle tones sticking up a bit. One is around 14.5 khz and a related one at 7250 hz. These are common in DACs for one reason or another. One FFT plot is a linear frequency view and the other FFT plot has log scaling on frequency:
So with noise in the picture the perfect obtainable null would be one where the difference between two files left only this noise floor. Even amplified such a sound file would only result in the whoosh of noise. Without amplification and lots of it you of course will hear nothing over your speakers or headphones.
But I never quite got that either. Even when running a piece of music or test tones through the same analog cable twice in a row changing nothing I never got two signals to null out with only this low level noise signature. Instead I got something like this:
This particular one is some music recorded first over Audioquest Diamond X3 and then over the cheapest, junkiest interconnect I have. Then the two files subtracted from each other. It doesn't matter, they looked the same even recorded from the AQ twice in a row. The highest remaining signals at -117 dB are right above -100 db from the original signal level. Above 2 khz I did get nothing poking out of the noise floor, but not from 50 hz to 2 khz. A practically perfect result above 2 khz only.
Now play this signal over your speakers and even at max volume you hear nothing at all. Making me think even these imperfect results are close enough the cable is not significant. Still, amp this up by 60 db or more and you hear the noisy hiss and at times fleeting bits of the original music. I am attaching it as a zipped MP3 if you wish to hear it. Labeled "musical imperfect null". It has been amplified 60 dB without which you would hear only silence. This is about the gain of a phono stage plus the gain of a power amp together.
This however kept bugging me. What was keeping me from getting just a noise floor? Even if cables differed, using the same cable twice should have given nothing more than residual noise. Yet the music at some very low level left traces in the result. I think I have figured out why, and will go further into these issues in part two.
Find part two here:
Just for reference for those wondering about the equipment used. Playback was with Win7 on a netbook via the Audiophilleo 2 USB/SPDIF converter. I also sometimes used a Laptop and desktop with no difference in the results. Playback software was Foobar 2000 in WASAPI mode which is a bit perfect playback software. Bit perfection was also confirmed by comparing a SPDIF recorded file to the original. The Audiophilleo 2 fed the input of a TACT RCS 2.0 room correction pre-amp set to bypass mode and unity gain. All done at 24 bit and usually 44.1 khz. The analog output of the TACT fed a TC Electronics Impact Twin. This firewire controlled device was clock locked to the TACT. The digital output after AD conversion in the Impact Twin was then fed into the digital input of an M-Audio 24/96 sound card. Audacity was the recording and analyzing software used.<p><a href="
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