Soundcard performace .......FFT pics and problems.

[A 8]

Hi Ojg,

You have made some very good progress in minimizing the noise-level of this ADC! However in my experience, the tricky part is minimizing THD. I think you will see that when you start measuring sinewaves with amplitudes close to full scale that the linearity of the ADC becomes the limiting factor.

Thx, the improvements are really surpricing and you are completely right about THD but I gave up on it @ full scale for this soundcard. It stops at around -90dB whatever I do.
This was basically the only parameter I could change and I wanted to measure my diy dacs and poweramps ability to resolve really low level data (beat the -122dB reading I showed you in another thread)

This thread have got me interested in buying an external soundcard. Does anyone have an overview of what ADC/DAC chips the different USB/Firewire soundcards use?
Let's ask Google first:
Creative Labs Audigy2 NX: Sigmatel STAC9460
M-Audio Sonica Theater: AKM4529
M-Audio Firewire 410: AKM AK4381, AKM AK5380, AKM AK4355
M-Audio Firewire Audiophile: AKM AK4628
Terratec Aureon 7.1 Firewire: WM8770

Another aspect in considering these cards are digital processing and DA conversion, my creative lab does a really crappy job at this and it shows as heavy frequency ripple at analog AND spdif out.

/Michael

[A 8]

I dusted off an old ACKDAC by HoltMods using a crystal 18bit AD and connected it through spdif to the same soundcard, shorting the analog input at the ACKDAC and get this...

[gigatron]

I have an internal sound blaster audigy 2 oem (using hacked audigy 2 zs drivers).

What kind of things can be done to this card, it's internal, to improve it.

[jewilson]

I don't believe your performance can get any better.

You test equipment and setup maybe misleading you.

[ojg]

Shorting the inputs to the ADC is a little bit like cheating

The proper way of doing this is applying an analog 1kHz sinewave at -60dBFS to the ADC. This signal is small enough that the noise will dominate over the distortion. Then measure the THD+N in spectralab and add 60dB to that number. This gives you the dynamic range.

To generate a -60dB signal, it is best to have your DAC output close to 0dB and then attenuate with a potmeter. This way you attenuate both the siganl and noise coming from the DAC and so you take the DAC performance out of the equation.

[jewilson]

A8,

ojg is correct, about the method to get S/N. Shorting the input act like you just getting the noise from the final amp out stage. In fact some converter stops clocking when the inputs are shorted, it a trick they use. So to get this kind of performance from an oscillator is very difficult indeed. I mean you will need a super oscillator a precession pot super cable that short and then turn off all light.

After all when you take the fundamental test frequency and run it through a distortion analyzer, the analyzer take the signal through a notch filter what left over is distortion- or noise. To measure this kind of noise levels you will a distortion analyzer and a spectrum analyzer or just one dam good audio spectrum analyzer. So if you can get the real S/N down to 110-115dB that great. The fact is that most converters don’t have true 24 bit operation or linearity.

[A 8]

Ojg, jewilson, I agree fully but there is more to it.

I started off trying to loose the spikes at 5 and 15kHz but to be able to see them clearly I had to let spectralab do heavy averaging.

I was to focused on the spikes to think about the consequences of heavy averaging, it cancels random noise to almost nothing and that is why the results get so good. If I use peak hold it is more in line with the specs at around -124dB with no signal and I am sure it gets worse if I would measure with a signal as you suggest.

The thread was to show the spikes and try to understand if that is as good as an adc gets or if I had particular problems due to the spikes that could be fixed.

Anyone know if there is some form of standard doing these type of measurements, i.e is it peakhold or should one apply some amount of averaging?

[ojg]

I think neither peakhold nor averaging will tell you the real DNR number. When I think about it FFT is probably not suitable for this measurement at all, because the FFT itself will also be averaging the samples. I.e. a 32768 length FFT averages over 32768 samples.

Here's what I would do: Record a few seconds worth of your -60dB signal at 24 bits into Cooledit or similiar program. Then use Cooledits filter function to notch out the 1kHz fundamental. Now select the statistics-menu and it will tell you the RMS value of the signal. This will be your DNR number.

I don't know if this is the standard way of doing it, but it comes close to how it would be done in the analog domain.

[JTT]

I don't want to hijack this thread, but with so many soundcard enthusiasts on board, someone may be able to provide me some advice.

First, considering the noise problem, you might look at the spec and reviews for the Echo Indigo IO PCMCIA soundcard (http://www.echoindigo.com/) - tho' it aint exactly cheap.

Now the question: I'd like to buy one of these for use with my laptop, mainly for LS testing, but I'd also like to use it with my desktop. Problem is, no one in the UK seems to offer an affordable PCI to PCMCIA adaptor card for the desktop, except for those offered for use with wireless LAN cards. I suspect that these cards would work OK as general purpose PCI - PCMCIA adaptors (they mostly use the same type of RICOH chip), but I'm not sure, and suppliers seem very leery about offering any confident info.

Anyone come across this problem, and a solution?

JTT