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D/A preamp FFT measurements for review plz

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Finally was able to get some decent measurements of the distortion of my equipment. Up to this point I had a Tek scope with FFT, but the noise floor was around -55 dB and was only able to detect gross problems. Now I am using a Focusrite Saffire FW box with Arta and Audiotester. Haven't decided which package to purchase yet; still under probation.

At least all the measurements I have obtained with both softwares are identical, so I believe my procedure to be reasonably correct. Now, I would like to know if my results are good/bad/average.

My preamp schematic is attached. Not shown is the K&K Audio Mark II DAC, with SPDIF input and differential current output PCM1794. The I/V is shown feeding the differential gainstage.

Three FFT's are attached. No calibrations were performed. First one is a loopback using soundcard analog out, just to see how the output looks. Second is output of the entire preamp at max volume. Third is output of the D/A taken at the grids, full volume.

Questions are hopefully easy to answer (sorry for the scaling differences):

1. For the loopback, what would be the expected cause for the peaks at 2.5k, 2.7k and 5.5k ?

2. Would you agree that the majority of distortion is introduced at (or before) the I/V stage ? Except maybe a little 3rd is introduced, typical of the differential stage.

3. Is it reasonable to expect a cleaner signal at the I/V? I can't know if the problem is the SPDIF source or D/A.

4. Results good/bad/average/useless ?
 

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The loopback looks very noisy to start with. The broadening of the fundamental on the plot is probably an artifact and suggests that you need to optimize sample rate versus number of points or the exact frequency of the test signal, as well as apodization. Get that one right before you start measuring external devices. Ritemark may be useful software for getting the measurement system in order.
 
There was a number of settings in Arta, where I just followed the instructions, but some of them were not explained very well. ASIO buffer size and F/W buffer size in msec. Biggest improvement was in selecting Kaiser5 over Hann, also per Arta.

FFT length for the screenshot was 16K at 96 kHz. Averages 64. Turning the averages down did increase the noise floor a bit, but not like it used to be.

I verified my output signal is not distorted, then turned it down 2-3 dB to make sure it was running clean. Input volume turned up just before entering the red zone, but definitely before the input circuit clips. Went through the voltage calibration with a Fluke 87, but this doesn't work well in Arta. Close enough, but not really accurate.

I am a little suspicious of the sample rate. My Saffire control panel is set for 44.1 kHz, and I am able to select 96 kHz through Arta, but I don't hear the internal relays clicking in the soundcard when this is done. When you select 96 kHz through the Saffire control panel, you hear the relays clicking. However, when I start up Arta it forces the soundcard back to 44.1. So I'll to mess around with it some more.
 
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There was a number of settings in Arta, where I just followed the instructions, but some of them were not explained very well. ASIO buffer size and F/W buffer size in msec. Biggest improvement was in selecting Kaiser5 over Hann, also per Arta.

FFT length for the screenshot was 16K at 96 kHz. Averages 64. Turning the averages down did increase the noise floor a bit, but not like it used to be.

I verified my output signal is not distorted, then turned it down 2-3 dB to make sure it was running clean. Input volume turned up just before entering the red zone, but definitely before the input circuit clips. Went through the voltage calibration with a Fluke 87, but this doesn't work well in Arta. Close enough, but not really accurate.

I am a little suspicious of the sample rate. My Saffire control panel is set for 44.1 kHz, and I am able to select 96 kHz through Arta, but I don't hear the internal relays clicking in the soundcard when this is done. When you select 96 kHz through the Saffire control panel, you hear the relays clicking. However, when I start up Arta it forces the soundcard back to 44.1. So I'll to mess around with it some more.

You need to correct for the equivalent noise BW of the windowing function as well. This can be as much as several dB. High order windows being the worst (in general). When this is done your true RMS meter and sound card will agree (as long as you brick wall the input to the meter at Fs/2).
 
What are the options for number of points? 16k points at 96kHz sampling is probably good enough for doing things like amplifier distortion, but if you want to get in tight for looking at low level sidebands, you'll want to use a larger number of points. 64 is a lot of averages, especially when the sample record gets longer- remember that the noise goes as the square root of number of averages, so you won't see night-and-day differences with only 16 averages. Not that more averages will hurt, it's just that it can be pretty time consuming.

Does ARTA also give you the option of Blackman-Harris apodization?
 
Scott:
Sorry, but you're losing me. Although I am an engineer and passed advanced math in college with a B, I dropped out of communications systems because it was worse than Greek to me. So a lot of terminology with algorithms and noise is above me. How does a windowing function affect measuring the output voltage of a soundcard? I would think the high frequency noise present from the sampling rate would be essentially unmeasurable by a multimeter.

Sy:
I can go from 4K to 128K. Blackman-Harris 3 or 4 (whatever that means).

This is a 128K BH4 with 4 averages. I confirmed I am running at 96kHz.

At the end of the day, I want to measure my preamps/amps distortion, maybe IMD and simple speaker measurements (tweeter polarity, room correction). Eventually, I will output SPDIF to my D/A, so is the assumption this will be a perfect sine wave for measurements?
 

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Scott:
Sorry, but you're losing me. Although I am an engineer and passed advanced math in college with a B, I dropped out of communications systems because it was worse than Greek to me. So a lot of terminology with algorithms and noise is above me. How does a windowing function affect measuring the output voltage of a soundcard? I would think the high frequency noise present from the sampling rate would be essentially unmeasurable by a multimeter.

Sy:
I can go from 4K to 128K. Blackman-Harris 3 or 4 (whatever that means).

This is a 128K BH4 with 4 averages. I confirmed I am running at 96kHz.

At the end of the day, I want to measure my preamps/amps distortion, maybe IMD and simple speaker measurements (tweeter polarity, room correction). Eventually, I will output SPDIF to my D/A, so is the assumption this will be a perfect sine wave for measurements?

I thought you wanted to measure the noise floor, the tones are not affected in the same manner.
 
So I end up with the 3rd at about 80dB down. I see at the I/V stage about 90dB down from the D/A alone. A little higher distortion than I was hoping for, but maybe I'm asking too much. What do you think about this?

These are with a 96 kHz SPDIF input into D/A.
 

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I agree with SY, you should be getting better results.
For a comparison, below is my M-Audio Fast Track Pro. This is an unbalanced loop back. 48KHz, 24 bit. Window Blackman4, 16K bins, 4 linear averages. H3 is at -104dB

The best I see noise on mine is about -135 using 131K bins.
 

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I don't know that my loopback is all that different from yours, Pano albeit maybe a little more distortion on the higher harmonics. My issue comes in when I drive my D/A with an SPDIF source.

If I have the calculations about right, 6n6p at 2.2k into a 9+9:2 transformer is (2.2*2)/9= 500 ohms, give or take. Input impedance of the Saffire is not specified, I'll guess at 20k balanced. Shouldn't be a problem.

I'll see what happens when I decrease the passive I/V resistor. To my eyes there is unacceptable distortion coming out of the D/A before it ever hits a grid. 6n6p may have high 3rd too, which is summed with the differential topology.
 
You're probably seeing the distortion of that circuit. If you expand the x axis around the fundamental, you should be able to see any power supply intermodulation effects. Move the test frequency around a bit to see how your transformer is working (especially in the bass and extreme treble). If you tune the cathode current, you'll be able to make that spectrum change- it's interesting to see how that happens with a given circuit, and it gives you a visual way of "tuning" the circuit.
 
Another screen shot. This time I injected an analog signal at the grids and monitored the output. Distortion from the diff stage, therefore. This is what you were referring to?

Yes, I can definitely see IM around the fundamental due to power supply. I don't know what you mean by "to see how your transformer is working", though. You talking about the power tranny or OPT?

I might have identified my next 6 month project: experiment with tubes and topology to find a lower distortion preamp.

Still, what about the D/A distortion? I would expect a clean voltage at the I/V. Further, where in the world is my even harmonic distortion coming from? I thought the diff stage put that almost into the noise floor.

Thanks for your help, too. I've done enough building and listening, now I want to start to understand if I have the performance desired. Just the next step in the evolution of a DIY nutcase.
 

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