I have used the ABX tool for Foobar. I don't use Linux. Besides, I have to switch USB cables and analog output cables to switch between dacs. There are no pots in the system due to their distortion effects. Instead there is one Goldpoint stepped attenuator. Someone would have to adjust the volume level manually.
Not only that but the dynamics of Andrea's dac makes your dac sound compressed. IOW, the perceptual volume level of the dacs changes dynamically with the music. There is no way to do perceptual volume level matching that will hold through a whole song. That's one of that things that makes the dacs sound obviously different.
Maybe I could get someone to swap dacs while everyone else leaves the room. But its simply not needed. The dacs plainly sound different.
EDIT: To make very fine volume level changes I would probably have to use HQ Player with different settings for each dac. I could do all these things if I thought in necessary in this case, but at the moment I am not planning on doing it tomorrow. You will get accurate feedback, but for tomorrow you can take my word for it or not. This assumes I can get people here tomorrow as planned.
Later, if and when the dacs sound closer to the same, then it might make sense to do more formal perceptual testing.
Not only that but the dynamics of Andrea's dac makes your dac sound compressed. IOW, the perceptual volume level of the dacs changes dynamically with the music. There is no way to do perceptual volume level matching that will hold through a whole song. That's one of that things that makes the dacs sound obviously different.
Maybe I could get someone to swap dacs while everyone else leaves the room. But its simply not needed. The dacs plainly sound different.
EDIT: To make very fine volume level changes I would probably have to use HQ Player with different settings for each dac. I could do all these things if I thought in necessary in this case, but at the moment I am not planning on doing it tomorrow. You will get accurate feedback, but for tomorrow you can take my word for it or not. This assumes I can get people here tomorrow as planned.
Later, if and when the dacs sound closer to the same, then it might make sense to do more formal perceptual testing.
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Thanks Marks, good stuff!
But compared to a native one this seems worse off, measurement-wise in some areas?
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An expanding DAC would sound more "dynamic"...
A distorting DAC would sound more dynamic.
Any tests using studio productions may just express how "nice" its sounds but never how correct.
//
A distorting DAC would sound more dynamic.
Any tests using studio productions may just express how "nice" its sounds but never how correct.
//
Wavebourn on what kind of distortion audiophiles do and don't like:
https://www.diyaudio.com/community/threads/euphonic-mechanisms-in-amplifiers.397918/post-7336457
Interesting post, seemed to me. We try to avoid that type of effect around here where I am.
https://www.diyaudio.com/community/threads/euphonic-mechanisms-in-amplifiers.397918/post-7336457
Interesting post, seemed to me. We try to avoid that type of effect around here where I am.
Acko, the two plots shown are just that what You see.. the phase noise plot of the the multiplied up source is 10dB better at 1Hz; ,~6dB better at 10Hz..
What You maybe are hinting at that the lower phase noise plot is getting more "rugged", less clean.
Now, if You have experience with looking at these things, that is again just a second confirmation of the quality of the DUT: at that close-in frequencies the applied instrument itself gets just too close to it's limits and gives a less stable output plot..
Reassuming: at 1Hz, 10Hz it's simply better than the 'native' 22MHz oscillator. At 100Hz, 1kHz they are equal. At 100kHz out - there is a slight advantage for the native oscillator, there one can observe the additive noise of the amplifier stage in the doublers.
What You maybe are hinting at that the lower phase noise plot is getting more "rugged", less clean.
Now, if You have experience with looking at these things, that is again just a second confirmation of the quality of the DUT: at that close-in frequencies the applied instrument itself gets just too close to it's limits and gives a less stable output plot..
Reassuming: at 1Hz, 10Hz it's simply better than the 'native' 22MHz oscillator. At 100Hz, 1kHz they are equal. At 100kHz out - there is a slight advantage for the native oscillator, there one can observe the additive noise of the amplifier stage in the doublers.
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Yes, understood🙏
Just the far out noise that I was looking at. Maybe this has not much effect
Just the far out noise that I was looking at. Maybe this has not much effect
The first one flutters, or stutters, what`s the word in English? The second seems not to and maybe because of that also sounds a tad vague in comparison to the first.
It`s not a small difference.
It`s not a small difference.
Early this morning while half asleep in my bed it dawned on me that I could just use MCK divided by flip flops as DCLK since MCU generates DSD signals from the same MCK.
Some solder-slingering and here are the results. This time only in DoP64.
First 1kHz:
That noise figure looks promising. BTW I switched to ES9822PRO to get to the lowest noisefloor.
Next 60Hz:
Still looking excellent.
Finally SNR:
SNR -116,8dB(A). So not only new PB but new WR and by a wide margin.
Some solder-slingering and here are the results. This time only in DoP64.
First 1kHz:
That noise figure looks promising. BTW I switched to ES9822PRO to get to the lowest noisefloor.
Next 60Hz:
Still looking excellent.
Finally SNR:
SNR -116,8dB(A). So not only new PB but new WR and by a wide margin.
So with a cleaner bit clock, you measure the lowest noise floor of all three of us (Hans, you and me). What does PB and WR mean?
Updated overview of measured noise floors:
-97.5 dB(A) for DSD64, bohrok2610, different board design, DAC + DIY STM32F723 board, MCU-generated bit clock
-116.8 dB(A) for DSD64, bohrok2610, different board design, DAC + DIY STM32F723 board, flip-flop-generated bit clock at twice the normal rate for DSD64
-105.3 dB(A) for DSD128, Hans Polak, DAC + Amanero
-103.3 dB(A) for DSD256, Hans Polak, DAC + Amanero
-95.7 dB(A) for DSD512, Hans Polak, DAC + Amanero
-100.8 dB(A) for DSD512 with my PWM8 algorithm, Hans Polak, DAC + Amanero
-104.6 dB(A) at 27 Mbit/s PWM8, MarcelvdG, DAC + FPGA board and some other logic from an earlier DAC
-97.5 dB(A) for DSD64, bohrok2610, different board design, DAC + DIY STM32F723 board, MCU-generated bit clock
-116.8 dB(A) for DSD64, bohrok2610, different board design, DAC + DIY STM32F723 board, flip-flop-generated bit clock at twice the normal rate for DSD64
-105.3 dB(A) for DSD128, Hans Polak, DAC + Amanero
-103.3 dB(A) for DSD256, Hans Polak, DAC + Amanero
-95.7 dB(A) for DSD512, Hans Polak, DAC + Amanero
-100.8 dB(A) for DSD512 with my PWM8 algorithm, Hans Polak, DAC + Amanero
-104.6 dB(A) at 27 Mbit/s PWM8, MarcelvdG, DAC + FPGA board and some other logic from an earlier DAC
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Sports terminology since this is a competition 🙂 PB=personal best, WR=world record.
One thing worth mentioning is that I only divided MCK by 4 with 2 flip flops. So DCLK is actually 6M for DoP64 (i.e. twice the required 3M). But apparently that does not matter.
With the doubled bit clock, you convert each bit twice and the FIRDAC doesn't suppress idle tones around half the sample rate anymore. That doesn't matter if there is nothing mixing down those idle tones.
Input signals were generated by Multitone Analyzer (SW used). I have no information about them. This is the homepage: https://distortaudio.org/multitone.html
/Martti
/Martti
Martti,
Looking at the pictures showing your gear, you use no reconstruction filter contrary to Marcel’s design but you let the ADC do the filter job, right?
That means that what you measure is the pure S/N from the DAC without added noise from construction filter board.
When comparing mutual results we will have to take that into account.
Hans
Looking at the pictures showing your gear, you use no reconstruction filter contrary to Marcel’s design but you let the ADC do the filter job, right?
That means that what you measure is the pure S/N from the DAC without added noise from construction filter board.
When comparing mutual results we will have to take that into account.
Hans