yes, I see it, you are right, and I think it uses for non-audio purposes(FPGA)
slightly left from the SIS570 you can see the 24M clock that uses for audio directly. probably from the downside of the board, you can find 22M also.
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@alecm
I think you are right, the DAC switches are fed by a pair of fixed oscillators at 22 and 24 MHz.
Moreover Nicolae of Rockna has bought a pair of crystals from me and he told me that he is interested on a gear to measure the phase noise.
Therefore I think that the Si570 is used for other functions, at least I hope so.
@TNT
There is not a question of sub-conscience, several Ian's FIFO users have experimented changing the oscillators and the Si570 was early abandoned because it performs worse than the Crystek (that's a poor oscillator)
I think you are right, the DAC switches are fed by a pair of fixed oscillators at 22 and 24 MHz.
Moreover Nicolae of Rockna has bought a pair of crystals from me and he told me that he is interested on a gear to measure the phase noise.
Therefore I think that the Si570 is used for other functions, at least I hope so.
@TNT
There is not a question of sub-conscience, several Ian's FIFO users have experimented changing the oscillators and the Si570 was early abandoned because it performs worse than the Crystek (that's a poor oscillator)
I think anyone in this game should realise that there is a lot of mind tweaking going on...
Strange that they choose an adjustable clock for the Ian board. How did they fix the i2c aspects?
I own an Ian Fifo buffer system and run selected NDK clock on them so I'm reasonably familiar with that gear.
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Strange that they choose an adjustable clock for the Ian board. How did they fix the i2c aspects?
I own an Ian Fifo buffer system and run selected NDK clock on them so I'm reasonably familiar with that gear.
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I think it's perfectly fine to try to optimize everything (and not insist on blind tests) if DIY is a hobby. If one tries to be very rational about this whole business, there is probably very little middle ground for audiophilic guesswork. Thoughtful engineering on Soren's part is still very valuable if one takes an interest in R2R for any reason. I thought I had a good reason to want R2R, not sure about it now. But I personally think it's also naive and foolish to dismiss the audiophile scene as "snakeoil" - no amount of training in science or engineering, nor intelligence in general makes us insusceptible to this type of influence. Perhaps gay men obsess less in general.
I don't know how important variable phase noise is, but compared to nanosecond level propagation time and rise/fall time deviations at the shift registers, I don't see how it should freak anyone out if we find the sound even close to acceptable as it is now. Not to mention the fact that it doesn't show up substantially on any of the standard measurements used to evaluate a DAC of any kind.
I don't know how important variable phase noise is, but compared to nanosecond level propagation time and rise/fall time deviations at the shift registers, I don't see how it should freak anyone out if we find the sound even close to acceptable as it is now. Not to mention the fact that it doesn't show up substantially on any of the standard measurements used to evaluate a DAC of any kind.
The problem is not loading the shift registers, for this job the phase noise (or the jitter if you prefer) does not matter.
Indeed, you can even load the registers quickly and then stop the bit clock (the best way to avoid interference).
The timing problem is related to the latch clock, the precise moment when the DAC switches change.
This clock is the most crucial in a R2R DAC (1704 aside), because even the slightest phase (or time) error during switching affects the quality of the digital to analog conversion.
You can try yourself, get a DAC with fixed oscillators (or a asynchronous FIFO) and then compare a few oscillators with different close in phase noise.
You will be very surprised with the result.
Most likely Rockna use fixed clocks where the DAC is the source, like USB input, and the programmable Si570 oscillator where the DAC need to track the source, like SPDIF....
This prompted me to finally try out the digital attenuator
The system where the 1021 lives has a nice tube pre feeding two SS power amps with more than 100k combined input impedance.
1. Bypassed the preamp and fed the 1021 raw output directly to the power amps.
Listened to this over 3 hours. Obviously clearer and less distorted than through the preamp, but with less guts and dynamics. Stereo images more diffuse. By the third hour I have had enough and longed for my beloved pre. Some sounds perhaps sounded better, but the majority of music did not.
2. Put the preamp back, but disabled the Shallco attenuator, which consists of a series z-foil and non magnetic shunt Holcos. Volume is controlled by the 1021.
Hard to believe but to me the digital attenuator of the DAM sounds better than the Shallco. Clearer, more detailed, less compressed. On top of that the volume position is further reduced by the preamp gain of about 9db so bits are getting eaten
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Thanks, understandable and logic. You like a buffer in between source and poweramp.
For me it was the opposite but I assume that is very individual. Like the DAM feeding poweramp direct.
Using the digital volume control in a good way beats probably every analog attenuator soundwise. Many believe it is not like that (I was one earlier) (tried some LDR, ladder attenuator with fixed resistors, Noble pots and TDK pots.) They are all being put to rest in a bag.
For me it was the opposite but I assume that is very individual. Like the DAM feeding poweramp direct.
Using the digital volume control in a good way beats probably every analog attenuator soundwise. Many believe it is not like that (I was one earlier) (tried some LDR, ladder attenuator with fixed resistors, Noble pots and TDK pots.) They are all being put to rest in a bag.
I do. Bothers me a bit as I don't understand the need for it.
In my second system a low distortion ss preamp performs the same function, so it's not just tube second harmonic, microphony and transformer core saturation... I also find an active pre more essential for digital sources as if the subtle hf filtration somehow improves musicality.
Your observations do not surprise me. The topic before was analog volume control. I do not exclude that a buffer could improve audio quality in certain configurations. Though it would need to be a very high quality one and preferably with no gain unless it is required to get enough output level. I would think it is best if such a stage would be in the same case as the DAC.
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Some amplifiers produce elevated common-mode distortion when driven by higher source impedances according to how their input stages are designed. Still, the 625 ohms of the DAM isn't exactly high.
tablet as source?
If I understand your question, here are two examples with Android and tablet or smartphone
Case 1:
Smartphone (TIDAL)-->USB/OTG-->cable USB-->I2SoverUSB v.III(Or Amanero)-->Soekris/DAM1021-->out analog
Case 2:
Smartphone (USB/PLAYER PRO)-->USB/OTG-->cable USB-->I2SoverUSB v.III(Or Amanero)-->Soekris/DAM1021-->out analog
If I understand your question, here are two examples with Android and tablet or smartphone
Case 1:
Smartphone (TIDAL)-->USB/OTG-->cable USB-->I2SoverUSB v.III(Or Amanero)-->Soekris/DAM1021-->out analog
Case 2:
Smartphone (USB/PLAYER PRO)-->USB/OTG-->cable USB-->I2SoverUSB v.III(Or Amanero)-->Soekris/DAM1021-->out analog
I was wondering if someone could help me and point out what's wrong with my dam1021?
This is one of the first boards, about 6 years old, and from one day to the next the audio started to distort.I recently updated from 0.99 to 1.06 and everything was ok. But after turning it on some days later the distortion started. I set it back to 0.99, no change. Then I updated to 1.21, same distorted sound. So I guess the updating is not the problem? But I am not sure.Is there "default" setting to which I could set back everything in case there is a bug in the FPGA?
Then I thought it might be a problem with the isolated input/I2S connection (I guess this is a switch which needs +3.3V to turn it on?). It is now always open with or without connecting +3.3V and measures +2.3V on the iso +3.3V pin if I disconnect the external psu with 3.3V. That doesn't make sense to me.
This is one of the first boards, about 6 years old, and from one day to the next the audio started to distort.I recently updated from 0.99 to 1.06 and everything was ok. But after turning it on some days later the distortion started. I set it back to 0.99, no change. Then I updated to 1.21, same distorted sound. So I guess the updating is not the problem? But I am not sure.Is there "default" setting to which I could set back everything in case there is a bug in the FPGA?
Then I thought it might be a problem with the isolated input/I2S connection (I guess this is a switch which needs +3.3V to turn it on?). It is now always open with or without connecting +3.3V and measures +2.3V on the iso +3.3V pin if I disconnect the external psu with 3.3V. That doesn't make sense to me.
I cleared up the grounding of my build some more. Correct grounding and good PSU quality is paramount for the DAC to work as it is supposed to.
Now the difference between the RME internal clock and it syncing to an external clock has become even more obvious. The former provides lower jitter and is probably closer to an accurate clock in terms of deviation from delivering precisely 44,1 khz.
The difference is most noticeably in the low end, where the internal clock sounds tighter and seems to go deeper. It also delivers an overall more real, tangible, 3D representation.
I wonder if the less accurate clock necessitates more corrections for the DAM1021 clock to make. This might increase jitter because of settling times, but it might also continually affect the phase relationship and thus (untechnically speaking) the dynamic integrity of the signal.
Anyway, I'd like to reiterate that I hope Soeren can fix this issue ASAP, as for me it's the only thing that seriously hinders what is otherwise stellar performance.