Only about -88 dB, so 0.004 %, D2 at 1 kHz in the unbalanced case with one DAC output open. Interesting, I would have expected about 0.04 % at 1 kHz, 0 dB DSD based on a calculation similar to the one of post #3903.
Maybe I misplaced a decimal comma/point somewhere. Do you use the normal 2 times 22 uF decoupling of the shift register supplies?
The distortion with everything connected normally is a lot higher than the figures bohrok2610 usually measures. Is the ADC driven too hard maybe? Or could it be a difference between differential and single-ended measurements?
Maybe I misplaced a decimal comma/point somewhere. Do you use the normal 2 times 22 uF decoupling of the shift register supplies?
The distortion with everything connected normally is a lot higher than the figures bohrok2610 usually measures. Is the ADC driven too hard maybe? Or could it be a difference between differential and single-ended measurements?
The first measurements has 6dB higher level and it looks like ADC is driven too hard. Many soundcard ADCs have steep rise in distortion above -6dBFS.
A couple of questions while I'm planning, as I would like to be involved in some modest manner.
Would a real high quality ADC like the one found in (eg) a Neumann MT48 work well as an analysis tool?
I will probably never have anything near to what someone like @Markw4 has available for listening testing. Not to mention ears and experience, I'm sure. I'm also likely to be one of those digital nomads, at least for the time being.
So I'm looking at headphones and a good amp as an alternative to high end monitoring and room treatment. The Neumann has apparently one of the very best amps. Would a pair of Dan Clark E3 work well for this application?
I'm looking at a one-time spend and want to try to get it correct the first time around. Any suggestions for high end equipment that will also be able to be used in a mastering context would be super helpful 🙂
The application is doing analog summing of stems, using a number of DSD DACs to feed the summing network. This will not be a passive summing solution, rather feeding a modified, fully-analog, good sounding board.
Would a real high quality ADC like the one found in (eg) a Neumann MT48 work well as an analysis tool?
I will probably never have anything near to what someone like @Markw4 has available for listening testing. Not to mention ears and experience, I'm sure. I'm also likely to be one of those digital nomads, at least for the time being.
So I'm looking at headphones and a good amp as an alternative to high end monitoring and room treatment. The Neumann has apparently one of the very best amps. Would a pair of Dan Clark E3 work well for this application?
I'm looking at a one-time spend and want to try to get it correct the first time around. Any suggestions for high end equipment that will also be able to be used in a mastering context would be super helpful 🙂
The application is doing analog summing of stems, using a number of DSD DACs to feed the summing network. This will not be a passive summing solution, rather feeding a modified, fully-analog, good sounding board.
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Guilty as charged! 😱 Yes, you're both absolutely right, In my haste I'd forgotten about the higher level and the ADC must've been clipping. I will redo that measurement in due course. The second two measurements should still be valid though.
Since in the measurements of post #3920 the differential (original) configuration has 6dB higher level I assume in the 2 SE measurements the filter output was taken of only 1 leg of differential output. This muddles the results as another variable is added (filter stage SE vs differential). Markw4's claim was that DAC SE output sounds better. If this claim is valid the difference should be similarily audible with both filter SE and differential output. With differential output the level remains the same as in the original which makes comparisons easier.
So my suggestion is to first try DAC SE output using filter stage differential output. If that sounds better than the original then it is more likely that DAC SE output is preferred. If the DAC SE output sounds better only with filter stage SE output then the preference is more likely due to using filter stage in SE mode.
So my suggestion is to first try DAC SE output using filter stage differential output. If that sounds better than the original then it is more likely that DAC SE output is preferred. If the DAC SE output sounds better only with filter stage SE output then the preference is more likely due to using filter stage in SE mode.
Just to be clear, I don't use either of the output stage boards that LTK and or Cestrian may be using. IIUC, LTK has tried a couple of different boards, not sure if its the same for Cestrian.
For myself, I prefer SE over balanced transformer outputs, where the transformers are proprietary prototypes manufactured by a well-known audio transformer company.
For myself, I prefer SE over balanced transformer outputs, where the transformers are proprietary prototypes manufactured by a well-known audio transformer company.
The calculation I did yesterday was incorrect. I forgot to take into account that the negative side of the DAC partly compensates for the signal current drawn from the reference, even when it is unloaded (except for the first filter capacitor that's on the DAC board). To be updated later.
I think both the single-ended and the differential OPA1632 output levels halve when you disconnect one filter input and terminate it with a resistor to ground.
I had to try this SE connection, so I modified the 1632 board by letting R6 (249 Ohm) go to ground instead of to - out from DAC and then placed a capacitor in series with the output from the filter. Now my preferences changed from the 2210-1678 board to the 1632. Now 1632 was the board with the more precise instruments and depth of field even though it was still a little softer, but as all the details are there it is just more pleasant to listen to.
I went a step further, partly because I do not like DC blocking capacitors in the signal path and partly because of the big thumb when starting playback. I placed the capacitor after the relay, so the every time the relay changed status, there was a huge DC shift on the capacitor.
So instead I placed a capacitor from the -in from the DAC to gnd, there by maintaining the same DC on the noninverting and inverting input on the filterboard and at the same time only the + signal from the DAC was present on the filterboard. This way there are the same load on the + and - outputs from the DAC.
If it is better to have the - output unloaded , just place a 100 K resistor in series with the - out before the capacitor to ground. Haven´t tried that yet.
I was wondering what kind of source material people uses when evaluating sound , so I want to contribute with two tracks from Schnittke´s Gogol Suite.
I worked , in the late ´90, as consultant for Pope Music (long time defunct now) modifying their Nagra D so I knew Gene Pope and how he made the his recordings.
He used two Bruel and Kjær microphones pretty close together and the Nagra D. The recording sessions were in Russia with russian orchestras.
The attachment is a link to a dropbox folder with the two tracks. They are examples of music in an acoustic space without any processing at all.
I went a step further, partly because I do not like DC blocking capacitors in the signal path and partly because of the big thumb when starting playback. I placed the capacitor after the relay, so the every time the relay changed status, there was a huge DC shift on the capacitor.
So instead I placed a capacitor from the -in from the DAC to gnd, there by maintaining the same DC on the noninverting and inverting input on the filterboard and at the same time only the + signal from the DAC was present on the filterboard. This way there are the same load on the + and - outputs from the DAC.
If it is better to have the - output unloaded , just place a 100 K resistor in series with the - out before the capacitor to ground. Haven´t tried that yet.
I was wondering what kind of source material people uses when evaluating sound , so I want to contribute with two tracks from Schnittke´s Gogol Suite.
I worked , in the late ´90, as consultant for Pope Music (long time defunct now) modifying their Nagra D so I knew Gene Pope and how he made the his recordings.
He used two Bruel and Kjær microphones pretty close together and the Nagra D. The recording sessions were in Russia with russian orchestras.
The attachment is a link to a dropbox folder with the two tracks. They are examples of music in an acoustic space without any processing at all.
Updated estimated values for the second-order distortion with only one DAC output connected to the filter, other filter input grounded via a resistor:
At 0 dB DSD, worst-case frequency (somewhere around 240 Hz to 300 Hz): about 0.051 %
At 0 dB DSD, 1 kHz: about 0.019 % to 0.028 %
It still doesn't match the measured 0.004 % at 1 kHz at all.
Circuit number three from post 2471, https://www.diyaudio.com/community/threads/return-to-zero-shift-register-firdac.379406/post-7635719 , is different from the circuit of post #3932, https://www.diyaudio.com/community/threads/return-to-zero-shift-register-firdac.379406/post-7870284 Which is the preferred one?
Interesting music; very dynamic. Wondering if that avoidance of processing included a lack of dither? Reason I ask is there a sort of a gritty/fuzzy low level distortion at least to my ears.
Also, kind of sounds like the mics were placed maybe a bit far back in the room? Its like the ratio of room sound to direct sound seems to favor more of the room, or something like that.
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