The output directly from the resistor ladder is single ended, hence you'll need two boards for "raw" balanced operation.
Why the sound at SE is better than Balanced, is probably because you have one additional stage. And as usual, less is better - there is of course always addition distortion in each extra step.
I am that slow
I have timed out after those CCCC characters showing up... isn't thata the DAM?
Start sending file using XMODEM/CRC protocol.
CCCCCCCCCCCCCCCCCCC
//
And the negative feedback in opamps is a plague on humanity, people don't realize how bad they are till they're gone.
Yes, i totally agree.. they give low distortion, but always found op-amp sound boring. A well designed discreete amplifier stage sounds alot more alive, and shouldn't give much more distortion either.
There's nothing superior about balanced. In fact in most cases it will be inferior as the signal will be passed through some opamps to generate the balanced out. With this DAC the unbuffered single ended output is the purest as long as you feed it to a reasonably high Z load.
Post #1758 by spzzzzkt
spzzzzkt! Your post was off topic, tasteless and provocative. It breaks diyAudio Rules # 1, 2 (Note 1) and is a perfect example of Trolling.
http://www.diyaudio.com/forums/site-announcements/167561-diyaudio-rules.html
spzzzzkt! Your post was off topic, tasteless and provocative. It breaks diyAudio Rules # 1, 2 (Note 1) and is a perfect example of Trolling.
http://www.diyaudio.com/forums/site-announcements/167561-diyaudio-rules.html
Given that spzzzzkt has been a very active and positive poster on this thread, I'd say that his post was anything but a good example of trolling. I know I thought it was both humorous and to the point. You were the one that made the "rubber meets the road" metaphor, and his post was an honest reaction to that. Lighten up!
Thanks Bob. I think I was thrown by the reference to J3 and didn't put two and two together.
Ray
Regarding the bridge rectifier, my philosophy is simply to not have components that aren't required to perform a function and with a DC input the bridge isn't required. It may or may not have an affect but without it who cares.
Regarding the BBB/Botic, have you looked below the surface of this yet? I already run this combo with a Buffalo IIIse DAC/Legato, using an Acko SO3 isolator/reclocker, and the results are excellent.
My questions with using a BBB with the Soekris DAC are around the clocking of the BBB; it seems obvious to turn-off the BBB's onboard clock and provide external clock signals to it as it avoids resampling of 44.1KHz data to 48KHz. I assume the Soekris DAC will accept the frequency select input from the BBB/Botic but am unsure whether the BBB/Botic will accept the MCKL output from the Soekris because the Botic software seems to assume discrete fixed frequency crystals. Any thoughts?
Ray
Does this mean that a download process will terminate if it lasts longer than 30 sec.? Because my download took way longer than that.
Can Søren confirm this?
Your 5 seconds really makes me wonder why download times vary that much? I had no other applications running other than PuTTy (+ of course various services running in the background) and certainly no other communication on the USB port. My system is W8.1 intel CPU and chipset.
Hi Paul,
Great work!
Have you tried implementing the so-called "apodizing" reconstruction filter that was all the rage among audiophiles a few years back?
The idea is to "hide" the time-domain effects of the successive reconstruction filters that the signal might have encountered during its production (ADC, resampling, etc.).
So as those filters must have a knee (and associated (pre)riging) around 22050Hz the idea is to be already down quite low (-20dB maybe, or even less?) at that frequency, and of course only use minimum-phase filters (at that frequency point at least).
Of course this implies cutting edges on the response up high, but this makes sens to use with "bad" recordings using bad filters (like the ones either recorded or "remastered" in beginning of the digital era...).
I think it would be best if Soren could implement a "preset" system, similar to the ones we see in some commercial DACs, to let the user choose and change filters almost "on the fly".
Of course the DAC needs to have a "flat" filter (within 0.5dB at 20kHz), and then other more shallow filters.
Regarding the 48kHz filters, you can directly use the 44.kHz one (I mean the generated coefficient, and let the DAC to the convolution at a higher rate), and same goes for 96 vs 88.2, etc.
Oh and by the way, please write either "pos" or "Pos" instead of "POS", as this one really sounds like the well known infamous acronym...
Hi pos,
Sorry, I've got used to typing three letter abbreviations as all caps... I was wondering about the name - and if you were a Dame Edna fan... pos being an abbreviation of possum ( https://www.youtube.com/watch?v=Dsp7fqj2RsA )
I purchased Peter Craven's AES paper on anti-aliasing yesterday and have a read through but need to have a proper look to understand what he's doing. It seems one of the key aspects is a null at 22050hz. I got a bit waylaid with octave's filter design tools - but that clearly has limits and a deeper understanding of dsp than I possess.
Søren has mentioned he plans for user selectable presets so I think something like you suggest is in the works. It would definitely make comparative listening a bit more straight forward.
Thanks for the tip on the 48kHz filters.
cheers
Paul
Hi Soekris,
1. You have already mentioned “then to +-4V by precision low noise medium current opamps”; “-4V reference is sent though an inverter with 0.01% resistors generating the +4 reference”.
What the low noise medium current opamps are you used?
2. Why you used opamps instead a simple resistive divider, for example?
1. You have already mentioned “then to +-4V by precision low noise medium current opamps”; “-4V reference is sent though an inverter with 0.01% resistors generating the +4 reference”.
What the low noise medium current opamps are you used?
2. Why you used opamps instead a simple resistive divider, for example?
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Filter downloads should be under 5 seconds at the moment, but will get longer as more, and bigger filters are added.
Firmware downloads take far longer, and I didn't see a time out issue.
I think the timeout is an issue while the DAM is waiting for input. I was searching around looking for a filter to upload today and ended up with a full line of C's and still didn't get a time out.
These were more usual with the current crop...
Code:
### Send (X) 1021LPSROv2.skr: 10752 bytes, 0:02 elapsed, 3604 cps, 31%
File downloaded and programmed, size 010752 Bytes.
Code:
# download
Start sending file using XMODEM/CRC protocol.
CCCCCC
### Send (X) MB1.skr: 10752 bytes, 0:03 elapsed, 3057 cps, 26%
File downloaded and programmed, size 010752 Bytes.What I've found is that using either Serial or ZTerm the DAC is a bit slow to respond after you exit from uManager. I've been switching between inputs to trigger a reload of the filters, and the DAC will consistently not respond to the first attempt to change inputs. The next attempt always works as expected. I think this might be a serial port wakeup issue on the DAC board.
Between Zterm and Serial main difference is that Serial deals with disconnecting the serial connection far better. You can replug and continue on with Serial, but need to restart ZTerm to re-establish a connection. If you leave both plugged in there is no difference, and ZTerm's ability to set a keyboard short cut for uploads wins hands down, as does the reporting of the uploaded file name.
Haha, that one is even worse
What does he call a null exactly?
I was under the impression that the attenuation needed to be high enough at fs/2, but that something like -40dB would be enough, or even less (-20dB is already quite low), as it was only there to mask an already subtle artifact.
There is no evidence that this ringing will occur exactly at fs/2 either, depending on the designs of the filters used in the production process.
So in the end all that remains is the idea of attenuating the signal before any other LP filter that might have occurred during production, and do it with shallow (less total ringing) and minimum-phase (only post ringing) filters.
Of course not, it terminate after 30 seconds of no valid activity on serial line....
If I understand the TP system, I think the Chronos board is designed to work with their cape and provide the clocking. But then, what (if any) synch happens with the DAM is not clear to me. I must admit to not understanding much of the basics involved in this--my DIY projects (at least in digital audio) have been the result of others' knowledge and sharing.
Alan