Modular multi channel DIY AD/DA Box

GroupDIY Audio Forum

Help Support GroupDIY Audio Forum:

This site may earn a commission from merchant affiliate links, including eBay, Amazon, and others.
Rochey said:
OPA1612 is the way to go.

I'm liking this one for this job since there's specific mention in the datasheet for use as a D/A buffer.

audiovisceral said:
Personally not a big fan of the Burr Browns. I'd prefer Analog Devices opamps like what Raphael mentioned. AD8620, AD8599, AD823, etc.

But I doubt we'd ever all agree on opamps.

So true! Ideally I'd like to get a few different BB and AD chips to do some A/B comparisons. The multi channel nature of this project will make A/B comparisons *so* easy.

-Dave
 
Yeah Raph, I really appreciate u sharing this with us. Id really be in for 24 channels, but would like a presolderd DAC, if it would be possible. Anyhow thank again.
 
snipsnip said:
Congratulations!

How do they sound?

Well, they do not sound. :D They simply play what you send to it. ;)
To be honest, I find it a bit hard to describe how they sound. The only thing I can compare to is my Presonus Firebox. I could say that my DAC sounds better then the Firebox but that would not be fair because I think if you build something yourself you'll always find that your self-made unit sounds better because you made it. ;)

Raphael
 

Attachments

  • DSCN3375.JPG
    DSCN3375.JPG
    239.1 KB · Views: 452
Haha. You know what  I mean!

Would be really great to hear some shootouts. Are these likely to match up to the quality of SSL, Lynx, Apogee etc?

 
Yes I somewhat agree but it may become a sort of summing box issue again where tone/colour v's pristine audio becomes a personal thing and there ends up being no right answer. Some ears may prefer a DOA coloured converter and others will find their ears rather clear as possible leaving their pre-converter gear to do their thing.
 
I've contemplated the idea of discrete converters a bit in the past and I think there may be better places to integrate DOA's. 

DOAs are big and expensive so I think it makes sense to put them where they will have the most audible effect. Technically perhaps this may be the ADC/DAC, since everything goes through them. But if the opamps are hardly applying any gain in the circuit, an IC might be near perfectly transparent, and you might be better off saving that discrete $$$ for use in the pres, comps, and perhaps eq's.

I don't know how much gain the opamps will be pushing through a circuit like this, so I don't know if it's even useful to think of it in these terms. But that's the first thought that runs through my head on the subject.

Also, remember this is surface mount, so even if you got some nice dual DIP-8 discrete opamps like Audio-gd OPA-Sun, you'd also have to buy SOIC-DIP adapters and hope the whole Frankenstein holds together without falling off the board. Price would be >$40 per opamp, or literally thousands of dollars for any sizable number of channels.

Considering you can get pretty great surface mount IC's for $5-10 each and they'll fit no problem, it seems like a more efficient way to go. Hopefully it should still sound awesome as well.

 
audiovisceral said:
I've contemplated the idea of discrete converters a bit in the past and I think there may be better places to integrate DOA's. 

DOAs are big and expensive so I think it makes sense to put them where they will have the most audible effect. Technically perhaps this may be the ADC/DAC, since everything goes through them. But if the opamps are hardly applying any gain in the circuit, an IC might be near perfectly transparent, and you might be better off saving that discrete $$$ for use in the pres, comps, and perhaps eq's.

I don't know how much gain the opamps will be pushing through a circuit like this, so I don't know if it's even useful to think of it in these terms. But that's the first thought that runs through my head on the subject.

I'm with you on this- since the opamps in this circuit *should* just be acting as buffers/ line drivers, they shouldn't really be adding any gain, so DOA's would be excessive.

-Dave
 
rkn80 said:
Well, they do not sound. :D They simply play what you send to it. ;)
To be honest, I find it a bit hard to describe how they sound. The only thing I can compare to is my Presonus Firebox. I could say that my DAC sounds better then the Firebox but that would not be fair because I think if you build something yourself you'll always find that your self-made unit sounds better because you made it. ;)

Raphael

Great project!

Where in Germany are you based? I've got some good converters here to compare it to.

I think the difficult part is to get really low jitter. But as long as the converter is the master in a given setup at least sophisticated reclocking circuitry won't be needed. A very clean power supply is important, too.

Is there a way to configure the converter chips to use minimal filtering, especially no linear phase filters? I'd prefer some phase shift from an analog filter (in both DAC and ADC) to compromises in the impulse response caused by (real time) digital filters. I think one of the reasons for the Lavry Gold's big latency is to minimize these problems.

Gregor
 
living sounds said:
I think the difficult part is to get really low jitter. But as long as the converter is the master in a given setup at least sophisticated reclocking circuitry won't be needed. A very clean power supply is important, too.

As far as word clocking, I am still not sure what we will need or Raphael will be building in.

I think the most economical connection unit for hooking this to your DAW would be the M Audio LightBridge. At $350, each unit can give 16 i/o 24/96 SMUX over firewire or 32 i/o 24/48. And they already feature BNC word clock I/O, so I think you should be able to set Raphael's D/A converters to ADAT slave clocking and send your master sync directly from the LightBridge along the ADAT.

However, I do not think this will work for the ADC's, as they will not sync via ADAT. For those, at a minimum, you will need 1 BNC word clock input per 16 channel unit.

Also, even with the DAC's, a dedicated BNC input is desirable. Many have expressed issues trying to sync via ADAT clocking, so having the choice to clock directly is good.

Will BNC word clock I/O be integrated into the unit design?

Is there a way to configure the converter chips to use minimal filtering, especially no linear phase filters? I'd prefer some phase shift from an analog filter (in both DAC and ADC) to compromises in the impulse response caused by (real time) digital filters. I think one of the reasons for the Lavry Gold's big latency is to minimize these problems.

That would be very cool. I would love to have some like those if we could.

But I believe it would require a fair amount of added analog circuitry if you want a different analog filter point for each sample rate, unless everyone will agree on a fixed rate of say 96. Plus I imagine the troubleshooting to get that working seamlessly so the ADC isn't being overfiltered or being fed extraneous frequencies from a not steep or low enough filter might be difficult.

A simple optional set-frequency LPF (eg. 70-90 kHz) in the signal flow might work if the chips can be so configured.

But anyway, it sounds like Raphael's already pulling it together pretty fast. So I think as long as it works, I'll be satisfied.
 
audiovisceral said:
But I believe it would require a fair amount of added analog circuitry if you want a different analog filter point for each sample rate, unless everyone will agree on a fixed rate of say 96. Plus I imagine the troubleshooting to get that working seamlessly so the ADC isn't being overfiltered or being fed extraneous frequencies from a not steep or low enough filter might be difficult.

A simple optional set-frequency LPF (eg. 70-90 kHz) in the signal flow might work if the chips can be so configured.

An analog filter that works for 44,1 khz (= 20-22khz) should suffice IMO. The additional high frequency content at higher sample rates isn't imporant anyway. At least the benefit from an analog filter (opposed to a digital one) would be greater than the loss of ultratrasonics caused by the fixed filter IMO.
 
Hi,

sorry the project was delayed for some days now due to a lot of work in my real life job (new product coming out).

I did not concern about external vs internal clocking so much. I simply removed the clocking from the DAC board because the intention of the project is to keep it modular and flexible. So you can simply decide how to clock it by adding the right board for it. The dac pcb offers the option to select other formats then 24bit I2S or to set the dac as master or slave, it simply offers every configuration option you can find in the datasheet of the dac.

@Greogor: I'm in Bremen, where are you?

Raphael
 
It's very important to have the clock as close as possible to the converter chips to reduce jitter. The modular setup gives the advantage to impliment a high quality clock, like this one:
http://tentlabs.com/Products/cdupgrade/xo2xo3/index.html


I'm in Cologne, so that's quite a journey. Maybe you could send over a prototype for a few days for testing?

Gregor

rkn80 said:
Hi,

sorry the project was delayed for some days now due to a lot of work in my real life job (new product coming out).

I did not concern about external vs internal clocking so much. I simply removed the clocking from the DAC board because the intention of the project is to keep it modular and flexible. So you can simply decide how to clock it by adding the right board for it. The dac pcb offers the option to select other formats then 24bit I2S or to set the dac as master or slave, it simply offers every configuration option you can find in the datasheet of the dac.

@Greogor: I'm in Bremen, where are you?

Raphael
 
living sounds said:
An analog filter that works for 44,1 khz (= 20-22khz) should suffice IMO. The additional high frequency content at higher sample rates isn't imporant anyway.


Wow.  :eek:

Crippling higher sampling rates with subpar filtering, seems to me to defeat the purpose.

I don't want to get into a long debate on sampling rates, but I can't let this one go.

A short example

A crash cymbal has 40% of it's spectral content over 100kHz. Cymbals in particular sound like sushi(*) at lower sampling rates because of the limited bandwidth.

In my experience, many instruments just sound better at higher sampling rates, this is not subtle.  8)

Some background.

http://www.cco.caltech.edu/~boyk/spectra/spectra.htm

A summary of this paper's findings. Column one refers to the figure showing the spectrum in question. Column two identifies the instrument. Column three gives the sound pressure level measured at the microphone. Column four gives the measured frequency extension: For instruments with harmonics, this is the highest frequency where harmonics are still present; for those without harmonics, the highest frequency where the sound is still at least 10 dB above the background. (See text.) The last column tells what percentage of the total energy is contained in the range between 20 kHz and the limit given in the previous column.

Code:
Instruments With Harmonics

Fig.   Instrument                  SPL     Harmonics       Percentage
                                         (dB)     Visible To        of Power
                                                   What Freq.?    Above 20 kHz

1. Trumpet (Harmon mute)    96.      >50 kHz           0.5 
2. Trumpet (Harmon mute)    76.      >80 "               2. 
3. Trumpet (straight mute)    83.      >85 "               0.7 
4. French horn (bell up)       113.      >90 "               0.03 
5. French horn (mute)           99.      >65 "               0.05 
6. French horn                    105.      >55 "               0.1 
7. Violin (double-stop)           87.      >50 "               0.04 
8. Violin (sul ponticello)          77.      >35 "               0.02 
9. Oboe                                84.      >40 "               0.01

 
Instruments Without Harmonics

Fig.   Instrument           SPL   10 dB Above    Percentage
                                  (dB)   Bkgnd. to      of Power
                                           What Freq.?   Above 20 kHz

10. Speech Sibilant        72.     >40 kHz          1.7 
11. Claves                   104.     >102 "            3.8 
12. Rimshot                  73.     >90 "               6. 
13. Crash Cymbal         108.    >102 "            40. 
14. Triangle                   96.    >90 "                1. 
15. Keys jangling            71.   >60 "              68. 
16. Piano                     111.    >70 "               0.02

(*)Sushi as in sounds like ass. :)

Mark
 

Latest posts

Back
Top