24-bit R2R DAC using miltiple 16/18/20-bit R2R chips

[Marek]

Hi,

I want tu bulid 24-bit non-oversampling DAC with passive I/V (resistor) and tube output stage.
Ofcourse first choice was PCM1704 but as you probably know this chip can't handle relatively high load resistances - with 10 ohm or lower you get best results =>very low output swing about 25mV. This is not suprise because this chip is designed to work with zero load impedance like input of OPAMP which I want to omit.

There are other R2R DACs which can handle higher resistances and provide more output current. So we got AD1865 AD1862 PCM63 TDA1541 etc.
Now, heve you ever saw a solution using multiple lower bit DACs to achieve higher one? I'm not thinking of pararelling chips to make ~17bit from 2x16bit chips but about making true 24-bit DAC in which lower significant bits are converted by one chip(s) and higher ones by another.

Anyone?

Thanks,
Marek

[Bernhard]

non os means you have only 16 bit data

[Marek]

No, you misleading terms.

nonos mean non oversampling - this has nothing to do with bit depth or sampling frequency. You can do oversampling or not with every mode 44.1/16bit 96/24 or even 384/32.

[mrfeedback]

16 bit played back on a 24 bit DAC is still only 16 bit playback, however with increased precision.
To make use of additional DAC bits capability requires interpolation and upsampling.

Eric.

[oshifis]

Marek,

What you want is not possible with chips designed for 16 bit. The accuracy of the MSB should be less than half of the LSB absolute value - in your case you want 24-bit accuracy from a chip designed for 16-bit accuracy.
Paralleling DAC chips is a different matter, because of statistical averaging of the errors. You gain 1/2 or 1 more bit depth (I am not sure) by paralleling 2 chips. So you need a lot of DAC chips to get 24-bit depth from 16-bit chips. But then you get the additional benefit of high output current capability...

[Ken Newton]

Marek:

There's an approach which meets all of your criteria and still utilizes the PCM1704. Before I explain, you should realize the even the PCM1704 does not provide anything near a 24-bit THD levels. At best the PCM1704 enables a significantly lowered quantization noise floor, but you only obtain using either a true 24-bit data source, or a 16-bit source with an oversampling digital filter.

Okay, with that said, you could still use a 10 ohm i/v resistor with the PCM1704 and have a tube output. With a fullscale output current of 1mA from the PCM1704 the peak signal output will only be 10mV. To obtain a standard 2VRMS output level, you will need to amplify that 10mV about 280 times, or 49dB. Such a gain is less than is typically provided by most two-stage tube phono stages. You can now see where I'm heading.

The necessary tube output stage would be comprised of no more than two stages. Just utilize about any moving-magnet based tube phono design or diy kit and omit the RIAA network. Or, better yet, replace the RIAA network with a analog image-rejection filter. In fact, you would likely have too much gain and would need to pad down the signal level between stages. Even a single tube stage could suffice. An 12AX7 or 12AT7 triode would have enough gain if you used an active current-source, or a large choke of you would prefer, to load the anode and a cathode-follower to buffer the output.

[SoNic_real_one]

Tubes are just not good to do I/V conversion. There is this popular belief that there can magically improve the antiquue DACs like the crappy TDA1541, but reality is that at the best a tube stage can attain only 80-83db Signal/Noise - Dynamic Range. That is equal or less than 14 bit of REAL resolution.
Trying to connect a 24 bit DAC to a tube stage is pointless.

[Marek]

Originally Posted by mrfeedback

16 bit played back on a 24 bit DAC is still only 16 bit playback, however with increased precision.
To make use of additional DAC bits capability requires interpolation and upsampling.

Eric.

I got many hi-res tracks 96/192kHz/24bit and even 384KHz/24bit (dxd files). I use lossless files through USB->I2S converter in asynchronous mode as a digital source. I do not use classic CD transport anymore.

Originally Posted by oshifis

Marek,

What you want is not possible with chips designed for 16 bit. The accuracy of the MSB should be less than half of the LSB absolute value - in your case you want 24-bit accuracy from a chip designed for 16-bit accuracy.
Paralleling DAC chips is a different matter, because of statistical averaging of the errors. You gain 1/2 or 1 more bit depth (I am not sure) by paralleling 2 chips. So you need a lot of DAC chips to get 24-bit depth from 16-bit chips. But then you get the additional benefit of high output current capability...

I was thinking about this solution (usign 16-bit chips):
-DAC chip feeded with 16 most significant bits loaded with eg. 100ohm resistor
-DAC chip feeded with remaining 8 bits loaded with 100ohm/65536 resistor

and then sum both output voltages (but I don't know how)

Originally Posted by Ken Newton

Marek:

There's an approach which meets all of your criteria and still utilizes the PCM1704. Before I explain, you should realize the even the PCM1704 does not provide anything near a 24-bit THD levels. At best the PCM1704 enables a significantly lowered quantization noise floor, but you only obtain using either a true 24-bit data source, or a 16-bit source with an oversampling digital filter.

Okay, with that said, you could still use a 10 ohm i/v resistor with the PCM1704 and have a tube output. With a fullscale output current of 1mA from the PCM1704 the peak signal output will only be 10mV. To obtain a standard 2VRMS output level, you will need to amplify that 10mV about 280 times, or 49dB. Such a gain is less than is typically provided by most two-stage tube phono stages. You can now see where I'm heading.

The necessary tube output stage would be comprised of no more than two stages. Just utilize about any moving-magnet based tube phono design or diy kit and omit the RIAA network. Or, better yet, replace the RIAA network with a analog image-rejection filter. In fact, you would likely have too much gain and would need to pad down the signal level between stages. Even a single tube stage could suffice. An 12AX7 or 12AT7 triode would have enough gain if you used an active current-source, or a large choke of you would prefer, to load the anode and a cathode-follower to buffer the output.

Yeah, I know that there is no audio DAC which provide real 24-bit linearity, but still there are sonic benefits of using 24-bit data with non-ideal linearity 24-bit DAC.

About 280x tube stage gain - that was my point.

Originally Posted by SoNic_real_one

Tubes are just not good to do I/V conversion. There is this popular belief that there can magically improve the antiquue DACs like the crappy TDA1541, but reality is that at the best a tube stage can attain only 80-83db Signal/Noise - Dynamic Range. That is equal or less than 14 bit of REAL resolution.
Trying to connect a 24 bit DAC to a tube stage is pointless.

From my experience tubes are great as output stage of DAC. I currently use SRPP circuit with 6N30P tubes with ESS Sabre32 DAC with impresive results but now I want to try one of top ladder DAC in nonos mode (only for higher sampling freq tracks due to known treble roll-off)


Another solution is PCM1704 with step-up trafo. I got LL1931 which has extremly low primary DCR ( 0,9ohm in 1:16 config.) so loading secondaries with 3k will results with effective 10ohm seen by DAC chip.

[Calvin]

Hi,

the named DACs are current output DACs which all feature similar or lower current amplitudes than the PCM1704 with its 2.2mApp@1kOhm. AD1865=2mApp@1.7kOhm, PCM63=4mApp@670Ohm and the TDA1541=1.2mApp@?kOhm. Lower current amplitude means higher IV-resistor-values for a given voltage amplitude, meaning higher noise figures.
All those DACs can´t handle high load impedances because of the restricted voltage compliance. Its just that DACs like the AD1865 feature additional OP amps on-chip, so that You can easily configure a voltage output (IV-converter stage) just by providing for external resistors.
But what´s the sense in using Tubes as output buffer and putting OP-amps into the signal path at the same?
Another issue is the number of Bits. Every doubling of chips results in just 1 Bit more resolution. To achieve 24Bit resolution with a 16Bit DAC chip asks for massive parallelization (8Bit=256). The second solution would be to work with two DACs using different internal reference voltages and to multiplex between the two pipelines.
So it´d be best to stay with the PCM1704 even though its quite costly.
You also might give the NOS-idea a second thought. How do You manage different clock rates and how do You switch bandwidth limits of the output aliasing filters? NOS has serious disadvantages technically as well as sonically (A piano player hits the fingerboard with his fingers...he dosen´t plug the strings with the blade of a saw). The PCM1704 is capable of clocking at up to 768kHz using a appropriate up-/oversampling filter. At such high clock frequencies one can safely omit with the output aliasing filter which accounts for much of the bad reputation of digital sound.
Lastly You might rethink about using tubes as they limit the capabilities of a DAC stage. It´d be hard if not impossible at all to achieve better than 12-14bit linearity and dynamic range.

jauu
Calvin

[Marek]

Yes those DACs has similar output current BUT diffenent tollerance to load resistance. eg AD1865 (current output!) perform very well with 100ohm resistor, with PCM1704 it is unacceptable.
About aliasing filter. When using high sampling freq there is really no need to use of aliasing filter. I'm using tube amplifier so filtering is done on OPTs.