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Delta-sigma vs “Multibit”: what’s the big deal?

DonH56

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Note (not for you John) that 45 bits was the width of the digital filters, not the final resolution, but yeah. I had completely forgotten about TOTL. Most manufacturers had jumped on the single-bit bandwagon pretty early for a variety of reasons (most of which you know better than I -- I had moved on to starting a family and such and lost track of audio for a while). Which then became multi-bit delta-sigma converters.

BTW, where in Australia do you call home? Quuensland the down-under spelling? ;)

The Sony CXD-1244S 8fs digital filter used in their TOTL CD players employed 45 bit noise shaping in 1989.

View attachment 17906
 

restorer-john

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Quuensland the down-under spelling?

I fixed it!

BTW, where in Australia do you call home?

We are up on one of the high points, backing onto the conservation area. (the koalas and wallabies don't complain about loud music)

our place.JPG
 

hvbias

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Not much to do with multibit vs deltasigma.

Had to switch speakers from main room to media room due to energetic 4 yo grandchild who threatened to flip my Gauder Akustik Arcona 100 speakers.

Anyway we're happy with the switch. Evolution Acoustics MicroOnes now in our main listening room for the first time. They used to reside there a couple of years back driven by 2 * Hypex nCore monoblocks and an AudioLab MDA DAC/Pre. Never sounded good to my ears. Tried them with an Audio-Gd Ref 5 partnered with ShengYa pre and amps, no luck as for SQ.

Anyway, here we are a few years later, Evolution Acoustics MicroOnes speakers in main room (blame kid in my avatar) partnered with my Norma integrated amp, Topping D50 DAC for the first time and they sound sublime. Now just need to find some subs ...

View attachment 17905

I always thought those EE MicroOnes were among the best looking monitors.
 

restorer-john

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...Definitely Hi-Bit...

With my subjective hat on, I agree with you. There was something not quite right about the Hi-Bit approach. They used a 16bit D/A and shifted 2 bits to improve low level resolution. I have a NIB TOTL CX-1000 preamplifier here which used their last and (supposedly) best implementation of the architecture. It's never impressed me sonically. A lovely piece of gear and expensive at the time (AU$2399rrp).

But to be honest, and even though I sold tons of Yamaha, their 'sound' didn't appeal to me. Maybe it was because we often paired Yamaha with their NS-xxxX series speakers at the time which were dry sounding mostly.
 

DonH56

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RayDunzl

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I have no idea what "free linear interpolation" means

I suspect, in this context, comparing multibit (jagged) output to DS (smoother) output, you get "linear interpolation" at no additional cost (free).

I reference to:

the way multibit is implemented - u get square shaped signals aka more noise . with deltasigma u get free linear interpolation and less noise.
 

solderdude

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Maybe he means what the output of the actual DAC chip looks like, not the actual DAC device output on its connectors ?

Found some nice pics on the web illustrating what comes out of a 1bit DAC chip, a 3-bit DS DAC chip and R2R DAC chip.
These pictures may well be the reason why some think R2R is not such a bad approach as it appears to be 'cleaner' and needs 'less filtering' afterwards while in reality it is the other way around:

DSD/1 bit DS DAC chip output:
1-bit_green.jpg

images

1-bit_green.jpg


3 bit DS DAC chip output:
images

3-bit_green.jpg


R2R DAC chip output:
images

16-bit_green.jpg

images taken from http://www.mother-of-tone.com/conversion.htm

But there are lots of NOS (Non Over Sampling) and 'filterless' DACs on the market that actually put out 'jagged' signals.
Here's a 'cheapy' Archimago measured at the device output.
attachment.php


There are people that think all R2R outputs look like this but is not the case. Over Sampling makes the steps much smaller and 'errors' are moved higher above the audible range.
Add to that most headphones and speakers are so 'sluggish' they will 'smooth out' the jagged response and if the speakers/headphones don't do this for NOS 44.1kHz signals the ears will anyway.
There are even expensive DACs with similar actual output signals that are thought of highly in the subjective audiophool world.
Metrum Acoustics being one of them. Below the analog output signal of the 'Metrum Quad DAC'. (Picture from OJ-neg at SBAF)
vCFTLJV.jpg



To confuse the non technical minded folks even more you often see these plots here and there:
1012AQDfig05.jpg

This is taken from: stereophile and shows 'jagged' (and noisy) squarewave outputs from a DS DAC chip.
Whaaaat ? Can DS chips also output (noisy) jagged signals. Yes, as that is (close to) the actual signal that it must produce at -90dB 16 bit.

But the UD501 for instance (measured by Archimago) in the (to me) wrong setting to use it in, being 'NOS setting' can also produce 'jagged' signals but produces very nice needle pulses and squarewaves which 'shows' how great the impulse response is and how wonderful it must sound... behold.

Time%2BDomain.png

Of course these artificial signals do not exist in recorded music so don't have to be perfect. Sinewaves need to be 'perfect'.
Some manufacturers (and audiophools), however, keep insisting filterless NOS DAC's are THE best way to reproduce a digitized recording.
Comparing the actual 'analog' waveforms that were recorded to the actual reproduced analog waveforms will show how wrong their thinking is.
Below a picture taken from Computeraudiophile and a bit 'misused' by me to make my point.
waveform.png.68bd33cac004229fa83d04e8c890fd23.png

Consider the signal upper trace the 'original' recorded signal and the lower one from a NOS filterless ladder DAC.
The upper signal is also what one gets when looking at the output of any (properly reconstruction filtered) R2R (ladder) or DS DAC.

The lower trace is what the 'digital value representation' of the digital signal that is actually sent to the DAC chip looks like as well.
The upper trace is the waveform that is represented by the lower trace as it is supposed to come out of a properly constructed DAC device.


The problem with a lot of measurements is that most people are not technically proficient enough to understand what they are looking at and even more at what point in a circuit and what the audible consequences are.
Even for technical people some plots are still a mystery when scales don't come with it.

I am prepared to say that measurements are great and can say a LOT to those that understand it but are confusing and often give the wrong signals to those who do not understand what they are looking at. Often they start to draw their own (flawed or completely wrong) conclusions and vent their opinions all over the web.
 
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restorer-john

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Very cool! Looks like you've got the best of everything.

Not quite, I could do with a couple of your loudspeakers at my place... :) (jealous) and a lot more room in the house. Oh, and a big shed so I can finally get my two offsite storerooms of gear onsite, and stop paying insane storage bills each month.
 
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mansr

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Let's try to make some slightly less misleading pictures. Here I'm using an iFi DAC with a DSD1793 chip.

Playing a 12 kHz tone at 48 kHz sample rate, this is what we see at the output pin of the chip:
tek00000.png


Oh look, stair steps and noise. The steps are due to the DSD1793 upsampling using proper interpolation only up to 8x the input rate, followed by a simple hold to reach the rate of the sigma-delta stage.

After a simple RC low-pass filter, we get this waveform:
tek00001.png


That's much better. While the high-frequency noise is much reduced, there's still a hint of the steps.

With a DSD64 source, the output from the chip looks like this:
tek00002.png


No steps, but much more noise. I don't know what's causing the banding. A single sweep of this waveform:
tek00003.png


It's a mostly sine shaped curve, but with random squiggles rather than distinct steps.

After the low-pass filter it looks like this:
tek00004.png


That's a rather smooth sine wave, though it has more residual noise than the PCM mode. Finally, a single sweep of this:
tek00005.png


My assessment of this DAC is that the low-pass filter is somewhat inadequate. Seems to be something of a trend these days.
 
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THW

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I am prepared to say that measurements are great and can say a LOT to those that understand it but are confusing and often give the wrong signals to those who do not understand what they are looking at. Often they start to draw their own (flawed or completely wrong) conclusions and vent their opinions all over the web.

Yeah that's actually the same feeling I got, I just don't buy the idea that detailed and good measurements can't convey how something will work in real use

However, since these measurements are nothing but raw data, they're useless and have no meaning by themselves, they only become useful when they're given context and meaning which is the sort of work that is being done here and I really appreciate it

I guess that's also why I really like this site, it doesn't just throw measurements of various DACs and amps at you, it gives you measurements but also explains how they will affect your experience with them which is great for someone like me :)
 

The Dragon

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Yeah that's actually the same feeling I got, I just don't buy the idea that detailed and good measurements can't convey how something will work in real use

However, since these measurements are nothing but raw data, they're useless and have no meaning by themselves, they only become useful when they're given context and meaning which is the sort of work that is being done here and I really appreciate it

I guess that's also why I really like this site, it doesn't just throw measurements of various DACs and amps at you, it gives you measurements but also explains how they will affect your experience with them which is great for someone like me :)

Very well said. This site does a great job of cutting through all the marketing hype and getting to the real truth about the performance of audio products. I absolutely agree that looking at a bunch of graphs without the proper context and explanation does nothing but add confusion to an already confusing hobby. The product manufacturers know that audiophile consumers who are not very technically oriented are easily swayed by marketing hype - some of which is just not very truthful or at best misleading. They are more than happy to take advantage of this. Then there is the entire field related to the psychology of marketing which is applied with vigor in such a niche market with so much competition. Real data doesn't lie.
 

solderdude

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Isn't the DSD1793 a hybrid chip which has a (form of) R2R ladder for the upper 8 bits and DS modulation for the lower 16 bits (total 24 bits) ?
Being neither a full R2R nor DSD chip ?

Of course the oversampling indeed creates the steps in between the actual 12kHz samples (48kHz SR) and lowers the step size as well as moving the step frequency far above the audible range.
I suspect the noise on the steps is the interpolated DS to create a 'smoother' transition (sort of dither) for the 8 bit ladder.
After not so steep (fixed at around 75-80kHz and I thought only 12 or 18dB/octave) post filtering the HF noise (DS) is removed/smoothed but the much lower frequency 'steps' are filtered less and thus are still somewhat visible.

I have not seen any R2R (or hybrid) DACs, that can 'do' multiple bitrates which have proper reconstruction filters for each bitrate. Those should be steep and change their -3dB point matching that of nyquiist of the applied sample rate in my opinion.
All the filter designs I have seen (but certainly did not see all designs) are fixed just before the maximum nyquist of the maximum bitrate and mostly between 6dB/oct to max. 24/dB/oct depending on the DAC chip used.
So I agree that most post filters in the analog path are totally inadequate for 44.1 and 48kHz sample rates and leave noisy signals. The noise is above the audible range and cannot be reproduced by speakers, headphones nor ears. They are very small in amplitude so think it is inconsequential for fidelity.
 

mansr

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Isn't the DSD1793 a hybrid chip which has a (form of) R2R ladder for the upper 8 bits and DS modulation for the lower 16 bits (total 24 bits) ?
Being neither a full R2R nor DSD chip ?
It is a hybrid, yes, but none of it is an R2R ladder. The top 6 bits are decoded to a unary (thermometer) code, while the low 18 bits go through a 5-level sigma-delta modulator. The two parts are then combined with appropriate weighting in the actual D/A stage.

I suspect the noise on the steps is the interpolated DS to create a 'smoother' transition (sort of dither) for the 8 bit ladder.
I'm not sure what you mean by this. The noise seen is, obviously, from the sigma-delta modulator. There is, however, no interpolation taking place here. Throughout each step, the SDM is targeting a constant value, specifically the low 18 bits of the output from the 8x upsampler.

After not so steep (fixed at around 75-80kHz and I thought only 12 or 18dB/octave) post filtering the HF noise (DS) is removed/smoothed but the much lower frequency 'steps' are filtered less and thus are still somewhat visible.
In this DAC, the external filter is a 2nd order RC low-pass. Not so steep indeed. I'd have to desolder the capacitors to determine their values, and I haven't done this, so I can't say what the cutoff frequency is.
 

solderdude

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Do you happen to know what type of (physical) Inverted Complementary Offset Binary Code conversion is used ?
Is it currents being added ?
In any case the effect is similar to a ladder (i.e. voltage steps like a ladder) where each step is DS modulated.

Any idea as to why the designers would DS each 'step' and go for an average step value and not do the interpolation between the current sample value and the next one ?
Your scope shot seems to suggest the average is targetted or are the traces overlayed so many times it only looks like the average is targetted ?
Maybe an 11kHz @ 48kHz would show more ?

In the averaging case the resolution would likely be only 6 bit (with 8x OS) when each sample value is targetted by the DS ?
How can they get 24 bit resolution unless the DS does not interpolate values between samples ?
Is the DSD1793 not functioning like a 6 bit SD DAC ?

Most likely Ifi just copied the LPF filter from the datasheet: 77kHz 12db/oct (2nd order).
 

mansr

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Do you happen to know what type of (physical) Inverted Complementary Offset Binary Code conversion is used ?
Is it currents being added ?
Look at the datasheet. My interpretation is that the binary code is first converted to a, still digital, 63-level unary representation, which is combined with the 5-level SDM output, and the whole is then converted to analogue in a current summing stage.

Any idea as to why the designers would DS each 'step' and go for an average step value and not do the interpolation between the current sample value and the next one ?
That would add complexity that they presumably found unwarranted.

How can they get 24 bit resolution unless the DS does not interpolate values between samples ?
Is the DSD1793 not functioning like a 6 bit SD DAC ?
The top 6 bits are readily converted via a unary code and dynamic element matching without getting unwieldy. The correction needed to reach the actual sample value, the low 18 bits, is supplied by the SDM. The advantage over using SDM for the full word is that the level of the modulator noise is reduced by 36 dB, putting that much less pressure on the analogue filter.
 

solderdude

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Ah ... see it now... mixed up the PCM1793 and DSD1793.
 

mansr

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Ah ... see it now... mixed up the PCM1793 and DSD1793.
Those chips are essentially the same. The differences are the control interface (I2C vs direct pin inputs) and that the PCM1793 lacks some features including DSD support and filter bypass. The actual DAC section is the same in both.
 
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