Low-distortion Audio-range Oscillator

Richard you can operate the dlbSVO as a high Q SVF. Just put it into that mode.
In the filter mode you have both band pass and low pass outputs. The Band pass is 6dB slope and the low pass output is a 12dB slope.

Take your next best gen and run a signal to the SVO/F input at -20dB or less input signal.

This will give you prove of concept with an analog process.

Just follow the instructions I sent you for tuning.
 

Resembles the story why JC doesnt use opamps much... all discrete. Cool.

David.... Dont get me wrong... I am pleased with the gen as it is. Great job !

Yes, i did some of the things you suggested and do get lower distortion.

Just wondering....... is there a technique to get to zero instead of ever lower and lower.... cancellation or filtering or ?



THx-RNMarsh
 
Apparently, there is a DAC with an SNR of -140dB and THD below this figure.
It is the Bruno Putzeys' DAC add-in to his Makua Preamp that boasts similar performance figures.
But there is also a price tag on it.

DAC

>> Even by today’s exacting standards, extraordinary care has been taken deal
>> with jitter. Mola-Mola’s DAC uses a home-grown asynchronous upsampling
>> algorithm whose input frequency measurement slows down rapidly until after
>> a few seconds of lock, the frequency ratio measurement is frozen. Frequency
>> stability is then wholly determined by the internal clock, a laboratory grade
>> 100MHz SC-cut oscillator. This is effectively an atomic clock sans the physics
>> package (which contributes nothing to spectral purity but quite a lot to cost).

Oh, yes. An atomic clock without the parts that make it an atomic clock, but we
had to drop the words to the gullible crowd. But we do have alternate facts.

Next year, we will redo the moon landing, just without that clumsy
Saturn-V that only adds to the cost.
 
Last edited:
Oh, yes. An atomic clock without the parts that make it an atomic clock, but we
had to drop the words to the gullible crowd.

Next year, we will redo the moon landing, just without that clumsy
Saturn-V that only costs money.

I'm disappointed with some of the write up on that site. Let us not forget that the DSP is virtually useless in the test signal case, a 1kHz sine wave sampled at 96kHz has only 24 unique values if you take into account sign/magnitude.
 
et tu, Scott?

anyone else notice Putzeys stated PWM vs PDM rise/fall time linearity effects exactly backwards in his reply?- I've been reading this for stuff for decades and could find multiple sources

in PDM (sometime additionally qualified as "return to zero") you get a fixed V*t area for each pulse as each has exactly one rise and one fall per fixed on time interval, this is the best for linearity

in PWM the rise and fall times are spread over differing on times, giving rise to nonlinearity

and DSD isn't generally implemented with PDM, "return to zero" output, DSD makes a decision to switch output at a fixed rate, but you can have many consequitive ones or zeroes in the output - giving you the rise/fall time problem again
 
Resembles the story why JC doesnt use opamps much... all discrete. Cool.

David.... Dont get me wrong... I am pleased with the gen as it is. Great job !

Yes, i did some of the things you suggested and do get lower distortion.

Just wondering....... is there a technique to get to zero instead of ever lower and lower.... cancellation or filtering or ?



THx-RNMarsh

I have extra PCBs here. I'll build up an SVF fixed tuned and see what we get.
 
That Mola Mola box kinda looks like that Symnet box
layout with out any space between components. All
crammed together. I wonder what kind of support they
can muster? or is it only a blog?

I think Scott is a lot smarter than he lets on.
I used to think, who'd not want to work for
Apple,...the best. Until I started to research
it more, and the movie, well that speaks volumes.
Now I understand your statement, that you didn't
want any of that (paraphrased)...kind of like
working for the Bill Gates....who want to work
with that kind of crap....of course they call there
competitors and agree not to hire them....
Good for you Scott, hope you enjoyed your
vacation too.

I'm wondering if that is The JC that Richard is mentioning....
got me thinking about how Gene Krupa got kicked outta St. Joseph College.

A pic of Jesus with "To Gene, from J.C."--Signed picture hanging over his bed.

Cheers,

Sync
 
anyone else notice Putzeys stated PWM vs PDM rise/fall time linearity effects exactly backwards in his reply?- I've been reading this for stuff for decades and could find multiple sources

in PDM (sometime additionally qualified as "return to zero") you get a fixed V*t area for each pulse as each has exactly one rise and one fall per fixed on time interval, this is the best for linearity

in PWM the rise and fall times are spread over differing on times, giving rise to nonlinearity

and DSD isn't generally implemented with PDM, "return to zero" output, DSD makes a decision to switch output at a fixed rate, but you can have many consequitive ones or zeroes in the output - giving you the rise/fall time problem again

I'm worried that marketing has pulled Putzes over to the dark side. He has gone from very rational discussions of things like feedback to pretty esoteric discussions like these with an remarkable amount of ego. I have heard the stuff about the small ripples from an independent source but I'm not so sure you can actually measure them.

On the pre-ringing stuff- what speakers can reproduce it or any related artifacts? Aside from the impossibility to record a 1 sample wide pulse from any acoustic event.
 
maybe just evidence of less respect for the audiophile press than his rubber duck

I expect we could converge on a common understanding of the engineering issues given time to exchange definitions, assumptions

but the soft speculation does worry - but that's what the audiophile press is in the business of pulling for
 
I'm worried that marketing has pulled Putzes over to the dark side. He has gone from very rational discussions of things like feedback to pretty esoteric discussions like these with an remarkable amount of ego. I have heard the stuff about the small ripples from an independent source but I'm not so sure you can actually measure them.

On the pre-ringing stuff- what speakers can reproduce it or any related artifacts? Aside from the impossibility to record a 1 sample wide pulse from any acoustic event.

Besides, what is or how it is written or edited, or explained to lay public ..... does his gear go as low as he said or have the dynamic range he says, etc?


-Richard
 
I'm worried that marketing has pulled Putzes over to the dark side. He has gone from very rational discussions of things like feedback to pretty esoteric discussions like these with an remarkable amount of ego. I have heard the stuff about the small ripples from an independent source but I'm not so sure you can actually measure them.

On the pre-ringing stuff- what speakers can reproduce it or any related artifacts? Aside from the impossibility to record a 1 sample wide pulse from any acoustic event.

I've had the good fortune to have dinner with Bruno several times and I almost always catch a conversation at the various audio trade shows. I have zero worries about Bruno going to the dark side, he is a gifted engineer and his tolerance for B.S. is very low...

I bumped into him at the recent AES show in L.A. and learned more in 10 mins than I did at the whole rest of the show.
 
Yes I can accept that a passive low pass cleans stuff up, but a miniDSP (or any other similar equipment)? That means your signal has gone through an ADC, then DSP-processed, then out through a DAC, lots of active stuff, and then all harmonics down to -130dB??

A tall order!

Jan

I mixed (passive) a -60dB 2.5KHz signal to a 0dB = 1v 1KHz signal. I already know there is an accumulated -1dB of system level error. But I am not going to bother to correct it. Here is that mixed together level as seen from the 725D analyzer monitor output without and with a LP filter applied.

-60dB signal.JPG





filtered.JPG




THx-RNMarsh
 
Last edited:
Anyone notice the new OPPO DAC has dynamic range of 140dB spec?

Uses ESS 9038 PRO



-RNM

The ES9038PRO SABRE DAC features ESS’ patented 32-bit HyperStream™ DAC technology with up to 140 dB – industry leading –
DNR in mono mode and –122 dB (0.00008%) total harmonic distortion plus noise (THD+N).

There are a bunch of 8 Channel DAC's out there anyone try summing/averaging them into a mono signal as an instrument output? I proposed this ages ago, a learning loop could null spurs saving everything into a file of coefficients.

• Customizable output configuration: Mono, stereo, 8-channel output in current-mode or voltage-mode based on performance criterion
• Customizable filter characteristics: User-programmable filter allowing custom roll-off response
• Programmable THD compensation to minimize THD caused by external components
 
Last edited:
There are a bunch of 8 Channel DAC's out there anyone try summing/averaging them into a mono signal as an instrument output? I proposed this ages ago, a learning loop could null spurs saving everything into a file of coefficients.

I think we stopped at the summing averaging process because there isn't anything simple around to do that. The hardware can get quite complex even just to do two channels.
I think everyone backed away from that one.