Which Capacitors for Audio?

[living sounds]

So much about perceived sound quality is subjective. There were (and still are) plenty of people who think that an SSL console driven into the red is "the sound of rock". People today use all kinds of distortion plugins to chase the technical imperfections of yesterday's gear.

For my own ears, the circuitry in my console, consisting of nothing better than well-implemented (that's crucial) RC4558 and NE5532/4 is good enough to run everything through. Plenty of coupling caps (with small film bypasses) in the signal path, that make no audible difference vs. what is coming straight out of my DC coupled discrete DAC.

I think digital audio is a far bigger problem, especially the converters availible today that often achieve great THD, noise and frequency response specs but still somehow manage to sound really artificial. And don't get me started about ITB processing with lot's of stages of real time up- and downsampling, that sonic signature is all over modern music and it's far worse than an 80s Rick Astley record to my ears.

 

[thor.zmt]

Abbey said this was not possible in many cases.

I would be happy to take any fun loving audio circuit and demonstrate how to do it. It is trivial. And most emphatically possible.

As always, it is a question of correctly defining the goals and then using appropriate technology.

What we see instead is often intellectual laziness that first of uses generic circuits and then fixes problems that pop up in the same intellectually lazy way and excuses it with expediency.

I am not a designer but you are suggesting most of the circuit designs I've seen in pro audio are no good.

I suggest they can be improved.

That's a big claim. I see capacitors used for this purpose all the time.

So do I. And it is trivial to demonstrate that correcting the application to avoid electrochemical capacitors in "open loop" positions objectively improves performance.

Now if this produces better sound quality or a preference for such a circuit is a different question and much harder to answer, having tried that too I found that usually the "better" circuit was preferred also subjectively.

Tell that to the record label spending $20K per hour.

Why? Because someone turned a knob which silently did what is supposed to?

If the record label spends 20k per hour and doesn't want high sound quality, their goals are radically different from, how we record (for example) classical music, or acoustic jazz, or any real music made by real people with real instruments, as opposed to one week at #1 one hit wonders who's "music" is entirely created artificially in a "Lab".

What is more important, such a label where sound quality is secondary or tertiary, a high quality parametric EQ is wasted anyway.

I have much less interest in either participating in such a "creation" (I quit after a few weeks) or listening to it.

At least outside of bars, clubs and similar venues, where the quality of the clientele of your preferred gender and the quality of the drinks is much more relevant than sound quality (actually some such venues tune systems deliberately to sound what I consider poor, as it supposedly makes people drink more).

Thor

 

[Gold]

Why? Because someone turned a knob which silently did what is supposed to?

No, when it didn't do what it was supposed to do because sound quality is more important than functionality.

If the record label spends 20k per hour and doesn't want high sound quality, their goals are radically different from, how we record (for example) classical music, or acoustic jazz, or any real music made by real people with real instruments, as opposed to one week at #1 one hit wonders who's "music" is entirely created artificially in a "Lab".

Well, I'm not a fan of milque toast music. So we'll disagree on that. I'm not too impressed with shiny objects.

What is more important, such a label where sound quality is secondary or tertiary, a high quality parametric EQ is wasted anyway.

I have much less interest in either participating in such a "creation" (I quit after a few weeks) or listening to it.

At least outside of bars, clubs and similar venues, where the quality of the clientele of your preferred gender and the quality of the drinks is much more relevant than sound quality (actually some such venues tune systems deliberately to sound what I consider poor, as it supposedly makes people drink more).

Got it.

 

[Newmarket]

Dear Lord! You really have no idea what horrible people you are, do you? How lucky I am that my tenure in this hellhole of lazy, dogmatic, close-minded, out-of-date egotists lasted only a week or so. As I said when deleting my account, it was like finding yourself enrolled at a rather unpleasant school.

This reply is shining example of the revolting and poisonous culture you have nurtured here.

Calm Down mate

 

[Gold]

So much about perceived sound quality is subjective.

Living Sounds and I have had correspondence about AD/DA converters. We hear things differently. I like Prism converters and he doesn’t. He likes Pacific Microsonics converters and I don’t. They are of course both very high quality and I doubt he would refuse to use a Prism like I wouldn’t refuse to use a PM. We haven’t insulted or killed each other either.

 

[Newmarket]

I suggest that sound quality is the Raison d'être for sound equipment. It is not optional.

Works better is actually optional!

However IME there is rarely a need to compromise this, though some re-learning may occasionally be required from the operator.

Thor

No. It clearly depends on usage. Ergonomics and workflow etc are clearly important in audio production. Maybe not so much in consumer playback where the consumer is willing to make the effort.
From someone involved in a superlative technical solution of its time. If the users need to RTFM to even route a signal - then you have a problem

 

[JohnRoberts]

Living Sounds and I have had correspondence about AD/DA converters. We hear things differently. I like Prism converters and he doesn’t. He likes Pacific Microsonics converters and I don’t. They are of course both very high quality and I doubt he would refuse to use a Prism like I wouldn’t refuse to use a PM. We haven’t insulted or killed each other either.

I don't like to question what people say that they hear on the internets, but it seems that premium AD/DA convertors should express pretty small objective errors.

My favorite technique for comparing analog paths is simple null tests, for convertors there could be errors related to clock synchronization but it might be interesting to see how well they null without fancy clock tweaks.

JR

 

[Gold]

My favorite technique for comparing analog paths is simple null tests,

You can't null a PCM stream with an analog output. You could null one converter against the other but that would only tell you they are different. Not which one is more accurate to the source. Small objective errors is what we are talking about. Whether is matters or not is subjective.

 

[JohnRoberts]

A null test can tell you how different they are.

If the difference is small enough it becomes a never mind...

JR

 

[Gold]

A null test can tell you how different they are.

If the difference is small enough it becomes a never mind...

JR

We already know it’s not a nevermind. We both hear differences and assign different value judgements to them.

 

[abbey road d enfer]

I am familiar with that. Personally I did variable Q simple parametrics, these tend to actually work better than fully parametric ones, for music.

It cuts down the number of Op-Amp's to one per band. The capacitors to block DC are also the actual filter response determining ones, so we get DC blocking for free.

It just shows you have an academic vision of the design of a music recording console.

Yes, I noticed. Mind you, most of these also did not fit 5 way fully parametric EQ's per channel.

4 is common enough.

 

[abbey road d enfer]

Well, I know some Pro audio cone midrange drivers with a single piece Alu former.

I'm not interested in midrange. Excursion is low enough to make current drive work.

What happens is called eddy current braking. If you want to be fiendish, you use a split former and bring out both sides of the split using classic tinsel connections. Add a variable resistor and you can adjust Qm fairly freely.

I"d be schocked if it worked for bass.

I find the way devialet does sub's on the "Phantom" much more interesting, to get low bass (at very low efficiency) in a small box.

That's the product I was talking about.

Absolutely not. You are (hopefully) not using the voice coil back emf, but dedicated sensing that senses (preferably) the actual sound pressure.

Or voice coil dsplacement/speed/acceleration. Works good enough as long as piston mode is valid.
And I maintain it's a form of electrical damping.

 

[thor.zmt]

I'm not interested in midrange. Excursion is low enough to make current drive work.

It was a 15" Driver. The funny thing, you could feed it a LOT of bass and the conf would barely move. For a while we had a bunch of them in big ass horn flares. More is a bit hazy, I think they where Gauss.

I"d be schocked if it worked for bass.

Why? A single piece metal voice coil former acts as shorted turn, very much like the voice coil, except, of course it is always 100% in the Air Gap of the magnet system. Using an Alu voice coil former also reduces midrange distortion.

That's the product I was talking about.

This uses motional feedback and more interestingly, it's a moving magnet system, not moving coil.

Or voice coil dsplacement/speed/acceleration.

Voice coil based sensing is not good, I found that out in the 80's and instead went to microphone on the cone for my experimental speaker which incidentally also applied current drive.

It used drivers easily available in East Germany, dual 8" Bass rated at 35W, 5" Mids that actually were Schulze Berlin Full Range drivers used in Studio monitors targeted at the applications as the LS-3/5 and those clear supranyl 1" Dome tweeters.

It started as an attempt to implement voice coil sensing on all 3 Ways, which was an utter disaster. Running all drivers via current and using microphone based feedback on the Bass section worked great. I had to drop the crossover from 500Hz to 200Hz to be sure the feedback was stable across full range the bass section covered.

And I maintain it's a form of electrical damping.

Electrical damping is using low impedance source drive (aka voltage drive) to act as "sink" for the back emf, causing a "breaking" action.

There is no explicit feedback loop involved.

What is more, as BL is not constant with voice coil movement this damping is highly nonlinear and variable.

It is possible to work around this to a degree and when making passive speakers that are expected to work with random amplifiers a voltage interface is likely the lesser problem in practice.

Motinal feedback on the other hand is in explicit feedback loop that uses a proxy for the actual driver acoustic output to correct all and any errors across the pass band of the feedback system. It is not somf kind of damping, though an undamped Resonance is ONE form of errors the system may encounter.

So sorry, but MFB <> DF.

Thor

 

[abbey road d enfer]

For my own ears, the circuitry in my console, consisting of nothing better than well-implemented (that's crucial) RC4558 and NE5532/4 is good enough to run everything through. Plenty of coupling caps (with small film bypasses) in the signal path, that make no audible difference vs. what is coming straight out of my DC coupled discrete DAC.

That's an interesting comment.
I guess many would sneer at you for claiming you don't hear the difference between a straight wire and a bunch of opamps, capacitors, whatever.
I'm with you on that.
So when people claim they can hear the difference between two similar circuits differing only by the type of capacitors or opamps, I am extremely suspicious, unless there is a serious degradation of performance in one of them.

I think digital audio is a far bigger problem, especially the converters availible today that often achieve great THD, noise and frequency response specs but still somehow manage to sound really artificial.

I don't have the same reservations about converters. I've quite often been amazed at how good a mix sounded, and figuring out it had been done with a M-Audio or Presonus interface

And don't get me started about ITB processing with lot's of stages of real time up- and downsampling, that sonic signature is all over modern music and it's far worse than an 80s Rick Astley record to my ears.

Up and down sampling in plug-ins is limited to factors of 2, so use the traditional method of inserting zeros in the data flux or averaging adjacent samples, which results in MSB rounding error. Since it is most often done internally in 32-bit floating, I can confidently claim I can't hear it.
Stacking 8 plug-ins all with up/down sampling would result in the 3 MSB's in error, for a clean 22 bit resolution (32-bit has a resolution of 25). Even tracking at -20dBfs would leave 18 bit resolution, or 0.0004% accuracy. I don't think I can hear it either.
I have found that most problems with digital processing are wrongly attributed to inherent faults in the principles. Most of the times, these problems end up being due to improper implementation, either in the digital algorithms, or in the analogue support circuitry.

 

[thor.zmt]

It just shows you have an academic vision of the design of a music recording console.

Not academic, but pragmatic and realistic.

4 is common enough.

Many analogue consoles I worked with had 4-Way tone controls, with two semi-parametric variable Q mids and classic Bass/Treble.

I have not actually worked on an analogue console with four or more fully parametric bands. I would also question the value of this in a mixing desk. I do not feel the extra overhead in use of having to control Q as well as frequency and level.

What I implemented back in the 80's as submission for the East German Radio & TV network incorporated four semi parametric bands and was accepted for production and a few made before the wall came down. It used a discrete Op-Amp for the main signal path and Gyrators based on early "audio Op-Amps".

It was considered a massive improvement in terms of flexibility while also delivering improved subjective sound quality over the existing units.

Variable Q lends itself to music use, as when making small level correction we commonly want a very wide bandwidth and thus low Q, as we are changing the "tone". While large corrections are typically made to compendate resonances or similar issues and thus need to be narrowband and thus High Q.

In a semi parametric EQ the Q adjusts itself in parallel with the level. So I only really twiddle one knob, in a full parametric EQ you force me to twiddle two knobs for the same result.

So I'd suggest that actually fitting fully parametric EQ's is not based in practice and usability, but on an academic view that something more precisely controllable and more flexible is a better choice than something simple that works much more intuitively for the user.

Thor

 

[abbey road d enfer]

It was a 15" Driver. The funny thing, you could feed it a LOT of bass and the conf would barely move. For a while we had a bunch of them in big ass horn flares. More is a bit hazy, I think they where Gauss.

Current drive works when cone displacement is low. That's why it's good for midrange, or bass speakers where displacement is low by virtue of acoustic loading.

Why? A single piece metal voice coil former acts as shorted turn, very much like the voice coil, except, of course it is always 100% in the Air Gap of the magnet system.

But the efficincy of a shorted turn decreases with frequency. That's the well-known Foucault effect.

Using an Alu voice coil former also reduces midrange distortion.

Essentially by reducing the effects of increasing reactance of Le.

Voice coil based sensing is not good,
I found that out in the 80's and instead went to microphone on the cone

You conclude a general principle, on the basis of a failed experience.

It started as an attempt to implement voice coil sensing on all 3 Ways, which was an utter disaster. Running all drivers via current and using microphone based feedback on the Bass section worked great.

Going from voice-coil sensing to microphone sensing and current drive is at least changing two parameters. Basic scientific approach says change only one parameter at a time.

I had to drop the crossover from 500Hz to 200Hz to be sure the feedback was stable across full range the bass section covered.

Of course, the transmission between speaker and microphone is not invertible, so the closed-loop needs to be strictly bounded.

So sorry, but MFB <> DF.

Still, MFB is a way to modify damping by electrical ways. If I were in samantics, I would say that damping factor is a physico-mathematical notion that is neither strictly electrical or related to whatever impedance.

 

[Gold]

So I'd suggest that actually fitting fully parametric EQ's is not based in practice and usability, but on an academic view that something more precisely controllable and more flexible is a better choice than something simple that works much more intuitively for the user.

Thor

I’m with you on this one.

 

[thor.zmt]

No, when it didn't do what it was supposed to do because sound quality is more important than functionality.

In the "Sound Quality over Functionality" Case, the knob cannot not do what it is supposed to, because it doesn't exist.

If it is there at all, it will work as designed. If not, that's a maintenance issue, not Design.

If, as engineer, you step behind an unfamiliar console, the first thing you do is familiarise yourself with how it works. There are always enough differences that no two models even from the same maker are identical in how they shape sound and are operated.

Well, I'm not a fan of milque toast music. So we'll disagree on that. I'm not too impressed with shiny objects.

I'm a Musical omnivore.

I "get" Mussorgsky, Beethoven and Tchaikovsky just as I get GWAR or Rush, or Dancefloor music.

But I'm not interested in contributing my effort to disposable music that is meant to top the charts, drop out and be forgotten.

So I never worked with the kind of people like Mike Posner...



"I'm just a singer who already blew his shot
I get along with old timers
'Cause my name's a reminder of a pop song people forgot"

Thor

 

[abbey road d enfer]

Not academic, but pragmatic and realistic.

OK, that means you haven't designed one.

Many analogue consoles I worked with had 4-Way tone controls, with two semi-parametric variable Q mids and classic Bass/Treble.

I know this was a popular format for medium market mixers.
The most reverred mixers had generally 4 full parametric bands.

I have not actually worked on an analogue console with four or more fully parametric bands. I would also question the value of this in a mixing desk. I do not feel the extra overhead in use of having to control Q as well as frequency and level.

One that uses his car for doing errands does not feel the extra overhead of an Aston-Martin.

What I implemented back in the 80's as submission for the East German Radio & TV network incorporated four semi parametric bands and was accepted for production and a few made before the wall came down. It used a discrete Op-Amp for the main signal path and Gyrators based on early "audio Op-Amps".

By "gyrator" I think you mean the synthesized inductors, probably associated with a swinging-input stage. All I can say it is functional. Is it flawless? Certainly not. If I was on a semy-desert island and it was the only available, I would use it, but it's not my go-to.

It was considered a massive improvement in terms of flexibility while also delivering improved subjective sound quality over the existing units.

Then one wonders what were the existing units... I know pretty well what broadcast mixers were in the 60's and 70's. Not made for creativity.

Variable Q lends itself to music use, as when making small level correction we commonly want a very wide bandwidth and thus low Q, as we are changing the "tone". While large corrections are typically made to compendate resonances or similar issues and thus need to be narrowband and thus High Q.

In a semi parametric EQ the Q adjusts itself in parallel with the level. So I only really twiddle one knob,

This is a compromise, and as such it's seldom the right one. That's the argument Rane used for their EQ's. It lasted only the time of novelty.

in a full parametric EQ you force me to twiddle two knobs for the same result.

Not the same result. A compromise guided by the designer's choices, not by the user's needs/wishes.

So I'd suggest that actually fitting fully parametric EQ's is not based in practice and usability,

Tell that to any decent SE.

 

[thor.zmt]

So much about perceived sound quality is subjective. There were (and still are) plenty of people who think that an SSL console driven into the red is "the sound of rock".

Well, they are wrong, the sound of Rock is an old Neve console driven into the reds...

Sound City (2013) | WatchDocumentaries.com

So many of the greatest rock recordings want through that desk, it's not funny.

People today use all kinds of distortion plugins to chase the technical imperfections of yesterday's gear.

Or they chase the original sound thinking utterly wrongly that it were those imperfections that madd the sound.

Every time I hear these software emulators, even the "VCM" modules in my yammie, I cannot but go: "What hilarious dummy sound".

For my own ears, the circuitry in my console, consisting of nothing better than well-implemented (that's crucial) RC4558 and NE5532/4 is good enough to run everything through. Plenty of coupling caps (with small film bypasses) in the signal path, that make no audible difference vs. what is coming straight out of my DC coupled discrete DAC.

Good job. Or bad DAC... Hehehe... No offense meant.

Still I cannot help wondering what it would be like if you changed the 4558 & 5532 in a few channels for OPA1678 and tries an OPA828 where'd you use 5534...

Back in the day when I still did that, I offered to repair/mod one damaged channel for a low cost, as taster. Then I used OPA627/637 and OPA2604 with adapters for the 074 replacements.

I cannot recall a case where I was not asked back to mod the subgroups and sum and at least a few channels. I especially wrought my revenge on desks designed by one DS where one also had to fix the grounding.

I think digital audio is a far bigger problem, especially the converters availible today that often achieve great THD, noise and frequency response specs but still somehow manage to sound really artificial.

I agree. One time in a massive queue at Beijing immigration (before the Olympics , a lot of rules suddenly changed and every foreigner had to get his paperwork resorted), I met an ABC (nominally American Born Chinese, usually also applied to US based and educated CBC's) who's job was head of the ADC/DAC chip development division for a famous US based maker, who's chip Ard one of the staples in pro audio.

We started chatting and found we worked in a similar field. In the end we went out for dinner at Yin Yong Restaurant for Duck, Cat Ears and Bear Paw.

And he came over to have a listen to the audio system we were making, which he thought sounded great.

We also talked about if he ever listened to the ADC/DAC Chips his people designed.

He looked at me quite like "is he mad" so I went and pulled out the EVM for the latest greatest chip out and plugged it in and set up level matching in the Amp ( < 0.5dB difference).

We played the same cut synchronised and switched between the two and his face fell. The EVM sounded, politely put, English way, "pretty good" (that's English for terrible).

He was even more shocked when I explained to him what DAC Chip my design used. We went to the famous mushroom hotpot after and talked a lot of shop, on what I thought were reasons for the difference is in sound.

There is a corollary here, quite a few years later this chip manufacturer introduced a fundamentally new range of DAC Chip's. And tarnation and blimey, they sound excellent. I actually designed them into several products. I also checked on the guy, he is still there.

I suspect, he started to listen, in more way than one.

And don't get me started about ITB processing with lot's of stages of real time up- and downsampling, that sonic signature is all over modern music and it's far worse than an 80s Rick Astley record to my ears.

True. Mind you, it DOESN'T have to be like that, that's all down to the wet ware, not the software.

In principle nothing wrong staying in the box, but it is tempting to overegg that pudding.

Thor