A chip-amp to rival Hi-End - design advice

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Hi,

..and create other degradations of their own.

Depends. If you get them designed for a specific power level and frequency range they can be made surprisingly good.

For example, I see no issue in a 3-Way system to use Hi Nickel Permalloy (Mu-Metal to old worlders) cores and the number of turns would be quite small. The Low Frequency section may need a 400Hz steel core.

In fact, due to the impedance conversion you would find that as long as this system is played at material amounts of attenuation the sound is likely better than than a traditional system with the volume control ahead of the power-amps.

The looks of that amplifier case are totally different from what one would (should?) expect from somebody in his position.

Why. Remember, he is an ex-diy guy too. For his personal system he decides looks don't matter, so what?

Ciao T
 
ThorstenL said:
I would also add that using ANY volume control ANYWHERE (digital domain or analog domain) degrades performance.
What about amplifiers with volume control based on variable gain? What's the trade-off there?
pacificblue said:
Isn't that the point of attenuation?
I think the point is (ideally, of course) to scale down the waveform, while preserving its exact shape. And since this is impossible, we have to ask another, very important question - which degradation (digital or analogue) sounds better?
ThorstenL said:
The MF/HF stuff definitly LM3875 and with "Class A Bias" on the output (2.2K/5W resistor will do) and on around 24V rails. For the Bass likely LM3886 or LM4780 with 35V Rails.
Let me just clarify this - does it mean that you advocate against using My Ref topology, or was that just "for the sake of discussion"?
 
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What about amplifiers with volume control based on variable gain? What's the trade-off there?I think the point is (ideally, of course) to scale down the waveform, while preserving its exact shape. And since this is impossible, we have to ask another, very important question - which degradation (digital or analogue) sounds better?Let me just clarify this - does it mean that you advocate against using My Ref topology, or was that just "for the sake of discussion"?

Do you mean active volume controls such as the Baxandall type ? implemented using one or two opamps.
If so results can be excellent, a big bonus is that channel matching is much improved as a linear pot is used and the gain is a function of the angular rotation of the pot. Any resistive mismatch between the two gangs doesn't matter.

You can think of attenuating digitally as using lower and lower bitrates as the level reduces... the quality worsens.
Attenuating in the analogue domain preserves the waveform essentially untouched... and with a good design that ideal can be reached very closely.
 
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Well, I think I meant using a pot in place of the resistor that normally sets the gain, but I'm not really sure... :p It would probably drive amp into oscillation in some settings... Like I said I don't really know much about audio electronics, I just read or hear things here and there, and then try to make sense of them :p

Well basically it is, but it's not suitable for use with a power amp IC.
The active volume control is a dedicated part of the circuit using two opamps for the best result.

This is the idea, here using one opamp. The gain is now a function of the rotational angle of the pot, it's value doesn't matter as such.
 

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Hi,

What about amplifiers with volume control based on variable gain? What's the trade-off there?

Well, the LM38XX chips cannot easily be used with a gain below 10.

Past that, variable gain requires an element that varies the gain. There is a limited set of choices and they boil down to the same as conventional volume controls. So you do not really change the device that varies the volume but simply where it is placed. As variable gain systems always have quite a bit of signal current flowing and most pot's seem to sound worse in such an application it may not be the best solution.

I think the point is (ideally, of course) to scale down the waveform, while preserving its exact shape. And since this is impossible, we have to ask another, very important question - which degradation (digital or analogue) sounds better?

My experience so far favours keeping the volume control in the analogue domain if analogue amplifiers are used and to place it as far back in the chain as possible.

Let me just clarify this - does it mean that you advocate against using My Ref topology, or was that just "for the sake of discussion"?

I personally do not feel that nesting multiple high feedback loops is such a great idea, you reduce some issues but bring in others. Plus, the LM38XX series is probably "good enough" (unlike some other chipamp's) on it's own. For example both stability and overload behaviour cannot help but be compromised. So I would be unlikely to use it myself. And I would recommend building it only to experienced DIY'ers, as such a design is hell to debug if there is some oscillation of other problem.

But in principle it makes no difference on my recommendations if you do a straight gainclone or the like or if you use a nested feedback loop design instead in it's place.

Ciao T
 
Dear All,

Interesting discussion. I am agree with Thorsten, that any kind of attenuation degrades sound quality in a way. It is debatable which topology harms worse. I can contribute some of my experience with this issue. For a multi-mediaplayer design, we came to the same issue. What to do? Do it in the digital domain? do it in the analogue domain? and if so, do it with relays, a chip or a potential meter. It was clear that many of the (audiophile) people involved with this Media player where already prejudged against doing it in the digital domain, cause of the resolution lost. Personally I think this issue is highly overrated. Yes you do loose resolution, but then we did some comparison on the meters in matter of THD, noise and signal integrity. You would be surprised to see what even happened to single tone sine wave with a potential meter on the lowest position. You loose more then, having the same attenuation in the digital domain. Then in this case we did blind tests, to find out if it really matters so much. Our prototype could do both. Do it with the PGA2300 BB chip, with a potential meter and digitally in the (Wolfson) DAC's themselves. From the 7 people we where with 6 preferred the digital domain every time. Strange enough even though the potential meter measures worser then the PGA2300 chip, no one liked the PGA2300 chip blind. It seems that this chip always added a sonic signature on any chosen volume level.

My personal opinion is, that you have the least noticeable degrading in sound quality if done (correctly) in the digital domain. Doesn't say this is "perfect". You can save many parts in the analogue circuit, and feed an amplifier almost directly from a DAC. I wonder why no one ever tried to use a chipamp as PowerDAC. Means the chip amp is placed directly after a voltage out DAC, and you do the lowpass filtering in the chip. I can't think of any shorter signal flow. Extra benefit, you can feed the chip amp symmetrical from the dac chip.

I see some hesitation about nested feedback systems here. And I can understand why. It add some extra complications to stabilize and to compensate. But it isn't the end of the world, and if you work careful, step by step and with a good plan you can do it, and end up with a very stable Amplifier. Again nothing is perfect so Nested Feedback isn't either. But if you see the reaction from people who actually build the REF, and/or other nested feedback amplifiers, you can see they are extremely positive with the results. My own experience is as well that it takes the level of the chip amp one level further, if you use a top-class driver opamp. The design and sound-quality now depends mostly on the first opamp and not on the chip-amp anymore. Then again, it is a bit comparing apples with apples, since both topologies have have many implementations.

With kind regards,
Bas
 
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From the 7 people we where with 6 preferred the digital domain every time.
Thanks Bas, that is most interesting. I felt that digital would be the best and chip would be the worst, but now you've confirmed it.

As for why are we hesitant to use nested design - well didn't you just say that a chip always introduces some colouration? We all know LM38xx on its own is very transparent - does it remain so, when used in conjunction with an opamp? Is all the "immediacy" still there?
 
Hi Bas,

Strange enough even though the potential meter measures worser then the PGA2300 chip, no one liked the PGA2300 chip blind. It seems that this chip always added a sonic signature on any chosen volume level.

Yes. This device and it's Cirrus Logic cousin are really bad.

My personal opinion is, that you have the least noticeable degrading in sound quality if done (correctly) in the digital domain.

I find that the only "digital volume control" I find largely transparent works only in 6dB steps. This is a little limiting. Maybe a hybrid solution?

I tried this, the other way around, with the analogue attenuation switched in 6dB steps and the maximum digital attenuation limited to 6dB. In the end, following blind testing, we went back to a complete analog domain solution.

I wonder why no one ever tried to use a chipamp as PowerDAC. Means the chip amp is placed directly after a voltage out DAC, and you do the lowpass filtering in the chip. I can't think of any shorter signal flow. Extra benefit, you can feed the chip amp symmetrical from the dac chip.

Tried it. Works quite well actually. Hence I recommended that exact approach. However I would still throw an analog domain volume control into the mix.

I see some hesitation about nested feedback systems here. And I can understand why. It add some extra complications to stabilize and to compensate.

And the result is much less stable, unless you give up most of the benefits of nested feedback by limiting the amount of loop feedback to that in the output device at higher frequencies, which is exactly where the chipamp would need help.

There are smarter ways to deal with these, if we already allow so much extra complexity (feed forward error correction anyone?).

Ciao T
 
Hi,

Thanks Bas, that is most interesting. I felt that digital would be the best and chip would be the worst, but now you've confirmed it.

If you (or rather Bas) compare using one of the worst sounding Chip's, what do you expect?

I spec a lot of BB/TI Parts in my commercial designs, but the PGA23XX series is pure and unadulterated low fi junk. Any 99 Cent pot does better.

There are a few Chip Volume control that are good enough (correctly applied) to not greatly fear comparisons to the usual stepped attenuators, though a REALLY GOOD ONE (say build on a silver switch, using Rohpoint Econistor non-inductive, precision copper-alloy wirewound resistors with 2 resistors per step) outperforms it notably, as does a good TVC.

Ciao T
 
And the result is much less stable, unless you give up most of the benefits of nested feedback by limiting the amount of loop feedback to that in the output device at higher frequencies, which is exactly where the chipamp would need help.

There are smarter ways to deal with these, if we already allow so much extra complexity (feed forward error correction anyone?).

Ciao T

Dear Thorsten,

There are smarter ways to do nested-feedback. In my example to keep it easy I only show compensation direct from the driver output back to the inverted input to compensate. In that case you reduce the total loop feedback at higher frequencies. But like you know yourself, you can also use compensate tricks inside the total loop, and still benefit global feedback at higher frequencies. But because this is a bit more tricky I didn't showed that in my example posted earlier.

With kind regards,
Bas
 
Bas,

There are smarter ways to do nested-feedback.

Sure. But is even smarter to not loop another feedback loop around an already too slow device. You are not making it any faster, but you are increasing what Malcolm Hawkesford (and LFD) call "fuzzy distortion".

Instead of looping feedback, why not non-looped feed-forward? It has non of the limitations of feedback and was actually part of the original 1920's Black patent that is now cited as the first application of negative feedback (wrongly, Paul Gustavus Adolphus Helmut Voigt invented and patented negative feedback nearly a decade earlier and sensibly applied it straight to the transducer and not the amplifier, though clearly it linearises the whole system, including amplifier).

Something like an LM6181 (sadly discontinued) or AD811 makes a pretty serious feed forward corrector if you know how to combine it's output with that of the main Amp (chip or discrete). Stochino wrote an interesting article on that in wireless world (*).

Still, I do not think (based on experience) the LM38XX series gains much from either methode. In fact, I was considering them as the "feed forward" correctors for some digital amp's, but that project is currently being extensively cryotreated (it is on ice).

Ciao T

* Stochino - "Audio Design leaps Forward" - Wireless World 10/1994
 
Bas,



Sure. But is even smarter to not loop another feedback loop around an already too slow device. You are not making it any faster, but you are increasing what Malcolm Hawkesford (and LFD) call "fuzzy distortion".

Instead of looping feedback, why not non-looped feed-forward? It has non of the limitations of feedback and was actually part of the original 1920's Black patent that is now cited as the first application of negative feedback (wrongly, Paul Gustavus Adolphus Helmut Voigt invented and patented negative feedback nearly a decade earlier and sensibly applied it straight to the transducer and not the amplifier, though clearly it linearises the whole system, including amplifier).

Something like an LM6181 (sadly discontinued) or AD811 makes a pretty serious feed forward corrector if you know how to combine it's output with that of the main Amp (chip or discrete). Stochino wrote an interesting article on that in wireless world (*).

Still, I do not think (based on experience) the LM38XX series gains much from either methode. In fact, I was considering them as the "feed forward" correctors for some digital amp's, but that project is currently being extensively cryotreated (it is on ice).

Ciao T

* Stochino - "Audio Design leaps Forward" - Wireless World 10/1994

Dear Thorsten,

I need to read and learn more about feed-forward to really understand every single facet of it. Our forum member Janneman, had some interesting design with it with his PAX amplifier. Ps. I think especially in multiple chipamps in bridge/parallel you can benefit from nest-feedback or might be feedforward error correction. When multiple chips are paralleled, with each it's own feedbackloop, there will be always differences in gains and bandwidth between each of the devices. The will try to correct each others ang will give small added distortions in the higher regions. Maybe that is why often one single chip sounds better then multiple. However if you take them all in a loop, those non linearities can be corrected.

One other interesting point talked about here was to put a LM3886 as DAC follower which take over the function as buffer and LPF filter. It saves one opamp behind the DAC. I think in search for the most direct signal-flow this is really interesting. I can think of an active speaker system, with 3 DAC's direct followed by 3 chip amps. When right implemented, one can have a very high resolution system.

With kind regards,
Bas
 
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Hi,

When multiple chips are paralleled, with each it's own feedbackloop, there will be always differences in gains and bandwidth between each of the devices. The will try to correct each others ang will give small added distortions in the higher regions.

As you seek sub milliohm output impedances you cannot use the usual cure, 0.1% resistors for the feedback loops of the Amp's and 0.5% power resistors for the summing.

I find this is enough to push the probelms to below audibility (though they may still exist.

The problem with nested feedback in the higher regions, if it actually offers any advantage over the original chipamp it is less stable and the relationship is proportional.

One other interesting point talked about here was to put a LM3886 as DAC follower which take over the function as buffer and LPF filter. It saves one opamp behind the DAC. I think in search for the most direct signal-flow this is really interesting. I can think of an active speaker system, with 3 DAC's direct followed by 3 chip amps. When right implemented, one can have a very high resolution system.

Agreed, as said, I'd reserve the 3886 for the Woofers though and leave the Mid/Hi where current is not so much in demand with 3875. Also worth doubling up chips for balanced in balanced out to the driver, unless you are religeous about output impedances (nudge nudge, wink wink).

Ciao T
 
Hi,



As you seek sub milliohm output impedances you cannot use the usual cure, 0.1% resistors for the feedback loops of the Amp's and 0.5% power resistors for the summing.

I find this is enough to push the probelms to below audibility (though they may still exist.

The problem with nested feedback in the higher regions, if it actually offers any advantage over the original chipamp it is less stable and the relationship is proportional.


Agreed, as said, I'd reserve the 3886 for the Woofers though and leave the Mid/Hi where current is not so much in demand with 3875. Also worth doubling up chips for balanced in balanced out to the driver, unless you are religeous about output impedances (nudge nudge, wink wink).

Ciao T

Dear Thorsten,

Well since I am that religious about output impedances :D :D, there is actually a way to make a bridged balanced chip amplifier (and I do prefer balanced drive) and make a correction in where you take the differential signal and feed that back to the driver chip. :D I am actually listen to such a configuration right now, but I will offer some details once I am really satisfied about it.

With kind regards,
Bas
 
Hi Bas,

First let me point out something. Over the years I have owned many commercial systems and made many of my own. In the long run I always come back to tubes (best single ended) and wideband (some call them "Full Range") speakers with bass and treble support.

This may be down to growing up where music and listening is concerned with beautifull big German Valve Radios (Dampfradio), full of tubes, transformers and wideband drivers.

I also very much like coaxial speakers and I very much appreciate good solid state gear (even if I du not use it).

I also have an abiding hate relationship with dome tweeters (the inverted Domes from Focal are about the only ones I spend real time with, they are not completely bad) but I like Magnetostats and ribbons and have a hate/love relationship with compression drivers and horns.

My current system (as of today), to provide context is as follows:

PC Source (Custom, Fanless), SPDIF from the motherboard chipset (Via Vinyl) with heavy modifications

TDA1541 DAC with Tube Output, while based on the diyhifisupply "Satch" it's designer (Thomas "Tubelover") would not recognise it, includes DEM reclocking and I2S attenuators and powersupplies so overbuild it is well past silly o'clock (sounds good though)

The SPDIF cable uses WBT Nextgen and Belden 1575 (I think) Video Cable (foamed PTFE core, Foil screen, close 75R impedance match).

For a long time I have been an "Analog Addict", but this digital setup gets me close to the best parts of analogue (LP) sound, without the drawbacks while delivering Digital's benefits very well.

Preamp with DS1666 Volume control and Tube stage (basic diyhifisupply fare - for testing, not bad at all though, even though as designer I should perhaps be quiet)

Poweramp is a "local feedback only" 6550 based Push Pull Class A Amp, around 25 watt, NOS Brimar ECC83/12AX7 for gain, NOS black plat mil spec 6N12 (think russian octal 5687) as concertina phasesplitter.

All electronics silver wired, silver capacitors for coupling and all that jazz.

Speakers have a special 5" Wideband driver (Alu Cone with oxide layer for damping) run with a first order 300Hz highpass (many big film caps). Bass is a 10" Alu Cone (with oxide layer) Woofer with a first order lowpass.

As supertweeter I use a ring radiator magnetostat, again first order highpass. This intentionally broadens the dispersion of the wideband driver in it's upper passband (at the cost of a litte on-axis treble boost which according to Jens Blauert stimulates the "hight" perception for sound, so singers vocals appear at the right hight in the sweet spot (read well above the top of speakers).

Speakers measure quiet flat in room for a passive, minimal system and have a decently controlled directivity.

Speaker Cables are Tempflex PTFE/Silverplated Copper Ribbon cables (basically High End SCSI II cables - cost 5 USD per foot as computer cable!).

This gives you an idea of what I personally listen to at home for relaxation and fun. Many of my commercial differ, as they aim to please many, not just myself.

The thing this system majors in is immediacy, directness and transmitting emotion. In classic audiphile terms the 30ish Hz LF cutoff is a bit high, the speakers are very slightly coloured, but the combination is very enjoyable.

Now with this long prefix, to the meat.

I have one question for you though. If you had to build such an high-resolution system, with 3 DAC's and 3 amps. Would you stick to the chip amps, or would you design discrete? Or semi discrete with the LME driver chips?

I PERSONALLY would use discrete hybrid amplifiers (likely fully balanced circlotrons with what I call Tubelington - a Tube follower mated to N-Channel Fet *& PNP Sziklai follower).

I would likely also use classic multibit DAC's (likely multiple PCM1704) without digital oversampling filters with Tubes as analogue stages and maybe even tube based rectification for the DAC's (definitly for the tube stages). I'd probably also commission our software people to make me a very minimalist IIR software crossover for this setup.

And I'd likely build up the speakers myself as true cardiode systems from basically 16Hz or so upwards with a directivity index of 6dB up to around 2KHz and then rising.

Of course, such a system means I have to take a year or two off.

If I had to build something more realistic and I would not require compatibility with commercial equipment for testing; the Behringer DCX and band-optimised Gainclone route with a nice 3-Way system (wideband midrange) would probably one I'd select as being well supported hardware-wise and in the community as well as offering very good results for the time and effort expended.

I'd probably still hanker after my big imaginary system (I'd probably shall call it MOAS) though.

Ciao T
 
I tell you what, guys. I don't fully understand all the technical terms used, but I like the way this discussion is progressing ;) It gives me lots of ideas.

I'm quite sure I won't be able to go all out on crossovers, wideband drivers and all that straight away - my budget is limited and audio gear has quite a competitor to my money, in the shape of this beauty:D
An externally hosted image should be here but it was not working when we last tested it.


But when I have ideas, I will realise them - or at least try to - sooner or later.

I think the best point for me to start, is to make the power amp section first, and take it from there. I could start easy, with my trusty Noble pot as a preamp. So at the minute, I would like to focus on choosing the best way to go about the power amp section.

I understand that there is little that clearly speaks for, or against one design over the other (although, having read a few threads, there was no one who said simple GC was better than My Ref; if anything, subjective opinions seem to favour My Ref). Which means that if I really want the better of the two, I will have to build both and compare them side by side.


On a side note: I've bought them Mordaunt Short System 442 speakers. I was auditioning them on seller's Naim amp, and I kid you not - I think my GC is better... :eek: Also, when I compared them to my Wharfedale MFM 5, it turned out that the difference between the two was far from staggering. Yes, they do pick up detail better, and the soundstage is more 3-dimensional, but Wharfedales, despite sounding slightly flatter in comparison, are very neutral, and hard to fault really.
I think I had a VERY good system as it was anyway - just didn't fully appreciate it, because of how cheaply it was put together :p

There is but one problem - the MS442 don't work very well with my GC... They are ok for the most part, but the bass illness is back, big time. I know they are capable of putting out some serious bass, and very accurate at that - because they did so on the Naim. Also GC does not do too badly on the bass, when paired with Wharfedales. But on MS422 with the Gainclone, the bass is really dissapointingly lacking. Still razor-sharp, but too thin, not "meaty" like on Wharfedales.
MS442's sensitivity seems comparable to Wharfedales (which are 89dB) - or maybe a tiny bit less. They are rated at 6Ohms vs Wharfedales' 8Ohm.

I'm going to do some listening tests over the next few days. I'll put the GC against my Denon AVR-1705 and my sister's Creek 4040 and the Wharfedales against the Mordaunt-Shorts; I may throw my sister's B&W DM602 S3 into the speaker mix. And then I'll see what's what.
Because as it stands now, I'm not sure whether I actually like them Mordaunt-Shorts...
 
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