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High-end chipamp build project

Posted 24th January 2013 at 08:21 AM by abraxalito
Updated 15th February 2013 at 06:12 AM by abraxalito

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Go here : Possibly the most frugal high-end sounding amp?

If you have comments, feedback, critique or improvements and don't want to put it on that thread, feel free to leave it in the comments here.

Here are the first measurements - the Nitro amp being fed by the Ozone DAC with a six-tone multiple, peak digital level -20dB. Peak output power around 100mW in an 8R speaker (connected for this test). It looks from this that the datasheet is a little pessimistic on THD, though here I'm measuring the TDA8566, not the 8561. No measurement graphs are presented in the 8566 DS and the 8561 only has plots for 2R and 4R loads, not 8R.

Schematic now added. BOM here : https://docs.google.com/spreadsheet/...Hc&usp=sharing

Additional schematic showing how to use TDA8566Q in place of TDA8561Q as the latter isn't available from Mouser. Mouser only has fewer than 300 TDA8566Q in stock so best to get in quick - I doubt they'll be getting more as this chip is going obsolete. The pins shown unconnected in the mod schematic aren't unconnected - it means they're connected the same way as for the TDA8561Q.

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Posted in Power chip amps

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Comments

1. 	Yes the CM choke is critical - I'll probably put up some more details to explain how to make one yourself. I need to see if there are any bare segmented cores available from the standard vendors.
   	Posted 5th February 2013 at 10:12 AM by abraxalito

2. 	Got my stuff. May try to throw together pcb and see how it goes.
   	Posted 8th February 2013 at 11:41 AM by buzzforb

3. 	[QUOTE]There seemed to be a fairly major improvement in bass dynamics in swapping PSUs, that's something I'd like to get to the bottom of. I think perhaps I'm not using enough electrolytics in the decoupling..[/QUOTE]. :D, :D, ;) ... Frank
   	Posted 26th February 2013 at 06:25 AM by fas42

4. 	Listening now with 40,000uF shared between the left and right channel - today plan to split that and wire closer to the pins of the chip itself. Hope you like the plots of PSU noise
   	Posted 27th February 2013 at 12:51 AM by abraxalito

5. Nice analysis ... :) There's lots to optimise in all that; I've used many approaches and techniques over the years, but the one thing that's stood out is the tremendous benefit that accrues from improving the quality of the power rails. If people heard the quality of my bass, in terms of tightness and gutsiness, they would fall off the chair if they comprehended that it was all being done with 20W and 50W chip amps ... Frank

   Posted 27th February 2013 at 02:49 AM by fas42

6. Its the bass area (only) where the PSRR of your traditional LTP architecture chipamps score over the one I'm using, since the open loop gain is relatively huge the PSRR even referred to the output can be around 80dB in theory. Mine tops out at 60dB so I'm attempting to make up for that with more microfarads. Next step up is some hexacaps I'm building out of 2,200uFs. Have just measured them at 8mohm ESR at 100Hz, down to 6mohm at 10kHz.

   Posted 27th February 2013 at 08:53 AM by abraxalito

7. 	Making more solid progress, good to see -- how are the dynamics now, can you push the amp to the point of clipping or even beyond without loss of tonality? And, what about sibilance, your old bugbear? The improvements along the way should have helped in that area ... Frank
   	Posted 7th April 2013 at 04:57 AM by fas42

8. Dynamics are to die for, fully befitting the name 'Nitro' - I hadn't realized how good the Ozone DAC was previously, improving the amp hasn't revealed any DAC limitations so far. Since the amp is only 10W per channel, I've been running it flat out since day 1, with digital full scale set to the clip point. Notice no change in tonality when I adjust the digital gain (on the DVD player feeding the DAC). I waved goodbye to sibilance last year, or was it two years ago now? That goes when LTPs are abandoned or when the LTPs are protected from HF hash. Curiosity now is on how many uF can be added while improvements in bass 'slam' are apparent. Currently about 30,000uF per channel.

   Posted 7th April 2013 at 09:40 AM by abraxalito

9. A good path of discovery you're on then! My adventures have always showed that there appears to be no limits to how good the sound can get -- which is why I use the phrase "there is no such thing as a bad recording". When the very "worst" album you've got finally reveals itself to be capable of creating a "magic" experience then you've pretty well got full control of the machinery ... The bass 'slam' works better when the supply is effectively stiffer - I've tried several approaches: large scale paralleling of small cap's; regulation in the fashion you recently tried; and most recently cleaning up the waveforms within the stages of a conventional linear supply. All have delivered the goods to a satisfactory degree, combining the techniques would probably take it to "the next level" -- something I've not yet tried ... You've monitored the supply "noise" while driving the system with a real signal -- I would be aiming to minimise that level of modulation by whatever means possible ... Frank

   Posted 7th April 2013 at 12:16 PM by fas42

10. I've still found 'bad' recordings to remain steadfastly 'bad' despite improvements to the amp. Was only this evening listening to one (Decca, Haydn Masses) - not at all 'bad' in a musical sense but plagued with IMD, presumably at the mic front-end, with spitty sibilance on voices and general top-end grittiness. What I've noticed in adding on electrolytic caps is that it matters where they're connected - with caps distributed along the wires between the power source and the amp chip, the soundstage isn't so deep. With caps connected at only a single point, the soundstage is surprisingly deep. I haven't figured out why this should be yet, but I shall soon be adding on more caps at the same 'star point' for capacitor connection.

    Posted 7th April 2013 at 04:17 PM by abraxalito

11. Hmmm, "top-end grittiness" to me says that there is still some "weakness" somewhere in there, there is some type of interference getting into the picture sufficiently to be audible. All my experience over the years has taught me that the last remaining "bad" recording is giving me clues as to the fact that the system has not yet been sufficiently optimised, that the difficult musical passage is a giveaway that more can be done. Have you recently tried experimenting with completely shutting the house down electrically, so see if that makes a difference? All mobiles, etc, off? The capacitor behaviour says to me that transient current draw is having significant impact on the amp, that PSR thing still. The parasitics of the wiring and caps when spread out change the shape of the current pulses as seen by the amp pins sufficiently to be audible. I would try recording the voltage waveform at the supply pins with the caps in the 2 states with the same music and compare, see if that says something ... Frank

    Posted 7th April 2013 at 10:33 PM by fas42

12. Sure there's ntereference getting in, it was getting into the equipment those guys used to record. I still haven't got an understandable explanation from you Frank as to how the interference in my apartment knows when I'm playing a 'bad' recording and only shows up then. Mobile noise does show up sure, uncorrelated with the recording quality - that'll have to wait for a complete re-build of the I/V front end on the next-but-one incarnation. That's not to say more can't be done, the single point capacitor banks will be on the next build... Can't follow your words about transient current draw - I'll not be comparing measurements on the supply noise in the two cases - don't want to go back to the poor soundstage!

    Posted 7th April 2013 at 11:44 PM by abraxalito

13. Of course there are poorer recordings in terms of the encoded or embedded distortion, you only have to listen to an "audiophile" recording to see how you can eliminate every shred of that happening on the "other side" ... but that's not the point of the exercise. I'm relying, and in my experience it works for others too, on absolutely minimising any emphasis placed on the recorded distortion by the playback process -- psychoacoustics in other words. So, different recordings will always have different degrees and styles of embedded distortion, but [B]my experience [/B]has been that the extra distortion imparted by the playback mechanism can either enhance, or subdue, subjectively, that encoded distortion. So, system working at 98%, put on one recording - sounds fantastic ; put a different style of recording - yuuck! The playback distortion is effectively intermodulating with the recorded distortion in the latter case to create an unpleasant listening experience ... I can say this because I've gone through this cycle over and over again; thinking that I have got to the heart of a recording and it's not so good after all, but then find or realise that my system is not as good as it can be, fix it, and that recording is then on song ... Why I mention the electrical shutdown is because that is still one of my important techniques to assess where the problem is; the recording doesn't sound too good, so switch off everything, and the SQ jumps major notches -- so, playback was the problem ... The current draw thing is in regard to the amp working in class AB, the +ve and -ve current waveforms are not nice things to look at when the amp is working hard, getting close to square wave shapes at output crossover points, lots of lovely high frequency harmonics in there. And this is what is stressing the capacitor banks and layout ... Frank

    Posted 8th April 2013 at 12:18 AM by fas42 Updated 8th April 2013 at 12:21 AM by fas42

14. Yes I agree that poorer recordings which already have an excess of distortion constitute a torture test for a system - that's because the more tones in the source material the more the IMD performance of the reproducing system is critical. Recordings which already have IMD embedded are harder not to distort further. So no disagreement there. But I didn't say the poor recording was an unpleasant listening experience - if it had been unpleasant I'd not have listened through to the end of the disc, I'd replace it very quickly with a more pleasant sounding disc. But a recording with embedded IMD is never going to sound 'on song', just as a DSD or 1bit recording never will...

    Posted 8th April 2013 at 12:38 AM by abraxalito

15. And that then comes down to the individual - how they listen to music in general. I enjoy the "texture" of music and sound a great deal; it imparts a sheen to the air so to speak, and that's what I particularly tune into at the moment. So, I can listen to to the most musically "nothing" pieces if they inject the right tonal balance into the space -- I'm thinking here of some extreme elevator music I have, ambient twiddling, designed to be acoustic wallpaper. Yet, muso's can't help themselves, they will inject subtle nuances into material when they don't have to; when the volume is turned up on these CDs all sorts of interesting sounds and effects become apparent, mastered at very low volume compared to the main "theme", these would be completely missed by 99% of people listening to the recordings. My wife, whose father played violin in a local orchestra, is very sensitive to poor string tone; at a live chamber music recital, while I'm enjoying that live "texture" element, she will be increasingly irritated by the intonation and instrument quality of the strings, will be glad when it's over. So when she approves of a CD playing chamber music at live levels I know I'm pretty close to the "zone" ... :) Frank

    Posted 8th April 2013 at 01:23 AM by fas42

16. Beautiful, just beautiful!! I've just enjoyed a great belly laugh seeing your latest 'structure' ... I've got a few of those hiding in dusty corners of my work boxes ... ;) Great for playing games of "Amplifier, find the amplifier! ... it's here, sir, hiding in a dark back corner of the power supply room ..." Of course, this points out the absurdity, of how most power supplies in amplifiers are hopelessly too skimpy for the job they're supposed to be doing ... and, how much clean information is buried in those 'dismal' recordings people disparage ... Frank

    Posted 9th April 2013 at 11:20 AM by fas42

17. Thanks - I felt the picture made it look even more beautiful than it is in real life - perhaps the effect of the camera flash making the colours more saturated Since that pic I've more than doubled the capacitance per channel (75,000uF up to 170,000uF) and can't say I prefer the sound with the larger number of caps. I did not like the sound as much though with the bigger 68,000uF caps added. So I reckon the limit of what I can hear has been reached, at least with this layout of caps - which I admit isn't optimized. Back to the DAC now for the next tweak, the power supplies for the I/V stage are due for an overhaul

    Posted 9th April 2013 at 02:48 PM by abraxalito

18. Yep, this is where some fine analysis of what the power supply modulations are about may be helpful; in essence it's optimising the high frequency impedance, while minimising bass signal sag, to suit the amplifying module -- adding more or "wrong" type of caps, in ways where the parasitics of the connecting matrix add too much to the mix will eventually reach a balance point for the overall setup. My DIY gainclone was very much a 3D exercise in sculpting of the bits to absolutely reduce the parasitics I was aware of to a minimum -- far less capacitance then what you used, but probably more stategically organised ... Frank

    Posted 10th April 2013 at 12:20 AM by fas42 Updated 10th April 2013 at 12:25 AM by fas42

19. Strategic organisation of capacitance is indeed planned for the future - but at present no strategy has presented itself to me. An initial rule-of-thumb seems that an amp needs about 50,000uF per amp of rated output current - mine is rated at 2A. That Boulder amp I think even with an 8R load is rated almost 10X as much so should come with 500,000uF just for its 8R rating. But they market it as being capable of 2R loads so that adds a factor of 4, bringing us to 2F. However there are other factors to take into consideration - my supply is fed by a linear reg, their's is fed direct from a transformer. They may well have achieved a better PSRR by regulating the small signal stages though, leaving only the output emitter followers fed from the noisy rails. I tried a quick LTSpice simulation to see what the PSRR of an emitter follower output stage was but couldn't find any decently modelled bipolar transistors in the library and gave up. TI's LME49600 has about 75dB at LF - if this is representative of a single pair of emitter followers, presumably the PSRR is going down 6dB for each one in a paralleled output stage?

    Posted 10th April 2013 at 01:00 AM by abraxalito

20. As an aside, this is an interesting description of what you have to deal with all the time in systems working at the high end level: [URL="https://www.whatsbestforum.com/showthread.php?10195-The-audio-journey&p=185430&viewfull=1#post185430"]The audio journey[/URL]. The poster's experience with the "better" cables swapped in exemplifies the lack of understanding that this industry is riddled with -- why was the sound changed so much by the change of cable? The usual explanations are of course nonsense, but something real is happening, so what is it? As very rarely acknowledged, the original cables were allowing interference effects to inflict themselves severely on the system. Or, the new cables were sharply attenuating interference, perhaps above audio frequency distortion, being passed down the chain from component to component. Nothing more than this, the "better" cables were simply allowing the parts of the system to work more correctly than otherwise. Frank

    Posted 10th April 2013 at 02:32 AM by fas42

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