Hi all,
I am about to build a new stereo power amplifier with two integrated circuits. I know that it is better to use two transformers with two individual rectifier circuits, but i would like to avoid the cost of a second transformer. I am wondering if I will get better stereophonic figure by using one transformer and two seperated rectifier circuits in order to have seperated grounds for the two individual channels, instead of using one transformer and only one rectifier circuit for both channels. I mean, will the difference be detectable?
I am about to build a new stereo power amplifier with two integrated circuits. I know that it is better to use two transformers with two individual rectifier circuits, but i would like to avoid the cost of a second transformer. I am wondering if I will get better stereophonic figure by using one transformer and two seperated rectifier circuits in order to have seperated grounds for the two individual channels, instead of using one transformer and only one rectifier circuit for both channels. I mean, will the difference be detectable?
I went down this path with my LM3886, I think it is a good middle ground. Provided that you have an appropriately sized transformer and a reasonable amount of capacitance in each channels separate rectifier/cap bank then I think that the result should be better stereo separation. (based on the premise that the modulation of the power supply by each channel will be isolated to their own individual cap banks), whether this premise stands up or not I'm not sure.
I did not however have separate grounds choosing to have the star ground common to both channels (but all returns separate).
As I did not build a single supply I can't make comments as to whether there's any audible difference (RMAA measurements should easily show if there is a measureable one).
I am going to rebuild the power supply for my other amp (which does have a single rectifier and cap bank) and converting to a rectifier and cap bank for each channel, effectively doubling the capacitance and will be taking measurements before and after, however with the speed I get around to things I wouldn't hold your breath waiting for the results
Someone else who has actually tried both, listened and measured might chime in
Tony.
I did not however have separate grounds choosing to have the star ground common to both channels (but all returns separate).
As I did not build a single supply I can't make comments as to whether there's any audible difference (RMAA measurements should easily show if there is a measureable one).
I am going to rebuild the power supply for my other amp (which does have a single rectifier and cap bank) and converting to a rectifier and cap bank for each channel, effectively doubling the capacitance and will be taking measurements before and after, however with the speed I get around to things I wouldn't hold your breath waiting for the results
Someone else who has actually tried both, listened and measured might chime in
Tony.
I haven't measured it, but when I moved from a single rectifier to a separate one for each channel I got a very noticeable improvement in sound quality, and not just the imaging. I also ended up halving the cap bank from 80,000uF for both channels to 40,000uF to each channel because I didn't add capacitance, and did not lose much of the bass.
I went down to 20,000uF per channel later without further loss of impact, whereas with a single rectifier and 40,00uF (my first attempt) the sound was still a little lifeless and would grind down at higher output levels.
I went down to 20,000uF per channel later without further loss of impact, whereas with a single rectifier and 40,00uF (my first attempt) the sound was still a little lifeless and would grind down at higher output levels.
I'm not so certain that it is so much better to use two transformers. One big transformer will have better regulation than two small ones. Also will be cheaper, alternatively the same money can be spent to get a more powerful supply.
I don't think crosstalk between channels is really a serious issue when running both left and right from the same power supply. Neither does Doug Self in his power amp book. Its true that grounding might be simpler with two separated supplies, but this issue is really very minor in my experience. I'm running a stereo pair of chip amps at the moment very happily from a single supply and haven't encountered grounding problems. I have compared this single supply system with a similar set-up using separate supplies and the differences are inconsequential.
Depends on your design targets. Consider the LM3886 as an example as it's one of the better chipamps around. Avol at 20kHz is 50dB so if its operated in a typical configuration with 30dB gain there's only 20dB excess loop gain available for error attenuation. If you assume an unregulated common supply for two channels with 20,000uF supply ripple will be about 12dB below the signal level. Vee PSRR at 20kHz is about 48dB. So, if the left and right signal levels have a constant power spectral density and are within 10 dB or so of each other (which in my experience is typically the case), then the channel separation is 48dB + 12dB + 20dB - 10dB = 70dB, or about 0.03% IMD. However, 1/f is a considerably better approximation of music's power spectral density so, if you start analyzing some slice of midrange or tweeter bandwidth as an IMD victim from the rest of the spectrum as lowpass filtered by the supply, that margin typically reduces by 20 to 40dB. 0.3% IMD has been shown to be audible and undesirable, 3% is pretty poor.
One can always run different analysis and come up with somewhat different numbers, but the finding getting a clean trebel is harder you might think has been a consistent one in my experience. The LME49810 makes better numbers in this regard than any monolithic chipamp I know. If you start looking at biamping or triamping with the tweeter channel as a victim operating the control loop on op amps like the LME49990 or LME49724 ends up being attractive for their higher Avol over the LME49810.
Work through the math above for your design options and then you'll know. The advantage of separate rectifiers is the ripple in the victim channel tends to be more spectrally compact so the IMD floor ends up being lower. However, every time I've worked through this I've ended up reaching the conclusion the extra USD 15 or so it costs to buy two smaller trafos and an extra bridge looked like cheap insurance.
Another advantage of paired trafos is you can mount them with opposing windings and get some amount of leakage flux cancellation; chipamp circuits tend to be compact enough they're not too susceptible to noise pickup from the trafos but I've seen amps hit surprisingly bad noise problems due to cap charging transients being being reradiated from transformers and picked up within input stages or the feedback loop. Shielding low frequency magnetics is hard, so getting a few extra dB of attenuation from field cancellation is a boon.
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I concur. One cure (IMHO) is good decoupling, I usually use 100nF SMD ceramics right at the chip pins - P2P, not PCB so they're actually *on* the pin - and a 100uF Panasonic FM at the PSU entry points. Even so, the treble is not as 'fast' and clean as say my Pass Labs F5, which has even less PSRR (but a more beefy power supply). In the case of an AC-coupled chipamp, the capacitors at the input and Ci position play a pretty important role IME. I find very good results with a Panasonic FM at the Ci position, far better than the exotic Black Gate NX and Nichicon Muse that I've also tried there, specially for clean treble (unfortunately it is still a little behind a good discrete though far ahead of the cheaper Class D amps and most commercial amps under $1000).
In my case, if using a center-tapped transformer where stereo grounding becomes a real nightmare with just one transformer and two rectifiers. Often the best option is to relocate the star ground to the location where the power supply lead enters the chipamp, decouple it really well with low-ESR caps, and keep the ground paths really short, with a very thick ground wire back to the PSU caps. I still think dual mono is the better configuration.
About flux fields - I've used a LM4766 an inch away from a EI-core transformer without any hum being picked up at all. Maybe just lucky, I guess, or maybe not sensitive enough hearing
Agreed. I think however there is more to dual supplies than crosstalk. I hope to get some test equipment soon, then I can have some measurements between dual mono and dual rectifier, but IME the dual mono just sounds a lot better!
So far, so good, I'm following your argument.
But here I start to get lost. For a 50W/8R amp, our peak signal level is 56V p-p (20VRMS). 12dB below this is about *0.25 so 14V p-p. I've never seen such high ripple on the power supply in any amp I've designed. A 14V drop in 8mS from a 10,000uF capacitor (remembering I = C * dV/dt) implies a current draw of 17.5A whereas the peak current for 50W/8R is 28/8 = 3.5A per channel, 7A total maximum. So somewhere these numbers aren't stacking up. Secondly, the ripple will be at low frequencies, smallish multiples of the mains, and will fall with increasing frequency because the res cap's impedance is falling with frequency too.
This part about IMD I don't follow either. How does PSRR translate into IMD? You're saying what's on the power supplies will intermodulate with the signal, rather than just adding to it?
I agree that a clean treble doesn't come so easily with chipamps, but my experience of that is that its caused by poor grounding and decoupling, not by interchannel crosstalk. I did have lots of treble grain on my active speakers (single channel per PSU - see my blog) but I fixed this by re-grounding.
yes - that's peak to peak. with 34V rails, we have both -34V and +34V so if the amp could swing all the way to the rails we'd get 68V p-p. But it can't, so in practice with 34V rails we get around 62V p-p. 56V p-p is just a conservative figure because in practice we get a little more than 50W into 8R on bursts.
Right, I know in theory that's what it should be but I rarely see anything above 14V before the sound starts to crack up real bad. Mostly 11-12V, though most of real listening its 7-8V. I assume these are RMS figures as they come off a DMM. I'm just not sure it can actually manage 20VRMS output cleanly.
Square, actually, but what I do is I set the level to what the maximum comfortable listening level is, then retain the same volume setting and run a signal gen into the amp input. The source is a PC, so it's pretty easy to do this without changing anything.
Maybe building a small attenuator and make a PC oscilloscope is a good idea now that you mention it.
Maybe building a small attenuator and make a PC oscilloscope is a good idea now that you mention it.
Sounds like a good idea. I don't have a PC handy near my amps so I recently got a Sony PCM-10 flash-card recorder and use this (with a 200:1 input attenuator) if I need to check distortion by looking at the recorded file in Audacity.
I'd assumed a 4 ohm load and 50Hz mains with the usual small signal ripple model for a full wave rectifier. From there applying Ohm's law and a little math gives you
Vripple / Vsignal = 1 / (2 f C Zspeaker)
where f is the mains frequency, C the reservoir capacitance supplying the output signal, and Zspeaker the impedance presented by the speaker+and passive crossover+whatever in a power spectral density weighted sense. With this model the ripple on 8 ohms at 60Hz should be around 20dB below signal, though your case is probably coming up against the limits of the small signal model's validity range.
The above assumes a constant current draw over the discharge and using peak to peak ripple so it's a bit conservative; adjust as needed for sinusoidal current draws, different sorts of transients, or different definitions of ripple. I'm guessing you're using sinusiodal draws which are symmetric about 0A. In which case the measured ripple'd be more like 23dB down. Might be another +/-3dB in there depending on how exactly our capacitance accounting lines up.
I think this gets into the mental model one uses to think about what's going on. We both agree the supply caps produce ripple that's a down modulation of the signal and the amp ends up with an output error that's the ripple attenuated by PSRR. Seems to me it's equally valid to think of the output error as additive noise or as a modulation of the input signal.
I should go back and check the PSRR measurement conditions on the 3886; it's possible the supply induced output error's attenuated only by PSRR and not excess loop gain, in which case I've overestimated the part's performance.
Do you have data on what the noise levels introduced by the grounding problems were? The 3886s I've listened to and measured haven't been particularly great either subjectively or objectively---roughly 0.5% THD at typical home audio tweeter levels in biamp, input stage limited. It's certainly possible the ground problems were above that level and interchannel crosstalk through the supplies is below that level, in which case performance is not supply limited and it'd make more sense to invest effort in changing to a higher fidelity part.
How much bypass capacitance, what mains frequency, which chipamp (in my limited experience the 3886's somewhat unsual in its ability to drive close to the rails), and is the test being done within the DMM's passband? Posting the schematic wouldn't be a bad idea.
twest: I use the reference schematic, with 22K/1K feedback resistors, a 470uF Ci cap, DC-coupled input and no zobel (tried, makes no difference).
PSU has 10,000uF per rail per chip, separate rectifiers and separate cap banks but a single 48VCT transformer.
I use a 1KHz square wave, so I guess it should be within the DMM's range.
Mains here is 230V/50Hz, give or take about 20% tolerance depending on the power company.
PSU has 10,000uF per rail per chip, separate rectifiers and separate cap banks but a single 48VCT transformer.
I use a 1KHz square wave, so I guess it should be within the DMM's range.
Mains here is 230V/50Hz, give or take about 20% tolerance depending on the power company.
To me, addition of noise isn't really the same as modulation of the signal. But I'm not sure that the difference is important here, we both agree the noise is unwanted. The result of modulation though is unwanted signals at different frequencies.
No, I don't have equipment designed to measure that stuff. But even if I had an AP, I'm not sure how I'd go about measuring it. I did try with just a PC soundcard and didn't see anything different in the two cases.
You're saying the LM3886's input stage is causing 0.5% distortion? Sounds to me like you've got a serious circuit problem or a half-dead chip.
Naw, it's right in line with what you'd expect from the datasheet. Typical output levels on a biamp's tweeter channel for home audio are around a hundred microwatts---with 90dB efficient drivers the total power per speaker is a few milliwatts to deliver the slightly loud 65dBish average SPL many folks tend to listen at---and the tweeter power tends to be 10-15dB down from the total channel power. The THD+N curves in the 3886 datasheet are in good agreement with Douglas Self's rule of thumb crossover distortion triples for every order of magnitude reduction in power. If you project past the datasheet's 10mW limit the results are in good agreement with the measurements I mentioned.
Also, if you look at the equivalent circuit of the 3886 you'll see it's not actually an op amp and hence has a few limitations around the input. tomchr and I found if the part's operated class XD there's no change in distortion, which confirms the problem is a control loop limitation and not crossover distortion in the output devices.
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Those are THD +N curves - to me, they look to be dominated by noise, not distortion.
I've read Doug Self's earlier book and this rule of thumb did not ring any bells. So I've had a peek in his latest edition, and found this:
reducing the output power from 25 W to 250 mW, which is pretty drastic, only increases THD percentage by six times
That's on pp178,9 of his 5th edition. So there's a two orders of magnitude reduction only giving six times. So we'd expect a bit under 2.5X, per order of magnitude, not quite as much as tripling. He also mentions a bit earlier in the chapter that at lower levels, the harmonic structure tends to be skewed away from the higher order ones so at lower levels the distortion becomes more and more benign.
The thing to remember here is that Self's measurements were done on a deliberately underbiassed amp to maximise the distortion and one with considerably lower noise than a LM3886 with its heavy input LTP degeneration.
<edit> I forgot to comment on your earlier remark that its the input stage which is causing the distortion. Clearly not, any crossover distortion will be caused in the output stage. Self says those measurements made above only became meaningful once other distortion mechanisms present had been sufficiently minimized.
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Neither of these is in agreement with the measurement data I have. Do you have data which shows otherwise?
Post it up by all means. If you have measurement data which shows input stage crossover distortion that would be really something.
No, because I'm happy to rely on National's measurement data in their datasheet for the LM3886. Let's look at my earlier claim that at low level the THD +N figures are dominated by noise, given all that I've so far introduced.
I'll take the page 12 top right graph of 1kHz THD+N into 8R. Here we see the distortion plus noise figure touch 0.002% just before clipping, which is shown as being 39W. Let's assume initially that this figure is distortion dominated.
Applying Doug Self's six times more distortion for a hundred times less power rule once, we reach 0.012% at 390mW. Apply it again and we obtain 0.072% for 3.9mW. The graph reads 0.035% for 10mW. Extrapolating the graph to 3.9mW we won't get double the figure as the slope of the graph isn't steep enough at 10mW. At 40mW it shows 0.017% - multiply this by 2.5 and we get a figure short of 0.072%. So I conclude its noise dominated here even if it was distortion dominated just before clipping - which is debatable.
The story is rather different at 20kHz though, I'm not claiming that the graphs are noise limited there. Were you only referring to the 20kHz plots when you claimed the input stage was causing 0.5% distortion?
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