Listening Experiences.

My greatest problem was trying to find a pickup cartridge with good sounding treble. All those I tried sounded different, but all produced the same 'spitting sibilance' on vocals. Trying different pre-amp circuits and load impedances seemed to help very little if at all. Reviews which mentioned the 'smooth treble' proved to be unhelpful, and walking round a local Hi-Fi Exhibition and listening to the exhibits, they all seemed to have the same problem to some degree. Some people seemed to be happy with this sound. I found it more obvious when listening on headphones, and one effect of poor high frequency channel separation is that vocal sibilance can become disembodied, appearing to come from a different location than the rest of the voice. To me the best sounding of the cartridges I tried back then was a cheap Audio-Technica (AT6S?) with spherical stylus, which had tolerable treble but also sounded a little distorted and muffled.
It was many years later I saw a review in 'Hi-Fi Choice' for the Technics EPC205C-Mk3. The high rating given in that review was not the deciding factor, I had seen plenty of good reviews for previous disappointing purchases. What attracted my attention was the flat frequency response, well maintained channel separation to beyond 20kHz, plus a claim to have the lowest ever tip mass for a mm-cartridge, and also an unusually low inductance, making sensitivity to load impedance less of an issue. When I eventually bought the Technics cartridge the resulting sound was an astonishing revelation, it was vastly better than anything I had previously heard from my records, and even some discs which I had assumed were seriously bad recordings were now perfectly listenable. I still have a turntable, now a Pro-Ject 1 Xpression, but a replacement stylus for the Technics cartridge is practically unobtainable, apart from a few very expensive re-tipping services.
There are some cheap replacements not made by Technics and not identical, but possibly better than nothing, including an example from Stylus Plus, which was at one time available at £16. I thought this would just be a waste of money, but then I found one of these on eBay for even less so decided to give it a try. The sound is certainly not in the same league as the original stylus, but nowhere near as bad as I expected. It still sounds better than any of my other mm cartridges.

For comparison the next photo is my Technics cartridge next to a more typical example, the Ortofon VMS20E MkII, showing the difference in size of stylus and cantilever.

My first CD player was a Marantz CD273SE, and it is still working well. Some of the first CDs I listened to were disappointing, with what seemed like a very harsh sound, but others I thought sounded near to perfect. Listening on the Marantz, or on a portable Panasonic player or on the very cheap dvd drive on my computer makes very little difference, the bad CDs still sound bad and the good ones still sound good, so it seems recording quality is far more variable than playback equipment differences. The computer uses an EMU-1820m soundcard, and initial checks with a test CD show the computer player to be close to perfect. I have always liked the Panasonic player sound more than the Marantz, although I am uncertain whether this is a real or imagined difference, and now, April 2012, I have bought a Technics SL-PG390 (£16 on eBay), which uses the same single bit technology as the Panasonic portable, and tests show it to be already quite good. Attempts at improvement made little measurable difference, but I was already happy with the sound, and my modifications appear not to have done any harm at least, either audible or measurable.

Slew Rate or Half-Power Bandwidth?

The advantage of the bandwidth figure is that we can more easily use it to compare different amplifiers with different power ratings. If we require a 100W amplifier to produce its full power at 20kHz it will need a much higher slew rate than a 10W amplifier needs to produce its specified power at the same frequency, so we need to calculate the equivalent figure for different power ratings. The half-power figure gives a simpler comparison with no need to adjust for power rating.

There can be some danger in measuring high power output at very high frequencies, for example the resistors in the output Zobel network may not have adequate power rating for such a test, and there may be other problems such as cross-conduction in the output stage. The dangers can be reduced to some extent by using a toneburst signal. I only tested my MJR7 up to 100kHz because that is the upper limit of my signal generator, and there was no obvious reduction in maximum output or any other problem at that frequency.

DSP.

ABX

There are many different ways to conduct listening tests, but the most convincing are those which can separate real from imagined effects. There are of course reasons other than imagination why listening test results can be wrong. Unfortunately arguments about 'subjectivism' tend to degenerate into accusations that those on one side are liars or fools, while those on the other side are deaf. Maybe these accusations are sometimes true, but without reliable testing methods how would we know? Reliability however is unlikely to be easily agreed, those who 'know what they can hear' naturally will regard failure to confirm this knowledge as evidence that the test method is in some way faulty, while for positive results those who think these audibility claims are nonsense will blame experimental error, and again maybe sometimes these assertions are true.

Personally I prefer an alternative approach, having devoted some time to error extraction testing I know that signals below a certain level when reproduced alone are inaudible, for example anything below 300uV at my speaker terminals is below audibility at 1m distance for even a test subject with far better hearing than mine. The lowest audible level I found is for frequencies around 3kHz as expected. Anyone claiming to hear effects below this level as part of a music signal is in effect claiming that these effects become identifiable in the presence of music played at their normal listening level. This implies some sort of reverse masking effect. Testing for such an effect is easily done by keeping the 'distortion' signal separate from the undistorted music signal, including using a separate amplifier and speaker. The distortion signal can then be played alone and switched on and off to confirm that there are no audible clues such as background noise or switching clicks. It is then a relatively simple matter to determine whether any test subjects are able to repeatably identify whether the distortion signal is on or off when accompanied by much higher level music.
Surprisingly there has been one test reported where inaudible signals apparently did become audible when another signal was also played, but that was for a highly artificial situation with signals close together in frequency applied to separate ears and beats heard (Groen, 1964, Acta Oto-Laryngol. 57 p224-230). This appears never to have been confirmed by other researchers. A paper published in 1995 was entitled 'Failure to hear binaural beats below threshold' (J. Acoust. Soc. Am. Volume 97, Issue 1, pp. 701-703, 1995), and as the title suggests the researchers failed to replicate these findings, and no difference was found between detectability of the beats and the audibility of the weaker sinewave.

I mentioned ABX double blind tests in the previous piece. Negative results from a single test, even if the listener is known to have good hearing, prove nothing other than that the listener at that time and under those particular circumstances was unable to reliably identify a difference. That tells us very little about whether an effect can ever be audible to anyone, at most it is evidence that extreme pronouncements, e.g. that anyone who can't hear it must be deaf, are gross exaggerations. If many listeners known to have good hearing fail to identify a difference we may deduce at least that it is fairly difficult to hear. If possible it helps to repeat the tests with the difference increased to the point where it becomes obvious and then decreased in steps until it is no longer distinguishable, this provides some practice and also demonstrates what effect is being investigated to help the listeners concentrate on the right thing. An example was featured on a BBC TV program about a professional violinist, to demonstrate that she could identify small distortions in violin music far better than the other listeners tested.

Even repeatable positive results are not necessarily correct if the same incorrect procedure is used. Another example I read about is an experiment comparing a 10kHz sinewave with a 10kHz squarewave. The only difference should be harmonics at and above 30kHz, which appear to be inaudible when produced alone. The experimenter explained that to ensure the 10kHz components were at the same level the peak voltages at the speaker input were set in a ratio 1.273 for the two test signals. The calculation appears to be correct, but one important source of error is being overlooked, and again this is thermal compression. The square wave with its additional harmonics causes a significantly higher dissipation in the speaker voice coil, which increases its temperature more and therefore increases its resistance and so reduces its output. To achieve the usual requirement of 0.1dB level matching additional adjustment may be needed. A better way to carry out this experiment would be to use two separate drive units, one with a continuous 10kHz sinewave and the other with only the harmonics needed to produce something as close as practicable to a square wave. This second signal could then be checked with the sinewave absent to ensure there is nothing audible such as background noise or switching clicks. If it is inaudible when heard alone but its presence becomes identifiable with the sinewave switched on, then this would appear to confirm a real audible effect. Maybe someone has tried this approach, which could be applied to many high frequency or sub-threshold effects, but I have so far found no examples.
It may be that a 30kHz component of a square wave really has some audible effect, but unless all alternative explanations, no matter how improbable, have been identified and excluded by further experiments then the conclusions must be considered unproven. I have suggested one explanation for this example, but there may be several more, for example intermodulation in the ear could produce a 20kHz component as the difference between the 10kHz and 30kHz components. A possible test for this would be to repeat the experiment at a range of square wave frequencies, e.g. from 7kHz up to 15kHz to find the upper limit for a positive identification of the harmonics. If 10kHz is close to the upper limit then an intermodulation effect becomes more probable. If it continues far beyond 10kHz then audible components within the conventional audio band are discounted.

Open-Loop Distortion. How low does it need to be?

Compare TMC-TPC

A while ago I bought Bob Cordell's excellent book 'Designing Audio Power Amplifiers', which gives examples of two pole compensation (TPC) and transitional Miller compensation (TMC) on pages 178 and 182, and I tried a simulation to compare these methods. The results are not exactly what I expected, at very high frequencies I thought the two capacitors in series would need to have the same value for the two circuits to have matching loop gain, so finding that the TPC version needed far lower capacitance than expected was puzzling. I don't see anything wrong with the simulations, so maybe it is correct, but I have never seen this mentioned anywhere, so I can easily believe I made a mistake somewhere. Using 20k in the feedback network in one circuit and 2k in the other may explain some or all of this difference. What may seem odd is that I have connected R3 (2k) to the nominally 1V 'output' rather than the actual output at node 5, which may look wrong , but connecting the other way the observed 'loop gain' no longer is second order, and attenuation starts at a far lower frequency. The connection shown gives the 'right answer' for the loop gain, but it takes a little thought to see why.

Initially, using AIM-Spice, I got a series of error messages, possibly because the dc gain was infinite, and I had to add 100k resistors to earth from both input and output of the VAS to get something like the TPC plots in the book. The TPC capacitors were then reduced to 25p, and a 0.035pF added between input and output of the VAS to match the peaks in the responses. Only small differences remain between gain and phase, showing that output stage distortion reduction will then be about the same for both methods. An optimum design of a real amplifier may of course lead to different conclusions, but these simplified circuits suggest that the only significant difference is the smaller capacitors in the TPC which will require less current from the input stage, and so could be an advantage.

Here is the simulation using AIM-Spice:

(In the first diagram for clarity I have left out the 100k resistors from input and output of the VAS to earth and the 0.035pF capacitor.) The gm signs look wrong, but that is how the voltage controlled current sources work, the current is into the output terminal not out. The 1V is the output of the output stage, and the gain and phase shift round the feedback loops are found by observing V5 (TMC) and V7 (TPC). There are of course other feedback loops, but it is these loops round the output stage which determine its distortion reduction. Both TPC and TMC are simulated together as a single diagram so that the results can be seen together on the same graph for direct comparison.

TMC-TPC-Compare.
VIN 6 0 AC 1
R1 1 0 1k
R2 1 6 19k
R3 3 6 2k
R4 2 0 100k
R5 4 0 100k
R6 11 0 1k
R7 11 6 19k
R8 9 0 20k
R9 8 0 100k
R10 10 0 100k
C1 2 3 36p
C2 3 4 150p
C5 8 10 0.035p
C3 8 9 25p
C4 9 10 25p
E1 5 0 4 0 0.96
E2 7 0 8 0 0.96
G1 2 0 1 0 -0.002
G2 4 0 2 0 0.04
G3 10 0 11 0 -0.002
G4 8 0 10 0 0.04

The red traces are TMC and green traces are TPC. At the top is the gain, and below is phase, or to be more precise, excess phase beyond the 180deg resulting from the inversion in the feedback loop.

The small differences could probably be reduced further by a few small adjustments. It appears there is no great advantage in either version, but both have the same dangerously high phase lag in excess of -170deg over a similar wide frequency range.
For comparison here is a recent result for the MJR7-Mk5 feedback loop phase response, first with 7R5 load:

The phase lag reaches only -110deg, which may seem excessively cautious, but now look what happens when we add a 2uF load:

The phase now dips a further 60deg to reach -170deg around 100kHz. With different capacitor values the maximum phase lag will be at different frequencies. Imagine now what will happen to the TPC and TMC examples in a real amplifier with a capacitive load. The same 60deg increase in phase lag would take the total far beyond -180deg and stability is then conditional, requiring that loop gain must not fall too far, as it will normally do near clipping, near slew rate limiting, and during switch-on or off. The effect of capacitive loads will not be the same for all amplifiers and so the effect may be better or worse than this example. It depends on several factors including open-loop output impedance and the value of the output inductor and its damping resistor.

Dielectric Absorption

A common example used to illustrate the effect of dielectic absorption is to charge a capacitor by connecting it to a constant voltage source for a few minutes, then discharge it by connecting a small value resistor across its terminals just long enough for the voltage to fall close to zero. Then, having removed the resistor the voltage is monitored by a high impedance voltmeter, and is found to slowly drift back up to a significant fraction of the original voltage. If the original charging period had been as short as the discharge period then the upward drift would not happen, or at least nowhere near to the same extent. From this we could conclude that the capacitor is in some way reluctant to accept a full charge, and then continues to store it long after it appears to be discharged, only releasing it at a later time. This entire behaviour can be produced using ideal capacitors with no dielectric absorption by simply adding a second capacitor with a large value series resistor in parallel with the main capacitor, as in the next diagram:

The parallel RC values are not necessarily correct for any typical capacitor, for example the values for a 1u mylar type are given by Bob Pease as 0.006u plus 1000M, and his more accurate equivalent includes several more parallel RC combinations. A single RC is however sufficient to explain the basic behaviour of a capacitor with dielectric absorption.

Suppose both capacitors are initially discharged to zero volts. Moving the switch to C (Charge) a high current will flow through the 1R resistor and the 1u will charge with a time-constant of 1usec, and it will be close to the full 10V after 1msec. The 0.1u capacitor however has by then barely started to charge because the 10M resistor only allows an initial charging current of 1uA, and after 1msec it will only have reached 10mV. If we instead leave the switch at C for a few minutes then the 0.1u will also become charged close to the full 10V. If we then move the switch to the discharge position, D, for just 1msec, then the 1u will discharge back close to zero in that time via the second 1R resistor, but the 0.1u will barely change at all in that time, it will fall by only 10mV, down to 9.99V. With the switch now open we have a 0.1u capacitor charged to almost 10V connected to a 1u capacitor at 0V via a 10M resistor, and current will flow back into the 1u, slowly increasing its voltage and reducing the 0.1u voltage, until both capacitors are at the same level, about 900mV.

That is about all there is to dielectric absorption, at least if all we need is to understand the practical effect in electronic circuits. There is not actually a separate resistor and capacitor inside a capacitor with DA, this is just a useful way to simulate the effect, and also it appears that capacitors with high levels of DA often, but not always, have higher nonlinearity than capacitors with low DA, which suggests that the actual mechanisms within the dielectric material may in some cases include nonlinear effects. For most capacitors the nonlinearity is very small, there are only a few types such as high-K ceramics which add unusually high distortion.

Capacitors used in amplifier circuits can be divided into two groups, those used for signal coupling purposes, which should have very little signal voltage across them, and those used for filtering and frequency compensation, which may have much higher signal voltage across them, and therefore can with some advantage be chosen as types such as polypropylene with low DA and very low distortion even at high voltages. For coupling applications DA is less of a problem, and the only significant effect appears to be a reduction in unwanted phase shift compared to ideal pure reactive components.

From a practical point of view this is all very convenient, because filtering and frequency compensation capacitors are usually of relatively low value, and polypropylene types are then both cheap and of small physical size, which helps avoid interference pickup. Input coupling capacitors are often of several uF, and then polypropylene types are both large and expensive, but fortunately unnecessary, and smaller and cheaper types such as polyester can work as well, if not better. In one of my own tests a large polypropylene picked up 12dB more interference across the audio band compared to a small polyester. This I suggest is more important than whether the third harmonic distortion added at 20Hz is at -140dB or -160dB.

Peak Current Requirement of Speakers

There is a widely quoted claim that the peak current requirement of a typical speaker is far higher than we would expect from its minimum measured impedance. Some experimenters have done real tests with music and speakers and found that current peaks are unexceptional compared to the voltage peaks, but even so it is interesting to try to explain why the theoretical prediction is either wrong or at least irrelevant to music signals. The explanations I have seen for the current peaks mention the effect of 'back-EMF', but this is already accounted for by the inductive component of the impedance. Assuming linearity, for any individual frequency the current is determined entirely by the impedance at that frequency, and any more complex signal can be analysed as a sum of different frequencies, and the individual currents added, taking into account their phase angles in the usual way. So if each frequency has the current expected from the speaker impedance plot, where do those predicted high current peaks come from?

To see the problem from a different direction, consider a periodic signal voltage, so this involves just a fundamental frequency plus its harmonics. Keeping the amplitude of each harmonic constant but adjusting their relative phase we could ensure all frequency components are at their maximum positive voltage at the same time, and then the total signal would have its maximum possible peak at that instant. When applied to a speaker the frequency dependent reactive impedance ensures that the frequency components of the current are shifted around in relative phase, and are not then all at their maximum at the same time, so the current never reaches its maximum possible peak level. We could, instead, adjust our test signal so that it is the frequency components of the current which are all at their maximum at the same time, then we will get the maximum possible peak current for this set of frequency components, but no longer have the phases we needed to get the maximum voltage peak. We can then observe a ratio of peak current to peak voltage which if we assumed them to be directly related might suggest that the speaker impedance is far lower than our conventional minimum impedance specification.

Real music signals however include many different frequencies, some being harmonics and others unrelated, which will invariably combine to give peaks in both speaker voltage and current, but at different times unless the speaker is a pure resistance, and it is these we need to compare because in practical use the amplifier gain will, or should, be kept low enough to avoid clipping of the voltage peaks, and then hopefully this will also keep the current peaks adequately low. Using a test signal designed to have high current peaks without high voltage peaks allows us to turn up the volume further before voltage clipping compared to a signal with the same frequency components but with voltage peaks, and so gives a false impression, not applicable to a typical music signal. It would of course be possible to include a section of this test signal close to the voltage clipping level in an otherwise normal piece of music, but to do so would be perverse to say the least, and it is hard to believe any music lover would find the result enjoyable, irrespective of whether or not their amplifier could supply the current demand.

A simplified calculation: Including harmonics up to the 29th, the peak amplitude of a square wave can be increased by a factor of 2.97 just by adjusting the relative phase of the harmonics. An example in Ref.1 has a load with sections of a 28V square wave applied, which if we take the average impedance as 8R would suggest a peak current of around 3.5 amps. An adjustment of the harmonic phases to maximise the peak current level then gives a peak current 2.97 times higher, i.e. 10.4 amps, near enough to the 10 amp peak shown to occur. The calculation method is not entirely correct, we should use the actual impedances at each of the harmonics rather than a very approximate average, but is good enough to suggest that the phase shifts of the harmonics of the current are able to account the observed current peaks. The factor of 2.97 is not a maximum for all functions, it applies only to a continuous square wave including only harmonics up to the 29th. If we included all harmonics up to infinity we could in theory get infinite peaks of zero duration. See reference 5 for more details.

It is easy enough for an individual to check the maximum current they need, just by adding a small value resistor, e.g. 0.1 ohms, in series with the speaker and looking at the peak voltage across the resistor using an oscilloscope. With my own MS-20 speakers playing typical commercial cd recordings at the highest level I would normally use the peak current was around 500mA. With less efficient and lower impedance speakers, a bigger listening room, or music with very high peak to average ratio, and for those who like their music loud, levels well above this figure may be needed, but even my 'low power' MJR7 amplifier has at least a 23dB margin relative to the 500mA level. There may be commercial reasons for amplifier manufacturers to suggest that hundreds of watts and extremely high currents are needed even for typical domestic use, but instead of just accepting this I recommend checking the real level needed for any given application.

Choosing Headphones

I wanted to buy some new headphones, but have problems finding reliable comparisons. Of course if it was possible to try a few of the short-listed items even for a few minutes this would at least narrow the search, even accepting the 'burn in' idea, but the models available for audition at local stores tend to be the fashionable and expensive variety, and those I tried proved seriously unlikeable to me, not even close to my old PX100s. Subjective asessments are sometimes difficult to take seriously, and separating the good from the bad can be a problem.