Their expertise lies in making coffins for monkeys
As always, a pleasure to read John Watkinson; his books on digital audio were my bible for years.
Thanks, John, keep up the good work!
Today’s loudspeakers are nowhere near as good as they could be, due in no small measure to the presence of "traditional" audiophile products. In the future, loudspeakers will increasingly communicate via digital wireless links and will contain digital processing. Indeed, the link between IT and loudspeakers is destined to grow …
But no progress can be made when science is replaced by bizarre belief structures and marketing fluff, leading to a decades-long stagnation of the audiophile domain.
No. The ludicrous "audiophile" market, thought annoying, is tiny, and should not prove a barrier to any serious new technology. It certainly didn't slow down the adoption of CDs, for example. And thankfully it has no influence over commercial audio. There are no oxygen-free cables in Sony studios.
I am surprised that commentators continue to talk about MP3s and compressing codecs. Storage capacities have already made music compression fairly redundant. A micro SD card can happily house a major CD collection uncompressed these days.
This article covers a lot of ground in a short space, and is therefore superficial in parts, annoyingly so. The author clearly believes that time delay is the overarching consideration in pretty much all audio design, even dismissing (he calls it "debunking") transmission line speakers in a couple of sentences. But this doesn't stand up. As Douglas Self points out in parts 6 and 9 (also numbered 11) of this article
...phase may be important for a base drum, but what about an electric guitar, where even the live performance is provisioned through "legacy" loudspeakers ?
I would add that that live music does not originate at a point source. In an auditorium, the audience members are in widely different positions relative to each instrument, with corresponding wide variations in time delay and sound direction and distance. So should the CD be based on the time delays at seat 34-A or those present at seat 6-C ?
Commercial music passes down an extremely long channel before it reaches your hi-fi. Perhaps hundreds of amplifiers, CPUs, mixing units, filters, DSPs. How likely is it that group/phase delay passes through unchanged at all these points ? Only if all these components have zero group/phase delay is it worth redesigning your loudspeaker crossover unit, and even them only if you think that the effect is gross enough to be audible.
My reaction to what you say is to suggest that what matters here is the distortion introduced by the loudspeaker. We can throw away frequency information—with my ears you can throw away a lot—but there are other elements in the signal, and the designers don't look at the whole picture. Loudspeakers are the place with the big distortions, and the fixes are the low-hanging fruit of audio technology.
My late father, who had terrible hearing, did get something from digital surround sound. It helped him distinguish the background sound from the speech. And sometimes the sound of footsteps had a direction which mattered, and which he could hear.
That just needed an amplifier and a bunch of ordinary speakers. but it was timing information that plain old stereo systems lose.
Dave, you are correct.
Audio design engineers with whom I have worked, very much do concentrate on timing, as well as frequency response. But they were not desigining speakers, they were designing mixing consoles with all the analogue (and digital) electronics in that, from active filtering using literally hundreds of operational amplifiers and many different types of active filtering circuits.
I haven't worked for any speaker manfacturers, so I can only assume the author is right when he says speaker manufacturers are not focussing on the timing.
But from discussions I had with colleagues (design engineers and recording studio engineers), there is something missing from all the specifications of audio systems, two systems can look the same on paper, in frequency response, in phase response, but yet still sound different. Generally speaking, our system of measurement, what we are measuring is incomplete.
No, I think you're jumping to the wrong conclusion about what you think the author believes.
He clearly is focussing on time delay has being a very important consideration, but that does not mean he believes that nothing else is important.
He does mention in the article about how the brain processes frequencies after the brain processes the transient information.
There is no doubt (without doing any research into his background), he's an electronics engineer, and being a fellow of the audio engineering society, I am highly confident he does understand about frequency spectrums, the representation of signals in both time and frequency domains and the transformation from one to the other.
I think he is focussing on one key aspect of audio design which is vitally important which he perceives (and possibly rightly so) is present in amplifier, filter design but which is missing from the last element of the chain in speaker design.
But this doesn't mean he doesn't understand that frequency response isn't important. I'm highly confident he does know it is.
>How likely is it that group/phase delay passes through unchanged at all these points ? Only if all >these components have zero group/phase delay is it worth redesigning your loudspeaker crossover >unit, and even them only if you think that the effect is gross enough to be audible.
First point: manufacturers of equipment (and certainly output stages of DACs) go to great lengths to design an active filter of high order, which buggers up the phase of the frequencies, and correct those phase errors using other types of active filter circuits. I know, because I've seen the circuit diagrams and spoke to the analogue design engineers undertaking the design.
Second point: You're making the assertion that because a system may not be of zero group delay up to the speakers that you don't need to design the speaker elements for minimal group delay. It's a flawed way of thinking. What matters is the total group delay of the entire system and if minimising it only in the speakers helps reduce the total delay for the system, then that's a good thing.
And by the way, from ex-recording turned electronic audio design engineers with whom I have worked, the author is entirely correct in his analysis.
All very lovely, scientific-sounding and possibly even correct.
However, it all falls by the wayside when faced with the modern fashion of compressing the dynamic range of the music source material to increase its "loudness" to make it stand out from the crowd (or now: blend is and not get left behind) when broadcast or streamed to the user.
So given that fundamental quality limitation, not to mention all the background noise in out lives, the quest for "perfection" whether in loudspeakers, amplifiers or dragging a piece of crystal across some plastic is largely futile.
As it is, for most audiophiles that I have met, the goal is not perfect audio quality. The real goal is to impress their other audiophile friends with the size and cost of their stack.
Indeed, one of the benefits of the CD (and therefore digital) revolution in music is that albums were remastered without (or at least with less) compression.
But it also depends on the artist. If you create your music using digital instruments with deliberately limited timbre, then all the speaker improvements in the world won't help make the music sound better than it would with earbuds.
@ Frankee Llonnygog and all
Isabella Stewart Gardner Museum Music Library
Fortnightly podcast of live classical (chamber) music. Mp3 128kB/s bitrate. Very listenable and creative commons licence. Sound a lot better than the bitrate (if you see what I mean)
Enjoy. Pass on to others.
vinyl? RIAA curve in/out. digital? hacked and (occasionally) regenerated.
this on top of the compression/expansion, noise gates, preamp distortion, microphone effects, and multistage processing loss of definition all throughout the recording and duplication chain. tape has a 50 dB range with nonlinearities on the low end, and processing like dbx adds additional artifacts. digital recording is itself a series of compromises.
so to start with, there is no "true" fidelity in a commercial source.
I have always had my doubts about the 30s and 40s RCA research that "7% distortion is the point at which the human ear detects." that's as good as they could measure pure tones. and every improvement such as the feedback loop, beam-power tetrode, and differential amplifier stages iimproves on that.
in the beginning and the end, we have analog. careful use of analog technology across the chain provides fewer machine artifacts and a more realistic experience.
I'm smelling more ways to try and push 1-inch transducers and chips and calling that excellent.
when it's just another ploy to get me to dump all my stuff and buy new.
the push side of the market is looking desperately for another gimmick, while the pull side is way tired of "it's all crap, but this is better." we don't want to stuff the landfills and spend all over again out here.
And any "new" speakers will immediately have magic woo applied to them. They will be made to look like space ships by designers who are considered more important then the engineers by the marketing department.
You just going to end up "needing" a thousand dollar power cord for each speaker instead of just one for your power amp.
Most modern music is just badly produced almost from day one. That's why I gave up buying CDs and ripping to a lossless format(*). It's just less hassle to download the MP3. My ageing ears help ensure that I don't hear the damage rendered by the studio 'engineers'. In addition most of my listening these days is in the car or via bluetooth headphones on the train or walking around.
I have a reasonable enough system at home but generally don't have the time to just sit and listen. It doesn't bother me too much. I enjoy listening to it - ignorance is bliss and all that :)
(*)WMA if you must know :)
Agree entirely, especially where radio is concerned. The compression and jiggery-pokery that is applied these days sounds almost painful to my ears. DAB was supposed to be the be-all and end-all with the promise of "near-CD quality" and user-adjustable compression (THAT fell by the wayside), and what do we get? More and more stations shoe-horned in at ever-decreasing bit-rates, hideous audio processing to boot (with Radio 4 Extra in glorious MONO, for pity's sake!) with the resultant sound as flat as the proverbial pancake. Why stations also find the need to compress their satellite feeds is beyond me, although they seem to be the best quality available at the moment (for what it's worth). I complained bitterly to to the late Radio Authority about excessive compression and processing some years ago and got back some waffle about catering for the majority of people who would be listening on portable radios or in the car. Poppycock, I say! I have some very old analogue reel-to-reel recordings of good old Alan Freeman doing "Pick of the Pops" on a Sunday afternoon, pre Optimod (or what ever they use now) and and they sound fine. Likewise, very early Capital Radio stuff. Put current offerings into an audio editing program like Audacity and all you get from the waveform is virtually a straight line! No dynamics at all. Absolutely appalling sound. So-called "re-mastered" re-issues seem to have suffered the same fate of heavy-handed compression. Nothing like the originals. Why should classical music be singled out for the non-compression treatment? This all makes a mockery of using high-end audio equipment if the input has been trashed to start off with.
"with Radio 4 Extra in glorious MONO"
Well a lot of the stuff on Radio 4 Extra was actually made in mono. (Hancock, Navy Lark, Much Binding in the Marsh, all the varieties of Round the Horne, Take If From Here) . and some of that has been reclaimed from off-air recording, so accurate rendition is not the prime importance
Mines the one with "Everybody down" written on the back.
I think there is a fundamental misunderstanding regarding compression.
An old friend and Grammy winning audio engineer, once told me that "Compression is like sex, it is absolutely necessary for reproduction, can be quite pleasurable between consenting adults, but will get you in to serious trouble if used inappropriately"
Because it is either:
1. synthesised using electronic instruments
2. subjected to autotune for vocals
3. deliberately manipulated for effect (many modern pop vocals sound like they're been stretched to make them extra trebly and reedy)
4. subject to massive dynamic compression to make it sound louder
5. mucked about with on a mixing desk, overdubbed, artificial chorus effects added etc
Only some carefully-recorded classical music on CD (not radio) might benefit from real high fidelity. Or maybe live pop, but in most cases the performers aren't all that hot without the studio trickery. For everything else, if it sounds good, it is good.
"You see, Piet, I can call you Piet, right? ... Right, anyway you see Mr Mondriaan, we sort of smoothed out the lines on those prints, and we changed the red for blue, because, you know, it was cheaper, and this modern art, it's all imagination anyway, and you know, I think it looks pretty darned good like this..."
If 90% of modern music is indeed just sythetic sound manipulated into a dischordant, over-compressed, over-processed mush before the listener gets it, it is still vital that the replay equipment reproduce all of this noise, compression and mush accurately.
Because that's what it's meant to sound like.
The artist intended it to sound a particular way. You're free to like it or not, but it'd be better if you judged it on the basis of it being the sound they heard when they said "yes, go with that", rather than a loose approximation of it.
Sadly this is in fact true. Well nearly.
I can think of three occasions on which I can say (I used to design Hifi and other audio gear) that I really understood what true HiFi was.
1/. Testing out radio 3 classical concert on the sort of speakers I could never afford, and hearing not 'applause' but individual people clapping. Clapping is exactly where the time domain is important. Otherwise its sounds like white noise.
2/. Listening to an album called 'Bass Culture' - probably the heaviest and most political reggae album ever made over a 2KW setup featuring 4x15" woofers in closed cabinets. I did manage to crack the studio ceiling. BUT every bass slap was a punch in the guts ..superb.
3/. Standing in the middle of a disco dance floor where we had rigged medium and HF horns to focus the sound exactly there, and actually pull the level down at least 30dB off floor where there was a restaurant. SPL up around 110DB with no distortion and it wasn't painful, just unbelievably clear treble with an amazing stereo image.
Yes, you can do good sound systems, but generally the cost is in 5 figures.
However, post processing does not murder the sound of hi hats and snare drums. Or an acoustic guitar.
Its worth while having the kit if you really can hear the difference. Most people cant.
The best setup I ever heard was an ambisonic array, 8 Kef 107 head units arranged in pairs; front, back, up and down and one very large sub woofer. The recordings were made on a Soundfield microphone in B format on 4 track reel to reel without any post processing. Ambisonics recreates sound pressures measured by a four capsule microphone time aligned to a single point to produce X, Y and Z direction and pressure W. Sadly not a very commercial proposition.
Speakers cant produce them, and the air cant transfer them. The fact that mp3 represents sine waves well and square waves badly is by design. The fact that you can draw a square wave on a computer screen does not mean you will ever hear one.
Just like you wont see a square wave on the ocean
If you put a square wave through a suitably designed high order 20Khz LP Bessel filter it still looks square, just with sloping sides and a bit of corner rounding.
The point is that what is at stake here is how much we hear in the time domain and how much in the frequency.
My experience suggests both. 'Clicky' sounds abound in the environment, and the information content and especially the location is not just assessed by how the relative volume is affected, between ears but by time delay. Mess with that and it doesn't sound 'real' any more.
'Classical' music is relatively free of sharp transients: Rock music is full of them. If all you listen to is opera, go for low distortion labyrinth.
but stick to IB for Led Zeppelin.. ;-)
Rubbish. To recreate a perfect square wave does require an infinite frequency response.
We have function generators which generate square waves, which are an important piece of test equipment, Sure, there is capacitance in the cables, capacitance in the load to which the function generator connects, and as a consequence, the square wave isn't an 'ideal' perfect square wave, but that doesn't actually matter. The less than perfect square wave is still extensively used and with good reason.
You seem to be adopting the attitude of "It's not a perfect square wave so it's useless". I don't honestly think you've even done any electronics experimentation at all with function generators, audio equipment at all.
Do you understand what a square wave is, and how to do a Fourier analysis (by hand on paper) on it? I'm not convinced you do. I'm not convinced you understand the importance of square waves and how and why they are used.
To recreate a perfect square wave does require an infinite frequency response.
Yes it does. It's an infinite series.
You seem to be adopting the attitude of "It's not a perfect square wave so it's useless".
An imperfect square wave is indeed useful, and does not require infinite frequency response. But look at the line from your post that I quoted above - a perfect square wave does indeed require infinite frequency response.
Posted by someone that doesn't understand Fourier analysis and waveforms.
Square waves (and impulses) are a very useful mechanism in the design of audio systems, it doesn't mean you are trying to design a system in order to reproduce a square wave!
If you understand what square waves are and the harmonics contained within, then you can easily see why they are so useful.
It's true that you can't hear them, but that remark fails to understand what square waves are.
Humans don't hear square waves (or any other waveform), what they hear is the summation of harmonics. A true square wave has an infinite number of harmonics, and we DO hear the harmonics up to around 20Khz. So in a way we do hear the square wave, but we don't hear everything in them, but then it's a physical impossibility for us to hear everything in the square wave.
"It's analogous to having an array of filtered microphones feeding into a DSP."
I don't think so. Each of those microphones would be sending a filtered version of the sound, whereas the nerve cells fire more frequently when their frequencies are heard, and these firings do not resemble any version of the sound waves. (Disclaimer: I'm not an ear expert, but a tinnitus problem made me read at one time more closely about how the ear works. Tinnitus is (or at least some forms of it are) caused by some of these sensor cells getting activated for no reason. Like a stuck pixel in an LCD. That is why one hears a whining sound at a certain fixed frequency, or frequencies).
Tinnitus is (or at least some forms of it are) caused by some of these sensor cells getting activated for no reason
I wouldn't say "for no reason"...
The cochlea is a wet environment. The oscillators within it will therefore have a low Q.
To overcome this, there is a positive feedback system.
When this misfires, you get spontaneous oscillation - that's tinnitus.
 Stereocilia, apparently.
There is a fair bit of localised processing in the ear, both mechanically and hydraulically in the fluid-canals, and then 'traditionally' within the neural nets that further pre-process the signals from the sensory hairs before going to the brain.
There's a heck of a lot of physically-distributed processing in an animal - for an obvious extreme example, the patellar reflex does not involve the brain at all.
This post has been deleted by its author
When I crank the gain on my guitar amp and plugin the Les Paul beast, I assure you, you can damn near hear a square wave ;-) (is there a smiley for humorous indignation?)
caveat: Transistor amp of course, valves are a totally different thing, I much prefer valves distorting rather than transistors clipping.
get yourself a function generator, and one of those crappy old speakers, you can quite clearly hear a square wave(well the fundamental & harmonics)... In fact get a bunch of function generators and an oscilloscope and you can make your own square wave! from scratch :-) and then compare it to the one on the FG.
you ears may vary
Yep, square waves are fuck-all use for helping to design equipment meant to reproduce music. We already use DSP to correct phase response, flatten frequency response and disrupt standing waves. Not much point trying to improve stereo image at this point, the industry gave up on two speaker solutions years ago.
So what do you use as a baseline during design if not a square wave? That is, if you create a design, test it, and then set about improving that design, and then re test it? How do you know if you have improved your design? You need a reference set of signals to apply to your design, measure the output, modify, re test with the same set of reference input signals. You have to use some sort of standard input signal which you can re-apply in order to determine if you have actually improved your design. That's where square waves come in (and frequency swept sine waves).
Your statement that square waves are F** all use is incorrect.
> In the future, loudspeakers will increasingly communicate via digital wireless links and will contain digital processing. Indeed, the link between IT and loudspeakers is destined to grow.
In the future? My speakers at work already have a box of tricks running a computer model of the actual drivers with a feedback circuit from the amplifiers to allow them to be driven to their design spec, with the risk of damage from overload and/or clipping virtually removed, while also performing the normal functions of a loudspeaker management system. And this isn't new technology either.
Whoever wrote the introduction to the article probably should have read it first.
I was interested in the subject of the article but the gross ignorance of the author was made clear on the very first page, saving me the bother of wasting my time with the rest of it.
The human ear is a masterpiece of sensor design, and that much should be glaringly obvious to anyone who has studied it in any detail; most of its workings have been known to science for generations. If the author has a better design, let's see it... I think we can safely file his work in the same dustbin Dawkins' rubbish about eye design was immediately filed by those who are actually scientific experts in that field and not ignorantly speculating science fiction writers.
The human ear does not have a flat response, either. All that external funny-looking cartilage gubbins (the pinna) has evolved to boost voice frequencies (around 1kHz) and attenuate others. So it's a clever transducer, but you wouldn't copy it for hi-fi, where what comes out is supposed to be more or less what goes in.
it's not because the frequency response of the ear isn't flat, that the reproduction of sound shouldn't.
In ideal circumstances, you'd want both the microphones and speakers to have a flat frequency response so that the original sound reaches your ears the way it was. Your ears will do the non-flat bit anyway.
Not only is the response not flat, the response is different for everyone. Fletcher-Munson is a best guess approximation.
In reality, everyone has different ears.
That aside - yes, perfect transient response would be a good thing. But isn't this the same John Watkinson who believes that 44.1kHz is perfectly adequate for audio?
Good luck getting perfect square waves out of any system with obvious band-limiting.
There's some debate about how much bandwidth you need for position estimation. It may only be a few kHz, so 44.1kHz is fine, and perfect square waves aren't necessary.
But also, it may be more complicated than that. One of the intriguing things about audio is how subtle it is, and how tiny changes can make such a huge difference to the sound.
HiFi is purposefully designed to have a flat response so that it plays back exactly the source material, however, it's actually more pleasing to the ear to have a slightly non flat response.
In the studio environment they're looking for equipment with a flat response for 'monitoring', but hi-fi in the home tends to not have an entirely flat response.
"it's actually more pleasing to the ear to have a slightly non flat response."
Really? You mean it is actually more pleasing to the ear if it doesn't hear the same sound that it would have done if the musicians were playing live? Hmm ... probably, but do keep quiet about it or else you'll upset someone.
"You mean it is actually more pleasing to the ear if it doesn't hear the same sound that it would have done if the musicians were playing live?"
Well of course it is. Otherwise how fo you explain the modern appetite for $2,000 valve amps with their distortion fueled 'warmth'?
And I've been to a number of concerts where hearing something different from what was coming from the musicians would have been an immense relief ;o)
"HiFi is purposefully designed to have a flat response so that it plays back exactly the source material"
Some HiFi is. But some manufacturers, notably Linn, consider the pace, rhythm and timing of music more important than a completely flat frequency response. Consequently, many of their speakers aren't exactly the last word in HiFi but what they do have is the "boogie factor" and surely that's what's most important. Their Keilidh model was a good example of this.
I think we've been reading different articles. In the one I read the author wrote:
"Sound in air suffers an impedance mismatch at the surface of a liquid, yet the ear has evolved to have remarkable sensitivity by using an impedance-matching mechanism consisting of a series of bones acting as levers between the ear drum and the transducer proper. Such an unlikely arrangement would appear to result in a score of Darwin 1: Intelligent Design Nil."
This is absolutely correct, and it is well known from what bones the present malleus and incus evolved. Reptils have a different system of sound transfer from air to liquid, but it works in roughly the same way.
The cochlea, on the other hand, is a very unusual sensor indeed in which the encoding is so complex I'm not even going to begin to summarise it here. It has to be, because the mechanism of signal transmission in nerves is complex - a kind of multiple parallel channel PFM - and there is no simple and logical way of converting sound waves to neural signals. The ear, like the eye, is quite good at what it does, but it certainly is not a way of transmitting sound to the brain with low distortion.
I can only speculate that you're an "intelligent designer", because what the author has written is scientifically uncontroversial, and any electronic engineer who takes the time to look at what is known about the human auditory sense will agree that it's a good way of getting perhaps the least worst result, but if you started out to design an animal from scratch you probably wouldn't start from there.
Intelligent design is about as scientific as cables with arrows on the outside to tell the electric field which direction to move in.
"Clearly there's an need for intelligently designed speakers. As listened to by God"
Quad electrostatics are the chosen speakers of the almighty, and all those of good taste and discernment. And they even sound fantastic playing MP3's off the phone into the pre-amp.
"As listened to by God" - you mean to say that the big-bang didn't blow his ears out like explosions do to those created in His image? Luck sod probably doesn't get haemorrhoids either; not fair!
grils girls only creator deity I know was Gaia and she hasn't been fashionable for years)
This post has been deleted by its author
Dawkins is one of the pre-eminent experts in the field. He may also be a writer, but he's a first rate evolutionary biologist with a large body of highly original and respected work behind him.
If you're going to post on things like that, perhaps preface with the disclaimer: "I have no knowledge of the following topic, but this is my opinion"
"Dawkins is one of the pre-eminent experts in the field."
If we're discussing the ear's suitability for the job then "the field" is actually quite complex. It starts, obviously, as an exercise in acoustics (and I doubt Dawkins is pre-eminent in that field) but we also have some engineering constraints:
It must be something that an embryo can create.
It must be something that the adult body can interface to and provide energy to.
It probably helps if it is also something that the adult body can maintain.
and as noted later in the article, mathematical fidelity of response is less important than being able to notice certain kinds of sound and locate their sources. As a result, a perfect microphone would be a totally rubbish ear. Pretending otherwise merely gives the ID-iots an easy target to shoot at.
This post has been deleted by its author
Wow ! Let me help you up again. That was a spectacular slip-up you had there.
Think again about what was written concerning time accuracy and the effect of bass reflex and transmission line designs. There is also mention of sealed loudspeakers. Then there is the stuff about compensating for room acoustic effects. Does that not seem like some real engineering going on?
The issue is, there are armchair engineers out there, then there are proper degree qualified electronic engineers that have spent three or four years studying this subject (along with a load of other stuff) at university, that understand the concepts of: frequency response, phase response, impulse response, Fourier analysis. The author is the latter, quite evidently so. Some people on this forum are the former, clearly so.
That's not a criticism of the former, but they need to know which category in which they reside, and be able to recognise which posters on this forum fall into the latter category.
Knowing a little about frequency response and 3db points and 6db per octave cut-offs doesn't make you a qualified electronics engineer.
Within a few lines of reading the article it became clear to me that the author is a professional engineer, almost certainly educated to degree level in Electronics I would suggest. I have spent a short period of my career working for a very well known professional audio company designing and building mixing consoles sold to very famous recording studios around the world (including the likes of Abbey Road, Air Lyndhurst, LucasFilm). Trust me, he knows exactly what he is talking about, and his conclusion that time delay is a key factor is absolutely correct.
I loved your line "gross ignorance of the author".
You do realise he has decades of experience in engineering speakers and audio equipment, and is the author of many of the reference textbooks used in the field? As well as being a guiding light behind some truly great speaker designs over the years, such as those electrostatics bearing the QUAD label?
And you might check out his 750 page reference book on digital audio...which I am sure will truly prove his "gross ignorance": http://www.amazon.co.uk/Art-Digital-Audio-John-Watkinson/dp/0240515870
I had an employer who insists his old tube-based amplifier is still superior to anything digital today. After hearing the quality at very high volume and very low wattage, I'm inclined to agree. Of course every few years he has to order new tubes from China or some eastern European country since there isn't much demand anymore in the west.
On the other hand, speakers have changed quite a bit over the years. There are even towers that you might not think capable of making any sound just by looking at them (google MartinLogan). Most people (myself included) still prefer a more traditional speaker for simplicity. When a simple speaker breaks, I know why and I can repair or replace it myself.
Valve amplifiers were used by rock musicians for a long time because they cope well with overload. The inherent limitations of transformer output (it's asking a lot of a transformer to work from 20 Hz to 20kHz) and the low maximum frequency of the sort of power pentodes used in audio amplifiers means that the negative feedback loop is fairly simple and so doesn't introduce a lot of phase error (which the author is writing about). Although MOSFETs ought to have roughly the same characteristics as pentodes, their capacitance is higher and their cutoff frequency much greater, so the driving circuit and the feedback is much more complicated and introduces problems of its own - driving thousands of pF with a strong Miller feedback is a PITA.
Aged people like me may remember the AD161/162 germanium power transistors which gave quite good results because their characteristics were a better fit to audio than were the silicon transistors of the day, especially the 2N3055 - but of course a couple of EL34s or KT66s could handle a lot more power and didn't die suddenly when the output was short circuited.
I believe that engineers in the Soviet Union did some excellent work on designing transistor power amplifiers, the only problem being that you needed to be sitting in a Government lab sorting thousands of transistors to find some good enough for your military circuit, and pocketing a few matched ones for audio use along the way. But they could then design with little negative feedback, so time distortion was minimal, and they were feeding into Klipsch horns so the power requirements were also small.
The Cambridge psychology department at the time was using Quad electrostatic speakers (which looked confusingly like portable electric heaters). It really seems that overall we've gone back from those days, and that integrated circuits and the miniaturisation obsession is really what has done for optimum sound reproduction, not the move away from valves.
Basically a straight through Class AB RF amp design, transformer coupled. I've built a few in my time, but for MHz rather than Hz.
I just wouldn't be able to bring myself to do that. The VI transfer curve of a MOSFET isn't very linear, so you need a lot of standing current to get any power out, unless you want your sine wave to have very high THD or you can sort hundreds of MOSFETs to find a couple with very good linearity and the same gain and threshold voltage. Which is what she seems to have done. I appreciate the purism of the design and tend to agree that 1% THD is perfectly tolerable if the tradeoff is good transient response and low phase error. If it was me I'd be trying to stabilise the bias, and use lots of small transistors in each side of the output stage to average the response. I would end up with a lot more components and might think I'd have been better off with KT88s. But I like the blue glow...even though I no longer have any valve amplifiers, just a box of old valves.
I have built headphone amplifiers in the past which were pure class A with careful attention to linearity and no feedback, but with modern audio sources there really isn't any point. In any case my ears aren't up to it any more, and everything I hear has been extensively reworked by Mr. Siemens.
"The Cambridge psychology department at the time was using Quad electrostatic speakers (which looked confusingly like portable electric heaters)"
"At the time"? I'm outraged. The original ESL 57s are still made new and sold by Quad Deutschland, for about €4k a pop. Or you can buy an original Quad Ltd pair on Ebay for a few hundred quid, and have them refurbed by One Thing Audio for about a grand including new stands, new treble panels, reworked and resprayed grilles and new electricals.
...and the low maximum frequency of the sort of power pentodes...
Here's me building HF transmitter output stages with me Dad's old EL35s and KT88s from the cinema P.As in the late 60s.
And we never knew...
PS: cheapo Denon box downstairs and Beyerdynamics and a phone amp upstairs in the man-cave. All toastie.
I gave up designing audio amps when essentially they got to be about 100-1000 times more accurate than the speakers and microphones that drove/were driven by/ them.
Valves distort in interesting ways without needing to be designed to do that thing: With transistors if you want that sort of overdrive you need to design it specifically. Having done so, they sound like valves. Apart from the microphony that is. But you can add that in as well.
I'll endorse just about everything the author says in the article. Time delay on a bass reflex or transmission line makes a total mockery of a bass drum or bass guitar slap.
You need IB and a damned big one. Preferably Infinite :-)
Oddlly enough a wall of speakers with open backs approximates..
Guitar amps, whether valve or solid state, have nothing to do with any kind of hi-fi sound reproduction. They are always driven well into distortion at every stage, from preamp to output. The important thing is how that distortion behaves. That is one reason why there is so much variation in sound even between valve amps of different design from the same manufacturer. The other major factor is the interaction of the speaker, its enclosure, and the output stage impedance and feedback mechanism. Try hanging a scope off the feedback node of an old valve amp and giving the case a good thump. Solid state amplifiers then to clip very cleanly and have low output impedances so that the speakers are highly damped and have less effect on the sound.
This post has been deleted by its author
But then that is not an amplifier! That's a high current output DAC.
It's a solution which obviates the need for an amplifier. But I go back to my original statement. An amplfier is an analogue circuit. It is. It has to be. An amplifier takes an analogue current or voltage and makes it bigger ( a simple class A amplifier with a simple transistor *is* an analogue circuit. There's nothing digital about it).
It's akin to saying "digital aerial for FreeView". It's not a digitial aerial, it's an analogue aerial.
I presume the capacitor is being used as an anti-alias filter? So all you've got here is a single order 20db per decade roll off filter. Depending on the sampling rate of the DAC, it doesn't have a high enough cut-off rate to deal with the images higher up in frequency, (the image spacing in frequency is determined by the sampling rate). I'm sure it would work, but it perhaps wouldn't sound too good.
Never heard of a digital amplifier. Amps have to be analogue.
They are called Class 'D' amps. I think I used to own a 5.1 receiver that was class D. I'm not completely sure but (this was about ten years ago) it was the same size as my DVD player. Despite being vertically challenged it could deliver 100w per channel.
I think this shows a failure to understand the definition of the word digital. The class D amplifier does not contain any digitial electronics and it does not represent information or signals using numbers.
The class D amplifier contains only analogue electronic devices. There isn't a logic gate, a flipflop in sight.
The class D amplifier contains only analogue electronic devices
That's only true inasmuch as all digital devices are made out of analogue devices.
There isn't a logic gate, a flipflop in sight.
Yeah, there is. It's all logic gates up to and including the PA stage. The audio is held as PWM info on a high-frequency carrier, meaning it can be processed in DSP or similar at any stage of the proceedings. The LPF at the tail end strips off the carrier - this is beyond the last active component of the amp.
> Never heard of a digital amplifier. Amps have to be analogue.
They don't. Class D Amplifiers are becoming more popular these days, as it allows you to be digital all the way through to the output, where your LPF strips the modulation away from the carrier and makes it analogue again.
One that isn't "I tried this new £500 plastic box and it sounded awesome". One with graphs, numbers and stuff!
I'm a bit mystified by some of the responses. Some commenters seem to reject the idea of good speakers on the basis that the commenter is happy to listen to shitty-sounding music. I guess after a few years listening to distorted music at high volumes on earphones, once tinnitus sets in, they won't be able to hear the difference anyway.
That's a pity.
If better gear becomes widespread and turns the Loudness War into a temporary aberration, surely that would be a good thing?
I grew up in the era of CDs, the 1980's. Lossless compressed music came about as a result of the internet as download speeds were relatively low (by today's standards).
The youth of today has rarely heard uncompressed music fed through a decent sound system.
Some years later, after graduation from university I spent a small spell working in a company which made large format mixing consoles used in recording studios and there I learned what music is!
We had a couple of demo rooms in which we could demo the equipment to our customers (including some big name artists but more often demo'd to studios).
I got off the bandwagon which I had been on since a teenager, repeatedly upgrading hif-fi, replacing individual units, I went out and bought a pair of active speakers, - professional grade for use in studios- with XLR balanced inputs. I bought the best. I don't need to upgrade (unless I get a bigger house and have a much bigger room).
You don't need to spend a huge amount of money, you just need to know what you are doing.
And key to that, is having heard a very high quality sound and getting used to it, hearing it every day, that then becomes your reference level of sound quality which you then use for comparison when you go out shopping for speakers and equipment.
Prior to that 're-education', my reference level was the existing hi-fi I had, and I was comparing prospective purchases with that as the reference. That resulted in incremental steps up each time.
I met one guy (the salesman) in a well known upscale audio shop who said to me upon hearing what I had purchased "They're the cheaper ones", implying that they can't sound particulary good because they weren't £6K ! Little did he know.
Don't assume that because it's expensive it must sound good. Don't assume that you need to spend £10K on speakers and amp to get anything decent.
Once bought some line level phono-to-phono cables a metre long costing £200. Then I started doing some experimentation using cables of the type used in studios, with cables purchased from the same vendor the studios used, the end result? I made my own interconnect cables (not exactly rocket science if you can solder!), which sounded equally good for something like £20, with the gold plated phono connectors costing way more than the cable itself! But I guess that company had to make a living.
But that's where the marketing BS comes in, sell the cables for £30 and people won't buy them, too cheap you see, they can't be any good.
Oh Christ, someone is about to do for loudspeakers what they did for radio, TV and telephones: make them more expensive and complicated while at the same time less fit for purpose.
Clue missile. People never walk into my 70s-era stereo sound and complain there is an impedance mismatch in the left tweeter causing a 2 decibel drop-off in the crossovers. They say "what're you playing?" so they know what to either compliment or complain about.
Give me back my analog TV. I'm sick of the periodic tiling the "state of the art" sends me or the set-top box that hangs every time the cable company sends new "info" to it (at midnight, every night, because, wow, rolling download schedules are so retro). At least ghosting didn't stop me seeing the f*cking news.
And don't put f*cking digital crap in my loudspeakers because I want them to work on demand not hang pending some driver update.
I have a pair of Dynaudio accoustics professional active speakers, all analogue, and I can assure you, even at your age, you *would* hear the difference. The industry rips people off with incredibly expensive consumer range speakers and amplifiers and Joe Public is lead to believe they need to spend £10K on a good system.
Most people haven't got a club what decent sounds sounds like, and it doesn't have to cost the earth, you just need to know what you're doing.
Actually, that statement (correcting for the autocorrection flub - yet another "benefit" of technology "improvements") couldn't be more wrong.
With the possible exception of profoundly deaf people (I readily confess to no knowledge of the specific issues faced by those with the condition) *everyone* knows what decent sound sounds like.
It just doesn't usually match up with your idea of what it sounds like. Or mine. Or Prince Charles's. Or Hyacinth Bucket's.
I've yet to meet a Hi-Fi fanatic who cared about things costing the Earth. I've been noticing the daft lengths some go to since the mid '70s. It's their money. They can spend it how they like. Kids in my neck of the woods spend a grand or more on in-car sound systems, then fit a sub-woofer that makes it sound like a hunnerd-buck boom-box turned up too loud. It's their money.
But when this sort of "new improved" idiocy takes root, choices I don't want are forced on me and options I prefer disappear.
Don't fit my 20 quid Wharfdales with digital upfuckery please. I want loudspeakers to work, not waste time asking the place that made them if there is an update this morning instead of playing Faust Tapes or Fairport Convention, which is what I bought them for, and I don't need the loudspeaker equivalent of "PC LOAD LETTER" messages displayed on them when I connect them to my amplifier.
Most of all I don't want the design specs drawn up by an IT twonk because they are universally terrible at understanding what real people want from anything, resulting in cellphones which refuse to yield your own number or that require seven operations to change the ringtone (in one step of which it is crucial you DON'T click the "change ringtone" menu item) or stereo players on which the volume adjustment is buried deep in some stupidly over-converged menu meaning that if you get an urgent phone call while enjoying Floyd at #11 the only option you have is to rip the amp's power cord from the wall.
Cell phones which don't yield their own number. Must be O2 network then.
They're the only carrier I've come across so far where you can't use the phone to interrogate the network to find out the phone number (phone number is held on the network not in the SIM or phone).
Cue my little problem a little while ago: was sent a phone and SIM on O2. But SIM hadn't been properly activated, emergency calls only. Couldn't dial a number to call someone for them to give me the number from the CLI. Couldn't look up the phone number because O2 network doesn't provide that facility.
Tried to get it resolved (not using O2 but via someone else, long story to go into):
they wanted the mobile phone number in order to resolve it! Duh!
Gave them SIM serial, IMEI, IMSI number (now f**ng go look up the mobile num from the IMSI number please!).
Two weeks to resolve...couldn't make this s*t up.
The no.1 rule when buying either is to listen to them together. Some amps sound dull and too bassy with some speakers, some sound too bright and thin. Getting speakers that sound good with your amp is far more important than getting components that are theoretically brilliant on their own. You could spend £500 and get great sound, or spend £5,000 and get atrocious sound.
> If you buy active speakers, then the problem goes away!
No it won't, unless you're using the bastardisation of "active" which used to be called "powered".
An active speaker is one which requires an external crossover to split the audio into bands, as often found in audiophile set ups. A passive speaker is one which has the crossover contained within, so just requires a single amplified audio signal.
A powered speaker is one which contains all the amplification and crossovers, and requires power and line level audio.
I'm using the word 'active' in in the way professional audio manufacturers use it.That is: mains power input, balanced (differential) analogue audio input, power amplifier, cross over and speaker drivers all contained in the same unit. Crossover filters comprising active filter circuits using operational amplifiers.
You may call it powered, but it's both powered and active.
"Of course every few years he has to order new tubes from China or some eastern European country since there isn't much demand anymore in the west."
I think you'll find there's still plenty of demand in the west - just ask any number of guitarists that use a modern tube amp (myself included).
Starts talking about "audiophiles" and then continues with "MP3" and "iPod" in the same breath.
Then, basically, it says that monopole drivers are bad and electrostats are better (well, it's kind of open knowledge, I am thinking of getting a pair of electrostats myself) but you can take the bad drivers and combine them with lots of digital tomfoolery and they will become good.
Replacing snake oil with a different flavour of snake oil?
No speaker is perfect. Cannot be, we know that. Yes, you can try to coax a better approximation of the source sound by tweaking the layout and mechanics or by finagling the signal. Neither of these methods is intrinsically better than the other.
I prefer the former, you - the latter.
The only thing that really counts is whether a particular speaker in a particular room suits a particular person.
Your last sentence says why you are wrong - "a particular speaker in a particular room".
The main benefit of "finagling the signal" is that it can be dynamically adapted - in near real-time or as part of a set-up procedure - to the acoustics of ANY room. Which you simply cannot do by "tweaking the layout and mechanics" in the vast majority of cases. Or at least not as well outside of an acoustically damped room.
Where the author is perhaps mistaken however is in assuming that such correction needs be in the speaker. Today's home theatre amps are incorporating dynamic room set-up DSPs of ever-increasing complexity and power, and as they use a calibrated microphone to measure room acoustics and dynamics, they obviously incorporate some level of speaker correction.
So it will be an interesting contest to see where the industry goes - self-powered and correcting speakers, or traditional designs with some level of correction incorporated in the preamp or receiver.
I have 4 Quads dating from the 1980's. Wonderful sound. The downside is the space they take up. Her indoors moans something rotten when I set them up in my Study and listen to some pre-1980 music. No digital recordings allowed here. All being played on my Transcriptors Saturn turntable or Akai Reel-to-reel recorder and through my home bedisgned and built Class 'A' amp.
Sure it is ancient but the sound reporduction is in a different class to anything coming out of an MP3 player or streamnig service (there might be a good quality one playing my sort of music but I've yet to find it).
Having recently returned to reading contemporary articles on hifi I was amazed and disappointed to find so little progress in the last thirty years (streaming and compressed sound sources notwithstanding).
Of course, one thing has changed - the price trend is ever upwards. Why haven't the advances in electronics and computing power made top-quality sound available for peanuts? This articles goes some way to answering that question.
> Of course, one thing has changed - the price trend is ever upwards. Why haven't the advances in electronics and computing power made top-quality sound available for peanuts?
They have. The world of professional audio is constantly becoming cheaper, lighter, smaller and higher quality thanks to these advances.
However, these improvements aren't being seen in the same scales in the audiophile world. As long as people are willing to believe that spending £2000 on a power cable is making a difference to the sound, nobody is going to be working towards making these 'audiophile-grade' products affordable.
Is 'Audiophile' a term that the Reg has adopted and changed the meaning of? Somewhat akin to the Reg having its very own definition for 'Boffin'.
I'd normally describe myself as an audiophile as the dictionary description does fit: "a person who is especially interested in high-fidelity sound reproduction". However this article seems to redefine it as "a gullible fool who's easily separated from his money for bragging rights".
Perhaps we need a Reg Dictionary page to clarify these changes to the English language that the Reg are attempting to introduce.
On a side note: A perfect square wave needs infinite bandwidth and hence will always be an impossible ideal. (If you don't agree with this then please do come back to me when you have calculated the complete Fourier series that represents a square wave.)
Nope, the "Gullible fool" definition has been around since at least 1990, when we were approached and asked if it was possible to make a 3 pin plug with gold plated pins and a fuse with gold plated end caps to sell to "audiophiles". That's a 230V 3 pin plug, by the way. He planned to advertise that the improved connection would prevent noise spikes from entering your kit via the mains. He didn't want to know that there were these things called "mains filters".
The guy was hoping we could do the job for about 40p and he planned to sell his plugs for over 7 pounds each. I wouldn't be surprised if he found somebody to do it for him.
In order to determine whether or not The Register is redefining the term audiophile, I suggest you read some issues of The Absolute Sound or even Stereophile, and then determine if their editorial staff and readership, to whom the term 'audiophile' is generally applied, fit the new definition or not.
This is like asking whether Communists are idealists who believe in equal sharing, or supporters of a murderous political system that rests on slave labor camps. Are you talking about the original dictionary meaning of the word, or what the people called by that name actually are in the real world?
Take any sound from the internet and burn to CD then the sound sucks.
So far every audio I have sampled and recorded to cd needed major editing to make it sound halfway decent.
A good example would be youtube. Loudness is messed up and BASS is almost totally missing.
What ever the sound engineers at youtube were thinking?
Waaaaahhhh!!!! why doesnt everyone want perfectly accurate sound reproduction, its not fair, its not fair i tell you!!!!!!
Sadly while much could be done to increase the accuracy of audio reproduction thats not what people want. Just look at the money Apple just paid for Beats. Beats as headphones go suck, they are exceedingly expensive junk with a spectacularly flavoured sound but its what people want.
This article is the aural equivalent of arguing we should all want perfectly nutritionally balance food pellets rather than the vast array of culinary delights we choose to partake of.
This article is the aural equivalent of arguing we should all want perfectly nutritionally balance food pellets rather than the vast array of culinary delights we choose to partake of.
I would rather think it argues you can have perfectly good stuff in the digital processing arena for cheap while people with extremely expensive rigs are barely approaching the bronze age, with all the superstition you would expect.
Nice article encouraging thought. I would only argue with one little point made about the amount of information in a CD being greater than that which speakers can re-produce. Based on my own experience in my music studio when I connect my cheap headphones directly through my budget amplifier to my professional microphone I can turn the volume right up and simply not be able to hear myself talking into my own ears - the reproduction through the system is so perfect because it is so un-touched. But when I record what I say at 16bit/44.1Khz PCM and play it back at that volume it blows my ears off. My advice for those people who want excellent reproduction is therefore to avoid the hi-fi shop/magazines and look at professional music production equipment, starting with studio monitors and a small mixing desk which can both be bought at a fraction of the cost of a high-end amp and speakers. Then spend the rest of your money on clean power and good quality source - preferably live and real.
The Linn thing did, I don't know about his uncle! And it wasn't a case of placebo (which is what I presume is being implied) because I wasn't expecting a change in any aspect of the sound from a motor change. Linns have long had a reputation for slightly "fuzzy" stereo imaging straight out the box (which mine pretty well was, albeit that box had been opened some years earlier as mine was secondhand) but it was something I was happy to live with for the rest of the sound. The reasons for upgrading were a) I wanted to play the extensive collection of 45RPM material I own (much of it containing tracks otherwise unreleased on vinyl) and b) the temperatures being achieved by the Linn Valhalla power supply with which my Linn was fitted (which were less Valhall and more Gotterdamerung!). I was not going to fork out the ridiculous money Linn wanted for a Lingo, so opted for the Origin DC conversion as a budget alternative, back in about 2003. I was surprised to find on listening following the upgrade how much more precise the stereo imaging was.
I was genuinely curious at to why, if as the author says, stereo imaging is dependent on speaker timing why a motor upgrade should make a noticeable difference?
running a green felt tip around the outside of a CD makes it sound better. Mostly because you think it does. There's probably some effect to the wow and flutter of the source, but high end HiFi folk can rarely spot the difference with their treasured improvement in a blind A-B-X test.
This post has been deleted by its author
I thought that audiophile kit had two main purposes.
The first, as already mentioned, is to impress other audiophiles. This is not new. Listen to a "Song of Reproduction", Flanders and Swann, 1957 - YouTube Link.
The other is to sound enjoyable, or at least impressive, to the audiophile (almost always male) and his friends. When I was younger, and green in judgement, I bought a pair of Koss Electrostatic headphones to go with my Linn LP12/Naim 250 system. They sounded very accurate, had great specifications, and certainly looked the part, but did not give any feeling of engagement in the music. My apparenltly far too small Linn Kan speakers were fabulous to listen to and, in my normal sized living room, almost nobody noticed that they were lacking in bass - We just enjoyed the music.
I am now old and broken, but still remember how good the direct-to-disk Sheffield Lab recording of Thelma Houston/Pressure Cooker sounded through my old kit.
And that surely is the point. If you actually spend time listening to music (i.e. minimal distractions and your attention focussed on the music rather than as background to your walk/drive/computer game/whatever)) then you want it to sound as good as possible for the money you have available. That, to my my mind, is the definition of an audiophile and does not deserve the perjorative use that the author has given it.
I do still enjoy sitting down and just listening to albums, be it alone, with my wife or with friends. For that reason I have built a system over the years that fulfills my needs but has not cost the earth. No component has been bought on its specs but because I have tested it and found a difference that justifies the price tag. I have a Linn LP12 because my primary source is vinyl, having been collecting for years before the CD came along, and I simply haven't heard a turntable that matches it. I bought secondhand, and spent some extra on upgrading the motor because of known issues with the Valhalla power supply and the fact I wanted to be able to play singles on it. I did not go for a Lingo because I would not have been able to justify that sort of spend to myself, never mind anyone else! I invested in a decent secondhand Naim CD player when the opportunity came along because my CD collection was growing, and the Linn was definitely outperforming the budget player I had; now the sources are about on a par. The amp is a Yamaha, and does a superb job; it was meant to be a stop-gap but I have no cause to replace it.
Hifi investment is a classic example of the law of diminishing returns; a £500 turntable probably will sound 10x better than a £50 one, but a £5000 will probably only sound 10% better than the £500 one (if that!). If you really can tell the difference, and have the money, then you should by all means buy that super piece of kit, but don't buy it because of the specs, or the price tag!
> That, to my my mind, is the definition of an audiophile and does not deserve the perjorative use that the author has given it.
It's not really the author who has coined the perjorative use, it's the "audiophile" industry that has caused it themselves.
So much of the market aimed at audiophiles is taken up by conmen, liars and frauds that the entire industry has gained a big black mark. And the big problem is the people who have been conned into paying £500 for a 1m twin phono cable are unwilling to admit they've been conned, and continue to perpetuate the myths.
You could just describe yourself as a hifi enthusiast - that hasn't attracted the same negativity.
With the advent of the Compact Disc, the bottleneck became the loudspeaker.
Personally I reckon the source that really shines a glaring searchlight on inadequate speakers is analogue-mastered vinyl....
I have previously related around here the tale of a mate who spent a small fortune on new HiFi kit, only to have me cast aspersions on his right speaker. He had the speakers biwired and had left the links on the right-hand one. After fixing this small configuration problem everything sounded right, but he couldn't tell the difference!
The first thing you need in an audiophile setup is audiophile ears.......
Another example: when I was working for the BBC, back in the eighties, I was asked to arrange a blind cable test for the local hifi club.
A fine time was had by all - particularly by me, since one of the cables they listened to was 13A mains flex with a one home resistor in one leg paralleled by a 1N4001 diode.
Not one of them noticed. Indeed, they couldn't agree which cable they preferred.
I remember a pretty famous audiophile magazine doing a blind cable test. They tested all kinds of exotic cables - but the source material and hardware all stayed the same - just the cables were swapped.
It was B&Q 13A solid copper mains Twin and Earth - the bog standard 1.5mm stuff.
I was thinking of a digital speaker solution only yesterday so this is timely.
Given that the time information is at issue here I was wondering whether dedicated speakers for each instrument might not be more practical. These could be either designed only to reproduce certain instruments (separate designs for violin, cello, guitar etc.) or be modified in their output through software depending on the use case.
Each speaker, (say typically four for a pop band or chamber orchestra) would recieve a digital stream time synced to each of the other streams. As the speakers themselves could be placed as desired the effect for the listener would be much more directional without the need for stereo capture/reproduction.
IE each instrument is recorded as a mono source, but the physical spacing of the individual speakers provides the reproduction of the sound space.
It might be more practical to arrange a number of panel speakers across a wall and assign them as required to allow maximum configurability with minimal physical intrusion.
This post has been deleted by its author
Wonderful article. Even some good science. Time accuracy is important, and many of the first CD recordings did not have reasonable phase-linear input filters, and were noted as being "blurry" at the high end. As we generate sound from our computer sound cards, typically the "5.1" system, the low frequency is made to be non-directional by design. Good accurate transducers wouldn't need such band-aids.
As for "audiophiles" (audio-fools), they seem to be members of a mutual admiration society, and to enter the "club" they need to spend lots, without thinking about it. When they start talking about multi dollar mains cords and the "unidirectional" connections, I just want to puke. Put a bunch of them in a room and do double blind tests, and they will only give excuses (while they pick the inexpensive amplifier as being the best). What you see advertised boggles the mind. I don't know if the marketeers actually believe in the products, or are laughing all the way to the bank. Somehow I suspect the latter. They seem like the over-the-phone psychics I hear on TV.
Life goes on, and educating the audience to "good sound" in an ongoing exercise, often thwarted by those who should know better. (*SIGH*)
Way back (when I was still young and good looking) I was a bit of an audio buff. Decided I wanted the Lecson pre and power amps with the matched Lecson speakers (double 8" woofers internal mounted with rear ports so room corner became the loading horn, stiff bedrexine 5" mid range horn loaded and metal ribbon tweeter dome dissipated). Matched with a direct-drive deck, Sure V-15 cartridge and SME arm. Sounded fantastic.
Managed to find a shop which actually had it all in stock. Got there to find they had a closing down sale so played it hard, each time I asked the price of the individual items (all heavily discounted) the owner got interested and said if I took an entire system he would add 20% more discount. Then I got an extra 15% for cash.
The Austin 1100 was quite top heavy as the speakers had to go on the roof rack, too big to get one on the back seat. Happy day! The neighbours (even two houses away) hated it.
You can hear it perfectly well, but what is it telling you? If it turns on and off, there is a period when there are a whole lot of other frequencies, at the start and end of each pulse; it isn't a pure sine wave any more.
It infuriates me because kids are still taught in A level physics that to reproduce a sound with frequencies up to F, you need to sample at 2F, the Nyquist limit. This is bollocks because as you approach the Nyquist limit the information tends to zero, i.e. if all the input sound was at F you would get anything from zero output to a triangle wave, depending on the phase at sampling. If reality is too hard for A level, don't pretend to teach it at A level! - and remind me again about that clever antialiasing filter which has a vertical response at F.
Scope manufacturers used to go in for this creative accountancy too, not good when trying to explain to management that no, even a 1GHz sampling scope could not tell you what was happening at 500MHz unless you were working under very special conditions (i.e. the input signal was invariant, which was exactly the case in which you had no interest.)
"to reproduce a sound with frequencies up to F, you need to sample at 2F, the Nyquist limit"
I believe the actual teaching is "a minimum of 2F to avoid aliasing" - not 2F to reproduce sound accurately...
Although the "minimum" bit tends to get lost usually in the drive to learn and churn formulae...
If you sample, such as what happens with D to A conversion, then you always get aliasing taking place, (if we're talking about the same kind of aliasing!). Increasing the sampling rate, results in those images being more greatly spaced out in frequency, allowing for anti-aliasing low pass filters to be made of lower order, = fewer components and cheaper cost.
If you sample, such as what happens with D to A conversion, then you always get aliasing taking place
Not so. If your sampling frequency is at least twice the maximum frequency in your input, aliasing cannot take place.
That's not the same as saying that the reproduction will be perfect as long as the Nyquist criterion is achieved - but it isn't aliasing that causes you problems.
...were the most ridiculous things a friend of mine bought.
He said he could tell the difference if he plugged them in the wrong way round.
Everybody else just kept quiet, in full respect of his madness.
He refused to listen when I told him most of the crappy music he's playing on this expensive setup was probably recorded in an average studio using old NS10 speakers.
"..were the most ridiculous things a friend of mine bought.
He said he could tell the difference if he plugged them in the wrong way round.
Everybody else just kept quiet, in full respect of his madness."
Are you seriously implying that none of you ever reversed the cables when he was out of the room?
I built a pair. Fantastic frequency response (separate sealed mid range units and horn tweeters). Brilliant Bass.
But complete rubbish to listen to actual music or speech on due to the temporal distortion. Presumably.
It's hard to make decent speakers without a giant sealed box. Murphy of olden days loved giant baffle boards, which is another route. But people don't want speakers larger than their 42" TV.
EV LT-12 loudspeakers were made for transmission line bass reflex enclosures and they were flawless if the cabinets were solid enough (2x12 doug fir sides w/ 3/4 cdx ply front, back, 3 horn baffle board separaters, 2x4 doug fir tapered horn glue blocks )...they were a tapered folded horn transmission line, exit bottom front...
Cabinet = 18in Wide x 30in High x 12in Deep...reflex slot 7 x 14 in, assemble w/bathtub calk and deck screws...RS.
So i forgot that the LT-12's were made to be mated to 35 watt tube type amps like the DYNCO Stereo 70, w/ P-P EL-34 tubes in a Ultra-linear amp setup...
IMHO= today's Hafler type DC amps for the TV room would instantly fry those things... think Infinity Towers / RS Mach-1's... both will handle the full output of a Halfler...and fill an exibit hall w/sound...RS.
This very minute, I am listening to Art Blakey on my Murphys. I have two baffle radios wired as speakers and they sound great. These were my poor man's Wharfedale SFB IIIs. Problem is, I just got a pair of SFBs that I'll be restoring. I don't have room for the Murphys and the Wharfedales. Dilemma!
John's been giving this same lecture for a few years now. I specifically recall an AES lecture a few years ago that has been well catalogued and discussed on various audio engineering forums. It isn't that what he said is in any way controversial, and there seems to be agreement that many aspects of his actual electrical designs are beneficial and sure to be in mainstream speaker design in the coming decades. No, I seem to recall the greatest amount of debate regarding his (slightly controversial) theory that omni-directional speakers eliminate virtually all of the issues that plague traditional speakers and render extensive control room treatment a moot point. He's the only one doing the science, so it isn't exactly like anyone can disprove him.
But I can't help but think that no matter how dogmatic audio design engineers can be - preferring the mechanical design side of things over the electrical engineering - and no matter how much more money they would (likely) make selling $24,000/pair B&W Diamonds over cheaper, yet still high quality alternatives, that if the technology was as proven as John believes that at least _someone_ would have implemented the technology, marketed it, and sold the resulting product to hi-fi consumers and audio engineers alike. Conglomerates love volume sales, so it isn't exactly like there isn't a business market for it.
Only the future will tell which of his predictions and technologies bear out.
I've recently had the pleasure of spending a good few hours in a professional recording, mixing, and mastering studio at the invitation of a forum member (not this forum) who got wind that I was screwing around with music as a hobby these days.
I learnt a hell of a lot while I was there, but that's by the by. The thing relevant to this article is this:-
Sitting on axis with top notch studio monitors I was played a vocal piece and it literally and I mean literally like the singer was there, this was way more convincing than any HIFI setup I've heard and I've heard some topping 50k. The problem was that after 30 mins or so with various musical pieces it became really tiring to listen to, simply because you could hear everything in all detail.
I mentioned this to the guy (a professional producer) and asked him what he used at home. He answered "you're right and I don't use these, you can't use them to enjoy music or a film, they're just too revealing, but they're what we need to do the job". He then mentioned that he uses far cheaper gear at home because it's more enjoyable and more forgiving.
With regards to the article I'm wondering if people really want hyper accurate speakers - I'm sure they think they do, but when the reality dawns that they're simply not that nice to listen to after a half an hour or so then they may regret their purchase.
Be careful what you ask for, because you just might get it.
Interesting, but was the sound fatiguing because of the high quality of the setup, or could it have simply been fatiguing because the highs were overemphasized (not necessarily by means of frequency response) as part of making the details of the sound audible? Or, conversely, maybe perfect sound is fatiguing, but that can be cured by rolling off the highs a bit, with the result then being an all-round improvement on ordinary hi-fi.
But then, it may be that audiophiles do find their high-end setups fatiguing, because I have seen articles by audiophiles where they talk about music as something special to be treated with respect, followed or preceded by railing against Muzak. So they may well make sure to be well rested before their brief music listening sessions.
In other words, since audiophiles did successfully discover transient intermodulation distortion, for example, the story may be complicated enough so that your experience doesn't totally discredit them.
That was always the problem with the Yamaha NS-10m's (the white coned ones) - they are amazing speakers, with stunning (and unmatched for a long time) time-domain performance.
However, you listen to them for a while and whilst accurate, they sound really tiring. Too clinical.
Similar things have been said to me by colleagues in the recording business, that the speakers used in studio setups are not suited to home use because speakers for use in a studio have different design goals, to have a totally flat response, to show every thing wrong with the sound so it can be corrected by the studio engineer.
I eventually opted at home for some active monitors from Dynaudio, and have found very, very little that can beat them. And I can listen for many hours at a time, and they compare to a separate amp and speaker configuration of home hi-fi costing many times more.
They are revealling, very revealing and if the original source CD hasn't been mixed very well they do show it up.
A classic example of this is an album by Garbage called 'Garbage' is just that, it's garbage.
The quality of recording and mixing is terrible.
But fortunately, few albums are this poorly produced, and whilst one or two may p**ss me off, on the whole, the very revealing nature of those monitors does give a greatly increased pleasure to the listening experience of much of the music I have.
An album by Tori Amos, with a conventional consumer grade equipment hi-fi, I wasn't particularly keen on her, play her album Under the Pink through my active monitors andy my opinion changed, and ever squeek of every piano key can be heard, the short intakes of breath heard, the detail that's there on the original recording will blow you away.
So, I think you can do a home setup with studio monitors, if you select the right studio monitors.
This is a rant about MP3s on one hand and idiotic audiophiles on the other.
In fact you mix them up in the same sentence.
Good audio is lossless - either PCM or with lossless compression.
As for "10 per cent of a CD" do you mean 1.6 bits/sample or are you discussing frequency? Is this with dithering?
"carpenters don’t know how to do that" is just patronising. I suggest you visit Kef and find out what happens in the real world.
"Only IT will break the sound barrier"? IT look after PCs, server and networks. To lump all the specialities involved in creating sound systems under that is just plain insulting.
Err.. materials science for example?
It's easy to bash audiophiles when your reference is MP3s, earbuds and bookshelf speakers. Even well-recorded CDs without compression have sonic limitations.
Unfortunately it will take more than new IT advances and software to bring the sound of a live orchestra to your living room. Sure DSP room correction will help some.
But what about the power and movement of air of a triple forte, the top-to-bottom coherence, unlimited bandwith and unrestricted dynamic range just to name a few things?
Listen to a pair of $200,000 Wilson Alexandria XLF speakers in a treated room with state-of-the-art electronics - and yes cables - to hear how close we can come.
Until then, you have no idea.
Was that for amplifiers, performance is measured, and for speakers, almost anything *but* performance is measured.
If you go look at stereophile equipment reviews, especially those for speakers, what's striking is how many measurements they make on things that don't matter. Impedance changes as a function of input frequency, for example. They always measure it, and it's a proxy for nothing predictable. Instead, they could do much more thorough measurements of sound pressure time and amplitude response to simple and more complex inputs, seeing as it's the sound pressure that we actually hear.
But what they really, really never do: compare, in the same measurement, recordings played through otherwise identical systems with only the speakers being different - this despite the fact that they often refer to a pair of speakers as "my reference speakers for years were..."
head to head performance metrics across vendors, as we're used to seeing for all other electronics? Nope!
Instead, volumes of meaningless stuff that sounds like wine snobs talking. Except that the number of people who are physiologically able to reliably (ie, measurably and repeatedly) distinguish some of the facets of wine being snobbed over is probably higher than those who can accomplish the same thing with audio.
The up side for the rest of us: the cast-off gear from two or three decades ago can sound as good as the very expensive stuff sold today. (All the moreso when the ebay seller misreads the label on what he's punting.)
Well balanced sound content, think Pitbull Concert at New Orleans, will give you a good sound wether listening w/ skullcandy cans or a Sanyo Mach-1 / Sony Infinity Tower lash-up...good clean sound...Yum.
IMHO = for me, Speakers are sort of redundant now days as there is seldom any privacy to use them, and, yes i know that is my choice of lifestyle...Bach Organ Concert in the Basilica at Kolobrzeg Poland ?? can do, just not every day...RS.
John's right about the communications part. But the task of amplifying the signal to drive the loudspeaker requires power and that is not really the realm of digital wireless links. And DSP is already in use in many amplifiers.
A better scenario is where the digital link feeds the speaker directly and the speaker contains the DSP, ADC and power amplifier. Speakers can only work on an analogue signal (or a high power PWM signal), but the DSP could be used to provide better optimized speaker crossover circuits and use a separate power amp for each speaker unit. This also means that such speakers will need another plug for the mains power feed. Where this technique wins is that the hunky power bit is up close to the actual speaker units and not being sent through those 'oh so special' speaker cables that claim so much performance. This is what many pro-monitor speakers do today, so what's really new?
The downside will be trying to get all interested manufacturers to agree the digital link interface specification.
DSPs in speakers are doing two things:
1) Making up for poor analogue design
2) Attempting to correct for the buggering up of the frequency response of the system caused by room acoustics.
I have some sympathy for 2 because not everyone can accoustically treat their rooms. But 1, do the job the right!
Suugest that the author reads up on the Linn Exakt system to catch up on what at least one audio company is doing about timing and phase issues. This may be enlightening.
Not saying this is the definitive answer, nor entirely unique (see Boothroyd Stuart Meridian), but its worth a look to see what is going on in the industry and from a company that has psycho acoustic experts within the team.
I logged on to say exactly this (if you will excuse the pun). Most of what the author is talking about is incorporated in Linn's Exakt system. The irony for those who use audiophile as a pejorative (for reasons I've never understood - why would anyone actively want to listen to something that sounds worse than it could?) is that it is one of the most expensive hi-fi systems ever made and one I will never be able to afford.
The irony for those who use audiophile as a pejorative (for reasons I've never understood
The reason the word is used as a perjorative is that the *vast* majority of people who describe themselves as "audiophiles" know precisely sod all about sound...
You beat me to it - Vandersteen made/makes all of their speakers to be time aligned across the multiple drivers. Having said that, I am still not sure that their drivers can match an electrostatic's for linearity. But it is important to remember that at least Vandersteen did emphasise their time cohesiveness...
Is it indoors or outdoors?
If indoors, what is the shape of the room and what materials make up the walls, floor and ceiling?
Sounds recorded for transmission, or sounds transmitted, are designed for a particular environment and deviation for this expected environment will sure cause the perceived sound to deviate from the result intended by the compiler?
In fact I believe that given reasonably well designed equipment, environment could well be a greater influence than the equipment.
I knew what was going to be said as soon as the statement about CD sound being perfect was uttered.
There will NEVER be such a thing as a perfect speaker because of several factors; the main two being :-
1/ Everyones hearing is different - and even that changes over time; I could hear 25KHz as a 20 something, and can still hear 22Khz at nearly 50.
2/ Everyones tastes are different.
The path to digital perfect has been one of alienation for many; modern recordings sound sterile and lifeless compared with a scratchy old recording from the 1960's, even when compressed into mp3 format; perfect doesnt mean better, if it did there would only be ONE recording of every piece of classical music for starters!!!
I listen almost exclusively to mp3 format these days - because my baby daughter cannot be trusted near an expensive HiFi.
I am always trying out new artists, but when it comes to the choice between a modern recording of a classical piece or a Jazz standard, and an old re-coded one, the old one nearly always wins.
I am THRILLED to see the cover of "Sparky's Magic Piano" on page 1. Am I the only reader that recognized it. My older brothers and I spent many a happy hour listening, and re-listening to it (on 78rpm disks if I am not mistaken). And then there was "Sparky in Orchestraville..."
As they say, "a blast from the past."
Does anyone have a digital copy?
Really interesting article. I spent 15 years designing DSP algorithms for audio processing so I have a bias towards the techy side of audio. The article certainly appealed to that.
But where all the technology stops is the ear. At that point it becomes bio-mechanics.
Just like it is impossible to define the perfect red wine, it is impossible to define the perfect sound. My "perfect sound" will be different to yours.
Sonar and radar are all about accuracy, music should be all about emotion and pleasure. It is very easy to get sucked in by the technology and forget the purpose.
My favourite music features all sorts of imperfectly reproduced dirac pulses caused by the scratches on my late fathers albums that I first heard as a child.
This is in touring sound, but the principles could be applied to HiFi if someone could be bothered. The idea is that no effort is put into getting the sound to line up in time/phase at the box, the dsp is used to line the sound up at the listener. So you're standing next to the box, it sounds rubbish. Sit in the audience area, and everything is aligned to your ear.
They used advanced modelling, and use a program to plot the audience area unique to each space., so you would need to measure and plot your living room, but that's small effort compared to getting all time/phase accurate!
The problem with touring sound is that the IM distortion caused by the non-linear compressibility of air at the high SPL produced turns everything to mush. One attempt to deal with this problem was the Owsley "Wall of Sound" used by the Grateful Dead when I worked with them in the early 1970s. Each element of the band--vocals, drums, guitars, and keyboards-- had its own channel, from microphone to speaker. The result was a very clean sound, particularly vocals. The problem was that they had no FOH mixer, but rather gave each performer their own control over their level with pots stuck onto the mic stands, so while it might have sounded good on stage the balance was often lacking in the house.
Interesting and stimulating article with a lot of cogent points. Agree there is much progress needed on Loudspeakers, but not so sure about the differences between bass enclosures.
At bass frequencies all loudspeakers can be viewed as 'motors' driving some sort of acoustic/mechanical filter. In a sealed box the filter is a resonant system comprising the cone mass and the combined springiness of the air and the cone suspension, if done badly there could be a resonant peak. A bass reflex box has an additional filter comprising the air in the port and the cabinet compliance. A transmission line is also some sort of low pass filter + a delay. So in all cases you hear a filtered version of what goes in, its just a matter of how the contribution of the filter is managed and trading that off against cabinet size, efficiency etc.
In regard to transmission lines and transients - Unlike a true transmission line, higher frequencies are attenuated in the line so the port is only emitting delayed low frequencies, this will still change transients (though less than shown) - but so will all other cabinets. Interestingly in the PMC cross section the driver is part way down the 'transimssion line' - so it will work a bit like a resonant tube as well, so there are two 'bass reinforcement' mechanisms being used.
The reason most modern music recordings are overly compressed with a very narrow dynamic range has nothing to do with codecs or lossy data compression it's because so much music is portable played on tiny earphones. If you play an old recording on modern iPod earphones it's so quiet you can hardly hear it as all the transient peaks massively limit your headroom. In the old days we'd have big speakers and headphones powered by big amplifiers, if it was too quiet, turn it up! You can't now as the kit doesn't have the headroom to do this so compressing the dynamics enables the whole thing to be played much louder on the small equipment. The reason classical music still has dynamics and isn't overly squashed is because the listeners of that sort of music still tend to play it at home on a larger system.
When I've mastered albums myself for music I've made as I much prefer the old sound and dynamics I've produced the music accordingly and it sounds great on my trusty B&W speakers, however try and convert it to MP3 and play it on an iPod then it doesn't know what to do with it, sounds all wrong and is very quiet. So I use limiters and compressors and all kinds of tweaking to make it as loud as possible, all the subtlety is gone but it works on the go. Active speakers aren't new and using DSP's to correct for room irregularities isn't new either, higher bitrates and better codecs were tried with SACD's and DVD-A's but they failed as most folk just don't care about the quality of sound and they have a very high tolerance for distortion. As long as they can hear the song then it's good enough for them. I think what's also interesting is that when MP3 first appeared it sounded all wrong, the compression sounded alien. Everything sounded electronic and fake. However after nearly 20 years of it we've gotten used to it. If I play a classic CD or LP to my teenage son or his friends who've grown up in this modern audio era the old less compressed & more dynamic audio doesn't sound better to them, it's not the nature of sound quality they've grown up with or heard.
But a wall of curved electrostatics focusing on the 'ideal listening point sofa' will more than blow your eardrums. You'll need to add distortion to make it actually seem loud! I've got a couple of electrostatics and a lot of people complained they produced no bass. A visit to the bathroom at the end of the hall and a couple of inches of water dancing in the bath proves otherwise!
A friend lived in an old maltings and the conical grain drier was the perfect place to hang a 24" fane 400W with baffle and a couple of satellites, the floor covered in wadding was probably about as good as you can get with moving coil. The sunloungers on stilts made great listening seats too! I'd recommend living in a horn loudspeaker if you like some serious rock!
But my idea of HiFi is just that - you should be able to hear the input to the system without any colouration at all. I've still got my els63's after many years and with a good length of mains cable and a decent amp they still outperform anything else - not that I've tried listening to much as I'm normally laughing too hard at the salesmen.
I'm surprised that people people believed that adding a delayed copy of a wave to the original wouldn't screw things up when the wwaveform varies... But then, I come from an electrical/electronic/engineering background, not a snake-oil one.
" it becomes immediately obvious how bad DAB, MiniDisc and MP3 are and that the only lossy codec that has any merit is AAC (at an adequate bit rate). "
This isn't a loaded question - legitimately curious - where does OGG vorbis stand here?
What's changed from that old, old layman's analogy of "digital" music being like the flickering florescent light bulb in that the harsher "digital" sound is transmitted as separate 1's and 0's while "analog" music is warmer and smoother because it is transmitted and received as a continuous "wave"?
Isn't this the reason for the continued interest in vacuum tubes and vinyl discs despite all of the hyped-up digital verbiage? Ask people who remember awesome "HiFi" when the term was first popular. Ear structure and sound physics haven't changed, but it seems the hype certainly has been "upgraded".
If all you want is earsplitting guitar twangs and thumping bass with a shrieking vocal, fine. But for classical music lovers, digital transmission is still harsh-sounding.
Take loudspeakers out of the room into the garden and their response will flatten. Try it once and hear the very obvious difference. Bass is generally much weaker, still somehow more pleasant, highs are more tame, other stuff depends on particular speaker design. But any of them will sound so much different compared to in-room sound.
Some psychoacoustic studies suggest that our hearing adapts to the fact of being in closed space (room) by rapidly taking into account its size and reflectivity after some initial sounds are made, and makes adaptive equalisation (evolutionary - to survive from bears that are met in caves and to better hear voices of companions at the same time). This aspect alone is hugely underrated. Yet it's difficult to explore. If widely exposed it could actually shakethe whole high-end industry as it generally relies on flat response which may turn out to be false target, at least in rooms. But lets omit this one and focus on what we think we know.
Assume we have phase-linear (better said time-coherent) self-built speakers that satisfactory reproduces square signal (mildly distorted squareform). Same applies to our correction software, plus it has 16K taps for high-precision bass correction and high-precision microphone that is omnidirectional up to 20+kHz and has its calibration file activated. Our source recording is brilliant balance of direct signal and spatial cues of both original recording venue (space) and placement of the instruments, rich in musical saturation, presenting instruments having great dynamics, almost uncompressed during post-processing. The source format is either analogue tape or its close digital "equivalent" of 32/384kHz FLAC combined with impeccable asynchronous DAC, having unmeasureable jitter level, noise and distortion values, and very fast amplifier with almost no NFB. We could either use one amp and 1-st order passive crossover (best transient response, wide co-band for wide, unobtrusive driver output blending) or brickwall digital (extremely narrow co-band, yet very low THD as all drivers are in their "safe" zone) with multiamping. Mains power is conditioned by very simple means (if any), and interconnect cables are under $100 (sorry). Its not very expensive rig overall (may be well under $10K if you know whom and where to ask).
Ok, now this is how vicious circle of infidelity in "normal room" will work even with all this great gear.
a) We do our measurementroutine and apply the correction to get out target frequency response flat.
b) If our correction software sums direct signal with the room reflected signal during measurements (for flat or gently declining sound power target), then even loudspeakers that has varying dispersion as a function of frequency aka non-linear polar response (100% of the designs, degree varies) will not spoil the final, corrected sound in regards to flat response. However, direct/indirect balance will be spoiled proportionally to dispersion variation. Which will be heard as distorted and blurry soundstage. Like main tone being there, but harmonics of the same instrument somewhere else;
c) if our correction software filters out reflections and uses only direct signal for correction, then we have the same problem as in b) plus non-linearities from room having varying reflection rate as a function of frequency (100% of the rooms, degree varies) distorting our perceived tonal balance.
d) in both b) and c) bass will be "smeared" timewise because of slow buildup rate of room modes (room dimension related resonances) after direct bass signal has initiated the pressure. Even with closed boxes it will behave like this. TL or bass-reflex will only smear it even further. Frequency-wise bass will be flat-corrected only in chosen measurement point (or averaged in preference area with multipoint measurements). Solid swings in bass pressure level depending of frequency are expected everywhere else in the room.
To my limited understanding only b) type of software combined with speakers having polar plot as close to constant as possible and combined with dipole bass (induces less of room modes) would shift sound more towards "right" and "even across the room". Perhaps including alignment towards right in-room psychoacoustics, that is yet TBD. At the moment this is 100K+ Ultra-high end market. Let's bring it to the common people. As you see, it's not that difficult with today's computational power. Some acoustics must be mastered too, though.
[I have no knowledge of the following topic, but this is my opinion]
I had hopes author would compare IIR filters against FIR especially considering their time response differences. And, by the way, what is miniumum tap (correction band) number to get the bass even enough? Hopefully in next article in series.
Besides that it's good, short (thus inevitably shallow) summary of some of major loudspeaker and reproduction culprits, hopefully stirring up the interest towards quest for better sound. Keep it up.
[I have no knowledge of the following topic, but this is my opinion]