I listen mainly to music that was originally generated using non-electro-acoustic sources.
So I too would like perfectly accurate sound (unless it be from the echo-affected seats in the Royal Albert Hall, in which case I would happily accept a measure of echo cancellation.
The situation could be different for "audiophiles" who do not listen to live, acoustically-generated music. I would prefer to call these people audiorasts (aka pederasts).
Unfortunately, no part of the reproduction chain is perfect, and this can easily lead to specification difficulty. For example, some of the best systems I have heard use speakers have total output acoustic power that correlates very well over the frequency range with on-axis density - but the intrinsic frequency response is not that flat; accordingly the amplifier frequency response has to be modified (with quite a fine resolution) to correct the overall response*. We are indeed degrading the amplifier to correct for the defects in the speaker - measurements of either would not look good on paper.
But the hardware is not the only problem - some "sound engineers" modify the microphone balance and even frequency response during the course of a piece of music. Even where they only do this where the featured instrumentalist is silent (i.e. in order to reduce background noise) you get a distracting change in the reproduced acoustic; but it also seems that some sound engineers think they know better than the conductors. BBC proms producers please note.
*In case anyone thinks otherwise, I am not referring here to any widely advertised system.
Being involved in instrumentation design, I was always a bit skeptical that a certain amplifier or speaker could sound better or worse than than measured results.
One day on a whim, I hooked up my distortion meters to the actual speaker terminals of my living room stereo, and I was in for a rude awakening. The distortion readings were a couple of orders higher than for a pure resistor load! Not only that, but the frequency response drooped quite significantly above about 6Khz using lamp cord.
When I tested various other power amplifiers, I found that their performance under a speaker load bore no resemblance to their performance under a resistor load.
Performance also varied wildly with the particular speaker systems used. A full range electrostatic by Quad made many amplifiers go unstable and most perform very badly. Not surprisingly, the Quad amplifier seemed unaffected by actual speaker loads.
Speaker cables did have a significant effect on frequency response and distortion.
So if ear tests don't correspond with instrument tests, you're not testing properly.
"... but the frequency response drooped quite significantly above about 6Khz using lamp cord. "
Long ago I heard a theory that the lower the resistance of the speaker cable, the better the low amplifier output impedance would dampen speaker cone resonances. Have you done any experimentation to compare 18 gauge lampcord with 12 gauge (household electrical) wire?
BTW, I cannot comment much on hearing vs measurement, I have "iron ears" caused by too loud a volume with headphones in my mis-guided youth. Those of you with kids, caution them to keep their music at a low volume.
How can you be certain that you are measurung the correct things, and you know what "Accurate" really means wrt sound quality? Here are a few historic examples where listeners uncovered gaps in the measurement methodologies available at the time:
Early SS amplifiers measured better than their tube conterparts, but many audio buffs found the sound to be clearly inferior. The culprit? Very high levels of crossover distortion having high peak/average ratios. Clearly audible. THD analyzers in the day notched out the fundamental and measured the average or RMS level of the residual. Thus a favorable objective measurement of a subjectively bad amplifier. We now know to look at distortion harmonic spectra at all amplifier power output levels and can easily uncover these flaws.
Jitter is another example of an audible defect uncovered by careful listeners and widely panned by many engineers as unmeasureable and so irrelevant. It wasn't until John Meyer developed and published a measurement methodology and started to correleate jitter with defects in reproduced sound that people started to take notice.
Historically these issues have been solved by savvy engineers and trained listeners who work together to achieve a beneficial result. Sometimes the engineer/listener is the same person, sometimes not.
The use of "audiofool" terminology is deragatory and prejuducial wrt to a group of individuals that historically have been able to uncover issues with audio reproduction that the then current engineering measurement methodologies were unable to uncover.
I'm not defending green pens, mpingo discs, quantum sinks, and the like. Far from it.
I'm suggesting there is a middle ground and broad brushed dismissal of the audibility of unmeasurable differences has been an historically false position to hold.
I pretty much agree with everything that has been written here so far...
So anything I add is intended as clarification rather than contradiction.
I think I can be certain that the things I measure are correct. On the other hand, I cannot be certain that I am measuring everything that is important. Similarly, I cannot be certain that the way I am measuring is appropriate, albeit physiological work means the situation should gradually be improving.
That was the case when crossover distortion was routinely ignored; however, the period during which accepted measurements ignored this is long gone - and audiorasts still claim that transistor amplifiers are never the equal of valve.
Known measurements that appear to be little-understood (or underused) at the present are mostly on speakers and on listeners.
Speaker issues include:
hangover, frequency-uniformity of angular distribution, and distortion (most speaker tests only measure high-level distortion - a potential mistake, particularly when organic and other so-called intelligent materials are used).
To return to "accuracy", I agree that listener tests are critical. But they need to be blind tests. The environment and performance method are also critical. For my money all tests should sporadically reference live music. The background is that the human nervous system is quite plastic, and for short-term tests people tend to prefer what they have become accustomed to. (The best counter-example was when a researcher brought a noisy change-over switch that was not connected in any way. In the absence of a difference in the sound, the difference between the "clunks" switching one way and another appeared to colour listeners' preferences.
Most published speaker measurements lie like a rug.
A speaker enclosure has 3 fundamental dimensions which will result in resonances in multiples of the half wavelength and multiples right up the audio spectrum. The Q of these resonances will typically range from 10 to 20. If one dimension is the same or a multiple of the other, the resonances are boosted. To evenly spread the peaks up the audio spectrum the fundamental dimensions of the enclosure should relate as the cube root of two.
To avoid revealing these nasty resonant peaks, speakers are typically tested with pink noise so the high Q resonances have no chance to build. Unfortunately music consists of frequencies held for significant time and the resonances will obviously color the music to the ear.
Every manufacturer wants his specs to look good. Testing realistically would significantly worsen the numbers. So most speaker and amplifier specifications are meaningless to real world performance.
So if ear tests don't correspond to instrument tests, you're not testing honestly.
Speaker and especially headphone transducers deteriorate rapidly with age due to temperature and humidity effects as well as fatigue. This deterioration doesn't have a direct relationship to original specifications or cost.
This is why a transducer that ages gracefully can sound much better to the ears than one that doesn't.
Hanging a curtain, rolling up a rug, or turning the speakers at an angle will noticably change the perceived sound in the room. A person's listening experience is subjective - measurements are not. So? The best advice I've been given is - "listen to the system, buy what you liked". Take into account that whay you heard in the padded listening room in the shop will sound different in your tiled floor living room.
Another audible flaw where the measurements had to catch up to trained ears was from the 1970's with transient intermodulation distortion (TIM or TIMD). It wasn't unil Prof. Marshall Leach (1940-2010) mathmatically defined it in 1977, tracing it to (lack of) amplifier slew rate. As he explained it to us in his EE4026 Audio Enginering class at Geogia Tech, amplifiers that have a high slew rate can still have TIMD, those that have slow slew rates *will* have it.
Three other factors come into play in the human auditory system for normal hearing people:
1) Ear canal resonance will vary greatly in both center frequency and Q: For my "guesstimating" for hearing aid and in-ear monitors (when I'm not using a probe tube mic (real ear measurement)), I use a figure of 2kHz for men & 2700 Hz for women, with a Q of 15; but varying that figure based on otoscopic examination for external auditory meatus texture. For more on this, google "Marshall Chasin"
2) You'll experience about 5% THD from the non-linear motion of the ossicular chain as it transforms tympanic membrane vibration into launching acoustic waves into the perilymph via the oval window;
3) Non-linear neural firing based on intensity, including direct stimulation of cochlear inner hair cells by the basilar membrane above 60dB HL (hearing level).
Editor, The Hearing Blog
As I recollect, the Acoustical Company (among others) was using both instantaneous and continuous signal slew testing at least as far back as the design phase of the Quad 405 (early 1970s). They also tested with realistic speaker loads (in addition to the electrostatics, which might be regarded as unreasonable).
Leach was formalising what he and the more meticulous of his colleagues had been doing for some years.
Other than maybe telling you what can and can't be ignored, I'm not clear about the relevance of the auditory system to this discussion - as it's a fixed part of the chain (albeit variable from person to person). The parts I think significant are the impact of sound level (to get subjective test conditions right), and (for design purposes only) of masking (to give an indication of criticality)
Thank you for elaborating on my point (the relevance is what can be ignored). Shame it reads as if it is intended to be a contradiction (especially as many authors treat inability to hear harmonic distortion as an aspect of masking).
As a side-issue: the physical ear is a multi-resonant system. Nevertheless, we perceive difference tones similar to those produced by second-order intermodulation. So we can live with moderate levels of 2nd-order intermods. But 3rd order can be excruciating. Have you seen a decent explanation?
P.S. For what it is worth...Langford Smith's team were able to detect 0.75%. My recollection is that other workers were able to demonstrate about 0.3% under the most "favourable" conditions; however, this could be a mis-remembering as I don't have easy access to check chapter and verse.
I once attended a course on studio recording at a well known recording studio.
I was appalled as to what was taught about "equalizing" the sound. The teacher tweaked every knob on each parametric equalizer for each sound channel for maximum "punch" to what I would call piercing. Monitoring was done at deafening sound levels. We were informed that this was standard studio recording practice.
This "equalizing" process was then repeated when it came time to make the master CD by another expert in this field.
It would appear that to earn his keep, a recording engineer is expected to adjust every single knob on his mixing board.
It's small wonder that so many recordings sound terrible when played on good audio equipment. How can one possibly judge the sound of an audio system when the the frequency and phase response of most sound sources has been so messed up?
Thank you. That is my biggest issue with the whole music industry. As they get more 'toys' to play with, they simply cannot leave well enough alone. People regularly condemn the 'sound' of Compact Disc, but if you have ever listened to a well produced CD, they are simply amazing. Unfortunately, well produced CDs always have been rare, and the situation has only gotten worse.
This post made me chuckle. A few years back I was in the studio recording music for a product. The "Engineer" on the board was constantly messing about with every knob on the "board". After a day of this I asked him to stop messing with things and leave everything "flat" while recording. He patently refused saying "This is what I get paid for". I just couldn't get him to understand that this was not the way to do things. I read once that Barbra Streisand would do "final mixing" WHILE THE GROOVES WERE BEING CUT in the master record. So much for RIAA, huh?
And don't forget that loud sounds are perceived differently due to direct stimulation of the stiff, less-frequency-selective inner hair cells by the basilar membrane, as opposed to softer sounds which are only detected by the softer, more selective outer hair cells~
The audio world is full of snake oil salesman. The worst I ever saw was cover plates for electrical outlets in the wall - this wa said to improve the quality of the sound in the room.
At least some people in the audio world has their heads screwed on correctly -
Check Peter Aczel's website www.theaudiocritic.com and read his older magazine articles on double blind audio testing and the influence of level differences of more than 3dB.
What about stickers to improve CD sound? But these "obvious" ones don't generally catch too many people.
Contrast this with "oxygen-free", "single-crystal" and "linear-dielectric" speaker cables, that continue to be highly-rated by some "hi-fi" magazines. Unless the cable has some serious flaw (break-down, mechanical weakness), adequately low resistance, inductance, and (for marginally-designed amplifiers) capacitance are all that matter.
Refresher: Inductance increases as the conductor spacing is increased, while capacitance increases as the spacing is decreased (and/or, of course as the dielectric constant e is increased)
(Conductor spacing is a common trade-off in overhead high voltage transmission lines)
I enjoy these conversations regarding "proper sound", "accurate", etc. I live a life of "dichotomy" with these things. I design and build my own line of guitar effects and amps. You can go though all the calculations, do everything "right" and then when all is said and done listen to it. Not always is the "correct" thing the "best sounding thing" (argued forever). I have both SS and tube amps. The SS amps are wonderfully convenient and portable. However, I prefer the sound of a tube amp (for guitar mind you). Given a "clean" SS vs. a tube I prefer tubes.
I've been listening to a lot of "live" band recordings lately and have noticed something. I will say I am "guilty" of this as well. When I record if there is a "silent period" it is not uncommon to edit it for "silence"; that is select the silent part and "clean it up" setting it to "zero". In modern recording this is done all the time. But having re-listened to some "old analog stuff" there is something to be said for the "silent parts" where there is some "ambiance" during the "silence". It gives the passage "continuity". With modern stuff it almost "starts and stops", especially if the Engineer gets over zealous and trims to much into the "sound portion".
Of course you want the vibration of the strings to be modified. That's just the electronic equivalent of the body of an acoustic guitar, or the horn of a saxophone; both of these filter and non-linearly distort the waveform.
Hopefully, however, you know how you want the performance to sound. The legitimate job of the recording engineer is to produce a faithful representation of the sound the artist wishes his audience to hear; if anyone is to modify the sound as heard in the venue it should be fully under the control of the artists (composer, conductor, performer.
Which reminds me of a recording sequence ca. 1963. The recording venue did not provide the acoustic desired by the composer. So the music was double-recorder using loudspeakers placed in different environments, and the composer and the engineers worked together until the composer was satisfied with the dynamics, resonance, and balance. The CD reissue (which closely recreates the original LP) remains a classic.
(I still have reservations about nearly all stereo sound produced via multi-miking, but that's yet another can of worms)
There's a renascence in vinyl recordings these days because many people say they sound so much better than CDs. What is it that is better - more noise, clicks, rumble, more distortion and poorer frequency response from this technically decrepit mechanical system?
Most studio recordings, because of their wild and piercing "equalization" settings have to tame things back down when cutting a disk so the stylus doesn't get launched out of the overlapping grooves. So yes, vinyl does sound more mellow, not because of the vinyl but because of the half deaf sound engineers "equalization".
A good demonstration of how bad the studios are is the sound of a "boom car". It goes BOOM tch tch BOOM tch tch. The tch tch is the energy in the 2Kz to 6Khz to give the recording punch and presence. I find in my home system I often have to turn off my sub-woofer because I don't appreciate chest compressions when listening to some sound sources.
So yes, the observant customer is always right even if she doesn't know the first thing about technology. It also goes to prove that most technologies can be turned into assault weapons.
Vinyl is a deeply flawed medium indeed, and I am enjoying a large collection I inherited nonetheless. Some of it (including things I've had for years) is still remarkably good and well-worth the fuss and bother. If you keep the records clean from the outset, the actual amount of wear from playing is quite small.
Some insist that the appeal of vinyl is one of antiquarian nostalgia, that somehow the clicks and pops key in the listener to bygone days. That may apply to someone, but for the most part I think it is deeply misguided. It is IN SPITE of the artifacts, not because of them, that it is still an enjoyable medium. And of course, much current pop music, as is noted, has vagaries of equalization and compression that are at best fashionable, at worst disgraceful. And the kids listen too loud!
The idea that people prefer inaccurate systems got a new lease on life not long ago when some professor insisted that his students liked lossy formats above lossless. This prompted Sean Olive to do controlled listening tests with trained listeners in the same age group. The results were contradictory as far as the notion advanced by Guttenberg and others: kids like accuracy too! see http://seanolive.blogspot.com/.
For information on loudspeakers and rooms, Floyd Toole's book Sound Reproduction is rapidly becoming the standard in the field. I recommend it highly.
All the literature I've read recently about room acoustics simply ignores the overriding effect of the fundamental room dimensions: length, width and height. The usual thing that's mentioned about dimensions is that the longest dimension determines the lowest frequency that can be effectively reproduced.
We know that each dimension will resonate frequencies and their multiples related to the half wavelength of that dimension. How can we make sure that the resonant frequencies of all three dimensions are evenly distributed up the octaves? The magic ratio is the cube root of two 1.25992! KISS
In our house we built two rooms that follow the formula. One is our living room, full of the usual sound absorbing furniture, curtains etc. The other is a sun room with hard glass on three sides and a hardwood floor. Recordings I've made in both rooms sound great. In fact the live sun room with it's long reverb sounds fantastic - like some large concert hall.
Some of the best sounding concert halls in the world are rectangular boxes with varying reverb times. I'm willing to bet that their fundamental dimensions follow the ratio of the cube root of two.
In my opinion the main purpose of the complex and expensive sound absorbers and reflectors is to damp down resonances caused by off the cuff room dimensions.
I'm not certain about there being a true optimum. In any event, the ratio you suggest would be rather surprising, as it results in a pair of closely-spaced (0.7%)resonances at only 5x the lowest room resonance. The traditional Fibonacci (= Golden Cuboid) ratios give resonance that are separated by at least 3% up to the 7th harmonic - and (for the same room volume) the maximum dimension can be increased by a factor of 1.57.
For large concert halls, naturally, the combination of low fundamental resonance frequency and diffraction due to hall features such as balconies makes numerical ratios pretty=much irrelevant. In this situation the harmonics in the critical frequency regions are closely spaced, and (paradoxically?) early arrival time time of first reflections becomes important. This can be achieved in many different ways, as you will see if you compare the dimensions of the "great" antique concert halls (from Grosser Musikvereinssall through Gewandhaus and Concertgebouw to Boston Symphony) with modern (or adapted) halls such as Royce the Bridgewater.
My favorite "snake oil" is the harmonic distortion in vacuum tube amplifiers: Whenever you have a tube amp you also have an impedance matching **iron core** transformer, which is also needed to keep plate voltage away from the output terminals. Because of hysteresis, the transformer I/O curve is NOT linear: Instead, the output is the convolution of the input current and the hysteresis curve (envelope).
Knowledge of EM in general and hysteresis in particlar is also how I separate out the men from the boys (& other pretenders) in the EE professionů
Editor, The Hearing Blog
Very interesting and informative discussion. As some have pointed out, and I believe most agree, is that much of the problem is with the source material.
I'm a retired engineer that's seen a lot of change over time and expect to see much more.
With sufficient processing power, and as much historical information we can dig up at our disposal, we should be able to recreate a model of the original performance. We could identify instruments, including the human voice, their locations and also some dynamics of recording environment.
With a rich descriptive language we should be able to transcribe the performances with any and all perceptible nuances. From there we could regenerate the performance.
While all of you possible sound engineers hark away at ear canal resonance, amp slew rate, mathematical dimensions of listening room and such esoteric funda, for the average listener who is a million mile away from such facts, what he likes when he hears is what matters. How in the world would one know how a royal philharmonic would sound when buying a CD. All that matters for one is how he likes when it plays on the device. So I would rather side with Audiofools saying accuracy matters but not at the expense of good sound which is the subjective part. Ironically people are willing to trade the latter for the former, go figure.
I don't know- you can have music played over $100,000 speakers with gold cables Vs a $1000 speaker with plain copper cables but one would be a fool to argue that the former is guaranteed to sound better.
As a live acoustic music and accuracy git myself, I would not argue with your priorities.
However, in my opinion you've confused the behaviour of some of the groups. There are in my opinion several major groups of purchasers - e.g.:
i) Those who are deaf to the blandishment of snake-oil salesmen, who trust only their own ears, and are wise enough to use a wide range of listening material when selecting systems.
ii) Those who belong to the first group, but also believe accuracy helps make things sound better, and that measurements are a means to this end.
iii) Those who think only the ears count, but happily use highly-distorted starting material to make their judgements.
iv) Those who think they only accept the evidence of their ears but are influenced by snake-oil salesmen; and
v) Those who will accept any received "wisdom" without making personal checks.
I have no argument with members of the first two groups - albeit the effects of sound level, and human suggestibility (or overcompensation for the latter) mean that it is extremely difficult to be a member of the second (unless you have loads of friends with appropriate systems who will loan them for a week at a time).
The "distorted material" group is likely to be disappointed when they get the system home; and the other groups will generally overspend on snake oil.
Accuracy measurements do not support high-cost cabling. Both measurements and blind listening tests suggest that only resistance and inductance are important. I probably go further than most, as I use standard "twin and earth" mains cable for speaker connections. Conductor area and number of paralleled cables depend on distance.
Re: Optimum room dimensions.
My research shows that the optimum ratio of room dimensions depends on the size (volume) and shape of a given room. So there is simply NOT one ratio that is always optimum.
My conclusions come from studying Oscar Bonello's work from 3 decades ago. Briefly, he showed how the frequency and strength, distribution and statistics of room frequency modes form a basis for catagorizing room dimension ratios.
My article on this subject appeared around 1999 in Speaker Builder magazine (I'm not at home to specify exact date).
John F. Sehring