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An Enticing Marketing Story, Theory Without Measurement?

Thomas_A

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A general question here. How many have tried to listen to a good speaker system where the front wall behind the speakers is treated with broad-band damping down to around 100 Hz and compared that to a ”live wall” and used EQ to adjust the effect of the room? Or for that matter in-wall speakers with traditional speakers? Perhaps forgotten now but e.g. the Carlsson/Sonab speakers worked together with the front wall to make it more invisible and I think the Allison speakers had similar ideas. With todays shallow horns or waveguides at least the tweeters are of less problems but you still have effects of the mid and woofers where you have boundary effects from the front wall. Whereas a good dispersion laterally fix the side walls you always have a front wall reflection that never has the same pattern as the side walls.

A second point is the level and delay of the side wall reflections. I am not sure you can extrapolate studies where the room is rather laege like the Harman test room with a much smaller room. If you want delays > 10-15 ms compared to the direct sound you would need a large room. Another solution is to have narrow dispersion or heavy toe-in so that reflections are coming more from the later reflections of the opposite side-wall.
 

RayDunzl

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@RayDunzl

So what's the "timbre" of electronic tones? Like how would "timbre" be measured.

I don't think you're understanding me. Everything you described is it's own concept, and when you're talking about harmonics and fundamentals. You can literally just speak about harmonics and fundamentals to paint a clear image of what sound you're attempting to reproduce or chase. You also are presenting this with FR graphs, which was as I said in the begining, using exclusively terms and scientific concepts we already have under our belt. And because you do that, you are forced to point out the cause of the flaws where timbre suffers, and explain it away using such concepts like FR to demonstrate the flaw. But if you're doing that, you're simply showing a system that has FR reproduction issues (or distortions/noise). There is no "timbre" here taking a hit, it only takes a hit because it's based on FR reproduction ability. And if that was the case, then this word need not mean any of this, and simply needs to mean "reproduction of sound accurately as possible from a few aspects involving FR".

Also, lets say you're recording a string being scraped of a cello or whatnot. Okay, and you have it displayed digitally in REW like you might for example using an ADC to import it digitally. How do you know the extent of a system that is reproducing these aspects? And more importantly, how do you quantify the "reproduction amount" actually occurring with playback being recorded? Better yet, how do you know any of it being sufficiently "good enough"? Like is there a system that can reproduce the timbre fully of one instrument, but utterly fail of another? And how is any of this verified? Because I get you can reply saying "sure there are speakers with no subwoofer that won't reproduce an explosion properly" or something to that effect, but in that case, simply make drivers to address each part of the FR range properly (from woofers to tweeter).

Like with distortions, theoretically we're done at 120db (if not 116 according to actual testing). What does the landscape of "timbre" even look like from a historical view from various playback devices (speakers or headphones)? Like how is timbre between speakers/headphones remotely tracked from device to device? To me personally, in any modern listening device - the only differences I can account for with my ears, ALL have to do with FR more than anything else. So if you want to say "yes yes, FR has to do with timbre quite a bit", well if it is quite a bit, why call it timbre, and simply not just as you explained "reproduction/fidelity preservation of harmonics/fundamentals/FR"

But again, I stress, how is any of this quantified and verified/used to determine what exhibits higher fidelity than the next thing? Do you simply compare a few tracks with their native ADC recordings plotted in REW for example, and then compare that with a recording of your speaker playing back the original recording? If so, how far off are with with respect to this "perfect timbre" ideal.. in the same way we now know how "far off we are from distortion-free listening"



So what's the "timbre" of electronic tones? Like how would "timbre" be measured.

With a synthesizer, whatever you will it to become. It would be measured as always, fundamental, harmonics, noise, attack, sustain, decay,, release, modulation, etc, etc.

If you want to give your electrified timbral experiment a name, make one up.

1570426991195.png



I don't think you're understanding me.


Probably not.

I'll stop here.

Maybe Timbre goes even farther than I've intimated - https://en.wikipedia.org/wiki/Timbre
 
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Tks

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With a synthesizer, whatever you will it to become. It would be measured as always, fundamental, harmonics, noise, attack, sustain, decay,, release, modulation, etc, etc.

If you want to give your electrified timbral experiment a name, make one up.

View attachment 35339





Probably not.

I'll stop here.

Maybe Timbre goes even farther than I've intimated - https://en.wikipedia.org/wiki/Timbre

The wiki article doesn't explain what I am inquiring about.

The problem is, you can say "timbre" and have that word substitute "sound fidelity", or simply "realistic sound" or "properties of sound". There are no metrics that can be used at all to have this word mean anything in reproducible fashion or any standardization of anything, simply because it can be used whenever someone wants and can only be present in objective form using already established objective understandings.

Now if you want to say timbre means any one of the three things I gave a synonym for - that's fine. But I don't see at all how there is any possibility of using it as a standalone metric to benchmark anything. Timbre itself can be the name of a benchmark that is summation of all other things you described explaining it. And if you can somehow assign a scoring system (which would need an ideal basis in theory to know what the "best" could even mean, per each benchmark), only then can you take the scores of each measurement, and apply a seemingly arbitrary averaging algorithm and only then would you be able to give a 1-100 score of the "timbre" of some device.

This is the only avenue a term like "timbre" has outside of things like music theory, or anything attempting to resemble useful predictive capacity with respect to objective scientific recording.

Even if you disregard all I just said, and somehow think "nah that's not the only way, we can use it normally as seen in this thread in single sentence descriptors of a products capability of faithful sound transmission", that only shoots you in the foot when someone rolls along and says: "How so sir, do you mean the spectral envelope metrics scored so high? Because that's the only way I would agree with you seeing as how the aspect of frequency modulation scored so low in this timbre evaluation of this speaker."

Or if we both disagree there is something "off with the timbre" how exactly do you propose we conclude which sub property of timbre was responsible for "it sounding off overall"? Because I'll tell you and anyone else right now... If your FR is a mess, there's no point in wondering if spectral decay was any good, since the FR is bad enough that other metrics pertaining to timbre cease to matter. And is yet again, another mark against using this word outside of laymen subjective shortening of using it, instead of saying "fidelity" or "faithfully reproduced" or any of the other terms I've mentioned prior.

Perhaps now I am a bit clearer? Sorry if I am even more confusing, but I cannot put it any simpler and comprehensive at the same time with my perplexity of using this word outside of laymen language.
 
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Cosmik

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Interestingly Genelec, up until recently at least - I have not checked lately - followed the guidance of the ISO and EBU standards (they are the same) because of their business with the broadcast industry. In brief, these standards require loudspeakers with anechoic (1/3 octave smoothed) on axis responses that are flat within +/- 2dB. OK. But then they tell users to measure steady-state in-room curves and adjust as necessary to make them flat (with a large tolerance). Unless one is listening in a non-reflective environment, or extremely close to the speaker, one cannot have both. There are more shortcomings of the standards, but this is a very bad start. In normally reflective rooms the steady-state room curve from well designed, neutral, loudspeakers will tilt downwards - they are flat and smooth on axis as measured in an anechoic space.
The results of this are mentioned unwittingly in one of Genelec's own blog posts:
Once GLM has served you a flat in-room frequency response on a silver plate, the result might be brighter than you like. While a flat frequency response is necessary as a reference, it may not be perceived as the best option with all types of content and environments, particularly when working long hours and monitoring above 80 dB SPL. I therefore typically roll off gently by around 3 dB above 10 kHz, but that's a question of personal taste, and may be influenced by room and listening distance also. Anyway, it's easy to add such manual trimming in GLM, once the response has been auto-calibrated to flat, so that's what I generally do.
In other words, flat in-room response is the reference, but may sound so bright it is painful - so you might want to change it arbitrarily. Your circle of confusion could not be better illustrated than this! I presume that some recordings are being EQ'ed and mixed using the "reference" and are therefore being turned down at the top end. I do think that old recordings from the 70s, say, often sound scintillatingly clean and open compared to modern recordings.
 

TimVG

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I have a feeling the EBU standard is getting misinterpreted and would benefit from a revision to clarify certain matters, as I posted before:

I've just read the EBU document (https://tech.ebu.ch/docs/tech/tech3276.pdf) and I see indeed guidelines for monitor (anechoic) frequency response tolerances, directivity guidelines, loudspeaker positioning, room treatment etc. They give an 'operational room curve example' and guidelines but there appears to be quite some tolerance to this, which would still allow, if I interpret the document correctly, the natural response of whatever a "good" loudspeaker does naturally in a treated room above the transition frequency. In fact in the notes of the operational room curve it states:

To avoid degrading the quality of reproduction, electrical equalization should be used carefully. It is advisable to make the corrections in the low–frequency range (f < 300 Hz) only.

Follow by a "All channels should be adjusted in the same way." .. I hope they mean adjusted to the same target ,below 300Hz.
 

UliBru

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Timbre is the amplitude of the components of the harmonic series in relation to the fundamental tone.
IMHO this explanation is too simple. Take a recording of an instrument and reverse it. The FR is still the same, you see the same harmonic series in relation to the fundamental tone. The playback of the reversed recording will sound different.
The ASA definition expresses some idea at least that there is more
ASA said:
"Timbre depends primarily upon the frequency spectrum, although it also depends upon the sound pressure and the temporal characteristics of the sound"
But obviously this is not a clear definition, there is no measure.
 

Absolute

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If timbre is undefined and unmeasurable, how come you can use a computer to replicate your voice and fool all your friends and relatives?
 

Juhazi

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Iron wire bending - timbre has a defined meaning for musicians, but it cannot be applied to hifi gear. Of course it is applied for marketing as well as eg. warmth which has nothing to do with sound physically.

There is no point at all to translate this hifi subjective and descriptive terminology to physics.
 

Hipper

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Like how would "timbre" be measured?

There are spectrograms of the timbre of different instruments. Some on Youtube and on the internet generally:

 

Thomas Lund

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“Circle of Confusion” has been used about two detrimental self-referenced systems when content is produced. One pertains to (hyper-)level: Mix&mastering -> distribution -> consumption -> mix&mastering; another to spectral balance: Monitors -> recordings -> mics&effects -> monitors. Both circles have been investigated at AES conferences for decades. The panel on 17 October will concentrate on the latter. Lossless with loudness-normalisation is now fortunately a given in delivery, so that circle should keep reducing.

http://www.aes.org/events/147/acousticssessions/?ID=6784

In-room frequency response (FR) of monitors is more ambiguous, outside an anechoic setting, and we agree about the importance of a smooth and controlled directivity, which indeed could be defined better in professional standards.

Anything but a flat on-axis direct sound FR would be unreasonable, but how do we reconcile reproduced sound with significant reflected power at the listening position? Part of the puzzle is what to count as direct sound, so a human with movement can’t just be replaced by a microphone when determining a neutral FR. Based on recent perceptual studies, we might not even (expect+receive+move) = hear the same. I.e. what we hear, even in a room, would be individual to some extent (BLV). However, unlike OSFA headphone listening, two persons in the same room are at least able to meaningfully discuss what they hear.

Most of the FR research quoted in this thread is about general consumption and likeability; which is different from critical pro listening with confounders kept at bay. GLM is aimed at the latter, and based on thousands of pro measurements accompanied by listening with notes. It guides to the best physical placement of monitors and suggests compensations without enforcing a particular FR. In case users wish to follow Floyd’s advice, only to compensate for low frequency room effects and to tilt the FR, this is easily accomplished, too. In some rooms, that method has arguably given me the best results. As we have discussed before, there can be several reasons why a professional, listening many hours per day, settles for a certain FR and listening level.

Coming back to the headline of this thread: With the point source 31-61 designs, one doesn’t get closer to ideal anechoic performance in both planes, going by Ilpo and by Floyd, so they should actually be the foundation for systematic tests of the most agreeable room compensation methodology in general.
 

Absolute

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There is no agreement on what "room-compensated" means. So you need to be more specific. Most commercial algorthms these days appear to improve the low frequencies - for a single listener at least. That is about 30% of the factor weighting in sound quality ratings. Above the transition frequency it is the "Wild West". Some algorithms use broadband "tone control" adjustments (good) while others pride themselves on flattening even small irregularities (probably bad). Masses of people, including many who should know better, think we hear waveforms. Wrong. Humans are greatly insensitive to phase, meaning that focusing on time domain errors is a "fools errand" to quote an earlier poster., especially if it compromises the amplitude response.
Fair enough, specifically I'm referring to systems that reduces excess/loss of energy caused by boundary reflections and room dimensions in order to achieve a flatter response in the listening area.

There are things I just can't seem to grasp, like the initial millisecond being very important for our sense of sound and timbre (direct sound), but tweeter, mid and woofer being 2-3 ms off each other doesn't matter while at the same time we can supposedly separate room reflections from the direct sound, even if the direct sound happens in the same time frame as some reflections.
Or the concept that we shouldn't equalize the speaker because direct sound is important, but broadband tone controls are good while small band is not. It does not compute, and it seems like many share my confusion on matters like this.

I have read your book, but not the third edition. I'll make sure that I do as it sounds like there's far more info on these things than the first edition :)
 

Thomas_A

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Fair enough, specifically I'm referring to systems that reduces excess/loss of energy caused by boundary reflections and room dimensions in order to achieve a flatter response in the listening area.

There are things I just can't seem to grasp, like the initial millisecond being very important for our sense of sound and timbre (direct sound), but tweeter, mid and woofer being 2-3 ms off each other doesn't matter while at the same time we can supposedly separate room reflections from the direct sound, even if the direct sound happens in the same time frame as some reflections.
Or the concept that we shouldn't equalize the speaker because direct sound is important, but broadband tone controls are good while small band is not. It does not compute, and it seems like many share my confusion on matters like this.

I have read your book, but not the third edition. I'll make sure that I do as it sounds like there's far more info on these things than the first edition :)

Just one q. What do you mean that drivers are several ms from each other?
 

Spocko

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I reckon 90% of the room acoustics and correction is a hoax, or is audiofool anxiety in people's heads. I get if there is a serious echo or wall placement that leads to sounds being out of time / phase, but most of what I see on YouTube by AV guys looks more like stuff to stroke their egos and show off how 'smart' they are with outlandish materials.
On YouTube, Audioholics has been pretty good with promoting caution when it comes to room EQ, with no corrections recommended above 500Hz.
 

Spocko

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The results of this are mentioned unwittingly in one of Genelec's own blog posts:

In other words, flat in-room response is the reference, but may sound so bright it is painful - so you might want to change it arbitrarily. Your circle of confusion could not be better illustrated than this! I presume that some recordings are being EQ'ed and mixed using the "reference" and are therefore being turned down at the top end. I do think that old recordings from the 70s, say, often sound scintillatingly clean and open compared to modern recordings.
Basically, Genelec is saying that for regular listening purposes (non-anechoic situations) tilting the treble down in the manner Dr. Toole has suggested is more pleasant. Thank heavens for GLM then, because you are given an EQ tool to change that frequency as necessary to taste.
 

TimVG

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Just to be clear - a downward tilting treble is what tends to happen -naturally- with neutral (on and off-axis) loudspeakers, when placed in a room, with some distance between speakers and listener(s)
 

Spocko

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The original sound source is the recording. Every form of source material is an artificial creation using a vast chain of equipment. All judged by playing back through monitors. If the playback is not true to the source, it becomes a guessing game.
I don't know if you realize, but even the "original sound source" that you identified as the "recording" - this is an artificial creation that is heavily dependent on the sound engineer's preferences that begins when the microphone is selected; microphones are as colored as speakers in that there are warm, neutral and bass heavy mics, etc. And without having the engineer's actual studio monitors for context as the "creator's intent", when you hear "warmth" in your speakers, you have no idea if it's your speaker adding the coloration or recording microphone. The beginning of this "vast chain of equipment" begins from the point of the microphone. So if a recording engineer's choices are too cool for your tastes through accurate neutral speakers, and your speakers happen to be rolled off in the highs or warm, then you are suddenly thinking WOW, my speakers sound great!
 

TimVG

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I don't know if you realize, but even the "original sound source" that you identified as the "recording" - this is an artificial creation that is heavily dependent on the sound engineer's preferences that begins when the microphone is selected; microphones are as colored as speakers in that there are warm, neutral and bass heavy mics, etc. And without having the engineer's actual studio monitors for context as the "creator's intent", when you hear "warmth" in your speakers, you have no idea if it's your speaker adding the coloration or recording microphone. The beginning of this "vast chain of equipment" begins from the point of the microphone. So if a recording engineer's choices are too cool for your tastes through accurate neutral speakers, and your speakers happen to be rolled off in the highs or warm, then you are suddenly thinking WOW, my speakers sound great!

Of course I realize that, and you are completely right. And it is exactly why neutrality and standards are needed from production in the studio to reproduction at home - recordings are simply all over the place, and I'm confident it's not always about artistic choices.
 

Cosmik

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Basically, Genelec is saying that for regular listening purposes (non-anechoic situations) tilting the treble down in the manner Dr. Toole has suggested is more pleasant. Thank heavens for GLM then, because you are given an EQ tool to change that frequency as necessary to taste.
They had tone controls 60 years ago :)

The issue is that the "reference" is wrong, but people in recording studios are probably thinking they are doing the right thing by using it. But of course in doing so, they are altering the recording. When released as CD, these recordings are known as Treble Under Recorded Discs (TURDs). Of course they may arbitrarily turn the treble down which is known as the Aural Analgesic Response Strategy (AARS).
 
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Thomas_A

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