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An Enticing Marketing Story, Theory Without Measurement?

edechamps

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if a room boosts a frequency by a room mode you can change the room. To change the living room to an anechoic chamber will certainly help.
But what if the spouse does not like to live in an anechoic chamber?
Is it better to leave the "neutral" speaker untouched or is it better to reduce the critical frequency by a filter? Of course the latter solution must be a fool's errand as it changes the neutrality of the speaker. I guess that the fool will not take care about this but enjoy the better sound.

By definition, room mode problems lay below the transition frequency (i.e. below around 300 Hz). No-one here is disputing that EQ'ing a speaker is beneficial below the transition frequency - we are all in violent agreement here, and that position has strong support from many studies. Above the transition frequency however, the deviations are smaller and the human auditory system is better equipped to separate direct sound from reflected sound, which is why this approach is ill-advised at such frequencies.
 

TimVG

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Simple example:
if a room boosts a frequency by a room mode you can change the room. To change the living room to an anechoic chamber will certainly help.
But what if the spouse does not like to live in an anechoic chamber?
Is it better to leave the "neutral" speaker untouched or is it better to reduce the critical frequency by a filter? Of course the latter solution must be a fool's errand as it changes the neutrality of the speaker. I guess that the fool will not take care about this but enjoy the better sound.

Below the transition frequency we can adjust the steady-state response to benefit, above it - it becomes more complex than that as been stated many, many, many times over in this thread
 

UliBru

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By definition, room mode problems lay below the transition frequency (i.e. around 300 Hz). No-one here is disputing that EQ'ing a speaker is beneficial below the transition frequency - we are all in violent agreement here, and that position has strong support from many studies. Above the transition frequency however, the deviations are smaller and the human auditory system is better equipped to separate direct sound from reflected sound, which is why this approach is ill-advised at such frequencies.
Indeed I do agree to a certain amount.
But:
- usually people use the speakers they own. Can we assume that all these speakers are neutral? Which percentage do you estimate to be neutral?
- if many speakers are not neutral above the magic transition frequency would it be great if we could get them more neutral?
- the auditory system can obviously separate direct sound from reflected sound. Can't we imagine that it is possible to separate direct sound from reflected sound by some numerical approach on an in-room measurement? More or less exact?
- if we can distinguish between direct and reflected sound can we imagine to check the direct sound for its parameters (or quality)?
- can we imagine to improve the direct sound by a filter, at least for all the less neutral speakers around there? Even above the transition frequency?
- if a filter is beneficial below the transition frequency can't it be beneficial for the full frequency range?
 

TimVG

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- if many speakers are not neutral above the magic transition frequency would it be great if we could get them more neutral?

You could by making your own anechoic measurements and correcting what can be corrected.

e auditory system can obviously separate direct sound from reflected sound. Can't we imagine that it is possible to separate direct sound from reflected sound by some numerical approach on an in-room measurement? More or less exact?

No, since an omni-directional microphone cannot seperate direct from indirect sound without loss of resolution, and it cannot also not discern the direction from which the directed sounds stem

if we can distinguish between direct and reflected sound can we imagine to check the direct sound for its parameters (or quality)?

You can by means of anechoic measurements

can we imagine to improve the direct sound by a filter, at least for all the less neutral speakers around there? Even above the transition frequency?

Sure, but it cannot be done from the listening position
 

RayDunzl

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Is there any accepted, reproducible demonstration of just what on Earth "timbre" is, in any sort of technical definition that can be applied across contexts?

Simple case, reproducible demonstration:

Pronounce the letters A E I O, you changed the timbre of your voice.
 

BDWoody

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Is there any accepted, reproducible demonstration of just what on Earth "timbre" is, in any sort of technical definition that can be applied across contexts?

1412440.jpg
 

UliBru

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You could by making your own anechoic measurements and correcting what can be corrected.
Wait a moment, I have to google for the next anechoic chamber near my house ;)

Sure, but it cannot be done from the listening position
An unproven claim can be contradicted without proof.

Talking about the (misnomed) room correction seems to be like hitting the play button of a video recorder. Immediately the same movie starts. The content: self-appointed fire-fighters full heartedly deny the possibility of room correction. :)
 

LuckyLuke575

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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.
 

TimVG

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Wait a moment, I have to google for the next anechoic chamber near my house ;)


An unproven claim can be contradicted without proof.

Talking about the (misnomed) room correction seems to be like hitting the play button of a video recorder. Immediately the same movie starts. The content: self-appointed fire-fighters full heartedly deny the possibility of room correction. :)


Ok then, enjoy the circle of confusion!
 

Tks

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Simple case, reproducible demonstration:

Pronounce the letters A E I O, you changed the timbre of your voice.

I don't get it. How would this be different than "pronouncing" anything else? I could easily just say:

Pronounce the letters A E I O, you changed the (insert whatever property of sound perception you want) of your voice. Which in conclusion could be something like:

"Changed the volume of various frequencies your vocal cords were producing... of your voice"

No revelation at all of what this "timbre" term means in the contexts it's being used (worse if attempting to being used in some scientific realm). If there is no spectrum to understand "timbre's" quantitative presence somewhere in relation to any other metrics that can be dialed.. then how would anyone know it even exists outside of a purely subjective machination?

People argue dark matter exists, while no one has detected any of it physically. The way they're able to do that is by the observation of the velocity of galaxies and the rates at which their motion occurs. That would be impossible with the current accounting of the amount of physical matter that exists, so then people rationlize there is some force doing this, detectable or not currently. But we come to this conclusion based on understanding of all the other properties that allowed us to come to this conclusion (mass, matter, gravity, etc...)

Since I've never seen timbre explain even remotely adequately in any objective term, does timbe even have it's own externalities like gravity/mass/velocity/matter does to concluding dark matter/energy exists?
 

Juhazi

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Discussion about EQ'ing above Schröder is weird here. Of course it can be done with good results! But not with automatic EQ!

Phase one:
Speaker's direct and off-axis responses above Schröder can be rather easily measured at home using quasi-anechoic semi-nearfield technique. Measurement artefacts can rather easily be distinguish and "normalize" with some experience of taking measurements. These can be EQ'd quite precisely with dsp using parametric or FIR EQ. DI however can not be changed this way, because it depends on physics and selected xo-points and slopes. ( A diy'er with a multi-way dsp speaker can control those too)

Phase two:
Measure each speaker at intended location and mic at listening spot (preferably several points or MMM/RTA) and full octave smoothing! Fine-tune placement first and yes you can suppress the lowest mode peaks if you want (preferbly half of them only!) Now you can set low-Q (Full-octave) peak or shelf corrections of .2-3dB and listen to the sound until you are pleased and response looks nice. Yes, left and right separately! You will be amazed how small changes are audible!

Remember that you need several measurements and first of all it must sound good! With dsp it is easy to save several settings and change them several times. Also remember to not eq small peaks and dips! When you get more familiar with measurements, check also decay and RT to find problematic frequencies that don't respond to EQ.

Don't get worried if you find controversial opinions and warnings of doing this. You are a free (wo)man and doing this doesn't cause global warming, kill animals or induce political crisis. Just stay out of hifi forums!
 
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TimVG

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Discussion about EQ'ing above Schröder is weird here. Of course it can be done with good results! But not with automatic EQ!

Phase one:
Speaker's direct and off-axis responses above Schröder can be rather easily measured at home using quasi-anechoic semi-nearfield technique. Measurement artefacts can rather easily be distinguish and "normalize" with some experience of taking measurements. These can be EQ'd quite precisely with dsp using parametric or FIR EQ. DI however can not be changed this way, because it depends on physics and selected xo-points and slopes. ( A diy'er with a multi-way dsp speaker can control those too)

Phase two:
Measure each speaker at intended location and mic at listening spot (preferably several points of MMM) and full octave smoothing! Fine-tune placement first and yes you can suppress the lowest mode peaks if you want (preferbly half of them only!) Now you can set low-Q (Full-octave) peak or shelf corrections and listen to the sound until you are pleased and response looks nice. Yes, left and right separately! You will be amazed how small changes are audible!

Remember that you need several measurements by first of all it must sound good! With dsp it is easy to save several settings and change them several times. Also remember to not eq small peaks and dips!

Don't get worried if you find controversial opinions and warnings of doing this. You are a free (wo)man and doing this doesn't cause global warming, kill animals or induce political crisis. Just stay out of hifi forums!

Agreed - but for me Phase2 for below the transition frequency only ;-)
 

Absolute

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Some people say "timing is an unscientific concept" and others say the direct sound is the alpha and the omega. Strange then that the relative timing between the individual drivers doesn't matter.

Smells like BS to me regardless of who says it.
 

TimVG

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Some people say "timing is an unscientific concept" and others say the direct sound is the alpha and the omega. Strange then that the relative timing between the individual drivers doesn't matter.

Smells like BS to me regardless of who says it.

Phase in crossovers matters of course, since it directly influences the amplitude response.
 

Hipper

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DSP above the transition frequency. My understanding is that those that claim their processors can do this rely on algorithms based on how a typical listening room behaves. In their multi position measuring process combined with the algorithm these systems can distinguish between direct sound and, presumably, all the first, second and maybe other reflections. In that way they can select the amount each frequency needs to be adjusted for the correct balance of the direct sound. Or something like that! They are therefore not exact but an educated guess. Better perhaps then just EQing flat based on one microphone measurement on REW.

Timbre. I just understood this to be the different sound we hear when the same note is played by different instruments, or the same instrument played in a different way. It seems to be caused by the different harmonics that each instrument causes in addition to the main, or first, harmonic. It can't be measured (like taste), just described. There are spectrograms of different instruments which can show differences.
 

Floyd Toole

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Hopefully all of the manufacturers are aiming for such speakers, albeit with different levels of other priorities.
Which brings me to some questions I've always pondered regarding such tests;

- Have any tests been done where the highest rated speakers were tested against itself only room-compensated?

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.

- Do you know if there's been noted differences in relative placements of highest rated speakers depending on where it was tested?
For example if nr 2 in one test became number 1 in a smaller/bigger/different environment, perhaps because of wider/narrower directivity.

- Would it matter how far away the listening positions were? Far off you have a significant drop in higher frequencies that you won't have up close.
Or are most things above 10k rather meaningless in the big picture?

Section 7.6.2 in the 3rd edition discusses a persuasive experiment showing the extent to which humans can adapt to different room and not alter their ratings of good loudspeakers.

The differences in HF rolloff due to different listening distances in domestic rooms is very small, very likely inaudible. Google should have lots of data on air attenuation vs. distance at different frequencies. Or, see Figure 10.12 in the 3rd edition.

Alot of people dealing in acoustics are talking about the first 5-7 ms as critical because that's perceived as direct sound. When you tested for preference for lateral reflections, did you find a particular point in time where the preference factor changed significantly?

See Section 7.6.5 in the 3rd edition and/or Olive, S. E. and Toole, F. E. (1989). “The Detection of Reflections in Typical Rooms”, J. Audio Eng. Soc., 37, pp. 539-553. In small rooms if the speakers are aimed at the listeners the first lateral reflections originate at close to 90 deg off axis - these sounds are greatly attenuated at mid and high frequencies.
 

Floyd Toole

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DSP above the transition frequency. My understanding is that those that claim their processors can do this rely on algorithms based on how a typical listening room behaves. In their multi position measuring process combined with the algorithm these systems can distinguish between direct sound and, presumably, all the first, second and maybe other reflections. In that way they can select the amount each frequency needs to be adjusted for the correct balance of the direct sound. Or something like that! They are therefore not exact but an educated guess. Better perhaps then just EQing flat based on one microphone measurement on REW.

Timbre. I just understood this to be the different sound we hear when the same note is played by different instruments, or the same instrument played in a different way. It seems to be caused by the different harmonics that each instrument causes in addition to the main, or first, harmonic. It can't be measured (like taste), just described. There are spectrograms of different instruments which can show differences.

" I just understood this to be the different sound we hear when the same note is played by different instruments, or the same instrument played in a different way." And, it is the different sounds of the same instruments and voices in recordings when they are played through different loudspeakers. These are definitely amenable to critical analysis and it turns out the the most preferred loudspeakers are those that add the fewest audible resonances thereby minimally altering the timbre of the instruments and voices. These are "neutral" loudspeakers.
 
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RayDunzl

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No revelation at all of what this "timbre" term means in the contexts it's being used (worse if attempting to being used in some scientific realm). If there is no spectrum to understand "timbre's" quantitative presence somewhere in relation to any other metrics that can be dialed.. then how would anyone know it even exists outside of a purely subjective machination?

Timbre is the amplitude of the components of the harmonic series in relation to the fundamental tone.

As you "speak" you are constantly changing the amplitude of the components of the harmonics of the fundamentals in your vocal sound, as well as adding some transients - generally consonants, to the vowel sounds. You also add "noise" - the letters C, F, S, H, X, and SH combination usually make noise (no defined frequency).

A clarinet, a trumpet, a flute, a stringed instrument all produce different harmonic series even if playing the same "note". That's what I call "timbre" in the basic case.

Bass Guitar (handy home-grown example), one note played:

The fundamental frequency is marked "1", the harmonics measured are "2" though "9". There are more, but the software (REW) chooses not to count any higher.

The harmonic series is multiples of the fundamental frequency.

1570402337483.png


Timbre may include more information, the noise a violin bow makes scraping the string, the (often) enharmonic series of drums, cymbals...

Timbre is the characteristic "sound" something makes, defined by the amplitudes of the frequencies (or noise) that accompanies the fundamental (if there even is one).

A pure tone - sine wave - is rarely encountered. That's a Fundamental with no harmonics. Its timbre is identified by the lack of harmonics. The closest I've come to producing one occurred when blowing across the top of a beer bottle:

1570402305031.png


If the reproduction system does not reproduce the recorded harmonic series correctly, the timbre of the sound will be changed.

I'd say the reproduction of Timbre is pretty robust - I can't remember mistaking a flute for a trumpet, on any system.
 
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Tks

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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"
 

jhwalker

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Timbre is the amplitude of the components of the harmonic series in relation to the fundamental tone.

As you "speak" you are constantly changing the amplitude of the components of the harmonics of the fundamentals in your vocal sound, as well as adding some transients - generally consonants, to the vowel sounds. You also add "noise" - the letters C, F, S, H, X, and SH combination usually make noise (no defined frequency).

A clarinet, a trumpet, a flute, a stringed instrument all produce different harmonic series even if playing the same "note". That's what I call "timbre" in the basic case.

Bass Guitar (handy home-grown example), one note played:

The fundamental frequency is marked "1", the harmonics measured are "2" though "9". There are more, but the software (REW) chooses not to count any higher.

The harmonic series is multiples of the fundamental frequency.

View attachment 35325

Timbre may include more information, the noise a violin bow makes scraping the string, the (often) enharmonic series of drums, cymbals...

Timbre is the characteristic "sound" something makes, defined by the amplitudes of the frequencies (or noise) that accompanies the fundamental (if there even is one).

A pure tone - sine wave - is rarely encountered. That's a Fundamental with no harmonics. Its timbre is identified by the lack of harmonics. The closest I've come to producing one occurred when blowing across the top of a beer bottle:

View attachment 35324

If the reproduction system does not reproduce the recorded harmonic series correctly, the timbre of the sound will be changed.

I'd say the reproduction of Timbre is pretty robust - I can't remember mistaking a flute for a trumpet, on any system.

I had written out a response on this very subject almost identical to your contribution, but as I was coming here to post it, I saw your very clear and concise contribution and realized nothing more was necessary ;) Good job.
 
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