At which sound pressure levels are loudspeakers calibrated to achieve a flat perceived loudness curve?

[Endibol]

From the Fletcher-Munson curves we know that the ratio between sound pressure level per frequency needs to vary with overall loudness level in order to perceive equal loudness over the entire audible spectrum. At low listening levels much more energy is needed for a bass note to sound equally loud as a mid frequency tone than at high listening levels.
This means that we only perceive equal loudness at one specific mutual ratio of the sound pressure levels of the woofer and tweeter (assuming a two-way system for simplicity).
I ask myself: at which sound pressure level do the loudspeaker manufacturers adjust the balance between the woofer and tweeter?

 

[abdo123]

It doesn’t matter because it all depends on what level the producer had in the studio during the production of a particular track.

However some speakers do have a loudness curve baked in, not a fan. Let me play with my toy instead of playing with it for me.

Speaker also don’t measure (significantly) differently based on input voltage. So without equalisation you can’t really tune any response to your liking.

 

[restorer-john]

I ask myself: at which sound pressure level do the loudspeaker manufacturers adjust the balance between the woofer and tweeter?

Typical listening levels where driver compression hasn't kicked in.

This, from a thread on loudness curve usage:

It’s funny that there’s some decent consensus that loudness compensation is important but speakers aren’t built to an implied listening level. Or maybe they are. I don’t have data to back it up, but I suspect people that like high SPL gravitate to flat speakers more than those preferring moderate volumes.

Should you use Fletcher-Munson loudness compensation?

I know many are familiar with the Fletcher-Munson or equal loudness curves. In a nutshell: “The Fletcher Munson Curve contains a set of graphs that show that when you listen to music at a lower volume, mid-range frequencies will be sound more prominent, whereas high and low frequencies will be...

www.audiosciencereview.com

 

[MaxwellsEq]

You are at a concert. A cello is playing very quietly. Because of the Fletcher-Munson curve, you perceive the the middle tones as louder than the higher and lower notes. It plays much louder, suddenly the lower and higher tones are clearer.

Next to you is a microphone. It doesn't suffer from the curve! It records faithfully to a machine, which also doesn't suffer from the Fletcher-Munson curve. You listen to the result on a perfectly flat set of speakers. The quiet bits, just like the concert has quieter lower and upper tones, because your hearing has the Fletcher-Munson curve. Whilst the louder bits, just like the concert have clearer upper and lower tones.

So the Fletcher-Munson curve is NEVER encoded in the signal, so amplifiers with a loudness button distorts the signal and a speaker that tries to correct for it will ALWAYS be wrong.

 

[Sancus]

(good) speakers are designed to be anechoically flat when measured by a microphone. Not perceptually flat. The curve of equal loudness perception is never flat at any SPL.

The real variable that matters is "at what spl is the content mixed/mastered?" since almost all content is going to be altered by EQ, compression, etc before it is released. And the answer there is usually about 80-85dB, it depends on the audio engineers, the type of content/standard they're mixing for(tv/film have real standards, music doesn't), etc.

Given that amount of bass/treble is also a matter of preference, as long as you listen around 80dB you're probably hearing more or less what was intended. If you listen significantly quieter, say 70dB, you likely need to add some bass boost via EQ or a room correction target curve.

 

[kemmler3D]

amplifiers with a loudness button distorts the signal

Agree that a loudness-compensated speaker will always be wrong except at the exact volume where that loudness compensation would be correct for a certain recording. This is not so much like a broken clock being right twice a day as a broken calendar being right a few times per millennium. Agree that it is a bad idea in speakers.

The amplifiers do distort the signal with a loudness compensation, but depending on the music and your playback level, the distortion might be more "correct" than not adding it.

To extend your cello example - if the live cello performance averaged 89dB SPL and you're playing it back at 80dB SPL, some loudness compensation *could* make the perceived tonality closer to the original performance.

The devil is in the details, but in theory loudness compensation filters can make sense.

 

[MaxwellsEq]

To extend your cello example - if the live cello performance averaged 89dB SPL and you're playing it back at 80dB SPL, some loudness compensation *could* make the perceived tonality closer to the original performance.

The devil is in the details, but in theory loudness compensation filters can make sense.

I understand this, but which curve would you employ to not corrupt the fidelity (assuming we don't mind wrecking linear phase to achieve the effect)?

You can never know what the correct shape is, because:
a) Each curve is a slightly different shape (and they are an average of many people, so each individual is different!)
b) The shape would be relative to the "correct" listening level for the recording. Bob Katz and his K-System assumes that monitoring level is at 85dB, so we could use that. But we know that each recording is potentially slightly different.
c) The curves (and the equivalent ISO versions), don't exist - they are like isobars. In other words, you can't switch from curve 7 to curve 8, because there are an infinite number of curves in between, each as slightly different shape and getting more extreme the quieter the listening level is.
d) There is often "gain-riding" by the mixing engineer and mastering engineer and sometimes they apply compression, which means very few recordings have louder quiet sections than would happen in real life, thereby reducing the Fletcher-Munson curve!

Ultimately, there is no way to reliable create a Loudness button which doesn't mess up phase and which actually corrects for lower listening levels. That doesn't mean people don't benefit from a bit of bass and treble boost at lower listening levels, but they are not accurately correcting for equal loudness and are messing up phase, so Loudness buttons always distort.

 

[Sokel]

The sloppy way I do it using foobar and Mathaudio plug-in is having 5 presets lowering the midrange area/boosting low's/etc, according to attenuation close to the known chart demonstrated above.

It's really easy to load them in real time and hear the difference and makes listening to lower levels much more enjoyable and at such levels critical listening is not really important.

 

[Andysu]

You are at a concert. A cello is playing very quietly. Because of the Fletcher-Munson curve, you perceive the the middle tones as louder than the higher and lower notes. It plays much louder, suddenly the lower and higher tones are clearer.

return of the jedi , cello THX sound system george lucas depressed

THX - George Lucas, Star Wars, & A Disappearing Cello: The Story Of THX Cinema Certification

All George Lucas wanted to do was hear his film soundtrack to Return of the Jedi in the way it was meant to be heard. The rest is history...

www.thx.com

 

[restorer-john]

Next to you is a microphone. It doesn't suffer from the curve!

Proximity effect in any directional microphone will change the low frequency balance depending on how near or far the mic is to the sound source.

 

[Cars-N-Cans]

From the Fletcher-Munson curves we know that the ratio between sound pressure level per frequency needs to vary with overall loudness level in order to perceive equal loudness over the entire audible spectrum. At low listening levels much more energy is needed for a bass note to sound equally loud as a mid frequency tone than at high listening levels.
This means that we only perceive equal loudness at one specific mutual ratio of the sound pressure levels of the woofer and tweeter (assuming a two-way system for simplicity).
I ask myself: at which sound pressure level do the loudspeaker manufacturers adjust the balance between the woofer and tweeter?

In the old days this was the "loudness" button on receivers, or more precisely, "fix-the-sh**ty-speakers" button as I, and I'm sure many others, really used it.

As far as speakers go, we don't want to do any such thing since this is entirely from us and not the speakers. In many ways, just like conventional music scale being in octaves, these "loudness" curves are already built into the music we make and it having a mostly 1/f spectral content. Someone like us hears it while they make it, thinks it sounds good, and we listen to it. The speakers just convey it.

 

[dasdoing]

loudspeaker manufacturer don't do this anymore it has been decades. a flat response is targeted.

 

[abdo123]

Proximity effect in any directional microphone will change the low frequency balance depending on how near or far the mic is to the sound source.

Good point.

 

[DVDdoug]

Proximity effect in any directional microphone will change the low frequency balance depending on how near or far the mic is to the sound source.

In a professional recording the mixing engineer would EQ that out. ...Or they might leave it for "effect". Most mics have a certain "character", most studio mics are not flat and the recording engineer usually chooses different mics for different sources. It's all "artificial"!

I do miss having a loudness switch. It was imperfect but IMO - Overall it helped a lot. And I'm too lazy to play-around with EQ all the time. If I want "good sound" I'll turn it up! Although not usually to "realistic" levels.

 

[kemmler3D]

I understand this, but which curve would you employ to not corrupt the fidelity (assuming we don't mind wrecking linear phase to achieve the effect)?

You can never know what the correct shape is, because:
a) Each curve is a slightly different shape (and they are an average of many people, so each individual is different!)
b) The shape would be relative to the "correct" listening level for the recording. Bob Katz and his K-System assumes that monitoring level is at 85dB, so we could use that. But we know that each recording is potentially slightly different.
c) The curves (and the equivalent ISO versions), don't exist - they are like isobars. In other words, you can't switch from curve 7 to curve 8, because there are an infinite number of curves in between, each as slightly different shape and getting more extreme the quieter the listening level is.
d) There is often "gain-riding" by the mixing engineer and mastering engineer and sometimes they apply compression, which means very few recordings have louder quiet sections than would happen in real life, thereby reducing the Fletcher-Munson curve!

Ultimately, there is no way to reliable create a Loudness button which doesn't mess up phase and which actually corrects for lower listening levels. That doesn't mean people don't benefit from a bit of bass and treble boost at lower listening levels, but they are not accurately correcting for equal loudness and are messing up phase, so Loudness buttons always distort.

Agree about all of this, although the importance of the phase distortion added by the filters is pretty debatable. You wouldn't really need to "switch" between curves though, there's no reason you can't have a system that continuously changes the FR depending on output level. There's plenty of reasons that it's not practical to have such a system, but it doesn't break the laws of physics or anything.

The biggest difficulty is knowing what the "right" reference level is. Would it be the 85dB supposedly used in the studio? Would it be what a cello's real SPL is, in a real hall? Or something else? That question makes the whole thing basically not worth it IMO.

 

[fpitas]

Ultimately, there is no way to reliable create a Loudness button which doesn't mess up phase and which actually corrects for lower listening levels. That doesn't mean people don't benefit from a bit of bass and treble boost at lower listening levels, but they are not accurately correcting for equal loudness and are messing up phase, so Loudness buttons always distort.

For the normal sort of filter, phase is defined by the shape of the filter response. If you want the boost, the phase comes along inevitably. And I'm not sure what negative effect that could have.

 

[-Matt-]

You wouldn't really need to "switch" between curves though, there's no reason you can't have a system that continuously changes the FR depending on output level.

This is exactly what Audyssey dynamic EQ tries to do.

The biggest difficulty is knowing what the "right" reference level is. Would it be the 85dB supposedly used in the studio? Would it be what a cello's real SPL is, in a real hall? Or something else?

This is the concept of "reference level", which In the AV world is firmly established. Sadly this is not so in the music industry.

 

[krabapple]

Of course F-M applies to the listener, meaning that you aren't hearing the recorded sound EQ balance 'as intended' unless you play it at the level it was created at.

*There's a good chance* that at home you are hearing it at a lower level than that. This is where modern 'loudness buttons' like Audyssey's Dynamic EQ come in handy. If it sounds right to you, use it. If not, turn it off. It's even adjustable if you are into fiddling. I find it quite useful.

 

[-Matt-]

So the answer to the original question (at least in an AV or mixed use system) is to calibrate at reference level and to use loudness correction if listening at a lower level.

Edit: And if you are listening to material mastered at a different level then you should use the appropriate reference level offset (a feature of Audyssey dynamic EQ).

 

[Marc v E]

Ime from my speakers a mild eq a low levels is very pleasent and so natural I don't even notice it anymore.

The rme adi has this kind of eq as one of its features. I've seen it mentioned multiple times on ASR that people love this dac because of this feature.

What I think is happening is that music recorded at a certain level, say 80db, must be played at a minimal level to hear the bass as was heard at 80db. Most probably 80db isn't necessary but say 50 db would probably be low enough for this effect to be real for a listener.