Testing audiophile claims and myths

[KeithEmo]

I agree entirely.

However, if you look them up in a dictionary, that's not what those words mean.
What the reviewer probably really meant was that "it failed to properly convey the sense of rhythm and pace".
And, by making up new meanings for words, or using them in a way that only insiders know what you really mean, they make it a virtual certainty that many readers will misinterpret their statement.

The whole object of communications is to convey meaning and information accurately.
This only works if we all agree that certain words mean certain things.
By making up a whole new language (by using normal words in new ways) they are (deliberately or not) creating a gap between "people who understand them and people who don't".
This may make "the insiders" feel cool, but it leaves the non-insiders trying to figure things out without much guidance.....
And, from what I read here and elsewhere, the result is often somewhere between misinformation and total confusion.
(For example, it would be natural for someone who doesn't read between the lines to believe that a device with "poor PRAT" had a clock that was running at the wrong speed.)

The proper way to describe that "PRAT" would be to say that "the speaker fails to convey a proper sense of rhythm on fast paced content, presumably due to excess energy storage at certain frequencies, which causes the output to continue after the input signal has stopped".
It may not sound nearly as poetic... but it's a lot more informative.
Stating it this way in fact serves several distinct and valuable purposes.
First, it is an accurate description that you can understand without asking an audio expert what the words mean (well, mostly).
Second, it tells a moderately informed person that they can probably see this as a quantifiable measurement - and they'll find it on the waterfall plot.
Third, it even tells an informed person where to look for a solution.... in this case, by looking for causes of excess energy storage and ways to correct them.
(Buying a speaker with a more solidly constructed cabinet, or a more well-damped cone, has a shot at reducing this problem; upgrading the clock in your CD player does not.)

I don't think an audio reviewer's reference to "rhythm and pace" (PRAT as some British reviewers call it) is related to the actual speed of a piece of music, but rather the ability of a speaker or headphone to keep up with the music. I think this is where driver speed and control comes in - where a driver may still be reacting to one sound when another comes in.

 

[jnorris]

I agree entirely.

However, if you look them up in a dictionary, that's not what those words mean.
What the reviewer probably really meant was that "it failed to properly convey the sense of rhythm and pace".
And, by making up new meanings for words, or using them in a way that only insiders know what you really mean, they make it a virtual certainty that many readers will misinterpret their statement.

The whole object of communications is to convey meaning and information accurately.
This only works if we all agree that certain words mean certain things.
By making up a whole new language (by using normal words in new ways) they are (deliberately or not) creating a gap between "people who understand them and people who don't".
This may make "the insiders" feel cool, but it leaves the non-insiders trying to figure things out without much guidance.....
And, from what I read here and elsewhere, the result is often somewhere between misinformation and total confusion.
(For example, it would be natural for someone who doesn't read between the lines to believe that a device with "poor PRAT" had a clock that was running at the wrong speed.)

The proper way to describe that "PRAT" would be to say that "the speaker fails to convey a proper sense of rhythm on fast paced content, presumably due to excess energy storage at certain frequencies, which causes the output to continue after the input signal has stopped".
It may not sound nearly as poetic... but it's a lot more informative.
Stating it this way in fact serves several distinct and valuable purposes.
First, it is an accurate description that you can understand without asking an audio expert what the words mean (well, mostly).
Second, it tells a moderately informed person that they can probably see this as a quantifiable measurement - and they'll find it on the waterfall plot.
Third, it even tells an informed person where to look for a solution.... in this case, by looking for causes of excess energy storage and ways to correct them.
(Buying a speaker with a more solidly constructed cabinet, or a more well-damped cone, has a shot at reducing this problem; upgrading the clock in your CD player does not.)

I agree with you also. Especially about the elitist terminology. It took a while to figure out that PRAT was Pace, Rhythm and Timing.

 

[Zapp_Fan]

Yep just wanted to chime in and say that, yes in theory EQ could improve perceived detail on the same transducer, if something was being hidden by masking effects before. Cut an unwanted peak at 500hz and suddenly something at 2Khz is audible? Plausible, although with any decent headphones or speakers, probably not a major effect to look for. Sensitivity / resonance / overall IR of the transducer should be expected to have a much, much larger effect. But let's not dismiss the masking effect, it's one of the major principles behind MP3 compression.

I also tend to agree that phase distortion introduced by (any reasonably implemented digital) EQ is very unlikely to be audible. It's very true that phase distortion introduced by all the mechanical components of the system should be expected to be orders of magnitude worse. Unless you're using extremely narrow bands at extremely high gain, I don't worry about it.

EQ is notorious as a source of placebo effect too. Many of us have sat adjusting an EQ for several minutes before realizing it was switched to "bypass". So switching the EQ in and out might also be expected to create a placebo effect with regard to expected audible changes in phase-related effects. Being subject to placebo effect doesn't mean you're a poor listener, it means you're human.

I do, in a philosophical sense, agree that less DSP is preferable, all else held equal. However, if you're confident enough that you have a problem with your headphone's or loudspeaker's frequency response that you can correct with EQ, I've come around to the idea that it can be worth the "decrease" in fidelity to the original signal.

 

[pinnahertz]

You're big on this "audiophile nonsense" thing. Fact is, all this stuff is "Audiophile Nonsense" to a non-audiophile. 1. Your willingness to screw up a perfectly linear signal with wanton equalization, 2. your decided lack of acknowledgement that additional signal processing is not a good thing, and 3. your arrogance to assume that you can properly tune out all the phase and amplitude errors introduced by "every single transducer known to man" with an audiophile-grade equalizer, indicate to me that you are indeed a non-audiophile.

1. Show me a "linear signal" coming out of a transducer.
2. You refer to equalization, and now any "additional signal processing" as if it's an on/ off thing. It's completely variable, and applicable as an inverse function. If EQ is so horrible, then perhaps we should eliminate it when playing records, tapes of all types, FM, AM, analog TV, every major recording studio, mixing desk, film dubbing stage, cinema....because they're all wrong?
3. What you think of as arrogance is actually engineering and science. Sorry. Yes, I'm proud to say I'm not an audiophile.

Do all your tweaking, tuning, and adjusting and listen to some well recorded stuff, then pull out the EQ and tell me if the music doesn't sound better - more dynamic, cohesive and transparent. You might even hear the detail that your equalizer is hiding.

Well, if turning off the EQ is an improvement, then it's set wrong. If you think "EQ" is a 1-octave or 1/3 octave graphic, then sure, but it's impossible to ever set those correctly, they're the wrong tool. I'm talking about precision EQ, IIR and better, FIR filters. I have found a rare situation when calibration accomplished little to nothing, but those are extremely rare, and well engineered systems.

 

[pinnahertz]

I don't think an audio reviewer's reference to "rhythm and pace" (PRAT as some British reviewers call it) is related to the actual speed of a piece of music, but rather the ability of a speaker or headphone to keep up with the music. I think this is where driver speed and control comes in - where a driver may still be reacting to one sound when another comes in.

More misunderstandings here: driver speed and control. For a driver to operate at any desired combination of frequency and amplitude, speed is already defined. Those parameters are inseparable. Resonance and, and the related properties of overshoot and damping, are more a function of driver and cabinet design and to a much lesser extent,amplifier interface. However, the concept of driver reacting to one sound when another comes in causing a problem is nonsense. The composite signal doesn't demand anything from a driver that any single signal within its design pass band demands. Velocity, frequency and amplitude are inseparable.

 

[bigshot]

Sorry, but the truth is that the phase errors and additional processing involved in equalization are detrimental to the sound, serving to mask rather than reveal details. If the amp/speakers cannot delineate fine detail, like the individual voice in a choir, then no amount of EQ will bring it out.

This may have been the case for a cheapie five band graphic equalizer back in the early 60s, but it isn't at all true for a halfway decent digital EQ. Equalization is essential to any speaker system. It can make midrange headphones sound as good as high end ones. Every mixing board I've ever seen has EQ on every channel and they studio engineers wouldn't use it if it did what you say. If you're dead set against EQ, you can spend a whole lot of money on your system, but odds are, it will still sound mediocre.

My AV receiver allows me to switch to "Direct" which cuts out all signal processing... DSPs and EQ. I can switch from my default calibrated position and direct back and forth very easily. The difference isn't subtle at all. My signal processing makes the sound more detailed, more dynamic and more natural sounding than playing the signal direct. The reason for that is because my signal processing has been carefully applied to correct for imbalances in my speakers and my room. The purity of the signal isn't a problem with modern solid state and digital home audio equipment. The problem is how that purity is translated to real world sound waves that you can hear. Tranducers and room are the key to a good speaker system.

I don't think an audio reviewer's reference to "rhythm and pace" (PRAT as some British reviewers call it) is related to the actual speed of a piece of music, but rather the ability of a speaker or headphone to keep up with the music. I think this is where driver speed and control comes in - where a driver may still be reacting to one sound when another comes in.

PRAT is one of those terms that you can safely use to discern if someone knows what they're talking about or not. If I hear someone use those terms, I know that I'm going to have to limit my responses to Audio 101 topics. Pinnahertz, I doubt he understands anything you're saying. If he's using PRAT, he's not going to understand many of the technical terms you're using.

I'm taking it one step at a time... Jnorris, have you googled "frequency masking" yet? What did you find? See the post just below this one if you want an example of what it means.

 

[bigshot]

Yep just wanted to chime in and say that, yes in theory EQ could improve perceived detail on the same transducer, if something was being hidden by masking effects before. Cut an unwanted peak at 500hz and suddenly something at 2Khz is audible? Plausible, although with any decent headphones or speakers, probably not a major effect to look for.

You would be surprised. I have a friend who is a sound engineer who gave me a vivid example of it once. He stood to block what he was doing on the equalizer and asked me to listen and tell him what I heard. I listened and I heard the high end of the cymbals going in and out. Muffled, sharp, muffled, sharp. I told him the treble was going in and out. He turned away from the equalizer and showed me the band he was adjusting. It was a midrange frequency, and the adjustment he was making in the midrange was quite small and almost inaudible. In music there are certain narrow bands that are very important... especially in the high end- consonants on vocals, high end on cymbals, etc. If there is an imbalance in the wrong place in the midrange, it can obliterate one of those narrow bands and the treble can be greatly affected. Treble is a big part of what people describe as "detail".

If you have an equalizer, try to isolate the frequencies of the sisss sound of the cymbal and then adjust a narrow spike one octave below it. You'll find the spot. It's really quite amazing. Not at all intuitive.

EQ is notorious as a source of placebo effect too. Many of us have sat adjusting an EQ for several minutes before realizing it was switched to "bypass".

I've never done that. I've waffled with a small correction back and forth, but that is usually because one piece of music I'm listening to is engineered a little different than another. When I find myself going into parallel park mode on a setting, I split the difference and live with it.

I think headphone users don't always understand what DSPs and EQ are for. It's easier to get a pair of headphones relatively flat than it is speakers in a room. And with headphones you don't really need DSPs to correct for room acoustics. With a speaker system, and particularly with a multichannel speaker system, DSPs and EQ are indispensable. If you get a really good set of headphones, you can get by without them.

 

[castleofargh]

Sorry, but the truth is that the phase errors and additional processing involved in equalization are detrimental to the sound, serving to mask rather than reveal details. If the amp/speakers cannot delineate fine detail, like the individual voice in a choir, then no amount of EQ will bring it out.

it's a misguided idea relying on the delusion that everything else isn't already altering the signal. and it's also IMO a vision of only one type of EQ which makes it a little dishonest. even objectively once you have defined an ideal target response, an EQ can improve the fidelity of the signal coming out of the headphone. it would probably become obvious to you if you went and measured a few situations for yourself.

@bigshot is right because it is strongly suggested that the best way to perceive the most information in music is a flat response(flat as in how our head would compensate for the response if the sound came from somewhere in front of us, so a subjective flat for headphone use). auditory masking alone fully justifies saying that we can get more details with a more balanced signature. so as far as we know, that is factual.

your stuff about phase errors, is hardly audible most of the time and when it is, it's usually because the FR compensation had to be drastic. which implies the sound would be crap without EQ anyway.
I'm not telling you anything new when I say that frequency response is a major aspect of how we perceive and interpret sound. IMO the very existence of crossover designs in most speakers is evidence that when presented with the choice, most designers went for tuning FR at the cost of phase and they didn't lose sleep over it given how for decades people have been very fine that way. the reason was that one single driver couldn't properly handle the entire audible range, but the fix ended up being several drivers and EQ in the form of an analog crossover. so speakers don't have flat phase response and music didn't die. because phase isn't the big thing some DAC manufacturers make it to be.

my hd650 doesn't have a flat phase response, I guess I should throw it away. and same with most of my IEMs(how I love reading a guy with a 1000+$ multidriver IEM saying that EQ degrades the sound). you're just presenting to yourself some idealistic vision where you can only see 2 options: ruining the phase response or not. so of course with that model you never see EQ as am improvement. but humans happen to care about FR a lot, and no headphone is fully flat for a user(well I guess it does happen at a statistical level for a few dudes somewhere). so you point of view is not the truth, because you look at the bad consequences of EQ and fail to notice how there are good consequences too and they often can result in overall improved sound.

 

[KeithEmo]

I agree with everything you said - except for one detail.

In speaker design, the basic intent of the crossover is NOT considered to be EQ.
The function of routing each range of frequencies to the correct driver is NOT considered to consist of "equalizing the sound going to each driver".
The BASIC purpose of the crossover is to send the right frequency to the right driver.... and, in its simplest form, it does NOT include EQ.
You are not "making it so the driver can handle more frequencies"; rather you are "preventing the driver from trying to handle frequencies it should not".
(The assumption is that the sum of all of the individual pieces will add up to the same unaltered total.)

In real life speaker designs, the crossover very often does ALSO includes EQ corrections applied to the individual drivers, or to the total.
However, this is considered to be a different operation than the basic function of separating the frequency content to the appropriate drivers.

it's a misguided idea relying on the delusion that everything else isn't already altering the signal. and it's also IMO a vision of only one type of EQ which makes it a little dishonest. even objectively once you have defined an ideal target response, an EQ can improve the fidelity of the signal coming out of the headphone. it would probably become obvious to you if you went and measured a few situations for yourself.

@bigshot is right because it is strongly suggested that the best way to perceive the most information in music is a flat response(flat as in how our head would compensate for the response if the sound came from somewhere in front of us, so a subjective flat for headphone use). auditory masking alone fully justifies saying that we can get more details with a more balanced signature. so as far as we know, that is factual.

your stuff about phase errors, is hardly audible most of the time and when it is, it's usually because the FR compensation had to be drastic. which implies the sound would be crap without EQ anyway.
I'm not telling you anything new when I say that frequency response is a major aspect of how we perceive and interpret sound. IMO the very existence of crossover designs in most speakers is evidence that when presented with the choice, most designers went for tuning FR at the cost of phase and they didn't lose sleep over it given how for decades people have been very fine that way. the reason was that one single driver couldn't properly handle the entire audible range, but the fix ended up being several drivers and EQ in the form of an analog crossover. so speakers don't have flat phase response and music didn't die. because phase isn't the big thing some DAC manufacturers make it to be.

my hd650 doesn't have a flat phase response, I guess I should throw it away. and same with most of my IEMs(how I love reading a guy with a 1000+$ multidriver IEM saying that EQ degrades the sound). you're just presenting to yourself some idealistic vision where you can only see 2 options: ruining the phase response or not. so of course with that model you never see EQ as am improvement. but humans happen to care about FR a lot, and no headphone is fully flat for a user(well I guess it does happen at a statistical level for a few dudes somewhere). so you point of view is not the truth, because you look at the bad consequences of EQ and fail to notice how there are good consequences too and they often can result in overall improved sound.

 

[castleofargh]

I agree with everything you said - except for one detail.

In speaker design, the basic intent of the crossover is NOT considered to be EQ.
The function of routing each range of frequencies to the correct driver is NOT considered to consist of "equalizing the sound going to each driver".
The BASIC purpose of the crossover is to send the right frequency to the right driver.... and, in its simplest form, it does NOT include EQ.
You are not "making it so the driver can handle more frequencies"; rather you are "preventing the driver from trying to handle frequencies it should not".
(The assumption is that the sum of all of the individual pieces will add up to the same unaltered total.)

In real life speaker designs, the crossover very often does ALSO includes EQ corrections applied to the individual drivers, or to the total.
However, this is considered to be a different operation than the basic function of separating the frequency content to the appropriate drivers.

I guess we're in a definition conflict here. to me a crossover even in it's most basic use applies a low pass or high pass filter, which to me is but one type of EQ.

 

[bigshot]

I would consider crossover to be EQ, but here's an even better one... A lot of audiophools who are vehemently opposed to EQ think vinyl LPs are the cat's pajamas. Do you suppose that they object to a phono preamp that applies the RIAA curve?

 

[Zapp_Fan]

I guess we're in a definition conflict here. to me a crossover even in it's most basic use applies a low pass or high pass filter, which to me is but one type of EQ.

I'd have to come down on the other side of that semantic debate. The purpose of the EQ is to give the user the ability to adjust FR, crossovers sometimes kinda fit this definition, (when they're the adjustable type) but usually don't, especially when built into loudspeakers or headphones. I guess it's not that big of a debate though. We all know the purpose of each device.

 

[castleofargh]

I'd have to come down on the other side of that semantic debate. A single filter (or even a few) does not comprise an EQ. An EQ is a series of filters designed to be user-adjustable, and has to have adjustable gain. Crossovers sometimes kinda fit this definition, (when they're the adjustable type) but usually don't, especially when built into loudspeakers or headphones. I guess it's not that big of a debate though. We all know the purpose of each device.

ok, I can live with that . the term crossover is clearly defined and it doesn't need to be called EQ. I was just implying that it does the same thing any analog EQ would do, alter FR and shift phase in the process. along with that idea, bigshot's vinyl is pretty good too, and the FR change of a colored amp(dismissing the distortion parts), is also showing the same behavior. which is why it can be so easy to correct both FR and phase by going the other way with the right digital filter or with an analog EQ.

 

[VandyMan]

I think that one of the issues with EQ is that most people have a poor understanding of what different frequency ranges contribute. I'm personally still learning. For example, it is widely believed on Head-Fi that "brightness" comes from the frequency extremes. I'm in my late 40s and I can't hear over 16.5K, but I can still perceive a "bright" headphone. The HD-800, a notoriously "bright" headphone, sounds that way due to the 8K peak, not because it has excess energy at say 20K. The problem is that if someone takes the HD-800, for example, and tries to fix it via EQ, they might roll-off the highest frequencies and be dissatisfied with the result. You need to understand certain fundamentals to do a proper EQ. That is why I'm looking forward to tools like the Realizer A-16 that will allow you to measure a headphone and automatically create the EQ curve.

 

[b0ck3n]

I'd never heard of masking frequencies before today but I am familiar with the phenomenon, truly fascinating stuff. The validity of EQ is beyond question but I've always had difficulties with its implementation when purely for music playback - I'm curious how do you guys use it? Do you ever listen to a recording and go "oh these Sennheisers totally mask the crap outta the snares on this recording" and adjust EQ accordingly?