Maybe "fast bass" is a very loose way to describe it, but nevertheless I do think there's a phenomenon here which involves bass lag. I'm actually a double bass player myself, and as I was saying previously the relationship of the bass to the rest of the music is important for swing and boogie factor.
As any musician knows, there's a difference between tempo and rhythm. The tempo may be constant but the whole feel of the music depends on how the accents are distributed within the rhythm - an LA samba is different from a Brazilian one, a New Orleans shuffle is different from a Chicago one. The accents are distributed differently.
In terms of bass, the question is where the bass "speaks". It's advanced in the beat because the bass speaks more slowly than the ride cymbal. The ride cymbal speaks the instant it is struck, but there is a lag before the middle of the bass note speaks. So a thin membrane like that of an electrostatic, a Maggie or a full range ribbon e.g. Apogee may possibly "speak" more quickly than a heavy cone with a much longer excursion. Hence the word "fast" in this context.
andy
As any musician knows, there's a difference between tempo and rhythm. The tempo may be constant but the whole feel of the music depends on how the accents are distributed within the rhythm - an LA samba is different from a Brazilian one, a New Orleans shuffle is different from a Chicago one. The accents are distributed differently.
In terms of bass, the question is where the bass "speaks". It's advanced in the beat because the bass speaks more slowly than the ride cymbal. The ride cymbal speaks the instant it is struck, but there is a lag before the middle of the bass note speaks. So a thin membrane like that of an electrostatic, a Maggie or a full range ribbon e.g. Apogee may possibly "speak" more quickly than a heavy cone with a much longer excursion. Hence the word "fast" in this context.
andy
Andy,
I'm actually intrigued by your idea of bass lag, as that might actually be a factor. The actual bass frequency, which admits of no deviation, can certainly be delayed, as many speaker builders know. The fact is that quite often "time alignment" (or the phase relationship) of the drivers in multi way speakers is adjusted at the crossover, but only at the crossover frequency as the acoustic centers change with frequency.
Best Regards,
TerryO
I'm actually intrigued by your idea of bass lag, as that might actually be a factor. The actual bass frequency, which admits of no deviation, can certainly be delayed, as many speaker builders know. The fact is that quite often "time alignment" (or the phase relationship) of the drivers in multi way speakers is adjusted at the crossover, but only at the crossover frequency as the acoustic centers change with frequency.
Best Regards,
TerryO
Is audiojoy perhaps referring to transient response when he speaks of fast bass? Please reread post 115
perhaps its as simple as having a bass lift at a certain frequency- ie 100hz.
On the basis that that frequency is normaly excited more at a later point in the decay or attack of a bass note maybe? I mean bass sounds subjectivly slow when its way off flat in room with a peak at 30hz, because in my mind when the note is exagerated, by nature it takes longer to occur and so decays out of sink with the rest of the harmonic information from the prospective of the human ear because it drops out of audiability (reduces in spl) at a later time than it should compaired to the rest of the information.
On the basis that that frequency is normaly excited more at a later point in the decay or attack of a bass note maybe? I mean bass sounds subjectivly slow when its way off flat in room with a peak at 30hz, because in my mind when the note is exagerated, by nature it takes longer to occur and so decays out of sink with the rest of the harmonic information from the prospective of the human ear because it drops out of audiability (reduces in spl) at a later time than it should compaired to the rest of the information.
Hi,
I quite agree, but requirements have to stated in design terms, not nebulous
"goals" to pontificated upon after the fact whether they are achieved or not.
The work done by Tool et al via blind listening tests is a classic example of
attempting to rationlise the subjective. I just get wound up by such drivel
as "I don't care about frequency response, I care about music", its simply
nonsense and predicates any further sensible analysis of the real issues.
Frequency response is the main issue, not least how it is defined, and from
that springs all further technical issues, because its not the only issue, if
you dismiss logic e.g. in your "philosophy" you do not have a philosophy.
"Hunches" I do not think so, knowing the subject and moving it on perhaps,
but not knowing the subject and "pure conjecture" simply does not work.
rgds, sreten.
FWIW IMO the secret to good bass is a system that has the resolution
to accurately represent the top end of a bass, that is the treble end
of the system is detailed enough to not lose the more subtle cues
lower in the mix coming from the bass player (and the bass drum).
Note I'm not talking bad bass, that is bass frequency issues, but to
be really good bass needs good top end transparency and detail.
For a good system the bottom end and top end "gel", one reason
I do not like biwiring, which tends to split them IM considered O.
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It seems that in most of the instances, where a speaker was said to have "fast bass" and was then subsequently measured, the results revealed a lack of the lowest bass registers and an exaggeration of a higher frequency, say 40-60 Hz which produces the overtones of the bass and can present the illusion of deeper bass with an accelerated (or quicker) response. All this is pretty well known to speaker builders, but to those, whose whole "knowledge" consists of regurgitating what some hare-brained, hack reviewer had written in a magazine, seem smitten with the idea of processing some sort of arcane knowledge that's been denied to the lesser folks.
Best Regards,
TerryO
this is nuts.
unless I got it wrong that is.
Is someone REALLY trying to say you can increase the 'boogie factor' (absolutely refuse to use the prat word PRAT) by leading or lagging the subs (woofer) in a playback system??
gee, I do hope I got it wrong because that is one of the silliest things I have heard in a long while, and on audio forums there are lots of silly things.
To take an example from musical creation (the drums and bass player being able to play on a slightly different time than the rest) and then conclude 'I can achieve the same thing by advancing/delaying some drivers' is crazy talk, and one (no doubt) that could only come from not having the slightest clue of which they are talking.
the ONLY place that can occur is during the recording..after that we have to recreate the recording as best we know how.
I mean, to think that delaying your woofer delays 'only the bass player' is absurd in the extreme.
why is time being wasted on this?
unless I got it wrong that is.
Is someone REALLY trying to say you can increase the 'boogie factor' (absolutely refuse to use the prat word PRAT) by leading or lagging the subs (woofer) in a playback system??
gee, I do hope I got it wrong because that is one of the silliest things I have heard in a long while, and on audio forums there are lots of silly things.
To take an example from musical creation (the drums and bass player being able to play on a slightly different time than the rest) and then conclude 'I can achieve the same thing by advancing/delaying some drivers' is crazy talk, and one (no doubt) that could only come from not having the slightest clue of which they are talking.
the ONLY place that can occur is during the recording..after that we have to recreate the recording as best we know how.
I mean, to think that delaying your woofer delays 'only the bass player' is absurd in the extreme.
why is time being wasted on this?
Perhaps you see it as a waste of time. Others see it as a timing issue. 😉
If the bass player plays slightly ahead or behind the beat and that changes the feel, why wouldn't delays in some frequency range change the feel? Sure, the harmonics of the double bass may end up in another driver, but that would just tend to confuse the issue. If the fundamental and the harmonics of the bass are offset in time, don't you think that would sound at least "different"?
If the bass player plays slightly ahead or behind the beat and that changes the feel, why wouldn't delays in some frequency range change the feel? Sure, the harmonics of the double bass may end up in another driver, but that would just tend to confuse the issue. If the fundamental and the harmonics of the bass are offset in time, don't you think that would sound at least "different"?
Terry o. My understanding was that notes had a fundamental tone frequency then harmonics as multiples of that tone. So sounds from a speaker from different frequencies should in that case be able to influence a fundamental tone???
Yes but you would be delaying the lower registers of piano, drums and bass or any other fundamentals in the woofers range all the same amount. How does that help?? I don't see how an audible timing shift would in any way be beneficial or natural sounding. I think it would make things sound disjointed for lack of a better word.
Rob🙂
oh, it was serious.
oh dear.
so you are trying to get the bass player to play 'just that little bit before' the rest, so advance the woofer.
Umm, what about the bass from every other instrument you are trying to get the bass player to play ahead of?
They are inextricably linked, it is on the recording, you cannot do anything about it. Advance one, you advance all, delay one, you delay all.
For sure, sit in the studio with the drums, guitar and bass etc all on individual tracks. When you mix them (I spose, not a recording engineer) you can slightly shift the time of a complete track, so in that case the bass player IS playing at a slightly different time in the finished recording.
Once it has been set in stone, that's it buddy. So, to tease maximum boogy factor out of the recording you need to have all the frequencies arrive at exactly the same time. (haha, going mad with italics and all eh?)
Have any of you even looked in the time domain?? Get on a diy forum, and everyone talks about FR, peaks and dips.
Bugger FR, give me correct time every day. (well, I'll take both thanks)
The holy grail of music reproduction is timing, to have all frequencies arrive at the same time and in phase, THAT (just to avoid another set of itallics) is where the magic lies, the ambient information, the correct tonal balance.
Do an experiment, those of you who can, run a sweep of your system, now delay the subs a few milliseconds, then run the sweep again.
Show us the graphs of the FR you get...huge variations from just changing the delay. Try it, I'm serious.
Delaying a driver (in the completely misguided attempt to infuse some sort of 'musicality' idea) will wreak havoc on the very thing you are trying to get, maximum extraction of musical enjoyment from a FIXED source, cd or lp etc.
Don't play anything, just sit in front of your speakers. Not very musical are they.
So ANY sense of musicality can ONLY come from the recording, it is no where else.
All we can do is degrade that information, and I assure you, one of the best ways to degrade it is to muck with the timing. In fact, the more you get the timing right the better it is.
Why? (geeze, it feels silly to have to state the bleeding obvious)..because you are best preserving what is on the recording, the only place musicality can ever come from.
oh dear.
so you are trying to get the bass player to play 'just that little bit before' the rest, so advance the woofer.
Umm, what about the bass from every other instrument you are trying to get the bass player to play ahead of?
They are inextricably linked, it is on the recording, you cannot do anything about it. Advance one, you advance all, delay one, you delay all.
For sure, sit in the studio with the drums, guitar and bass etc all on individual tracks. When you mix them (I spose, not a recording engineer) you can slightly shift the time of a complete track, so in that case the bass player IS playing at a slightly different time in the finished recording.
Once it has been set in stone, that's it buddy. So, to tease maximum boogy factor out of the recording you need to have all the frequencies arrive at exactly the same time. (haha, going mad with italics and all eh?)
Have any of you even looked in the time domain?? Get on a diy forum, and everyone talks about FR, peaks and dips.
Bugger FR, give me correct time every day. (well, I'll take both thanks)
The holy grail of music reproduction is timing, to have all frequencies arrive at the same time and in phase, THAT (just to avoid another set of itallics) is where the magic lies, the ambient information, the correct tonal balance.
Do an experiment, those of you who can, run a sweep of your system, now delay the subs a few milliseconds, then run the sweep again.
Show us the graphs of the FR you get...huge variations from just changing the delay. Try it, I'm serious.
Delaying a driver (in the completely misguided attempt to infuse some sort of 'musicality' idea) will wreak havoc on the very thing you are trying to get, maximum extraction of musical enjoyment from a FIXED source, cd or lp etc.
Don't play anything, just sit in front of your speakers. Not very musical are they.
So ANY sense of musicality can ONLY come from the recording, it is no where else.
All we can do is degrade that information, and I assure you, one of the best ways to degrade it is to muck with the timing. In fact, the more you get the timing right the better it is.
Why? (geeze, it feels silly to have to state the bleeding obvious)..because you are best preserving what is on the recording, the only place musicality can ever come from.
As Terry says, you can't go back and change what's put on a CD. What happens in the studio is that it is the responsibility of the bass player to make sure his bass "speaks" in such a way that the music swings. It's well known that if you want to kill the swing, a bassist who drags will do it quite effectively. So we have to assume that we start with the timing correct.
But what if subsequently and for many reasons the bass lags? First of all, we can't at a later stage just advance the bass player like we can do in quantising software. As others have said, we could advance the subwoofer but that's not the same as advancing the actual bass - that opportunity is gone. It may turn out to be a "fix" of some kind (just as rolling off the bass or exaggerating its upper transients may be) and may get some of the job done but it's not perfect.
Reproducing bass requires more energy, so over-rated power supplies may be a factor. Components like DHT tubes and polypropylene PSU caps may help maintain a quick reaction to the signal. Reproducing bass in speakers requires more excursion to move air, so mechanical factors apply.
We can't change the fact that bass frequencies are longer wavelengths. If the bass player himself or herself has to make adjustments for this in the timing of finger on string, then why is it difficult to imagine that some sort of extra care may be required to ensure that the timing out of the speaker is - as much as possible - the timing achieved in the studio in the first place.
Think of it this way - the absolute timing is in the cymbals, snare and hi-hat of the drum kit. That's easy to reproduce, requires little energy from the reproduction system and is also in the optimum audible range of the ear+brain system, which has to decode the mechanical signals coming from the eardrum. The task is to create a reproduction system such that the ear drum receives all signals over the full frequency range with exactly the same timing that they were created in the studio and consequently the brain - at the end of the chain - decodes the signal from the studio with the correct timing.
andy
But what if subsequently and for many reasons the bass lags? First of all, we can't at a later stage just advance the bass player like we can do in quantising software. As others have said, we could advance the subwoofer but that's not the same as advancing the actual bass - that opportunity is gone. It may turn out to be a "fix" of some kind (just as rolling off the bass or exaggerating its upper transients may be) and may get some of the job done but it's not perfect.
Reproducing bass requires more energy, so over-rated power supplies may be a factor. Components like DHT tubes and polypropylene PSU caps may help maintain a quick reaction to the signal. Reproducing bass in speakers requires more excursion to move air, so mechanical factors apply.
We can't change the fact that bass frequencies are longer wavelengths. If the bass player himself or herself has to make adjustments for this in the timing of finger on string, then why is it difficult to imagine that some sort of extra care may be required to ensure that the timing out of the speaker is - as much as possible - the timing achieved in the studio in the first place.
Think of it this way - the absolute timing is in the cymbals, snare and hi-hat of the drum kit. That's easy to reproduce, requires little energy from the reproduction system and is also in the optimum audible range of the ear+brain system, which has to decode the mechanical signals coming from the eardrum. The task is to create a reproduction system such that the ear drum receives all signals over the full frequency range with exactly the same timing that they were created in the studio and consequently the brain - at the end of the chain - decodes the signal from the studio with the correct timing.
andy
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Well, if I might add something...
Some of "my" design goals, not (only) achieved by measurements, but from experiences with trying out quite a large range of different amps, cd players, d/a-converters and speaker drivers in the past (and the present).
1. More important than the qualities of each component on its own (cd player, preamp, amp) is the way these components get along with in terms of their technical or electrical specifications. Take a few different cd players and amps and connect them and see (or better: hear) what happens. Maybe at some point you connect a certain cd player "a" to amplifier "f", which you heaven't done before, and you press the play button and you think: "BINGO!" - This is exactly what happens to me from time to time. A source I considered mediocre before suddenly shines in a completely different light.
The components of a good stereo system have to be able to "melt together" when playing music. The whole system has to become "one thing".
Important for that is, for example: To really take care of the particular specifications of each unit. What works in most cases is a source with a very low output impedance and the ability to deliver a fair amount of current to drive the next stage properly. If you have one of those nos-dacs with passive i/v-conversion, you will need a preamp that fits in. If you have one of those cd players with a discrete or opamp based output stage, you have good conditions to drive a power amp directly via a 10k pot. The amp itself should have a rather highisch input impedance (22kOhm, better 33kOhm or above) and a high input sensitivity (around 0,5 to 0,7V). These are some of the circumstances when a source is able to drive the next stage without trouble, and this is something that makes music sound speedy, agile, quick, effortless, dynamic and alive!
Some people stated that using just a passive pot instead of a preamp will "suck the life out of the music" - this is in fact true, if someone misses the important technical aspects mentioned above! (One reason why so many people around here build the Pass B1 Bufferstage.)
2. I think we agree that some recordings are really, really bad! Some cd's just don't sound good or involving or anything, no matter what, and regardless of on what system you listen to them. But then there are recordings that have the "potential" to sound good, but do not on certain systems. This is when diy audio starts to improve things for a better, more involving and live-like presentation.
3. Regarding speakers, not only a crossoverless or minimum crossover design might help, but I think it's also about sensitivity. Almost every speaker I was listening to in the past that had a low spl (around the common 82 to 86dB/w/m) was sounding quite liveless to me. These designs to me tend to sound like the speaker cones are strapped or bounded. There is so much energy on its way from the amp to the human ear converted to heat and so little to cone movement, it's simply not worth it. Other, much more sensitive designs tend to sound like they really WANT to play, just awaiting the next electical impulse to convert it to a mechanical impulse for your pleasure 🙂.
This, like the first point, is also something that has to do with a source driving the next stage. In this case the amp is the souce and the speaker is the next stage. If your amp needs hundreds of watts to make your speakers sound good, then this also means that you have to play music quite LOUD to get involving and moving sound out of them! Try to listen to quite low volume levels with your chain and see if the music is completely there and if it grabs your attention (or, if you will: if it delivers the message!). If not, there might be a problem in one of the sources and their way of driving the next stage.
4. One more thing about the frequency response. I think a nearly flat frequency response is important if you intend to reproduce the informations of your cd or tape or whatever in the most realistic and accurate way possible. But it's not that important if you intend to be shaken and moved by the music. Rhythm and timing is not lost if there is a slight dip in the range of 2,8kHz, if you know what I mean. Of course a nearly flat response is a requirement to call a hifi system a good hifi system, but "nearly" flat might just be enough.
So far for the moment...
Regards!
Martin
Some of "my" design goals, not (only) achieved by measurements, but from experiences with trying out quite a large range of different amps, cd players, d/a-converters and speaker drivers in the past (and the present).
1. More important than the qualities of each component on its own (cd player, preamp, amp) is the way these components get along with in terms of their technical or electrical specifications. Take a few different cd players and amps and connect them and see (or better: hear) what happens. Maybe at some point you connect a certain cd player "a" to amplifier "f", which you heaven't done before, and you press the play button and you think: "BINGO!" - This is exactly what happens to me from time to time. A source I considered mediocre before suddenly shines in a completely different light.
The components of a good stereo system have to be able to "melt together" when playing music. The whole system has to become "one thing".
Important for that is, for example: To really take care of the particular specifications of each unit. What works in most cases is a source with a very low output impedance and the ability to deliver a fair amount of current to drive the next stage properly. If you have one of those nos-dacs with passive i/v-conversion, you will need a preamp that fits in. If you have one of those cd players with a discrete or opamp based output stage, you have good conditions to drive a power amp directly via a 10k pot. The amp itself should have a rather highisch input impedance (22kOhm, better 33kOhm or above) and a high input sensitivity (around 0,5 to 0,7V). These are some of the circumstances when a source is able to drive the next stage without trouble, and this is something that makes music sound speedy, agile, quick, effortless, dynamic and alive!
Some people stated that using just a passive pot instead of a preamp will "suck the life out of the music" - this is in fact true, if someone misses the important technical aspects mentioned above! (One reason why so many people around here build the Pass B1 Bufferstage.)
2. I think we agree that some recordings are really, really bad! Some cd's just don't sound good or involving or anything, no matter what, and regardless of on what system you listen to them. But then there are recordings that have the "potential" to sound good, but do not on certain systems. This is when diy audio starts to improve things for a better, more involving and live-like presentation.
3. Regarding speakers, not only a crossoverless or minimum crossover design might help, but I think it's also about sensitivity. Almost every speaker I was listening to in the past that had a low spl (around the common 82 to 86dB/w/m) was sounding quite liveless to me. These designs to me tend to sound like the speaker cones are strapped or bounded. There is so much energy on its way from the amp to the human ear converted to heat and so little to cone movement, it's simply not worth it. Other, much more sensitive designs tend to sound like they really WANT to play, just awaiting the next electical impulse to convert it to a mechanical impulse for your pleasure 🙂.
This, like the first point, is also something that has to do with a source driving the next stage. In this case the amp is the souce and the speaker is the next stage. If your amp needs hundreds of watts to make your speakers sound good, then this also means that you have to play music quite LOUD to get involving and moving sound out of them! Try to listen to quite low volume levels with your chain and see if the music is completely there and if it grabs your attention (or, if you will: if it delivers the message!). If not, there might be a problem in one of the sources and their way of driving the next stage.
4. One more thing about the frequency response. I think a nearly flat frequency response is important if you intend to reproduce the informations of your cd or tape or whatever in the most realistic and accurate way possible. But it's not that important if you intend to be shaken and moved by the music. Rhythm and timing is not lost if there is a slight dip in the range of 2,8kHz, if you know what I mean. Of course a nearly flat response is a requirement to call a hifi system a good hifi system, but "nearly" flat might just be enough.
So far for the moment...
Regards!
Martin
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I get the idea that audiojoy has no idea what he is talking about; ambiguous terms, no objectivity, no attempt at understanding any of the science behind it and a willingness to criticise others for their lack of understanding of a completely nebulous notion.
But he feels the music and perceives the fast bass. I make no apologies for saying as it appears.
It reminds me of motoring journalists who write about the feel of cars, the magic of the response on the road, the intuitive nature of the steering and the feedback of the brake pedal. ALL of this is achieved through the work of engineers, primarily in the design phase, and the "magic" of engineering calculations. Once you know what the customer wants, the design aspect is made easier (though not easy).
Unfortunately, audiojoy doesn't know what he wants other than something magical but can't qualify, and certainly not quantify, what that is. It seems pointless to discuss the unknowable.
Frank
Hello Andy
The differences between time variations of a person playing an instrument and time variations due to physical driver offsets are different by orders of magnitude. Any delays in a speaker system are fixed delays and usually only 2-3 msecs worst case. The delays have to be long enough for your ear to be able to hear them as distinct sounds, if not they get merged into a single event.
Hello Terry
That would be the ideal but there are very few time aligned systems out there. The ones that are using multiple drivers are only time aligned in a defined listening window at a set distance. If you are listening to a time aligned system off axis does the boogie factor change as you move on axis into the window??
Rob🙂
one of the limitations of stereo I'd say Rob! It only works in the spot.does the boogie factor change?
Initially I was going to say yes (ie the comment above), but on a tiny bit of reflection I have concluded 'no'.
The boogie factor is IN the music, we all hear the boogie factor even on the worst systems, car radios, that sort of thing.
If anything, I reckon *we* get too analytical as soon as we sit in the spot. anyone else agree?
There are things improved by sitting in the spot, things which naturally cannot be conveyed elsewhere, everyone can fill those blanks in for themselves, but strangely enough I don't think boogie factor is one of them
there's a sudden turn up for the books!
definitely agree on the very few comment. I meant what I said above, 'most' people seem to be fixated on FR, perhaps overlooking the entwined nature between Fr and time, after all what is comb filtering as an example.
Most will do a sweep (as I mentioned above), see a few dips and peaks and proceed straight to eq (if that's what they do).
Switch to looking at the arrival times between the drivers, fix that first last and always (if you are able to), THEN do any eq.
maybe time for a graph??
here's one I captured a while ago, I found it quite 'amazing'.
these are the results of that sweep idea I mentioned above, the frequency covers the range where my subs cross to my mains.
note the large difference at the crossover point (around 50 hz or so). (from memory, that difference is 10 db or so, quite a bit no??)
I did not change anything at all, except the one with the dip had the subs delayed by, well quite a bit actually, around 15 ms or so.
Yeah, you COULD throw eq at it and end up with a nice flat looking graph, but the subs timing would still be wrong none the less.
Ie you can only fix a timing issue with a timing solution.
Happily it seems we have hit the 'advance the woofer' monster on the head.
Initially I was going to say yes (ie the comment above), but on a tiny bit of reflection I have concluded 'no'.
The boogie factor is IN the music, we all hear the boogie factor even on the worst systems, car radios, that sort of thing.
If anything, I reckon *we* get too analytical as soon as we sit in the spot. anyone else agree?
There are things improved by sitting in the spot, things which naturally cannot be conveyed elsewhere, everyone can fill those blanks in for themselves, but strangely enough I don't think boogie factor is one of them
there's a sudden turn up for the books!
definitely agree on the very few comment. I meant what I said above, 'most' people seem to be fixated on FR, perhaps overlooking the entwined nature between Fr and time, after all what is comb filtering as an example.
Most will do a sweep (as I mentioned above), see a few dips and peaks and proceed straight to eq (if that's what they do).
Switch to looking at the arrival times between the drivers, fix that first last and always (if you are able to), THEN do any eq.
maybe time for a graph??
here's one I captured a while ago, I found it quite 'amazing'.
these are the results of that sweep idea I mentioned above, the frequency covers the range where my subs cross to my mains.
note the large difference at the crossover point (around 50 hz or so). (from memory, that difference is 10 db or so, quite a bit no??)
I did not change anything at all, except the one with the dip had the subs delayed by, well quite a bit actually, around 15 ms or so.
Yeah, you COULD throw eq at it and end up with a nice flat looking graph, but the subs timing would still be wrong none the less.
Ie you can only fix a timing issue with a timing solution.
Happily it seems we have hit the 'advance the woofer' monster on the head.
Could i just add a subjective observation here regardig passive pre amps and PRAT. I have an integrated exposure amp (one of those companies respected for the PRAT thing) with a passive preamp section and to my ears it boogies better than a lot of the active preamp systems i have owned in the past.
another intersting observation from my perspective is that when i add my rel subwoofers at minimal vol and crossover at 30hz, voices and instruments have more obvious perceived 'warmth' and fullness, which in termds gives the voices more charcater and emotion. I am not talking PRAT here. Back to harmonics?? Afterall this thread is not purely about PRAT (its about 'musicality' in general)it is also about getting the timbres/range of 'tonal colours' of voices and instruments right which i believe are also very important in portraying the emotional content of what a musician is trying to portray.e.g a vocalist uses pitch overtones and tonal range preferences and changes to convey feeling. However, i guess if the prat is right and the system is therefore assumed to be set up right then the latter factors will automatically also comeinto focus???
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Was a non-starter any way since, as many have pointed out already, bass instruments contain a surprising amount of treble like the click of a kick drum beater, plucking sound of bass strings etc.
But I also fail to grasp how FR could possibly not influence the timbre of an instrument.
The difference between brass and wood wind instruments for instance is purely in the timbre ie the collection of overtones and harmonics produced by each instrument. Take them away and a saxophone, flute, clarinet etc playing the same note would sound the same. Consequently a drooping hf response would tend to make a trumpet sound a little bit more like an oboe and vice versa.
As for a speakers timing issues I find that an advanced treble (most flat-fronted cone/dome designs) is more damaging/unpleasant than the reverse as is the case with most cone/horn designs. But this is purely subjective.
That said Tannoys designers did say that the all-pass they fitted to time align them in the '80s caused more problems than it fixed.
But I also fail to grasp how FR could possibly not influence the timbre of an instrument.
The difference between brass and wood wind instruments for instance is purely in the timbre ie the collection of overtones and harmonics produced by each instrument. Take them away and a saxophone, flute, clarinet etc playing the same note would sound the same. Consequently a drooping hf response would tend to make a trumpet sound a little bit more like an oboe and vice versa.
As for a speakers timing issues I find that an advanced treble (most flat-fronted cone/dome designs) is more damaging/unpleasant than the reverse as is the case with most cone/horn designs. But this is purely subjective.
That said Tannoys designers did say that the all-pass they fitted to time align them in the '80s caused more problems than it fixed.
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