Ooh, that's funny!
Goes along with my opinion that 90% of US males have no hearing above 7000 hz.
...however I think that Linkwitz was right when he wrote that:
"Unbiased listeners have no difficulty recognizing accurate sound reproduction, even with hearing damage or with hearing aids"
Haha, you are quite right my friend
Cause what i really wanted to do was simply call BS
We truly have different takes on SPL and dynamics, huh
You don't need to pull punches with me but to say what I said is BS needs qualification to be anything other that hot air. If you want to point to something and discuss it go ahead otherwise lets leave that alone.
I also doubt we have much different thoughts on loudness and dynamics and your comment again has a tone which suggests you think I am sitting at home in my slippers listening to elevator music sipping a club soda.
I have played the drums since I was 14 (as that was the first time I could convince my parents to allow to have a set). To play a louder and more dynamic instrument I would probably have to take up the starting pistol.
The loudest music I have ever experienced was at a Pantera concert in the early nineties back when big acts still toured smaller venues and you could literally put your nose on the grill of the PA system. After a few hours of being physically pummelled by Vinnie Paul's double kick I left feeling like my ribs had been broken and loved it.
I'm not a teenager anymore things have changed and so have I, the drumkit is now a hybrid electro acoustic and I don't want my ribs broken by my stereo.
The limit on SPL is now me and I am interested to know why which is part of the reason I am designing a two way waveguided speaker with the sort of drivers you might choose.
With the system I have now tuned at the SPL I prefer to listen to I don't have to touch a volume knob or tweak an EQ as I am listening, I can just sit back and enjoy. Like everyone I wonder if it could be better.
Sorry fluid,
when i read "I choose to listen at about 70 to 75dBc on a slow SPL meter at the listening position just under 3m from the speakers. They can go quite a lot louder but I can't"
I just went wow, that's very low volume casual listening for me.
And the idea that the TC9 line can produce 10Hz just isn't at all realistic, it barely pulls much lower than 100Hz, imo/ime.
But none of that matters... very last thing i want to do is offend such a helpful, knowledgeable, peace making contributor like you. I say that in complete sincerity and thanks.
Switchin gears, I'm almost 69, and maybe the biggest kid on the block...i still love my ribs getting kicked !!!
when i read "I choose to listen at about 70 to 75dBc on a slow SPL meter at the listening position just under 3m from the speakers. They can go quite a lot louder but I can't"
I just went wow, that's very low volume casual listening for me.
And the idea that the TC9 line can produce 10Hz just isn't at all realistic, it barely pulls much lower than 100Hz, imo/ime.
But none of that matters... very last thing i want to do is offend such a helpful, knowledgeable, peace making contributor like you. I say that in complete sincerity and thanks.
Switchin gears, I'm almost 69, and maybe the biggest kid on the block...i still love my ribs getting kicked !!!
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That's could be as much as 12 dB less than what I would say is valid and that is dB(C) fast.
In another current thread, i just found the link to a thread you started years ago
SPL targets for speaker design
I kept finding myself nodding in agreement with you throughout that thread
Anyway, I've come to trust SPL flat more than A/C weighting when trying to 'mentalize' the SPL meter. Especially when i want to meter in time, fast, slow, peak, etc.
And equate it to real world loudness as I perceive it.
Do you see a reason to defer back to weightings?
I think dB(C) is correct for perceived loudness levels - not for spectral measurements. It eliminates very low and very high frequencies which don't affect loudness very much. But yes, unless you know for certain that a specific weighting function is correct in your situation, it's best not to use any. But most SPL meters don't leave you much choice. I never ever use dB(A), it's that bad.
dB(A) is for noise measurements where you want a lower number (marketing.) It is completely inaccurate, but it gives lower numbers. Reminds me of THD.
dB(A) is for noise measurements where you want a lower number (marketing.) It is completely inaccurate, but it gives lower numbers. Reminds me of THD.
Camplo, that is an example of pure ignorance imo.
Latency, or rather pure constant time delay in its most common expression, is the same as distance to listener.
It appears he uses what he calls latency, however he is manipulating the sound, to make a splash.
A BS splash......
Do you think sound changes like the youtube vid, just because of distance?
The dude is smoking something.
Latency, or rather pure constant time delay in its most common expression, is the same as distance to listener.
It appears he uses what he calls latency, however he is manipulating the sound, to make a splash.
A BS splash......
Do you think sound changes like the youtube vid, just because of distance?
The dude is smoking something.
Thank you Camplo, i had a good laugh when i heard slapback echo!
Latency mixed with direct signal.... hummmm the guy just discovered ...echo..
Next step is to bring feedback in and hey he'll name himself 'Lee Scratch Perry' and claim the title of ' great upsetter'...
Then and only then he could use ( smoke) pot!
For now he just don't smoke enough to post something like that ...in public domain!
That said with a bit of ingeniosity he could had made a point on minimising Early Reflection but...no...definitely he doesn't smoke enough weed!
Latency mixed with direct signal.... hummmm the guy just discovered ...echo..
Next step is to bring feedback in and hey he'll name himself 'Lee Scratch Perry' and claim the title of ' great upsetter'...
Then and only then he could use ( smoke) pot!
For now he just don't smoke enough to post something like that ...in public domain!
That said with a bit of ingeniosity he could had made a point on minimising Early Reflection but...no...definitely he doesn't smoke enough weed!
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Call in the Calvary you missed the objective of that video completely! lol
The first ms layer of the audio represents 0ms....layered on top is an identical copy spaced at specified interval. Its to give an auditorial reference for the amount of space between 0ms and the specified....So as I've specified before...I can tell a 10ms delay and for video and for bass its about the highest to tolerate....
Can't believe you guys missed the point of this video......whoops
looks like a 48db linear filter (mid to tweet) is within 10ms though so its a go for my system tolerance vs video...
If you play midi keyboard...in particular drums...anything above 10ms sucks.....as a drummer...my timing tolerance is minimal...though my coordination is pretty good I can play with like 20ms latency and keep on tempo (meaning id have to play consistently 20ms ahead of the tempo)...its crazy.
When you listen to heavy bass track with headphones vs a system with the huge 20-25ms group delay within its passband (bad reflex?)....you(I rather) can tell the diff....the only thing is without the hearing of it without distortion....you'd never know... specifically I can hear it while focused on a kick drum. I can hear the difference between the spread of the attack transient and the sub bass. I've argued too many times that I could hear it...funny how easy it is to prove it with the right material...and a little bit of instruction on how to use the material lol!
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^ Sorry but i don't see your point: it is well known that the limit is 5ms for direct monitoring.
I once tracked a guitarist ( death, black, whatever metal) which was bothered at 3,5ms but it was something like 190bpm 16th note with 'composed measure' ( something like 11/8 3/4 chained pattern).
Of course as long as the delay is constant you can play 'over it'. I'm sure you can try 50ms and still be able to play.
For reproduction you answered yourself in the last section: without reference...
( it always make me laugh when friends performs live: 'i've made a big mistake in the 3rd track' track which isn't released and he is the only one to have ever heard before.... 'well yes i agree, i heard it and leave the place!' is my answer. Always. )
I once tracked a guitarist ( death, black, whatever metal) which was bothered at 3,5ms but it was something like 190bpm 16th note with 'composed measure' ( something like 11/8 3/4 chained pattern).
Of course as long as the delay is constant you can play 'over it'. I'm sure you can try 50ms and still be able to play.
For reproduction you answered yourself in the last section: without reference...
( it always make me laugh when friends performs live: 'i've made a big mistake in the 3rd track' track which isn't released and he is the only one to have ever heard before.... 'well yes i agree, i heard it and leave the place!' is my answer. Always.
)
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Well I think at the 5ms section of that video its obvious why 5ms is a nice limit for "direct monitoring"...
I think I can handle 10ms....5 would be nice...with 48db at my general XO area latency says 7.33ms
I'm also speculating how "behind" 5ms is because without smoothing the group delay bump of my horn is about 4.5ms.. little peak at 340hz lives in between 300-400...
I think I can handle 10ms....5 would be nice...with 48db at my general XO area latency says 7.33ms
I'm also speculating how "behind" 5ms is because without smoothing the group delay bump of my horn is about 4.5ms.. little peak at 340hz lives in between 300-400...
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Really Camplo this is a non issue: you'll use dsp so for tracking you'll just need to go with IIR and load the dedicated setup... then you control using FIR afterwards. Going from one to another should not brings issues especially if you are composing/playing ( your attention is focused on differents tasks and skills. At least this is true to me. ).
).
I think you are saying design around minimum phase and use fir correction downstream?
In particular I think I’m getting better crossing with the steeper slopes (needs more investigation) so I appreciated keeping group delay down around xo...
While I have your attention do you know a good piece of hardware to run the filters I’d generate with acourate/rephase?
In particular I think I’m getting better crossing with the steeper slopes (needs more investigation) so I appreciated keeping group delay down around xo...
While I have your attention do you know a good piece of hardware to run the filters I’d generate with acourate/rephase?
^ you still have my attention! I'm not acting when i have nothing to say but that don't stop me reading. And you know where to find me if you want.
What i'm saying is compensating for the group delay of your horn through the use of FIR. Target flat frequency response for half an octave around xover point then use FIR profile to only affect phase and compensate for gd.
Overall what the first plugins availlable for 'linearisation' did ( 18y ago iirc) with regard to bass reflex gd. Tryed it, didn't like it. Unatural to me ( it should be obvious than for a bass reflex there is no 'counteracting' from another complementary filter so i suppose this is what i've heard as 'unatural').
Anyway i thoughts gd have been approached a year ago in there and that conscensus was that it is treshold/freq related. So maybe you'll gain from it, maybe not.
For the hardware i think you'll gain more from Mark and Fluid: they are up to date about gear (i don't care as much once i have something i'm pleased with. And to be honest i'm into console and analog gear atm more than loudspeakers as i've scored 8 vintage API output transformers recently and play with them to see where they'll end up in my chain ) and non locked on filter duties only as i am ( i still need to be definitely convinced / hear about room correction that i like).
Anyway i haven't changed my mind since last time we talked about it: pc/soundcard is the most powerful and economically rational choice in my view.
What i'm saying is compensating for the group delay of your horn through the use of FIR. Target flat frequency response for half an octave around xover point then use FIR profile to only affect phase and compensate for gd.
Overall what the first plugins availlable for 'linearisation' did ( 18y ago iirc) with regard to bass reflex gd. Tryed it, didn't like it. Unatural to me ( it should be obvious than for a bass reflex there is no 'counteracting' from another complementary filter so i suppose this is what i've heard as 'unatural').
Anyway i thoughts gd have been approached a year ago in there and that conscensus was that it is treshold/freq related. So maybe you'll gain from it, maybe not.
For the hardware i think you'll gain more from Mark and Fluid: they are up to date about gear (i don't care as much once i have something i'm pleased with. And to be honest i'm into console and analog gear atm more than loudspeakers as i've scored 8 vintage API output transformers recently and play with them to see where they'll end up in my chain
) and non locked on filter duties only as i am ( i still need to be definitely convinced / hear about room correction that i like). Anyway i haven't changed my mind since last time we talked about it: pc/soundcard is the most powerful and economically rational choice in my view.