Purpose of flat response below 20Hz - Why does it matter? - Page 9 - AVS Forum
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post #241 of 585 Old 07-21-2011, 05:16 AM
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Originally Posted by DS-21 View Post


Wait, 14Hz on human voice? That seems rather obviously to be an artifact of something.



You mean that you have no idea on what a close microphone setup can pickup?

Air movement from the mouth can produce very, very high levels of ULF. It seems that some people do not want that ULF filtered out based on their own theories!

Same process applies to anything that has been close microphoned. These same ULF transients may be noticed by musicians playing an instrument, but the same effect is not going to be noticed by a listener in the audience.

Same thing as hitting a baseball with a baseball bat. If you hit the ball with a bat, you can feel the impact in you hands. The pitcher does not get the same sensation. So if you took two recordings at each location, which spectrogram reflects reality?
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post #242 of 585 Old 07-21-2011, 05:23 AM
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Originally Posted by bossobass View Post


If you take everything below 20 Hz out of this scene, it becomes a complete non-event, certainly nothing like a real pair of low flying 20mm canons strafing you. Conversely, it's a whole new experience as recorded.

Bosso



When was the last time you were being strafed by 20 mm canons? Were you in the plane, on the wing (ahh,feel that ULF), or on the ground with a bullet in you head?

You never hear the one that hits you!


Once I did have an occasion to hear a B-24 flying over my house at a very low altitude (maybe 500 feet). Cool sound that you can not duplicate!
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post #243 of 585 Old 07-21-2011, 07:25 AM
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Originally Posted by mojomike View Post

Cool. Let me know how well that setup works out.

I connected everything last night. I have not powered anything up as I don't have to time to do it twice. I am waiting on the filters. However, is there a reason that there are no 40hz high pass filters? I know I have a lot to learn but this stumped me. Is this achieved via a combination of other available filters? I am asking because I realized last night that the ULS-15 crossover starts at 40. I will have to engage it and I want to minimize bass overlap(30-40hz range). My preference is 40hz and up on the Titans and about 45(upper limit) and below on the ULS-15. I want the Titans to sound as good/close to the Empire with music. Why did I ever visit the site! . I haven't even touched REW as yet. Thanks in advance for any suggestions.

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post #244 of 585 Old 07-21-2011, 08:20 AM
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Originally Posted by J_Palmer_Cass View Post

When was the last time you were being strafed by 20 mm canons? Were you in the plane, on the wing (ahh,feel that ULF), or on the ground with a bullet in you head?

You never hear the one that hits you!


Once I did have an occasion to hear a B-24 flying over my house at a very low altitude (maybe 500 feet). Cool sound that you can not duplicate!

On the same token I hear a live orchestra at my Mother in laws house every June and there is no way we can duplicate that either, but the goal is to come as close as possible.
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post #245 of 585 Old 07-21-2011, 08:39 AM
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Originally Posted by J_Palmer_Cass View Post

You mean that you have no idea on what a close microphone setup can pickup?***Air movement from the mouth can produce very, very high levels of ULF.

Good point. I hadn't considered that the mic would be so close to the mouth, though now that you mention it, it is obvious.

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Originally Posted by J_Palmer_Cass View Post

Same process applies to anything that has been close microphoned. These same ULF transients may be noticed by musicians playing an instrument, but the same effect is not going to be noticed by a listener in the audience.

That is exactly my point. Nobody on this thread has come close to showing that any stray ULF captured is perceptually relevant to music reproduction.

I am entirely open to changing my mind on subject if anyone provides relevant data to suggest whatever ULF may be captured in the recording process is perceptually relevant to music and not just a recording artifact. Thus far, that data has not been presented.

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Originally Posted by morrischestnut60 View Post

I connected everything last night. I have not powered anything up as I don't have to time to do it twice. I am waiting on the filters. However, is there a reason that there are no 40hz high pass filters? I know I have a lot to learn but this stumped me. Is this achieved via a combination of other available filters? I am asking because I realized last night that the ULS-15 crossover starts at 40. I will have to engage it and I want to minimize bass overlap(30-40hz range).

Why do you want to minimize "bass overlap?"

It is precisely that overlap that leads to smoother upper-bass response! It's what takes a system from low-fidelity mush in the upper bass:


(single sub in a corner)

to high fidelity:

(properly configured three subwoofer system, using the above subwoofer from 20-120Hz, and the other two subs from their ~40Hz low-end cutoff until 120Hz as well.)

Of course, one does need to take measurements to properly set up a subwoofer.

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post #246 of 585 Old 07-21-2011, 08:53 AM
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Any idea what that response looks like with REW (without the significant smoothing), or at higher levels? Curious to see how it holds up. Though IIRC you're temporarly renting an apartment so likely doesn't matter right now (much higher levels anyhow).

 

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post #247 of 585 Old 07-21-2011, 08:57 AM
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Originally Posted by DS-21 View Post

At what level?



Wait, 14Hz on human voice? That seems rather obviously to be an artifact of something.

And more generally, what do your mV values correspond to in terms of level? How far below (if any) are they from the average levels on the tracks? I ask because what I want to know is whether there's anything perceptually relevant.
(I don't care about content at 30kHz, either.)



Most of the time, what people think are "very low frequencies" are not ULF. Do you have numbers and relative levels?

For instance, I once read a study that measured the bass from Tube line under Royal Albert Hall - anyone who's ever heard a concert there will know what I'm talking about - at something around 25 Hz.



No musical program material will be ignored, except for the above-mentioned "sonic spectacular" disks that people only buy (myself included; I own both the Kunzel/CinnPops 1812 and, sadly, the Wellington's Victory) to demo the effects once or twice and maybe beyond that to show off the system to guy friends - never met a girl who gave a fig about bass from ordinance... - not for actual listening.

The point of my inquiry is to determine whether designing a system with the pointed goal of ULF reproduction (below 16Hz organ pedals) is useful for the reproduction of music. Not door slams, not 20mm cannon strafes, not dinosaur stomps, etc.

Alrighty then,. Out of curiousity:

Claudia Bonarelli - Boredom is counterrevolutionary


Stilluppsteypa - Confused bear thrown into the sea


David Kristian - Mumbling heights


Alva Noto & Ryuchi Sakamoto - Uoon II


Thomas Köner - Daikan


Kanye west - robocop


Lustmord - The Place Where the Black Stars Hang
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post #248 of 585 Old 07-21-2011, 09:00 AM
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And just for the heck of it
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post #249 of 585 Old 07-21-2011, 09:13 AM
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Originally Posted by DS-21 View Post

Good point. I hadn't considered that the mic would be so close to the mouth, though now that you mention it, it is obvious.



That is exactly my point. Nobody on this thread has come close to showing that any stray ULF captured is perceptually relevant to music reproduction.

I am entirely open to changing my mind on subject if anyone provides relevant data to suggest whatever ULF may be captured in the recording process is perceptually relevant to music and not just a recording artifact. Thus far, that data has not been presented.



Why do you want to minimize "bass overlap?"

It is precisely that overlap that leads to smoother upper-bass response! It's what takes a system from low-fidelity mush in the upper bass:

[(single sub in a corner)

to high fidelity:
(properly configured three subwoofer system, using the above subwoofer from 20-120Hz, and the other two subs from their ~40Hz low-end cutoff until 120Hz as well.)

Of course, one does need to take measurements to properly set up a subwoofer.

I dislike overlap for the same reason that I dislike double-bass on receivers. Also, I do not like my dedicated LFE channel above 80. I hate the "echoey" sounds that come from the sub. I also hear dialogue bleeding into the sub during cable viewing on occasions(with LFE above 80). I wish Kenwood made a line of Excelon HT receivers. Life would be so much simplier.

If multiples are supposed to reduce stress on your subs, why the boost at 20hz in the second pic? I understand that the subs crossed at 40 flattened out the midbass but the boost at the lower frequency me. Do you have the first sub doing the entire spectrum(when summed with the other two) which taxes it more?

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post #250 of 585 Old 07-21-2011, 09:23 AM
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Originally Posted by morrischestnut60 View Post

I dislike overlap for the same reason that I dislike double-bass on receivers. Also, I do not like my dedicated LFE channel above 80. I hate the "echoey" sounds that come from the sub. I also hear dialogue bleeding into the sub during cable viewing on occasions(with LFE above 80). I wish Kenwood made a line of Excelon HT receivers. Life would be so much simplier.

If multiples are supposed to reduce stress on your subs, why the boost at 20hz in the second pic? I understand that the subs crossed at 40 flattened out the midbass but the boost at the lower frequency me. Do you have the first sub doing the entire spectrum(when summed with the other two) which taxes it more?

It's not boosted (long story short before you get "yelled at" as I did ). He's simply using the SMS software to measure the response and doesn't actually have the SMS in the chain.

 

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post #251 of 585 Old 07-21-2011, 09:40 AM
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Originally Posted by morrischestnut60 View Post

I dislike overlap for the same reason that I dislike double-bass on receivers. Also, I do not like my dedicated LFE channel above 80. I hate the "echoey" sounds that come from the sub. I also hear dialogue bleeding into the sub during cable viewing on occasions(with LFE above 80).

That stuff simply doesn't happen with competent setup.

With incompetent setup, yes, any audio system is a parade of horribles. Some of those "horribles" are less noticeable than others.

Which is why once the gear reaches a rather minimal level of competence, setup dominates.

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post #252 of 585 Old 07-21-2011, 09:41 AM
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Originally Posted by pbc View Post

It's not boosted (long story short before you get "yelled at" as I did ). He's simply using the SMS software to measure the response and doesn't actually have the SMS in the chain.

Thanks for explaining that! I hope DS divulge how he achieved that even response.

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post #253 of 585 Old 07-21-2011, 09:45 AM
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Originally Posted by DS-21 View Post

That stuff simply doesn't happen with competent setup.

With incompetent setup, yes, any audio system is a parade of horribles.

Which is why once the gear reaches a rather minimal level of competence, setup dominates.

You have one sub running 20-120 and two running 40-120. What are you using to eq them since you only use the SMS to measure the system response? I assume you level match via your receiver. Dumb it down for me. You only have to do it once.

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post #254 of 585 Old 07-21-2011, 09:55 AM
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Originally Posted by morrischestnut60 View Post

Thanks for explaining that! I hope DS divulge how he achieved that even response.

Click on his name then look for threads he started, probably called "Geddes" or Multiple sub setup, or something like that. Or search his name for threads he's posted in as I think he's posted it a plethora of times now in various threads...

 

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post #255 of 585 Old 07-21-2011, 09:59 AM
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Originally Posted by pbc View Post

Click on his name then look for threads he started, probably called "Geddes" or Multiple sub setup, or something like that. Or search his name for threads he's posted in as I think he's posted it a plethora of times now in various threads...

Cool. Thanks.

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post #256 of 585 Old 07-21-2011, 10:31 AM
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Originally Posted by morrischestnut60 View Post

You have one sub running 20-120 and two running 40-120. What are you using to eq them since you only use the SMS to measure the system response? I assume you level match via your receiver. Dumb it down for me. You only have to do it once.

The credit goes to Dr. Earl Geddes, who came up with the setup method I use. It's an iterative process. Except for the part about moving the mic in an arc (taking multiple measurements and letting hardware or software average them is better, see here for more on the method: http://mehlau.net/audio/multisub_geddes/

I do my processing (level/delay-"phase"/EQ) with a cheap box called the miniDSP. There are three bands of parametric EQ in that setup. Most of the magic, and most of the work, is in getting the levels and delays right.

To save you the trouble of searching for more setup details, pretty much everything is here: http://www.avsforum.com/avs-vb/showthread.php?t=1340062

The two subs that are only running to ~40Hz are not highpassed, that's just what their passive radiators are tuned to. In the revised version (forthcoming) all of the subwoofers will be closed boxes, and all will contribute to ULF. Thus far, unloading hasn't been an issue.

For the record, I've used the same methods to achieve similar results in other systems as well. Ones with greater scale. This one is just the most recent.

It works even better (subjectively, not necessarily measured) when the mains are running full range as well! Unfortunately, the scale of these mains does not allow that. Here's the response of the first system I did using Dr. Geddes' setup methods, pre-miniDSP, in and old home, when the SMS-1 was the only processor I had. It used ZERO EQ, just three mains fullrange and three subs with relative levels and "phase" (really delay) controls used to optimize the response.

(The main sub was tuned to the high-20s, hence the rolloff.) I think the subs came in at either 120 or 140 Hz, but don't remember.

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post #257 of 585 Old 07-21-2011, 11:03 AM
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Originally Posted by DS-21 View Post

At what level?

Wait, 14Hz on human voice? That seems rather obviously to be an artifact of something.

And more generally, what do your mV values correspond to in terms of level? How far below (if any) are they from the average levels on the tracks? I ask because what I want to know is whether there's anything perceptually relevant.

The point of my inquiry is to determine whether designing a system with the pointed goal of ULF reproduction (below 16Hz organ pedals) is useful for the reproduction of music. Not door slams, not 20mm cannon strafes, not dinosaur stomps, etc.

When he is talking it could be an artifact. That is why I listed the recording when no instrument is played. Every time the bass drums are hit (in both songs the ULF level is increased). The ULF content of the first track is at least the same or greater than the less than 100 Hz (4.5 mV) while at ~ 200 Hz it is about 12.5 mV.

Here is a recent recording so it is recorded hotter but still not as dynamically compressed as most modern music. 550 Hz 71 mV. 220 - 285 Hz 43 mV peak. 110 Hz 235 mV. 30 - 40 Hz 100 mV average. 20 Hz 235 mv. 12 Hz 125 mV. 4 Hz 433 mV.

Does bass on a recording help develop a three dimensional soundstage?

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post #258 of 585 Old 07-21-2011, 11:07 AM
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Originally Posted by DS-21 View Post

The credit goes to Dr. Earl Geddes, who came up with the setup method I use. It's an iterative process. Except for the part about moving the mic in an arc (taking multiple measurements and letting hardware or software average them is better, see here for more on the method: http://mehlau.net/audio/multisub_geddes/

I do my processing (level/delay-"phase"/EQ) with a cheap box called the miniDSP. There are three bands of parametric EQ in that setup. Most of the magic, and most of the work, is in getting the levels and delays right.

To save you the trouble of searching for more setup details, pretty much everything is here: http://www.avsforum.com/avs-vb/showthread.php?t=1340062

The two subs that are only running to ~40Hz are not highpassed, that's just what their passive radiators are tuned to. In the revised version (forthcoming) all of the subwoofers will be closed boxes, and all will contribute to ULF. Thus far, unloading hasn't been an issue.

For the record, I've used the same methods to achieve similar results in other systems as well. Ones with greater scale. This one is just the most recent.

It works even better (subjectively, not necessarily measured) when the mains are running full range as well! Unfortunately, the scale of these mains does not allow that. Here's the response of the first system I did using Dr. Geddes' setup methods, pre-miniDSP, in and old home, when the SMS-1 was the only processor I had. It used ZERO EQ, just three mains fullrange and three subs with relative levels and "phase" (really delay) controls used to optimize the response.

(The main sub was tuned to the high-20s, hence the rolloff.) I think the subs came in at either 120 or 140 Hz, but don't remember.

Glad you returned. I found the thread. You gave me the embedded link but I missed it. Awesome stuff. Quite surprised that you mentioned the mains because I started wondering about my crossovers reading the thread. My setup is

Boston VR3s main(raised from full to 60hz after audyssey)?
Boston VR920 center(raised to 80 after audyssey)?
Boston VRMX x 4(crossed at 80)
Dual ACI Titans (will fully commit and let rolloff)?
ULS-15 (will fully commit and run full)
Denon 1908(Audyssey XT)

I did override the Audyssey crossovers after calibration. It appears that I should not touch it after reading the thread. This is a 180 degree for me. I guess it's like sub 20 bass. You won't know until you try it. Can I get by without the MiniDSP and Mics($800) if I get the velodyne(since it will provide the test tones and measurement as well)?

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post #259 of 585 Old 07-21-2011, 11:29 AM
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Thanks for that post Bosso. I have seen spectral analysis for cymbals and was looking for those for other instruments. You saved me a few hours today .

I have a Mac and went through at least 10 different programs yesterday before I liked one for spectral analysis but it does not accept 24 bit songs. It took me less than half an hour to find one and compare it to the others. Took me more time to convert my files from FLAC to WAV and then run an analysis. I still need to find a way to save the graphs as pictures. There is content below, it just depends if the mixer added a filter (some are brick walls).

I have some tracks from the same recording company. I've compared an unfiltered track to one of their filtered track. Both have content to at least 6 Hz. There is a drum test in a large live room also. The content when he announces the distance from drums to microphone has a peak of 1 mV at 14 Hz. When he is actually playing the drums, there are recorded values of 4.5 mV at 7 Hz. Another track with a drummer has almost no content below 20 Hz (.2 mV peak) but when the bass drum is used in the second half there is content at 9.6 mV at 7 Hz. Those are just some examples for instrument related.

As for outside noise, isn't that what connects a person to the event? Recordings of real events contain massive amounts of content below 20 Hz. Whether or not the person listens at a volume loud enough or if the noise floor in the environment is low enough is another matter as is the recording.

A larger 3 dimensional soundstage is generally attributed to the extension of the speakers. Does someone know at approximately what Hz this no longer pertains to?

These are excellent points. Thanks for the contributions to the thread.

Yes, outside noise, or ambiance, is indeed a big part of what connects the listener to the event. Place a drum kit on an elevated platform and its spectra changes dramatically.

Whether or not the real event is used with no processing or the sound is mixed through a complex algorithm with synthed sounds and then processed, the best way to reproduce that sound is in its entirety through a system that's carefully level matched and giving a flat response at the LP.

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Spectacular post, bossobass. Thank you!

Have you come across measurements other than dB/Hz and Hz/s that you have felt should be taken into account when reproducing sound?

Do you do anything in particular to accurately reproduce along the Hz/s axis, similar to what the *flat* part of the subject advocates for dB/Hz?

Do you know if there is something equivalent to Equal Loudness Curves for Hz/s that good recordings align themselves with?


edit: replaced dB/s with Hz/s throughout; apologies for any confusion.

More good points.

Monitoring your systems ability to track transients is very important. Transients are hard to measure. Generally, the shorter the duration of a transient, the more power is concentrated. Most every recording software bundle includes a waveform view option. You can use this to examine the source vs a mic'd version of that source through the sub at the LP.

This is the main reason I advocate plenty of amplifier power. It's for tracking transient peak bursts across a wide BW. I have not seen a subwoofer made commercially available that isn't amplifier limited. That's because, unfortunately, end users would abuse the available power to increase average playback levels to a point where the system will fail (and it increases the retail price).

A word about the amplifier:

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* Amplifiers are a voltage source, in theory, they should be able to produce any current on demand, in order for the voltage not to collapse under load. Well, that's the theory. In practise, SS amplifiers come very close to simulating an ideal voltage source but only to a certain current limit, after which the voltage collapses very rapidly, often with entire parts of the waveform missing, not just being clipped. This is normally due to the protection circuits. These are of course designed to kick in at current appreciably larger than those needed to drive the rated RESISTIVE load (usually at least double). That being said, there are great differences in how amplifers deal with overload of this kind. This is important for the topic because the current LIMIT of a SS amplifier almost as a rule depends on the actual output voltage at any given time. Amplifiers using bipolar junction transistors in the output stage (i.e. 'regular' transistors) have to cater for one cumbersome peculiearity of the latter - the transistor cannot withstand high voltage and high current at the same time, even if voltage x current = amount of heat produced is below the rated limit. This is important because speakers are not a resistive load. In a resistive load, current increases proportionally with output voltage. As output voltage increases, voltage across transistors decreases. So, current through transitor increases, and voltage on transistor decreases - hence the rule that you cannot have high voltage and high current across the transistor at the same time is easy to satisfy.

But, as I said, loudspeakers are not resistive loads, they are reactive loads. In a reactive load, the current waveform can lead or lag voltage. An extremely reactive load will have it's maximum current in the precise moment when there is the maximum voltage across the amps transistors, which would make it very difficult for the amp to survive, if it were not for it's output protection. And this is where output protection and current capability come into play in the reproduction of transients.

High power output (i.e. voltage capability) does not per se guarantee high current output. This fact was heavily abused during the receiver power wars, because power was measured on a dummy load, which is normally purely resistive. For a realistic load, protection would kick in way below rated power, because a realistic load is reactive.

With subs, the current trend is leaning towards higher efficiency contraptions, like horns and large ported subs and use of higher sensitivity pro sound drivers that require less amplifier for equal dBSPL output. This is the equivalent to increasing amplifier power without having to buy a bigger amplifier, but its price is loss of bandwidth, increases in certain non-linearities and, of course, huge size.

This is one reason lots of folks currently like the higher sensitivity systems. Most all of them have not experienced much <20 Hz playback, but immediately appreciate the dynamic tracking capability increase which betrays the lack of amplification capability with the other subs they're used to experiencing.

DS-21 has experienced this and has mostly confused the phenomenon with a low inductance-induced increase in transient response from the driver, which has little to do with dynamic tracking of transients. He's come to the point where he is preaching that any sub driver with higher than .6 mh inductance is incapable of fidelity, which is not necessarily (or even generally) true. He has described the lower dynamic tracking capable systems as being "thick in the mid bass", which is actually a pretty good description, but applied to the wrong phenomenon.

Most low end amplifiers and all plate amplifiers either have no clip indicators, have limiters that squash transients or clip transients that are so fast the limiter can't catch them and/or the clip indicator can't react at all.

Since transients in dynamic recorded source are hard to measure, the simple insurance comes in designing a system that has enough power to avoid collapse/distortion of the waveform. Again, higher sensitivity systems can do this without increasing the power plant, but they sacrifice bandwidth and add magnitudes more non-linear distortions out of band.

Lower sensitive systems, which actually have much higher sensitivity in the ULF band, can give both proper dynamic tracking and wide BW.

In answer to the Q: "Do you know if there is something equivalent to Equal Loudness Curves for Hz/s that good recordings align themselves with?":

Years ago I asked a similar question of several audio producers. One guy said "My hearing is not according to the ELC, the ELC is according to how I hear". I said, "Huh"? He said; "The ELC are fashioned after my hearing ability, so they're built in. If the low end sounds weaker than the higher frequencies... I boost the low end in the mix. Most well produced recordings will automatically follow the 80-90 phon ELC".

Simple enough, and certainly an extremely sensitive measurement system, not to mention the price is right.

Bosso
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post #260 of 585 Old 07-21-2011, 02:28 PM
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Originally Posted by kwarny View Post

When he is talking it could be an artifact. That is why I listed the recording when no instrument is played. Every time the bass drums are hit (in both songs the ULF level is increased). The ULF content of the first track is at least the same or greater than the less than 100 Hz (4.5 mV) while at ~ 200 Hz it is about 12.5 mV.

One explanation could be the fact that there's a phenomenon happening when using reproduction chain components (amps) that by some reason are producing a lot of second order distortion. Due to dynamic modulation of the input signal, the second order distortion components result in more than insignificant low frequency ("mixdown") artifacts for certain recordings (usually having some sort of periodic LF "beat" to it) -so called "phantom bass" (not seldom rendering visible ULF motion to woofers).

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post #261 of 585 Old 07-21-2011, 02:46 PM
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Originally Posted by J_Palmer_Cass View Post

I never said that I hate your subs (or any other system designs). I was just stating the obvious. Those are rather large subwoofers.

I am kind of disappointed that you are afraid to push them when you suspect that there may be higher level ULF below tune. Why not high pass those speakers at the proper frequency to protect those drivers?

They are. It's really the only thing about them I don't like.

Well, I'll tell ya. It's kind of a story but I guess I can sum it up.

It was a different time in 2006 when I had originally thought up of building such things. At this time I knew I wanted a significant upgrade from my single vented 15" plate amp power subwoofer. Bass charts were around just as well but not nearly as much documented back then. I wanted something that could be "bottomless" from 10hz and up when used with amp I was going to use. I didn't really care about any significant ULF <10hz at the time even though I knew it was there. There weren't a lot of affordable options for ultra low bass besides a sweet multi-sub sealed system but from my POV at the time, I didn't think it was a viable option, financially. Anyway, I never got any sort of filter for the <10hz cuz I didn't think it was a significant issue for me with the system configuration as it was.

Now that I am moving on to quad sealed, I don't think it will ever be an issue and would prefer it never be.

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Originally Posted by J_Palmer_Cass View Post

Ah yes, that underwater type sensation. Like at the end of this clip.

http://www.youtube.com/watch?v=wru7XVriBdQ&feature=fvsr




Anyhow, how large is your room and how loud do you play that movie to get that effect? In other words, whre do you fit in with this protocall as far as room size and listening level is concerned? Is -10 to -15 below calibrated reference the level that you listen to get this effect?


http://www.audioholics.com/education...ofer-room-size

My room is quite small, about 2,000cuft (or about the same as MK's but not made of concrete. This helps me a ton with low bass and I designed my subwoofer system to take advantage of this. I usually listen around -10 to -15dB but it really depends on the movie, time of day, day of the week, who is in the house at the time, etc. I may go louder on occasions, definitely. I don't listen to stuff at reference cuz I don't have all my walls treated (I still have bad slap echo. i need acoustic treatments. this is part of a large umbrella of projects that i am working on) and I find that my mains speakers to sound kind flat and shouty (in this room) at the highest levels. They can not keep up with the subs. This will change soon.

I perceive all these <20hz effects even at low volumes levels, btw. You become conditioned to the effects and signature of super deep bass once you've lived with that capability for some time. It's quite hard to miss once you know what it feels like. Bosso knows what I mean.

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Originally Posted by Steveo1234 View Post

Alrighty then,. Out of curiousity:


Kanye west - robocop



Kayne West _ RoboCop

Seems that the FFT and the RTA look different than whatever you are doing. The RTA is a representative picture shown in the manner that we hear.



Here is the FFT of that song. Peak hold and long term average was used.






Here is the RTA of that same song. Peak hold was used.





Song sample taken from here:

http://www.amazon.com/RoboCop/dp/B00...1304674&sr=8-1
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The SMS is good for a quick and dirty setup but it's not very telling.
At 1/3 octave it's way too coarse.
REW is far more accurate and use the SMS to correct what REW is telling you.
You'd be surprised at what the SMS isn't showing. (ask me how I know )

Quote:
Originally Posted by DS-21 View Post

Good point. I hadn't considered that the mic would be so close to the mouth, though now that you mention it, it is obvious.



That is exactly my point. Nobody on this thread has come close to showing that any stray ULF captured is perceptually relevant to music reproduction.

I am entirely open to changing my mind on subject if anyone provides relevant data to suggest whatever ULF may be captured in the recording process is perceptually relevant to music and not just a recording artifact. Thus far, that data has not been presented.



Why do you want to minimize "bass overlap?"

It is precisely that overlap that leads to smoother upper-bass response! It's what takes a system from low-fidelity mush in the upper bass:


(single sub in a corner)

to high fidelity:

(properly configured three subwoofer system, using the above subwoofer from 20-120Hz, and the other two subs from their ~40Hz low-end cutoff until 120Hz as well.)

Of course, one does need to take measurements to properly set up a subwoofer.

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post #264 of 585 Old 07-22-2011, 11:09 AM
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Originally Posted by kwarny View Post

Does bass on a recording help develop a three dimensional soundstage?

In general, yes. I think we've all observed the "turn the subs off, feel the soundscape collapse" phenomenon. Though the degree of improvement also depends on how well the bass is set up in the room.

However, that doesn't actually answer the question I've presented, which is quite narrow: the perceptual effect of extreme ULF, not bass generally. It does not necessarily follow that a system with a low-end cutoff of 16Hz will sound less spacious than one with a cutoff at 3Hz.

Quote:
Originally Posted by bossobass View Post

DS-21 has experienced this and has mostly confused the phenomenon with a low inductance-induced increase in transient response from the driver, which has little to do with dynamic tracking of transients. He's come to the point where he is preaching that any sub driver with higher than .6 mh inductance is incapable of fidelity, which is not necessarily (or even generally) true.

First, that is a really gross exaggeration of my position. The rule of thumb that seems to track my observed reality is 1mH Le per Ω of Re. (Assuming a driver that uses shorting rings well to control inductance linearity over stroke; obviously, drivers lacking shorting rings entirely are low fidelity units.) Above that, and things start getting bad. Below that, inductance probably isn't a factor.

But if your "dynamic tracking" theory is right - I'm very happy to accept that it may be - how come using the same amp a high-inductance sub will sound thick and a low-inductance sub will sound like music? Even if the low-Le sub is in fact rather inefficient broadband. The Aurasound NS18, for example, has a 1W/1m rated sensitivity of just 86dB.

[NOTE TO MODERATORS: would it be possible to break off the post from which I quoted below and my reply into a new thread? It seems way off the stated topic.]

Quote:
Originally Posted by morrischestnut60 View Post

Glad you returned. I found the thread. You gave me the embedded link but I missed it. Awesome stuff. Quite surprised that you mentioned the mains because I started wondering about my crossovers reading the thread. My setup is

Boston VR3s main(raised from full to 60hz after audyssey)?
Boston VR920 center(raised to 80 after audyssey)?
Boston VRMX x 4(crossed at 80)
Dual ACI Titans (will fully commit and let rolloff)?
ULS-15 (will fully commit and run full)
Denon 1908(Audyssey XT)

I'll be honest, I don't know any of those speakers, and a quick search didn't tell me what the drive unit complements are. I assume they are vented, in which case the first thing I would recommend doing is plugging the ports with foam plugs, or failing that rolled up towels or something. The mostly 2d-order rolloff of a leaky closed box is much easier to integrate with subs than a 4th order rolloff with unloading below that.

Quote:
Originally Posted by morrischestnut60 View Post

I did override the Audyssey crossovers after calibration.

I don't know. That depends more than anything else on the cone area and volume displacement of your mains. I wouldn't run anything a mains with less than a 10" woofer (or dual 8's) full range in a "typical" room (scale that up if your room is very large or leaky, and perhaps down if it's smaller). You don't want the mains to overload.

Quote:
Originally Posted by morrischestnut60 View Post

Can I get by without the MiniDSP and Mics($800) if I get the velodyne(since it will provide the test tones and measurement as well)?

Well...you can get (depending on placement constraints, room construction, etc.) between maybe 80-100% of the way there with just continuously-variable level/lowpass/phase controls and good measurements. If all your subs have those, and not just phase inversion, then it might be worth trying to calibrate without EQ. If not, the miniDSP is cheap and easy to use. And there are other solutions as well that provide individualized EQ for each channel. That's just the one I believe to currently be the most cost-effective considering OSX compatibility, capabilities, price, and ease-of-use.

I would not buy a Velo SMS-1 today. I use it because it's quick to use, with the MIC-5 take spatially averaged measurements in one go - without the MIC-5, I bought mine years ago when it was the only game in town. Today, it's simply too expensive (even at the current going rate of $300-400; when it was introduced it routinely sold for double that!) and too limited.

I would instead start with the Audyssey mic and FuzzMeasure Pro., which will generate tones and take measurements. (There is also not-a-Mac software that can do similar things. I have never used any of it.) You then take 5-6 measurements and the software will average them. Later on, if you want better data, buy a FireWire sound card (I use the M-Audio FireWire Solo, but don't know what's come out in the last several years) and a calibrated mic. But honestly for most consumer uses, the Audyssey mic and a Mac should be enough. (I cannot vouch for the mic-in ports of not-a-Macs, because I don't use them.) Since I discovered how closely data from just plugging the Audyssey mic into my MacBook tracks the data with the better setup, I've basically stopped using the better measurement setup.

Quote:
Originally Posted by sputter1 View Post

The SMS is good for a quick and dirty setup but it's not very telling.
At 1/3 octave it's way too coarse.
REW is far more accurate and use the SMS to correct what REW is telling you.
You'd be surprised at what the SMS isn't showing. (ask me how I know )

You're simply wrong. Well, mostly wrong. I agree with you if one's talking about the SMS-1 without the MIC-5 spatial averaging kit, or another means of effecting spatially averaged measurements with the SMS-1.

I suspect you're chasing phantom high-Q artifacts of improper measurement procedure (single-point rather than spatial average) instead using valid measurements to focus on things that actually matter.

I've confirmed my results with the SMS-1 using spatially-averaged measurements through more sensitive equipment on several previous occasions (see supra) and the results are simply not materially different. So I feel no need to continue to do so. When used properly - with the MIC-5 to produce a spatial average, and the five mics set up around the primary listening position - the SMS-1 provides sufficient data resolution for home system setup. If I were a professional, I would likely use something else, if only because I'd want the graphs I showed my clients to look prettier than phone pics snapped off a TV screen. But for consumer use, the problem with the SMS-1 is that it's overpriced and lacks capability (only global EQ/phase/level, when it should be able to provide that for each channel in addition to the global control).

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post #265 of 585 Old 07-22-2011, 12:23 PM
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Originally Posted by DS-21 View Post

In general, yes. I think we've all observed the "turn the subs off, feel the soundscape collapse" phenomenon. Though the degree of improvement also depends on how well the bass is set up in the room.

However, that doesn't actually answer the question I've presented, which is quite narrow: the perceptual effect of extreme ULF, not bass generally. It does not necessarily follow that a system with a low-end cutoff of 16Hz will sound less spacious than one with a cutoff at 3Hz.



First, that is a really gross exaggeration of my position. The rule of thumb that seems to track my observed reality is 1mH Le per Ω of Re. (Assuming a driver that uses shorting rings well to control inductance linearity over stroke; obviously, drivers lacking shorting rings entirely are low fidelity units.) Above that, and things start getting bad. Below that, inductance probably isn't a factor.

But if your "dynamic tracking" theory is right - I'm very happy to accept that it may be - how come using the same amp a high-inductance sub will sound thick and a low-inductance sub will sound like music? Even if the low-Le sub is in fact rather inefficient broadband. The Aurasound NS18, for example, has a 1W/1m rated sensitivity of just 86dB.

[NOTE TO MODERATORS: would it be possible to break off the post from which I quoted below and my reply into a new thread? It seems way off the stated topic.]



I'll be honest, I don't know any of those speakers, and a quick search didn't tell me what the drive unit complements are. I assume they are vented, in which case the first thing I would recommend doing is plugging the ports with foam plugs, or failing that rolled up towels or something. The mostly 2d-order rolloff of a leaky closed box is much easier to integrate with subs than a 4th order rolloff with unloading below that.



I don't know. That depends more than anything else on the cone area and volume displacement of your mains. I wouldn't run anything a mains with less than a 10" woofer (or dual 8's) full range in a "typical" room (scale that up if your room is very large or leaky, and perhaps down if it's smaller). You don't want the mains to overload.



Well...you can get (depending on placement constraints, room construction, etc.) between maybe 80-100% of the way there with just continuously-variable level/lowpass/phase controls and good measurements. If all your subs have those, and not just phase inversion, then it might be worth trying to calibrate without EQ. If not, the miniDSP is cheap and easy to use. And there are other solutions as well that provide individualized EQ for each channel. That's just the one I believe to currently be the most cost-effective considering OSX compatibility, capabilities, price, and ease-of-use.

I would not buy a Velo SMS-1 today. I use it because it's quick to use, with the MIC-5 take spatially averaged measurements in one go - without the MIC-5, I bought mine years ago when it was the only game in town. Today, it's simply too expensive (even at the current going rate of $300-400; when it was introduced it routinely sold for double that!) and too limited.

I would instead start with the Audyssey mic and FuzzMeasure Pro., which will generate tones and take measurements. (There is also not-a-Mac software that can do similar things. I have never used any of it.) You then take 5-6 measurements and the software will average them. Later on, if you want better data, buy a FireWire sound card (I use the M-Audio FireWire Solo, but don't know what's come out in the last several years) and a calibrated mic. But honestly for most consumer uses, the Audyssey mic and a Mac should be enough. (I cannot vouch for the mic-in ports of not-a-Macs, because I don't use them.) Since I discovered how closely data from just plugging the Audyssey mic into my MacBook tracks the data with the better setup, I've basically stopped using the better measurement setup.



You're simply wrong. Well, mostly wrong. I agree with you if one's talking about the SMS-1 without the MIC-5 spatial averaging kit, or another means of effecting spatially averaged measurements with the SMS-1.

I suspect you're chasing phantom high-Q artifacts of improper measurement procedure (single-point rather than spatial average) instead using valid measurements to focus on things that actually matter.

I've confirmed my results with the SMS-1 using spatially-averaged measurements through more sensitive equipment on several previous occasions (see supra) and the results are simply not materially different. So I feel no need to continue to do so. When used properly - with the MIC-5 to produce a spatial average, and the five mics set up around the primary listening position - the SMS-1 provides sufficient data resolution for home system setup. If I were a professional, I would likely use something else, if only because I'd want the graphs I showed my clients to look prettier than phone pics snapped off a TV screen. But for consumer use, the problem with the SMS-1 is that it's overpriced and lacks capability (only global EQ/phase/level, when it should be able to provide that for each channel in addition to the global control).

You're correct, I was referring to a single mic setup and not the 5 mic.
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post #266 of 585 Old 07-22-2011, 12:25 PM
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Originally Posted by DS-21 View Post

It does not necessarily follow that a system with a low-end cutoff of 16Hz will sound less spacious than one with a cutoff at 3Hz.

See, this is where the problem is. You make these declarations, but what are they based upon? You've admitted that they aren't based upon personal experience, so we can eliminate that. Please show some evidence, data, studies, papers... anything that might add some credence to your proclamations.

Quote:


First, that is a really gross exaggeration of my position. The rule of thumb that seems to track my observed reality is 1mH Le per Ω of Re. (Assuming a driver that uses shorting rings well to control inductance linearity over stroke; obviously, drivers lacking shorting rings entirely are low fidelity units.) Above that, and things start getting bad. Below that, inductance probably isn't a factor.

But if your "dynamic tracking" theory is right - I'm very happy to accept that it may be - how come using the same amp a high-inductance sub will sound thick and a low-inductance sub will sound like music? Even if the low-Le sub is in fact rather inefficient broadband. The Aurasound NS18, for example, has a 1W/1m rated sensitivity of just 86dB.

I see no exaggeration. You assert; "Drivers without a shorting ring are low fidelity".

Sensitivity of the driver at 1k Hz is irrelevant to the point.

My dynamic tracking statement isn't based on theory. Regardless of the sub driver used, if the amplifier used can't provide enough concentrated power to amplify the input signal of a transient without clipping, the dynamics of the source will not be properly tracked. Period.

Most amplifier manuals suggest: "If the clip lights flash momentarily, there is no need for concern. If they flash constantly, reduce playback level immediately".

Most everyone has mistaken that comment to mean that clipping is not deleterious to fidelity if it's momentary. It really has nothing to do with fidelity and everything to do with safe operation of the amplifier.

Can you show me the data that suggests a 'low inductance sub' sounds different from a sub using a driver with no shorting ring, all else being equal?

If you take the Tumult MKI and cross it at 100 Hz (an octave above where it was designed to be crossed), then you'll see a dip in response at cross because it's down -8dB at 100 Hz before the LPF is engaged. That has nothing whatever to do with your argument. It's just a case of using a driver to extract performance from it that it wasn't designed to provide. You simply lower the cross point and ask the mains or MBM to handle the transition.

If you cross the MKI properly at 50 Hz and then compare it to the MKII (with shorting ring and flat to 500 Hz) with the MKII properly crossed at 100 Hz (properly in both cases assumes a system designed for each of those scenarios)... there will be no difference in fidelity.

The transient response of the MKII sub will be superior to the MKI, but that's related to the cross point only and has nothing to do with inductance.

OTOH, you can properly set up both scenarios and under amplify one of the subs and you will have audible loss of fidelity in that system, through either gross waveform distortion through amplifier clipping or compression by the amps protection circuits.

Bosso
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post #267 of 585 Old 07-22-2011, 01:01 PM
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Originally Posted by bossobass View Post

See, this is where the problem is. You make these declarations, but what are they based upon?

What are yours based upon?

Please show some evidence, data, studies, papers... anything that might add some credence to your proclamations.

I've yet to see a shred of anything on this thread or elsewhere that's actually relevant to the posited question. Mere existence is not the same thing as either intent or perceptual relevance.

You're not even willing to correlate that ULF you've found in that Sting track to the music. That would be anecdotal, but it would at least be something.

Quote:
Originally Posted by bossobass View Post

Sensitivity of the driver at 1k Hz is irrelevant to the point.

Then please explain how your following statement:

"This is one reason lots of folks currently like the higher sensitivity systems. Most all of them***immediately appreciate the dynamic tracking capability increase which betrays the lack of amplification capability with the other subs they're used to experiencing.***DS-21 has experienced this and has mostly confused the phenomenon with a low inductance-induced increase in transient response from the driver, which has little to do with dynamic tracking of transients."

squares with the fact that lower-Le subs sound better even if sensitivity and efficiency is similar. And the same amp is used.

Take two multisub systems in which one of the subwoofer cabinets is of a constant volume and contains either:
A) a high-inductance driver (TC3000, ur-Tumult, Fi-Q, Ascendant Avalanche, JL Audio W7, etc.), or
B) The equivalent volume displacement in one or multiple standard overhung drivers of competent design and moderate but not Aurasound NS- or JBL GTi- low inductance (Peerless XLS, Dayton Reference, TC2+, etc.).

Compared to Subwoofer B, Subwoofer A must be severely bandwidth-limited to be of any use at all. And a multisub system anchored by Subwoofer B may need more subwoofers than one using Subwoofer A: it's down one pressure-source in the modal region, because Subwoofer A just can't help there. Too much inductance to be useful.

Now, lets say there's also Subwoofer C, which is the same cabinet holding the equivalent volume displacement in ultralow-inductance drive-units (Aurasound NS, JBL GTi, etc.).

How would that compare to Subwoofer B? (We can reasonably assume that Subwoofer A is not competitive.) Honestly, I don't know. My strong hunch is that, given equivalent in-room response, there will either be no difference or such a small difference that few if any will have reason to care about it.

Quote:
Originally Posted by bossobass View Post

My dynamic tracking statement isn't based on theory. Regardless of the sub driver used, if the amplifier used can't provide enough concentrated power to amplify the input signal of a transient without clipping, the dynamics of the source will not be properly tracked. Period.

OK, but that's outside of the scope of your argument that I'm conflating low inductance with dynamic tracking, though. Best I can tell, you distorted my position to believe that I was conflating inductance with sensitivity, and then proceeded to riff on something unrelated to anything previously discussed.

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Originally Posted by bossobass View Post

Can you show me the data that suggests a 'low inductance sub' sounds different from a sub using a driver with no shorting ring, all else being equal?

I don't need to, when you provide it for me.

Which is precisely what you do in this statement:

Quote:
Originally Posted by bossobass View Post

If you take the Tumult MKI and cross it at 100 Hz (an octave above where it was designed to be crossed),

If it's only designed to be used below 50Hz, it is (depending on perspective) either an extremely limited-ultility device or just plain a poorly-designed sub.

Given that modern best practices (i.e. the methods and procedures that provide the overall smoothest response in small rooms) require subs to be used well over 50Hz (typically 120-150Hz), it logically follows one should care how a sub behaves to at least an octave above that, given the sensitivity of the ear in that region.

Or, to turn your exact argument for ULF performance on its head, the LFE channel has a specified bandwidth of 3-120Hz, correct? So how can a sub be any good, if it cannot reach reproduce the top of that bandwidth?

You'll note that I have repeatedly said that if all one wants is a ULF air pump, then inductance is a non-factor. For movies, where fidelity is unimportant, that can be fine.

Quote:
Originally Posted by bossobass View Post

If you cross the MKI properly at 50 Hz and then compare it to the MKII (with shorting ring and flat to 500 Hz) with the MKII properly crossed at 100 Hz (properly in both cases assumes a system designed for each of those scenarios)... there will be no difference in fidelity.

Not true. The system with the well-designed driver (Tumult Mk. II) will be higher fidelity than system with the poorly-designed driver (Tumult Mk. I), assuming competent setup for both. The system with the T2 will have an additional pressure source to smooth out bass response in the room, whereas the system with T1 just has an air pump. The system with T2 will be better still if it is used a bit higher, depending on the size of the room and the volume displacement of the mains.

The really odd thing about this debate, given its heat is that in the end we basically agree on the practical ideal bass subsystem in most rooms: multiple closed box subwoofers, with enclosures sized such that the amp power on hand can push the individual woofers to their full linear travel but not much beyond that, with EQ to shape the response profile to taste. And if there's not enough output/headroom, add more subs.

Funny thing is, we've come at it from two radically different perspectives (you seem to say "ULF first!" and I say "smooth out the upper bass and then everything else will fall into place adequately because of the measures taken to smooth out the upper bass") but we've reached more-or-less the same solution.

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A higher sensitivity driver equals a lower sensitivity <20 Hz. When I say people are preferring that choice currently, I don't mean to say that they're doing so from an understanding of transient response vs dynamic tracking of transients.

It's just that they get a better result in that type of drivers BW with a less capable amplifier. Don't read more into it than that, which is what you are doing.

I've said many times and posted the evidence to back it that a flat, full BW FR equals a) better accuracy (reproducing what's on the disc vs not doing so really comes under 'duh'), b) a better listening experience.

The evidence is in the comparison of the digital feed to the mic'd result and the subjective conclusion comes from A/B listening by many people over the years.

Yes, we are in agreement on many aspects of the low end of things. The difference is that I have 2-1/2 octaves more BW and lots more headroom, which equates to a more accurate system. The subjective difference is irrelevant. Debating the validity of non-filtered recorded source is irrelevant.



It's not that I haven't presented a shred of evidence. It's that you've decided to shred the evidence I've presented.

In the real world, all transients have content to DC. In the real world of recorded source, there is intended content to <3 Hz. If your playback system rolls off at 20 Hz, an audible form of distortion exists.

You need more than what I've presented in this thread to convince you of the veracity of these simple truths? Dunno what to say to that.

Bosso
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post #269 of 585 Old 07-22-2011, 02:52 PM
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Just wondering here, Bosso. Medical science knows that various organs in the body have certain resonant frequencies (low) and sundry studies have indicated correlations with frequencies and perceived sensations (paranormal comes to mind but there are others). In your opinion, do you think the reason for the presence of low level information in movies is an attempt to tap into these perceptions?

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post #270 of 585 Old 07-22-2011, 04:15 PM
 
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I read something once about "the brown note", is that what you are referring to?
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