You got it right.The test is this: play a known sample at preferred max volume and measure = the max voltage your speakers will need - then post your results
Please please please tell me thats right
It really is that simple. I don't know where the confusion comes from, sure wish I did.
How can we know? We don't have enough info.
Why not? It's simple. If you set your present amplifier to the level you find loud, but not awful - then measure it, we will have an idea of your peak voltage. From that we can give you an idea of what you might build.
If you do decide to build, then your amp should be able to deliver a steady sine wave 9dB more than the test tone, minimum. No clipping, no significant rise in distortion at that power. That done, it should not clip or distort at your highest playing levels. It may not sound the same as another amp, but it should play just as loud without getting into trouble.
Mr vd Gevel's numbers should indicate that Pan-Oh-Man's riddle is not a half bad practical approach.
25.46V would normally correspond with 105dB (~19dB + 86dB/W/m), which is something the ESL63 is not capable of.
(so much for the oldy 303 and it's ~22V max)
LOL... ESL 63 will do 105 db ( peaks) ......
Tom, with all due respect, you simply do not understand this test. I respect your experience and designs, but you are not understanding the test - and I don't know why.
Please read post #277, all you need is contained therein.
I think you and Mr. Danley are talking past each other a bit.
Here's what I think your test actually measures: how much power does it take to get your speakers in your room either to the edge of their linear performance to a subjectively-pleasing level of overdrive, on program material you actually listen to.
One thing I've noticed over time is that speakers with linear frequency response and enough efficiency/cone area make it very hard to gauge how loud they're playing without instrumentation. Often, the only way to tell how loud it is without measurements to try to talk to someone. With smaller speakers, one "hears" that it's too loud because of distortion creeping up.
That's, I think, Tom's real objection. But I think also think the data you're collecting are novel and useful, and the question it could address is interesting.
I think it would be useful to add some metric for radiating area and efficiency to your data. Perhaps people who have already answered could provide that information. Also, it would be interesting to see someone with a "small rig" and a "big rig" do it on both, and compare the results.
Last edited:
Hi 5th
I guess the real issue is the title and the implied connection to what the speakers need.
For me, from a loudspeaker design /usage / measurement stand point, it is rather silly to assert anything more than an arm waving argument about “how much they need” without having the faintest idea of the distance, sensitivity, linear capacity listening level and so on but hey, it’s hifi where science and measurement is often a dirty or marketing words..
Please explain the connection between the various signals one might listen to and the level one sets the stereo too?
My purpose in writing was to alert those interested just how complex the issue really is from a technical point of view should accurate reproduction of the input signal be desired. For me, it was an eye opener examining the sounds I wanted to capture and reproduce and to see how often that is lost at the reproduction stage.
Unrealistic or not reproducing the original signal has been a concern for a good while, that’s why our commercial speakers are precise enough in time to reproduce a square wave over a broad band and generally have no evidence of crossover phase shift too. They are as close as I could get to the ideal single source in time and space Richard Heyser described in his paper on loudspeaker arrival times.
Hi pushpull.
A software based scope is fine too, the only issue is to make sure you can isolate / attenuate the input and always watch to see where ground is. What you would want to see is the time record, not the harmonic content.
The time record will show the momentarily clipped peaks if they exist.
Keep in mind too that the bandwidth is very limited compared to a hardware scope, these at a minimum work to hundreds of KHz where the sound card hits a brick wall at a little over 20KHz. The features of a high frequency signal in clip are well over 20K.
Mooly’s link to an amplifier / circuitry designer may be of some insight / help too. Being concerned about the actual signal and not doing it by ear, here, he refers to measuring one signal with only a 14 dB crest factor (peak to average). Post #27 here;
http://www.diyaudio.com/forums/soli...ly-need-domestic-listening-3.html#post2128057
Maybe it would be better for those still curious to poke around in the circuitry area for insight about signal dynamic range etc. Loudspeakers being by far the most non-linear and the weakest link to faithful reproduction may not be the best area to figure out what they should do.
Best,
Tom
I guess the real issue is the title and the implied connection to what the speakers need.
For me, from a loudspeaker design /usage / measurement stand point, it is rather silly to assert anything more than an arm waving argument about “how much they need” without having the faintest idea of the distance, sensitivity, linear capacity listening level and so on but hey, it’s hifi where science and measurement is often a dirty or marketing words..
Please explain the connection between the various signals one might listen to and the level one sets the stereo too?
My purpose in writing was to alert those interested just how complex the issue really is from a technical point of view should accurate reproduction of the input signal be desired. For me, it was an eye opener examining the sounds I wanted to capture and reproduce and to see how often that is lost at the reproduction stage.
Unrealistic or not reproducing the original signal has been a concern for a good while, that’s why our commercial speakers are precise enough in time to reproduce a square wave over a broad band and generally have no evidence of crossover phase shift too. They are as close as I could get to the ideal single source in time and space Richard Heyser described in his paper on loudspeaker arrival times.
Hi pushpull.
A software based scope is fine too, the only issue is to make sure you can isolate / attenuate the input and always watch to see where ground is. What you would want to see is the time record, not the harmonic content.
The time record will show the momentarily clipped peaks if they exist.
Keep in mind too that the bandwidth is very limited compared to a hardware scope, these at a minimum work to hundreds of KHz where the sound card hits a brick wall at a little over 20KHz. The features of a high frequency signal in clip are well over 20K.
Mooly’s link to an amplifier / circuitry designer may be of some insight / help too. Being concerned about the actual signal and not doing it by ear, here, he refers to measuring one signal with only a 14 dB crest factor (peak to average). Post #27 here;
http://www.diyaudio.com/forums/soli...ly-need-domestic-listening-3.html#post2128057
Maybe it would be better for those still curious to poke around in the circuitry area for insight about signal dynamic range etc. Loudspeakers being by far the most non-linear and the weakest link to faithful reproduction may not be the best area to figure out what they should do.
Best,
Tom
Hi Tom
(And the rest of you as not piking on any one in particular I was folowing this tread at the begining and now I am so bored with it that I only get to it once in a while so most probably I missed quite a bit of the good and bad stuff)
I can see the point you are making and agree there are beter way to mesure response
But I have found the test as proposed by Pano quite simple and easy enough to be performed by any one with very litle Instrumentation.
From it I got a rough idea of when my speakers start clipping (in my case possible more often than I would like 2 )
I think that this was what Pano was proposing and IMO hat off to him for such a simple and elegant test.
One thing I do not understand from your post.
You mention speakers able to reproduce a square wave.
The way I see it You place a voltagge either positive or negative to the speaker therminals whit and unspecified duration.
A bit like conecting a batery to cek phase on the woofer
Woofer go out (or in) prety quik stay out for the duration of the pulse and then, if not blown and the coil has melted go bak to the original position a bit like when an amplifier clip and the top of the sine flatens out but with much more sleve rate.(fast raise pulse edges)
Realy?
How does one record a square vave on the gove of a LP I supose you get Tump at rise and then silence as the nidle stays out.
Digital maybe possible I don't realy know or am interested(but then you get loas of litle square waves of not changing voltagge at 44or such KhZ sampling frequency)
In Nature could you please give me an example of anithink that make a noise like square wave that as I understand you tried to capture?
Or would it be and aproximation made by a very large number of Sine waves and armonics?
Nature is very good at making aproximations
Have you tried to ask a recording late operator (The ones that make the sound go from tape to round metal plate) to record same square waves and survived?
Have you found one that did try and did not brake the late?
Could we please carry on this discussion on another tread as....
I know now I am one of the Guilty ones making this tread a 100 pages long and mudding the water with issues far above the scope for which I understand it was started.
Most probalbly and quite easily I should have satarted my own tread ranting and raving about sometink or at least my perception of it....
Pano I am realy realy sorry please don't let me put you off and If you get another Idea 0.00001 % as good as the one you started with please post I will be listening.
(And the rest of you as not piking on any one in particular I was folowing this tread at the begining and now I am so bored with it that I only get to it once in a while so most probably I missed quite a bit of the good and bad stuff)
I can see the point you are making and agree there are beter way to mesure response
But I have found the test as proposed by Pano quite simple and easy enough to be performed by any one with very litle Instrumentation.
From it I got a rough idea of when my speakers start clipping (in my case possible more often than I would like 2 )
I think that this was what Pano was proposing and IMO hat off to him for such a simple and elegant test.
One thing I do not understand from your post.
You mention speakers able to reproduce a square wave.
The way I see it You place a voltagge either positive or negative to the speaker therminals whit and unspecified duration.
A bit like conecting a batery to cek phase on the woofer
Woofer go out (or in) prety quik stay out for the duration of the pulse and then, if not blown and the coil has melted go bak to the original position a bit like when an amplifier clip and the top of the sine flatens out but with much more sleve rate.(fast raise pulse edges)
Realy?
How does one record a square vave on the gove of a LP I supose you get Tump at rise and then silence as the nidle stays out.
Digital maybe possible I don't realy know or am interested(but then you get loas of litle square waves of not changing voltagge at 44or such KhZ sampling frequency)
In Nature could you please give me an example of anithink that make a noise like square wave that as I understand you tried to capture?
Or would it be and aproximation made by a very large number of Sine waves and armonics?
Nature is very good at making aproximations
Have you tried to ask a recording late operator (The ones that make the sound go from tape to round metal plate) to record same square waves and survived?
Have you found one that did try and did not brake the late?
Could we please carry on this discussion on another tread as....
I know now I am one of the Guilty ones making this tread a 100 pages long and mudding the water with issues far above the scope for which I understand it was started.
Most probalbly and quite easily I should have satarted my own tread ranting and raving about sometink or at least my perception of it....
Pano I am realy realy sorry please don't let me put you off and If you get another Idea 0.00001 % as good as the one you started with please post I will be listening.
LOL! No problem, no problem at all.
Come on Tom! Where have measurements or science been treated as dirty words in this thread? Please show us. This thread is all about science and measurements. You've fallen on the classic Straw man argument.
Not the faintest idea of distance, sensitivity, listening level and so on? I don't understand how you can make a statement like that, if you have read the test and the replies.
Those taking the test know the distance to their speakers, most seem to know the sensitivity and they certainly know the listening level as they set it with their own ears.
I do understand your argument from the point of view of a speaker designer, but that is NOT the subject of this test. The object of the test is to find out what voltage levels people are actually using when playing their own systems in their own homes. There is no SPL goal, no mention of how things should be, only how they really are. This is not a speaker design exercise, it's a real world measurement of how much voltage people are actually using. It's a measurement of practical reality, not of design goals.
This may not be a test that is useful to you.
Again, this is NOT a test for designing speakers or anything else. There are no standards or goals. It simply measures what you use in a real, day to day, situation. It's a measurement of what is, not of what is supposed to be.
That is what my proposed test will find out.
You tell me what power of amp I should use.
If it's one that results in the final test tone measurement of 1Vac then I will build a 1W amplifier and do the test for you.
If the majority think the result will be 2Vac, then I can build the 4W amplifier.
You tell me and I will build it. Then I will carry out the test and relay the results as described earlier for everyone to see and consider.
That may lead to a conclusion or series of conclusions.
I have given you a start point, speaker sensitivity and my typical average signal levels at the speakers.
I think if you're asking this then you don't get what's going on here, which is fine, unless I have misinterpreted your question, but I shall do my best to explain it once more.
I am assuming that these various signals that you are referring to are the sine waves that we've been mentioning. The important thing to note here is that we are not listening to them. When the sine waves are indeed used, we could be doing this from a remote location in outer space. How loud the system sounds subjectively when a sine wave is being played through it does not matter because we're not interested in this. In fact you could disconnect the loudspeakers for this part of the test if you so desired.
The relationship between the sine waves and the given volume setting is simple.
The sine waves are reproduced at a known signal level, that is -12dB as is referenced to the digital maximum of 0dB.
If we now play this signal through the system at the given volume setting it will produce a sine wave on the output of the amplifier with a given rms voltage as measured on your multimeter.
Lets say this voltage is 5Vrms.
So at your loudest ever volume setting, when a -12dB sine wave is played through your system, we get 5Vrms on the output of the amplifier.
Now 0dB digitally, represents the maximum possible signal level that the system could ever be faced with. This means that in the source music or source material, the largest signal level encoded within it will be 0dB, it cannot be any more. Hence if the amplifier is clip free when asked to reproduce a 0dB signal at the given volume setting, then it will always be clip free because the signal cannot get any bigger.
We may have used a -12dB sine wave above, but what we are really interested in is the rms voltage on the output of the amplifier when it is asked to reproduce a 0dB sine wave at the given volume.
Pano chose -12dB so as to protect the loudspeakers, but this doesn't matter. We know that at -12dB the amplifier will produce 5Vrms, so it is a simple matter of mathematics to scale this 5Vrms up to what it would be given a 0dB signal.
To get a gain of 12dB on a voltage signal you multiply the voltage by 4. So in this case our 5Vrms ends up being 20Vrms. This is what we want to know.
Having done that, we know that at our loudest ever used volume setting, that if we were to play a sine wave of 0dB amplitude through the system that it would produce a signal of 20Vrms on the output of the amplifier.
The relationship between the sine wave and the given volume setting is in figuring out the maximum output voltage that the amplifier would ever have to reproduce given the maximum signal of 0dB of the digital format.
As you should be able to see 0dB = the maximum that any transient encoded within the digital medium can reach. If your amplifier is clip free at your maximum ever volume setting, when playing a 0dB signal, then it will indeed be clip free regardless of the digital signal you choose to throw at it. This is all we are doing here.
In fact Pano's use of a -12dB signal serves two purposes here. The first is that it will protect your loudspeakers from being potentially over driven. But if Pano had used a 0dB test signal instead and your amplifier was too small then it would cause it to clip. This in itself is bad for the loudspeakers, but it would also screw up the accuracy of the reading on the digital multimeter as the peaks of the signal would be flattened.
This in itself is completely redundant, if you know the size of the signals that your loudspeakers see then you know precisely the voltage swing the amplifier need be capable of to reproduce even the loudest sound that it could ever be faced with.
I mean as an example using the Ricky Lee Jones track as mentioned before. The largest signal came in when the snare drum was whacked. Now lets say the snare drum reached -0.5dB on the digital scale. If we set the volume control to the loudest position that you would ever listen to the system at, we could now send a -0.5dB signal through the system and see if the amplifier has the voltage headroom to reproduce a signal of this level without clipping. If it doesn't clip then all is well. Of course another track could come along with a 0dB signal in it and as this is larger then the -0.5dB signal in the RLJ track, it has the potential to clip the system. Of course if we had tested the system using a 0dB track before and the amplifier didn't clip, then we'd know the system wont clip, at our given maximum volume setting, for any digital media because it's impossible to exceed 0dB.
We all know this test doesn't include how difficult the loudspeakers are to drive, but unless you're pushing the amplifiers voltage limits, or have a pair of loudspeakers that are mismatched to the amplifier, then this is unlikely to be an issue.
Sorry, maybe I didn't understand your question. Can you explain it to me better?
Hi Pano
My issue with a semi scientific approach to this is that it is based on a subjective judgment where the actual issue can be measured or estimated based on the input signal and speaker behavior.
I don’t see a compelling scientific or engineering argument for or connection between the technical issue at hand and a subjective judgment and rule of thumb, especially when one can easily examine the actual signals and Voltages instant by instant and know what is required based on fact and a theoretically idea system..
In the subjective a change of 10 or 20 dB isn’t that big when the span of hearing is more like 120dB and room noise floor typically in the 20 to 40dB range.
Also, there is a place for math but for it to be useful it must be in the proper context too.
For example, the simplest look says a 200W amplifier and 80dB efficient speaker can produce the same peak sound level as a 100dB sensitive speaker and 2W amplifier.
In reality, a commercial sound speaker made for a higher power level and sensitivity will typically be much more dynamic sounding than the small speaker driven at higher power that theoretically equaled the same level.
Loudspeakers are for sure the weakest link at least if one compares the signal going in to what comes out, so why shouldn’t this be examined, being hard or inherent in the approach is not an excuse I don’t think..
If one is looking at what comes out vs what goes in, the reason for old time efficient speakers like Altecs, Klipsch horns an such starts to become more understandable, while they have their own various sonic warts, they can preserve much more of the dynamics at living room levels because they can go so much louder than required.
While many are stuck on “this means it’s loud” and these systems can play more loudly too, it’s only the peaks that are any louder at home levels than more dynamically hobbled systems.
If you have ever sat down and listened to a set of efficient, powerful speakers like these, you already know what I am talking about, add enough power and gain structure adjusted so that one never clips anywhere and you only have the loudspeakers and program content limiting the dynamics.
My approach personally has evolved to use much more power and very efficient but modern horn speakers and subwoofers with sensitivity about 100dB 1W but can also handle a great deal of power.
I don’t listen loudly normally but once in a while I do turn it up, have company or if I use the system for movies or a synthesizer etc.
With this approach, I know by measuring I will never be Voltage limiting or reaching significant loudspeaker non linearity in my home. A frequent comment from first time listeners is "i never heard anything like that before".
The exception is the fireworks recording, I can instantaneously peak both the sub woofer and full range amps (5300 Watts) if I have it much louder than it was in my backyard when I recorded it. The shuttle launch recording i can't play as subjectively loudly as the roof of the building i saw it on but still a lot of fun and scares the heck out of the dog, cats and Guinea pigs.
Maybe the thumb rule is useful and the LAST thing i want to do is suppresses anyone's curiosity about the technical side of life, especially when there is so much BS and mumbo jumbo used to sell things in hifi now days and to me appears to have slumped into something reminiscent of a dismal downward spiral.
Best,
Tom
My issue with a semi scientific approach to this is that it is based on a subjective judgment where the actual issue can be measured or estimated based on the input signal and speaker behavior.
I don’t see a compelling scientific or engineering argument for or connection between the technical issue at hand and a subjective judgment and rule of thumb, especially when one can easily examine the actual signals and Voltages instant by instant and know what is required based on fact and a theoretically idea system..
In the subjective a change of 10 or 20 dB isn’t that big when the span of hearing is more like 120dB and room noise floor typically in the 20 to 40dB range.
Also, there is a place for math but for it to be useful it must be in the proper context too.
For example, the simplest look says a 200W amplifier and 80dB efficient speaker can produce the same peak sound level as a 100dB sensitive speaker and 2W amplifier.
In reality, a commercial sound speaker made for a higher power level and sensitivity will typically be much more dynamic sounding than the small speaker driven at higher power that theoretically equaled the same level.
Loudspeakers are for sure the weakest link at least if one compares the signal going in to what comes out, so why shouldn’t this be examined, being hard or inherent in the approach is not an excuse I don’t think..
If one is looking at what comes out vs what goes in, the reason for old time efficient speakers like Altecs, Klipsch horns an such starts to become more understandable, while they have their own various sonic warts, they can preserve much more of the dynamics at living room levels because they can go so much louder than required.
While many are stuck on “this means it’s loud” and these systems can play more loudly too, it’s only the peaks that are any louder at home levels than more dynamically hobbled systems.
If you have ever sat down and listened to a set of efficient, powerful speakers like these, you already know what I am talking about, add enough power and gain structure adjusted so that one never clips anywhere and you only have the loudspeakers and program content limiting the dynamics.
My approach personally has evolved to use much more power and very efficient but modern horn speakers and subwoofers with sensitivity about 100dB 1W but can also handle a great deal of power.
I don’t listen loudly normally but once in a while I do turn it up, have company or if I use the system for movies or a synthesizer etc.
With this approach, I know by measuring I will never be Voltage limiting or reaching significant loudspeaker non linearity in my home. A frequent comment from first time listeners is "i never heard anything like that before".
The exception is the fireworks recording, I can instantaneously peak both the sub woofer and full range amps (5300 Watts) if I have it much louder than it was in my backyard when I recorded it. The shuttle launch recording i can't play as subjectively loudly as the roof of the building i saw it on but still a lot of fun and scares the heck out of the dog, cats and Guinea pigs.
Maybe the thumb rule is useful and the LAST thing i want to do is suppresses anyone's curiosity about the technical side of life, especially when there is so much BS and mumbo jumbo used to sell things in hifi now days and to me appears to have slumped into something reminiscent of a dismal downward spiral.
Best,
Tom
I'm a little confused.
Does this mean that if I got 3.4V rms at my "loud" listening level, that I...
1) Square that and I get my output voltage for calculating rms watts needed? Or...
2) Multiply by 4, then square that product, then divide by the nominal speaker impedance?
For 1) I get 3.4^2/8 = 1.445 watts
For 2) I get 3.4(4)^2/8 = 23.12 watts needed to reproduce 0dBFS
Does this mean I'm coasting along at 1.5 watts with >20 watt max peaks?
In that case, I think I'd want a 20 watt per channel amp. Maybe a 15wpc amp with nice clipping behavior would do.
Or I could swap in a more sensitive pair of speakers. I have 95dB efficient ones (Klipsch). An increase of 3dB SPL = roughly double the amplifier in, correct? So swapping to those 95dB/1w/1m speakers should reduce my "loud" level to about 1.75V rms.
Looks like I have something to test...
Thoughts on that?
--
Does this mean that if I got 3.4V rms at my "loud" listening level, that I...
1) Square that and I get my output voltage for calculating rms watts needed? Or...
2) Multiply by 4, then square that product, then divide by the nominal speaker impedance?
For 1) I get 3.4^2/8 = 1.445 watts
For 2) I get 3.4(4)^2/8 = 23.12 watts needed to reproduce 0dBFS
Does this mean I'm coasting along at 1.5 watts with >20 watt max peaks?
In that case, I think I'd want a 20 watt per channel amp. Maybe a 15wpc amp with nice clipping behavior would do.
Or I could swap in a more sensitive pair of speakers. I have 95dB efficient ones (Klipsch). An increase of 3dB SPL = roughly double the amplifier in, correct? So swapping to those 95dB/1w/1m speakers should reduce my "loud" level to about 1.75V rms.
Looks like I have something to test...
Thoughts on that?
--
Hi bksabath
It might seem hard to picture but a loudspeaker radiator producing a square wave does not move like it had a battery connected to it. For the simple case of a horn, it is the Velocity that corresponds to the pressure and so the radiator motion is actually a triangle wave, not square.
To record / reproduce a square wave in theory takes an infinite bandwidth as it is composed of an infinite series of harmonics. To make a square wave that looks perfect on an oscilloscope, requires flat magnitude and phase about 10 X higher and to 1/10 the fundamental frequency of the square wave.
The usefulness of the square wave is not that it is like a natural signal but that to make it look right, requires the system to be well behaved and only have one time of origin throughout the range of harmonics.
One of the other things loudspeakers do generally (if they can’t preserve the input wave shape like a square wave)is spread out a broadband impulsive signal to that what emerges has the various harmonics in a new order and spread out in time.
Best,
Tom Danley
It might seem hard to picture but a loudspeaker radiator producing a square wave does not move like it had a battery connected to it. For the simple case of a horn, it is the Velocity that corresponds to the pressure and so the radiator motion is actually a triangle wave, not square.
To record / reproduce a square wave in theory takes an infinite bandwidth as it is composed of an infinite series of harmonics. To make a square wave that looks perfect on an oscilloscope, requires flat magnitude and phase about 10 X higher and to 1/10 the fundamental frequency of the square wave.
The usefulness of the square wave is not that it is like a natural signal but that to make it look right, requires the system to be well behaved and only have one time of origin throughout the range of harmonics.
One of the other things loudspeakers do generally (if they can’t preserve the input wave shape like a square wave)is spread out a broadband impulsive signal to that what emerges has the various harmonics in a new order and spread out in time.
Best,
Tom Danley
Tom, you appear to be refusing to realise what it is we're trying to say to the point of appearing to simply wanting to be obstinate. This isn't a personal attack, but you are an intelligent fellow and every single one of your posts in this thread only reinforces the idea that you simply do not get what's going on.
Either that or you do get it, but are simply trying to dismiss the idea out of hand because it doesn't fit into your way of doing things when put into the real world of installing pro rigs in demanding environments with clearly specified SPL targets.
Pano and I both completely understand every point that you've been making. All of your points are correct and cannot be argued with, given the context you are talking about them in. However the context of this test and this thread does not fit in to what it is you've been trying to say. From our point of view it simply seems like you don't understand what it is we're trying to say. If you do get what we are trying to say then perhaps you should do so with an example so we can stop wasting our time in explaining what it is we're actually achieving by doing this. In fact this might be a good idea. You explain what it is you think we're trying to do here with examples as if you were doing this test yourself and then point out where you think the flaw is.
A.wayne you seem to have missed the point also.
Lets take Pink Floyd and the 1812. You seem to be implying that if you ran the test for the PF track and then for the 1812 that you would arrive at a different result. Damn straight you would! But no one is trying to argue otherwise.
The 1812 would no doubt place much higher demands on your system because its average to peak levels are very high. Ergo you set the volume control much higher for 1812 then you would for Floyd because it's necessary to get the quiet start of the 1812 up to realistic levels. Naturally when a peak in 1812 comes along it is far louder then any peak in Floyd so is more demanding of the system. So far I assume we are both in agreement on this point and so far we don't have a problem.
Now prior to running this test you are instructed to set the volume using your most demanding piece of music at the highest possible volume you would ever listen to it at. Ergo, if you only ever listened to Floyd then your volume control might only ever reach 4, but if you listen to 1812 then it could reach 7. This is obvious. But if you do listen to 1812 then you obviously run the test with the volume set at 7.
Given two identical systems in two identical rooms, the person that only ever listens to floyd will end up running the test at volume 4 and the person who listens to 1812 will end up running the test at 7.
Person floyd will arrive at say 5Vrms. For this person this is all they need as they never listen any louder and they only ever listen to Pink Floyd. As a result they decide to change to a high quality class A amplifier of 10 watts.
Person 1812 however will arrive at 15Vrms. For this person they need more amplifier output swing because they listen to more dynamic music.
Obviously if person Floyd suddenly decided to listen to the 1812 on their revised system with an amplifier of only 10 watts, they would clip their tiny amplifier, but this is to be expected. They deliberately chose that 10 watt amplifier based on the assumption that they would only ever listen to Pink Floyd and indeed when listening only to Pink Floyd it was clip free.
This is the point of the test. Run it having set the volume control to the maximum position that you would ever set it to using your most demanding piece of music. This will determine how much amplifier output swing you need. If I came along I might need more, but it isn't my system, we're sorting it out for you with your music and your listening level preferences.
Either that or you do get it, but are simply trying to dismiss the idea out of hand because it doesn't fit into your way of doing things when put into the real world of installing pro rigs in demanding environments with clearly specified SPL targets.
Pano and I both completely understand every point that you've been making. All of your points are correct and cannot be argued with, given the context you are talking about them in. However the context of this test and this thread does not fit in to what it is you've been trying to say. From our point of view it simply seems like you don't understand what it is we're trying to say. If you do get what we are trying to say then perhaps you should do so with an example so we can stop wasting our time in explaining what it is we're actually achieving by doing this. In fact this might be a good idea. You explain what it is you think we're trying to do here with examples as if you were doing this test yourself and then point out where you think the flaw is.
A.wayne you seem to have missed the point also.
Lets take Pink Floyd and the 1812. You seem to be implying that if you ran the test for the PF track and then for the 1812 that you would arrive at a different result. Damn straight you would! But no one is trying to argue otherwise.
The 1812 would no doubt place much higher demands on your system because its average to peak levels are very high. Ergo you set the volume control much higher for 1812 then you would for Floyd because it's necessary to get the quiet start of the 1812 up to realistic levels. Naturally when a peak in 1812 comes along it is far louder then any peak in Floyd so is more demanding of the system. So far I assume we are both in agreement on this point and so far we don't have a problem.
Now prior to running this test you are instructed to set the volume using your most demanding piece of music at the highest possible volume you would ever listen to it at. Ergo, if you only ever listened to Floyd then your volume control might only ever reach 4, but if you listen to 1812 then it could reach 7. This is obvious. But if you do listen to 1812 then you obviously run the test with the volume set at 7.
Given two identical systems in two identical rooms, the person that only ever listens to floyd will end up running the test at volume 4 and the person who listens to 1812 will end up running the test at 7.
Person floyd will arrive at say 5Vrms. For this person this is all they need as they never listen any louder and they only ever listen to Pink Floyd. As a result they decide to change to a high quality class A amplifier of 10 watts.
Person 1812 however will arrive at 15Vrms. For this person they need more amplifier output swing because they listen to more dynamic music.
Obviously if person Floyd suddenly decided to listen to the 1812 on their revised system with an amplifier of only 10 watts, they would clip their tiny amplifier, but this is to be expected. They deliberately chose that 10 watt amplifier based on the assumption that they would only ever listen to Pink Floyd and indeed when listening only to Pink Floyd it was clip free.
This is the point of the test. Run it having set the volume control to the maximum position that you would ever set it to using your most demanding piece of music. This will determine how much amplifier output swing you need. If I came along I might need more, but it isn't my system, we're sorting it out for you with your music and your listening level preferences.
Ok pano, i lugged stuff around and did your test, well somewhat, instead of your downloaded signal i ran my own test tone, 1/3 octave warble covering both the bass decade/mid/treble 20-20k @ -20 DBFS NOT -12DB...
Volume was set at moderate but good level ( low-z speaker not attempting to kill the amp) but normal listening level for me when using this amplifier, the results :
1. bass decade 20-200hz ..... 2.5 v
2. mid decade 250 -2 k .... 2.4 v
3. treble decade 2.5-20k ..... 2.0 v
What is this telling you ...?
Recording used was Paul Brown and friends track 7 , to set reference. my point when using non-symbiotic recordings you can achieve desirable levels with no clipping on one and not so on another, IE, if i try to do so with 1812 it will clip and blow out the fuses or eagles hotel california , the audience response at the end would take out the amp if i played it at the same level i did the Paul Brown track ...
ughhh...
Volume was set at moderate but good level ( low-z speaker not attempting to kill the amp) but normal listening level for me when using this amplifier, the results :
1. bass decade 20-200hz ..... 2.5 v
2. mid decade 250 -2 k .... 2.4 v
3. treble decade 2.5-20k ..... 2.0 v
What is this telling you ...?
Recording used was Paul Brown and friends track 7 , to set reference. my point when using non-symbiotic recordings you can achieve desirable levels with no clipping on one and not so on another, IE, if i try to do so with 1812 it will clip and blow out the fuses or eagles hotel california , the audience response at the end would take out the amp if i played it at the same level i did the Paul Brown track ...
ughhh...