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Why you need 500Wpc for good sound


Islander

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Well, it looks like Dr Who is frightened by clipping. I'll confess that it scares me too.

If you are averse to clipping and you are averse to mega-watt amps, there is still a solution. Try bi-amping!

Go ahead get a smallish amp (SET, Pass etc) and use it for the spectrum above 500 Hz or so and use a slightly beefier amp for the low end of the spectrum (let's ignore sub-woofer regions for this example). You will need have the crossover BEFORE the amps to get the benefit. But you are in effect decreasing the need for a mega-watt amp in order to decrease clipping. The bass will be better handled (this is the usual weakness for smallish amps) and the wonderful mid and highs can be achieved with the tiny amp (the reported strength of smallish amps).

The usual objection is that, hey this will double up the costs for the amps. Not really. when you do this you would be surprised how good some rather commonplace (ie, affordable) amps can sound. There are plenty of outboard crossovers that can be used, although this aspect is probably where most will have difficulty in the implementation.

Good Luck,

-Tom

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Tom... along your line of thinking... (and this is probably better if I made its own thread, but be it as it may...)

Would you (or anyone reading) suggest that if you have two different amps when biamping, that you might get two different flavors of sound and as such, have a 'funky flavor' at the crossover point?

Example: Some people might say solid state is sterile... but suitable for bass work. Ok fine! What if we put a solid state from 1hz to 500 hz and then a 2A3 or OTL or (insert) from 500hz on up.

The sound I heard with my little SE-OTL's was a more real sound than I had with the other amps. To try to be concise, it was perhaps a matter of 6 verses 7 shakes of salt on a steak, but there was still a perceived difference.

So... if I put that SE-OTL on my K402 crossed about 500 hz and put a (tube push/pull or solid state) on the bass bin.... would mixing those two different technologies cause any sound issues that some might feel is negative?

Seems to me I once read Deneen saying soemthing about the 'voicing of an amp' and now I'm wondering if this is what he was talking about?

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I know I am being an unreasonable stubborn PITA but dang it my ears don't lie my brain perhaps but not my ears. What DrWho writes makes sense but it doesn't jive with my listening experience. I wouldn't be so stubborn if I hadn't lived with such a system for several years. I have not heard SE-OTL. As for mixing and matching topologies I like to keep it simple but Mikes2a3 does what your suggesting. I think he uses 2A3 SET on the HF and PP 2A3 on the LF. BTW when I get my Jubilee lfs I'll be saving mymoney for TADs.

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I bet if I picked the source material and we used speakers that actually played low enough that you would be able to notice your 8W clipping at about 80dB (even with better sensitivity than your belles). I'm betting that you might have become accustomed to your ears using the upper harmonics to fill in the lower notes not being reproduced, which is going to be a similar exeperience to when your SET amp clips "softly". The increase in inner detail and all that schnazz is just the quieter information becoming relatively louder...

This is what is going on a lot of the time with SET and other small amps. The soft clipping that is produced by these amps is percieved as euphonic, and that sound is preferred to a less distorted sound. Distortion may be defined as frequencies in the output not present in the source that were generated by a piece of equipment. These additional frequencies increase the loudness perception. All else being equal, the louder sound is usually judged to be the better sound. Aural exciters are used in recording and live sound situations for that purpose and have been shown to increase clarity and intelligibility. Like it or not, it's still distortion.

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Don: I agree that some folks will actually perfer an added amount of distortion. However I can not agree with you r argument that the increased loudness is the mechanism.

If I have a 1 volt signal and I add 10% distortion (0.1 volt) the combination will now produce only 0.8 dB more SPL (or voltage if you prefer). This would barely dent any increase in loudness, in fact you would be hard-pressed to call it a "louder" signal.

Yes, there is a common finding that a louder signal will be preferred, but this typically requires a much greater increase in SPL (or ultimately loudness). My example of 10% is arbitrary, but it does represent a large amount of distortion.

-Tom

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Because dynamic transients consist of more than one frequency.







Btw,
I'm being pushed into making some red herring arguments to get the
point across...I'm not suggesting that a 110dB speaker is going to need
10,000W for accurate reproduction of most source material. I'm trying
to point out that there is more involved when dealing with music. The
sensitivity calculations are absolutely true when dealing with single
frequency test tones, but the single frequency condition doesn't hold
true for music.







I am saying, however, that even 10W doesn't cut it for even a 110dB speaker.




Btw,
I would love for 10W to be enough because it makes the amplifier design
incredibly easier. Also, the resolution of the system always increases
with decreasing gain in the amplifier (assuming a system with enough
headroom). That said, I don't think you'll see me making any amps with
less than 50W power handling and you'll probably always find me with at
least 100W in my living room (assuming 110dB speakers). But hey, I like
very dynamic music at full (not loud) volumes...

I agree that I need lots of power for drums at loud levels. But when I listen to a jazz trio at normal levels, the volume dial sits at -40 dB. I disagree strongly with your 40 dB headroom number. There is no way that I am remotely close to clipping at any time. It all depends on content for sure.

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If I have a 1 volt signal and I add 10% distortion (0.1 volt) the combination will now produce only 0.8 dB more SPL (or voltage if you prefer). This would barely dent any increase in loudness, in fact you would be hard-pressed to call it a "louder" signal.

Perception of loudness does not depend on power levels exclusively. 1 dB in a narrow frequency band is easily heard. If that 1 dB is harmonically related to the fundamental the fundamental frequency will appear louder by more than 1 dB.

Precedence can make sounds of equal intensity sound different. The earlier-arriving sound will dominate. This may be one reason that Khorns sound "bright" even though they measure relatively flat. BBE maximizers use this principle in their operation.

When first introduced, Aural Exciters had many folks confused as to how they worked. As you said, signal levels were changed little compared to the percieved increase in loudness. A little bit of distortion goes a long way... you need to dial enough distortion to do the job without sounding badly distorted.

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A little bit of distortion goes a long way... you need to dial enough distortion to do the job without sounding badly distorted.

I like a little "clean" distortion when listening to rock music. Listening to smooth jazz however, I want it as clean as possible.

"A little bit of distortion goes a long way" sounds right to me........at least that's what my ears are telling me. [:D]

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Don, you have confused a couple of issues.

First, the loudness summation across critical bands will not be an extra 1 dB. The harmonic is only 10% re: the "signal" (or down 20 dB SPL). There will be very little "summation" due to this asymmetry in level. Even if it were a 1 dB increase in level, this would not be very noticeable in terms of an increase in loudness. In fact it would be very close a threshold difference and not a big deal.

Second, "precedence" would apply to signals arriving in asynchronously in time. Specifically, the effect pertains to the "spatial" information in the signals (binaural cues). It does not pertain to other aspects of the sensation like loudness, pitch etc. Besides the distortion would not be temporally asynchronous.

Perhaps you are confusing the ability to detect a signal with the listener's sensation of its loudness. These are very different things.

-Tom

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First, the loudness summation across critical bands will not be an extra 1 dB. The harmonic is only 10% re: the "signal" (or down 20 dB SPL). There will be very little "summation" due to this asymmetry in level. Even if it were a 1 dB increase in level, this would not be very noticeable in terms of an increase in loudness. In fact it would be very close a threshold difference and not a big deal.

I was using your figures showing that 10% distortion increases output by approx 1 dB. The point I was trying to make is that the 1 dB difference in level sounds louder than 1 dB if it is caused by harmonically related information, such as harmonic distortion. Try an experiment - play a pure note on a synthesizer, then add 10% 1 octave up. The fundamental will appear to be louder by quite a bit.

I mentioned precedence to illustrate that sounds can appear to be louder with no increase in actual level. Of course, this is not what is occurring here with distortion. BTW, if you delay a signal by 25 ms relative to another, you will need to increase power 10 dB (10X) to make both sources appear to be equally loud, assuming that you are equidistant from both sources.

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" My original point (and it is not an original idea) is that bi-amping can acheive many of the goals."

I agree with that....bi-amping and/or bridging results in the addition of power supply reserves.

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" My original point (and it is not an original idea) is that bi-amping can achieve many of the goals."

I agree with that....bi-amping and/or bridging results in the addition of power supply reserves.

Actually the advantages can go beyond that. A note of caution regarding bridging an amp, this may also increase the amp's distortion.

-Tom

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" this may also increase the amp's distortion."

yes..more componets...and then there's the buisness of which approach to use....high impedeance speakers should be connected in serial across the + connections of the amps while low impedeance speakers should be connected to parallel wired amps....I'm personally would not parallel wire SS amps.

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How much power compression does it take to cause 10% distortion? I honestly don't know.

The
issue with compression is that the loudest parts don't really get any
louder, but the "inner detail" (or sounds inbewteen the loud parts)
become louder. I'll go ahead and hypothesize that 10dB of compression
would correlate to about 10% THD...but I'd be interested to know where
that actually lands. A possible 10dB is a rather dramatic increase for
the quieter source material and I would expect it to appear much
louder.

I also agree that bi-amping reduces the strain on the amp.

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I also agree that bi-amping reduces the strain on the amp.


Could it also be that bi-amping greatly reduces harmonic distortion, since the treble amp cannot be getting any harmonics of tones below the crossover point? If I understand harmonic distortion correctly, a 100Hz tone has a first harmonic of 200Hz, a second harmonic of 400Hz, a third harmonic of 800Hz, and so on. If an amp's HD is "heavily third-harmonic in nature", as I've read in a number of reviews, I take that to mean that spurious harmonics from the low end will be generated well into the midrange.

If, for example, the crossover point in a 2-way system is 500Hz, the highest third-harmonic frequency generated by tones below that would be 4000Hz. If the system is bi-amped, the treble amp won't have any of those harmonics, since they'd be generated by the bass amp and its output is cut off above 500Hz. This implies that there would be no third-harmonics between 500Hz and 4000Hz, meaning the midrange would sound cleaner.

Am I understanding this correctly? I do notice that a bi-amped system does sound more clear, but I'm not sure of all the factors that give it that clarity.
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I agree that I need lots of power for drums at loud levels. But when I listen to a jazz trio at normal levels, the volume dial sits at -40 dB. I disagree strongly with your 40 dB headroom number. There is no way that I am remotely close to clipping at any time. It all depends on content for sure.

Maybe it's a Francophone thing, but we agree 100% on this one. Mine sits at about the same place. My original point is still about rating amps in dbWatts insteat of Watts, which are misleading. Even though I don't think I need more than 10 Watts (since normal is in the 10-100 milliwatt range) on my 108db/Watt setup, my receiver still has dynamic headroom in excess of 100 Watts per channel, so it's never a concern.

I think we have beaten this topic to death.

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I disagree strongly with your 40 dB headroom number.

I second psg on this as I have never in my life heard of 40db headroom on any amp period. Crown has always been noted for having gobs of headroom and my last Crown amp had 4db of headroom which is easily noticable. I've heard people say they are getting 20db headroom before but I have never seen that measured or heard it[:|] I just find 40db of headroom to be far fetched but I yeild to all you DIY'er builders as I'm always ready to learn something new[;)]

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Don: I give up. We don't seem to have a common language to discuss this topic.

Besides the original topic is about clipping and the need for mega-amps. My original point (and it is not an original idea) is that bi-amping can achieve many of the goals.

-Tom

In the Dope From Hope, Vol.15, No.4, dated 5th January, 1976, PWK wrote:

"Distortion makes a given sound pressure 'sound louder'."

For a good write-up on amplifier clipping and it's audibility:

http://www.moultonlabs.com/more/what_is_the_sound_of_one_amp_clipping/P0/

I hope this clears everything up for you.

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