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Old 22nd January 2011, 02:57 PM   #21
AP2 is offline AP2  Italy
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Hi,
This thread start with.."simple way to reduce power (<15w)".
I write a simple,easy way for it.
Yes, JBL used in 100w mini-box since 1990.

Regards
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Old 22nd January 2011, 03:41 PM   #22
maurycy is offline maurycy  United States
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Thanks to all who replied and provided insight. I think I am going to go with the bulb solution as I really need something easy. My electronic skills are also very limited so messing with gain is definitely beyond my knowledge.

AP2, since I am noob in electronics, by "in series to load" you mean just before the speakers?
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Old 22nd January 2011, 03:48 PM   #23
AP2 is offline AP2  Italy
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Quote:
Originally Posted by maurycy View Post
Thanks to all who replied and provided insight. I think I am going to go with the bulb solution as I really need something easy. My electronic skills are also very limited so messing with gain is definitely beyond my knowledge.

AP2, since I am noob in electronics, by "in series to load" you mean just before the speakers?
yes, bulb connected as "fuse" in speaker wire (or inside of speaker).
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Old 22nd January 2011, 04:31 PM   #24
maurycy is offline maurycy  United States
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Thanks. I guess I can put the bulb in the "munny" speakers so they will sort of glow. It would be a fun project for the kids.
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Old 22nd January 2011, 11:30 PM   #25
savu is offline savu  Romania
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Quote:
Originally Posted by AP2 View Post
Hi,
A very simple,very good system is a 6V lamp 1,5A in series to load (or try other current).
this , (becouse thungsteno change R with current) ensures an excellent automatic adjustment of the current

Regards
HI...

BOSE uses this method in they're systems for the woofer and satellites and it works flawless.

regards,
Savu Silviu
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Old 24th January 2011, 05:56 AM   #26
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Originally Posted by Eva View Post
Limiting the gain is useless. The extra gain is required because different recordings may vary in level as much as 10dB.

So then set the 0dB point at your peak level (with digital, peak level is equal to your highest possible average level) and let the -10dB fall where it may. This is not rocket science ... you can't do a digital recording without figuring this stuff out.

And, although it depends somewhat on what you're listening to, it's an extremely rare modern digital recording that is not pegged pretty much at 0dB with, at most, 3 dB of dynamic range over most of the track (ie +0, -3dB). Most, in fact, clip the digital waveform and basically just sit at 0dB for the entire song. Pop in your favourite CD (or, if you must, an mp3) and look at some waveforms. No dynamic range at all. Zero. Nada.

So, open any sound recording app on a computer, make an 0dB CD-R, and pop it into a CD player connected to your amp and speakers. Set the level. Done. The level will be lower on any other source, including a CD with actual dynamic range (if you can find one), so nothing to worry about.

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Originally Posted by maurycy View Post
After building my first gainclone amp based on LM3875 kit and two TA2020 amps (from pre-assembled boards), I would like to venture into a tube DIY. I know how to solder, identify parts, use DMM, etc ...
Thanks.
Quote:
Originally Posted by maurycy View Post
Thanks to all who replied and provided insight. I think I am going to go with the bulb solution as I really need something easy. My electronic skills are also very limited so messing with gain is definitely beyond my knowledge.
Hey, you can use the bulb. It's safe and will work. So what if it wastes power for no apparent reason? So what if it produces a load that the amp almost certainly was not built to drive. That's why you asked people who know, right? So you could choose the most lame possible way to do it. But, according to you, you have the skills to do it right.
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Last edited by Johnny2Bad; 24th January 2011 at 06:24 AM.
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Old 24th January 2011, 07:21 AM   #27
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Originally Posted by Johnny2Bad View Post
And, although it depends somewhat on what you're listening to, it's an extremely rare modern digital recording that is not pegged pretty much at 0dB with, at most, 3 dB of dynamic range over most of the track (ie +0, -3dB). Most, in fact, clip the digital waveform and basically just sit at 0dB for the entire song. Pop in your favourite CD (or, if you must, an mp3) and look at some waveforms. No dynamic range at all. Zero. Nada.
That's actually not quite correct. Although it's true than modern recordings are much louder than older recording, and thus have vastly reduced dynamic range, most will still have a average level at -10 to -9 dB which is todays de facto standard.

In the "good ol' days" before the "loudness war" recordings followed the RIAA guidelines that dictated between -18 to -12 dB depending on wanted dynamic range. Where -18 dB was called uncompressed and -12 dB was called maximum compression.

Please note that we here talking electrical dB where each +3dB is a doubling in level and not acoustic dB (Sound Pressure Level) where each +6dB is a doubling in level.

Todays de facto standard of tested amps reflects this in that they're tested against gaussian pink noise, usually normalized to -1 dB max peak which gives the -10 dB average level against RMS value.

So effectively you can pretty much always always divide the max RMS output power of an amp by 8 or 10 to get the actual maximum average power output with music signals.

This is why I say that unless he wants to play continuous sine waves, he don't need to lower the output at all, or even protect the speakers in any way. They'll not see more power than they're rated to.
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Old 24th January 2011, 08:39 AM   #28
Pafi is offline Pafi  Hungary
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Please note that we here talking electrical dB where each +3dB is a doubling in level and not acoustic dB (Sound Pressure Level) where each +6dB is a doubling in level.
This is a misconcepcion. First of all, level means a dB value, so this is a self-contradiction. Further, there are no acoustic dB and electrical dB (or level) as such. The 2 different kinds of levels you could have heard about are levels related to voltage (or current, pressure, etc...)
http://upload.wikimedia.org/math/9/0...fabacb463a.png
and level related to power (or intensity, etc...).
http://upload.wikimedia.org/math/3/5...fea2b719b0.png

One of the most important question is: what is the reference value (A0, P0)? In case of CD Audio it can be the physically allowable maximum (2 Vpeak in analog, 00h or FFh in digital), or the maximum of the actual song. Both are useful, but in other aspects...

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which gives the -10 dB average level against RMS value.
Level of what, and RMS value of what? You compared 2 undefined values wich are different in many aspects (but none of them are mentioned by you).
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Old 24th January 2011, 05:04 PM   #29
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Originally Posted by Saturnus View Post
That's actually not quite correct. Although it's true than modern recordings are much louder than older recording, and thus have vastly reduced dynamic range, most will still have a average level at -10 to -9 dB which is todays de facto standard.
I suggest you look at some waveforms of your favourite CDs, or even better an LP transcribed to digital. On older, tape-mastered disks you can normalize at -13dB Average and avoid clipped samples on 90% of the music recorded prior to 1990. Those that cannot be normalized at that level will have increased dynamic range (ie you would have to normalize at lower than -13dB to avoid clipped samples), not less.

On modern digitally recorded CDs the overall level will be reduced significantly if you were to attempt to normalize at that average level, indicating that even that small amount of dynamic range doesn't exist in the recording. [However, when playing back music, the modern and the older recording would have the same perceived loudness if normalized at the same average level, which is much more pleasant to listen to when playing multiple songs in a random order; it's what iTunes does when you enable "sound check", although sound check does a poorer job, sonically than manually editing the file would].

Eg: Katy Perry's "I kissed a girl"
Maximum dynamic range 4.9 dB (peak level measurement); -6.1 dB (average level measurement). In other words, you could normalize at -6.1 dB average and not clip any samples (that were not clipped on the original disk).


Quote:
Originally Posted by Pafi View Post
This is a misconcepcion. First of all, level means a dB value, so this is a self-contradiction. Further, there are no acoustic dB and electrical dB (or level) as such. The 2 different kinds of levels you could have heard about are levels related to voltage (or current, pressure, etc...)
http://upload.wikimedia.org/math/9/0...fabacb463a.png
and level related to power (or intensity, etc...).
http://upload.wikimedia.org/math/3/5...fea2b719b0.png

One of the most important question is: what is the reference value (A0, P0)? In case of CD Audio it can be the physically allowable maximum (2 Vpeak in analog, 00h or FFh in digital), or the maximum of the actual song. Both are useful, but in other aspects...

Level of what, and RMS value of what? You compared 2 undefined values wich are different in many aspects (but none of them are mentioned by you).
I just assumed you understood that a recording level is one thing and electrical signals are another.

With an analog recording, the 0vu level would be set based on an electrical level, although that level was a standard, the standard varied depending on whether your meter was calibrated to consumer of professional levels. Exceeding 0Vu was possible at the expense of increased distortion when recording a session to tape, but the engineer would use his good judgement, so large variance in average levels wasn't common. LP records would require certain constraints be adhered to to avoid the cutter making extreme excursions and creating an unplayable record, but that would be accounted for in mastering.

Broadly speaking an LP record had a fairly consistent level due to the desire to maximize level versus the practical needs of the cutter and the physical limitations of the LP. If you were to have large excursions encoded in the LP (deep bass), for example, the length per side had to be reduced. So the desired length of the LP by the label would limit overall levels and probably involve equalization to limit LF content. If you looked at an album and it had 25 minutes per side, you knew there could not possibly be much low bass encoded. If it were perhaps 18 minutes per side, it could possibly have extended LF information. And so on.

With a digital recording, 0dB is the maximum record level. This level cannot be exceeded regardless of how you try; all that will result is clipped samples and a higher average level (less dynamic range). All levels on a digital recording are referenced to that level; so -6dB is six decibels down from the maximum level possible. In contrast to an LP record there is no practical limit on LF response since there is no penalty to include it that matters to the artist, engineer, or label ... you can always create a CD Master of the same length, for example, with or without extended LF response.

In practical terms analog recordings were mastered for LP, so the cassette would use the same master, as would the CD release. Similarly, modern digital recordings are created for CD and the mp3 would use the same master, although there are some engineers who do master for each format independently today.

"In the case of CD Audio it can be the physically allowable maximum ... or the maximum of the actual song."

On a modern mastered disk, those values will be the same.

For more on dynamic range and current mastering practices, see:
http://www.cdmasteringservices.com/dynamicdeath.htm

Note that the most recent example in that article is from 2000, I can assure you it's become worse in the last decade.

When played back by hardware, that level will correspond to some maximum output level electrically. Exactly what voltage that is depends on the device and the load it's driving.

Since most CD players are supposed to output 2V, and since no other conventional line level signal is that high (eg, an am/fm tuner, typically 0.775V) then you can use the output of your CD player as your maximum output level expected.

All other line level signals will be lower. It's best to use your own CD player since few actually output exactly 2V and there is variance from unit to unit (it's extremely common for it to be above 2V), and you want to set the level necessary based on the gear you will be expected to use with the amplifier and speakers.
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Last edited by Johnny2Bad; 24th January 2011 at 05:33 PM.
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Old 24th January 2011, 05:18 PM   #30
macboy is offline macboy  Canada
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the obvious (to me) answer is to lower the rail voltage to the chipamp. Reducing it to about 2/3 will give less than half the unclipped output power.

Another way is to put a power resistor in series with each speaker. This makes a voltage divider with the speaker. Half the voltage gives 1/4 the power. This will affect the frequency response of the speaker, if the speaker impedance is not flat across frequencies (it never is). Given the lo-fi nature of this sytem, it should not be a problem.
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