John Curl's Blowtorch preamplifier part II

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I still can't understand how to get the thermal noise in each channel (or the 1/f noise or any other excess noise) to be correlated with the other channel without using a Bybee. Could someone help me? (N.B. noise is random - the clue is in the name).

First off random noise is random, however in acoustics noise is any unwanted signal. It need not be random. The fire engine siren going by you is noise. If it stops at your residence it is signal.
 
FM is usually bandlimited to 15kHz, so music doesn't fiddle with the pilot tone at 19kHz.

DAB, with a 48kHz sampling rate, ought to be able to go to at least 20kHz. Your example stops at 12kHz so I wonder if it was half-rate? I believe 24kHz is allowed in the DAB standards. Maybe they just filtered out all that nasty transient stuff which the encoder can't handle, such as cymbals, applause etc?
 
Reducing the randomness of the system reduces randomness of the resultant output signal, but not the original signal.
Eg, reducing the temperature of a system will reduce the system intrinsic noise (randomness), and hence reduce the resultant output signal randomness.
Or, clean up the power supplies and there will be less alteration of the through putted signal.
Where does the 2nd Law fit ?.

Dan.
 
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Mooly you mean: can you here the 19kHz stereo pilot?
I am surprised it is in the spectrum at all. Wouldn't the MPX filter be expected to make short shrift of that??
FM is supposed to be sharp filtered at 15kHz.

Jan

I used to be able to in the quiet bits, yes.

I'm not sure just how well Audacity handles looking at a spectrum like this but you can see the roll off around 15kHz. The level is getting way down in the noise at that point.
 
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FM is usually bandlimited to 15kHz, so music doesn't fiddle with the pilot tone at 19kHz.

DAB, with a 48kHz sampling rate, ought to be able to go to at least 20kHz. Your example stops at 12kHz so I wonder if it was half-rate? I believe 24kHz is allowed in the DAB standards. Maybe they just filtered out all that nasty transient stuff which the encoder can't handle, such as cymbals, applause etc?

I wouldn't like to say on that. All I can say is that it is what it is. Seem to recall the BBC have kept faffing with the encoders to try and overcome criticism of the 'quality' of DAB which was particularly obvious on strings.
 
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When I set the volume to normal listening level with no input I hear nothing at all, what's the problem? Someone somewhere in time made a pathologically bad amp/pre-amp so what?

Scott
When the audio chain has 3-4 equipments, flipping the (non polarised) plug of one or two of them, may make a difference in the ‘noise’ (*).
I’ve done it on several set-ups on many homes (and with no diy equipment in the chain :D)
The explanation of Ed (capacitive coupling of primary side through the core of the transformer) is valid.
Measuring leakage current confirms this.
Alternatively, with the equipment unplugged from the wall outlet, measure the capacitance btn each prong of the power chord and the chassis of the equipment.

(*) and yes it’s not noise in the strickt sense (random spectrum), call it a weak buzz


Go back and forth and you will quickly learn the affect of channel separations importance to high-end music reproduction.

In other words Vinyl and FM front ends are out :p


FM is supposed to be sharp filtered at 15kHz.

At the modulation stage (transmitter side).
The receivers are usually equipped with a 19kHz notch filter. Some with an additional low-slope LP filter past 15kHz.


Might be quite a bit more for some.

Place the bone conduction into the picture.:)

George
 
And just in case there is not enough misinterpretation in this thread I give you It's Official: People Can Hear High-Res | AudioStream

Didn't take long to be wrong. He shows a perfectly band-limited signal sampled above Nyquist and says that there is information lost between the samples. :rolleyes:

Figure 1 shows the result of sampling a signal. The upper trace is the continuous-time signal, while the lower trace shows the signal after being sampled once per millisecond. You may wonder why the lower trace shows no signal between samples. This is because after sampling there is no signal between samples—all the information that existed between the samples in the original signal is irretrievably lost in the sampling process.
 
(*) and yes it’s not noise in the strickt sense (random spectrum), call it a weak buzz

George

Funny when you say it it's OK but when I do I'm wrong. Yes a long series of higher line harmonics added at a fixed phase relationship does not sound like random noise. Just ran into this yesterday trying to get a sample of a customers transducer to work. My office has horrible line trash, taking the noise floor and notching out the first few harmonics just changes the base buzz tone but it remains distinct from the hiss.
 
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Funny when you say it it's OK but when I do I'm wrong. Yes a long series of higher line harmonics added at a fixed phase relationship does not sound like random noise.

It only makes it up to buzz at levels high enough to have the lower frequencies rise above the threshold of hearing. Then most folks can identify the issue.

I did a demo CD where we recorded hum, buzz and noise to explain that hum was almost always a magnetic coupling, buzz a resistive differential mostly and "Noise" a high frequency interference often through capacative coupling.

Lost the files in the Hurricane Ivan flood otherwise I would post them for you t0 see if you can hear the difference.

(Also did popcorn noise from a particularly bad microphone preamp and a bunch of other common distortions. Did send Ed Dell a copy but they never released it as the issues were more pro oriented.)

And yes I disagree with Georges word usage but it is cute how you are trying to turn the issue around.
 
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