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33
3rd Party Plugins - (fb2k) / Re: JScript Panel script discussion/help
Last post by ApacheReal -
In the end I managed to find the right settings with a static image.
Now even the vertical alignment is optimal.

Spoiler (click to show/hide)
34
Scientific Discussion / Re: What is the pre-amp input window range for dynamic microphone voltages
Last post by Maxotics -
You may have been confused by the tricks used in power amplifier output stage design, where extra voltages are needed to ensure there is no cross-over distortion
I believe we're talking past each other.  My point is that ALL amplification must make a trade-off.  It's a matter of thermodynamics...friction...noise...whatever you want to call it.  When one uses extra voltage to "ensure there is no cross-over distortion" that isn't, wasn't, ever desirable.  It's a kludge.  If transistors were 100% efficient it wouldn't be needed, right? 

Every component of electronics has a side-effect, so to speak.  So another component is added, not to remove it completely (which it can't) but to reduce it to the smallest cost possible.

Let me put it another way, I use a microphone.  I can't get the signal I want.  I ask why.  How much can't I get.  I get answers about all kinds of "this is the way the world (electronics) works."  But seldom, how do I put it, an honest answer.  So I say, there must be an answer because we have "laws" in how to measure currents so we must know where and how much isn't coming through.  Limitations in the materials used to make microphones, the amplifiers, etc.

It's like some people feel I'm attacking electrical engineering or something.  That I'm questioning their knowledge. 

Recording form microphones is OBVIOUSLY difficult which anyone who tries it understands.  There's a gain knob.  In effect, it is only a TRADEOFF knob.  You trade off one problem for the other. 

My question is, what are the tradeoffs as a percentage of theoretically perfect amplification and quantization.   Thoughts?
37
3rd Party Plugins - (fb2k) / Re: JScript Panel script discussion/help
Last post by ApacheReal -
My settings:

lines 17-26

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lines 110-115

Spoiler (click to show/hide)

result:



39
Scientific Discussion / Re: What is the pre-amp input window range for dynamic microphone voltages
Last post by fooball -
the way I understand amplification is, to simplify, if you have a really weak current you're going to need to seed or bias that current to get the amplification going.
No.  That is simply incorrect.  Amplifiers are internally biased to their "centre point", the input signal disturbs it away from that centre point whether the input be positive or negative.  That is the reason the signal path is AC-coupled, so that DC offsets do not affect the centre point.  I hope I've made that understandable.

("Centre Point" is not a standard technical term, I've used it to try to explain.)

You may have been confused by the tricks used in power amplifier output stage design, where extra voltages are needed to ensure there is no cross-over distortion, or the bias signal required to overcome hysteresis in magnetic recording.  These are compensated for in the design of the overall unit, do not affect the input or output (especially on properly-designed modern kit), and are not relevant to the likes of a pre-amp.

Something else which might be confusing you is the input bias current specification for op-amp ICs.  These are a consideration for the designer of the amplifier (if op-amps are used in the design) and not for the user of the amplifier.  If you are trying to design a high-performance pre-amp, then good luck with that!

No (audio) signal input should draw a DC current from the signal source.  Frankly, if the input did require "seeding to get the amplifier going", it would be a terrible amplifier!
40
Listening Tests / Re: Personal blind comparison of the Bluetooth codec, AAC and SBC and LC3.
Last post by magicgoose -
> For SBC, I'm not sure if there's much psycho-acoustic optimization in play.

AFAIK, there is a little room for choices based on psycho-acoustics.
In each slice, there's a number of frequency bands, which are encoded "independently", but their total size in bytes has a limit.
SBC encoder must choose which band gets how many bits, this ideally must be based on where the human listener will notice the distortion the least.
And for this choice I saw at least 2 distinct algorithms being mentioned (and only one widely used as the default).
There is probably some room for improvement in these algorithms to "squeeze the maximum quality" out of SBC, but I would guess it won't make a huge difference. There's also the problem that it's not practical to upgrade hardware SBC encoders anyway, so any work on improving a SBC encoder will not be delivered to some (arguably the most important) targets.
(Or does SBC encoding actually happen in special hardware? It's probably cheap enough to run on CPU in smartphones, too - but I don't know about that.)