(http://i1350.photobucket.com/albums/p769/striker_amplification/rendergifPLEXI_zps8233a6f1.gif)
ground plane?
im sorry, what are you asking?
you have access to and use of google, or some other search engine, yeah?
Im on the internet talking to you arent i?
use that clever a little bit further and you'll go far
Ahh i gotcha. ;)
i doubt it
And i doubt you have any real idea how to do half of what you think you do, my designs are original not copyright, cant say the same for you.......have a nice day artifart.
Thats what happens when you approach people with ill intentttttt :icon_lol:
Quote from: STRIKER AMPLIFICATION on July 10, 2013, 07:04:14 PM
And i doubt you have any real idea how to do half of what you think you do, my designs are original not copyright, cant say the same for you.......have a nice day artifart.
you got me! well done you!
enjoy your life my friend - better luck in the next one! :P
:icon_lol: OK!
ha! you are funny.
What is supposed to make this sound like a Plexi? No tone stack? What advantage does a second transistor have over just building an Electra distortion?
You have clipping diodes to ground connected to the output, so my guess is that the output level of this thing is going to be very low
Not really, the signal has still been preamplified, without clipping its too loud, even with the output clipped its still louder than bypassed signal.
And as for "how this will sound like a plexi"? its a very clean yet extremely saturated signal, jimmy page all the way buddy :icon_twisted: don't knock it till ya rock it brotha.
Quote from: STRIKER AMPLIFICATION on July 10, 2013, 09:24:26 PM
And as for "how this will sound like a plexi"? its a very clean yet extremely saturated signal, jimmy page all the way buddy :icon_twisted: don't knock it till ya rock it brotha.
I know what transistors plus hard clipping diodes sound like, and I'm not commenting on the sound, which may very well be just as good as other germanium transistor + hard clipping setups. I'm asking why you're saying it sounds like a plexi when there is no obvious attempt to mimic the tonal response of the amp or include amp's tone stack. There isn't even a gain control identified. The input cap cuts a HUGE amount of bass, even more when you consider the low impedance created by gaining up the transistor like that (maybe a couple hundred K at most?)
You also haven't identified the diodes used. This is beside the fact that symmetric hard clipping diodes bear very little resemblance to tube triode breakup.
So why not just call it an overdrive instead of attaching the name of a (highly recognizable) amp to it?
Join
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if you want to reach a larger diy audience with your designs.
Am I missing a trick, here, or is there no bias voltage getting to Q2 base at all? In which case, Q2 collector is going to get stuck at +9V and you won't get anything out of this circuit as drawn at all... Well, a tiny bit of heat, maybe...
The input cap is suposed to block almost all bass signal ,the bass signal is then reformed in the second driving stage in a much larger wave form, then brought back down to about 50% then clipped, this will give the drive a very sonic sound, the bass is produced WITH the guitar signal not MIXED within it.
Bias, not bass
Since he's using germanium transistors, the leakage could be enough to bias the second transistor into working, but every build will be different and highly dependent on the individual transistor used and it might be hard to get consistent final results. In an other thread someone suggested posting schematics instead of layouts. That would be much beter if you want more people discussing the details of your designs.
J
so, just socket the damn transistors and swap 'em around til it sounds good.
this looks like it may be an interesting sound with the right transistors, but *ANY* ac176 sure ain't gonna work.
i don't see how you can "re-add" bass tho. once you take them frequencies away you can't get 'em back. just like an mp3. once it's gone, you can't replace it. just doesn't work like that.
you MAY get sub harmonics added in, but that's not the same as bass frequencies.
what kind of amp did you test this with?
it may sound GREAT thru a ruby or small amp, but how will it hold up at stage volume? i build all kindsa ridiculous shit just to try it, and it often sounds great at low volume...
but until i try it thru one of my live rigs at real volumes, i don't really know what it sounds like... small amps homogenize the shit out of tones. what may sound great at a high gain thru a low volume amp often don't fly so well in real life use.
believe me, i've built a LOT of stuff... most sounds great quiet. get on stage, and well, that's not necessarily the case.
can you post examples? video/audio?
and original design?
it's a mis-biased emitter follower to my eye, the fuzz face it's modeled on was the original.
what you seem to have done to my newb eyes is take a fuzzface, and misbiased it into a clean booster.
i like to run my collectors at higher voltages too, but this is only gonna work with some transistors....
can you post their gains and voltages? be curious to look at.
it's almost like ya wired the emitter follower like a cathode follower. cool.
but straight b+ to the collectors is gonna be a little weird.
btw...as for artifus?
dude, you are dead @#$%in' wrong.
like to say welcome to the forum... but YOU gotta understand, you're gonna get constructive criticisms, sometimes props, sometimes flops...
and getting pissy, even in jest, isn't gonna get ya a lot of friends here. this isn't harmony central with people trolling and shit. none of us will tolerate that crap.
we're all in this together.
enough of that shit...
welcome, dude and thanks for sharing!
What part # are the diodes?
i test all of my pedals on a...(get this) 300watt sony stereo reciever with a 1968 fender 2x 15 concert series cabinet.
iam sorry for anything i have done, it wont happen again.
dude, it's all good man.... like i said, welcome to the forum!!
a 300 watt power amp?
lol
;)
not the same as a guitar amp.... distortion on the order of about a million times less, even with a "clean" amp. very tonally different, but may explain why it sounds as you describe.
rock on dude!
well after i have them tested on the power amp, i use a 68 silverface bassman, if it passes the power amp test, it will most certainly pass the guitar amplifier test, the breakup compared to the guitar amp is at least 50% more than the power amp, that wonderfull extra preamplifier does it all for ya! ;)
And i did in fact mean BASS not BIAS. I cant see how that would make any sence in that post. If i were refering to bias i would have used the word BIAS. Thanks: Ryan. ;)
you'll be alright ryan, anybody that can goof on themselves can't be all bad. ;)
but...that said,.. WE NEED CLIPS!! VIDEO OR IT NEVER HAPPENED... :icon_mrgreen:
Cool thang man ;)
Quote from: STRIKER AMPLIFICATION on July 11, 2013, 05:22:32 PM
And i did in fact mean BASS not BIAS. I cant see how that would make any sence in that post. If i were refering to bias i would have used the word BIAS. Thanks: Ryan. ;)
Yes, but my post immediately before that was about bias, not bass. Forgive me for assuming that you were answering my question.
Quote from: pinkjimiphoton on July 11, 2013, 12:47:09 PM
i don't see how you can "re-add" bass tho. once you take them frequencies away you can't get 'em back. just like an mp3. once it's gone, you can't replace it. just doesn't work like that.
from my experience, unless it actually does "completely reject" all of a specific frequency, any passive or active tone controls do greatly reduce a frequency, but they're still there, albeit at far reduced amplitude. As long as there's even .0000001 volts of that frequency there, it's theoretically possible, if not practical, to gain that specific frequency back up to the same amplitude as the rest which weren't tone controlled out. I've played with the treble booster schem from Muzique, and Even though there is much reduced dB for bass frequencies, those frequencies are definitely still there, even though it's something like a 0.002 input cap.
I don't want to be pedantic, but does this thing actually completely erase all of that frequency? As far as MP3, if you use software to remove the frequency completely, you cant get it back, but if you're just adjusting equalizer levels, it's still there way down. Right?
You are 100% correct ,The comedian. ;)
That's not jimi's point, comedian. His point is that, once you have digitised and compressed an analogue signal to turn in into a lossy mp3, you can never get a truly accurate analogue signal from it. Similarly, if you remove all bar a vanishingly small amount of any part of the frequency spectrum from a signal, you can never get it all back into balance without compromises. Like noise, for example. You alluded to that yourself, when you raised the question of its practicality.
@Ryan - Any insight into how Q2 is being biased? I realise that it's only the third or fourth time that I have mentioned it, but I'm not alone in wanting to know.
Again, what are the diodes?
sorry, 1N5404
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?
There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.
http://fuzzcentral.ssguitar.com/schematics/fz1.gif
That one starts off with a no-bias emitter follower! The transistor starts to conduct when the input voltage swings start to pull some current through the cap. But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.
hello.
my name is artifus and i am a human being.
thank you for your compassionate tolerance and understanding.
much love.
art,
we love you too, artifus, for sure... you're a very valued and respected member of the community. ;)
Quote from: pinkjimiphoton on July 12, 2013, 05:16:26 PM
we love you too, artifus, for sure... you're a very valued and respected member of the community. ;)
I believe you hit on a point that a lot of newcomers fail to pick up on.
This is not a blog where people log on to rant/bark about something/anything!
DIYstompboxes is a community with many long term members that should be treated as such. Respect should be shown to those that were here before, because without their loyalty and encouragement, this site may not have survived.
I often find answers to questions in the archives from years ago, written by a member who no longer frequents this site.
One can "register" to access all areas of DIYstompboxes but it is my observation that it takes much more than a simple registration to become a "member."
This site and help from it's members has allowed me to build, modify, and design things that I wouldn't have imagined before!
I think at the end of the day my best advise is......stick around and assimilate! You just might learn something!
Quote from: armdnrdy on July 12, 2013, 05:45:44 PM
This is not a blog where people log on to rant/bark about something/anything!
You do need to at least consider that Something/Anything is one of Todd Rundgren's most creative endeavors and has some heavy guitar, e.g. "Black Maria". ;D
appreciate it guys but i think i was a little out of order - think i was just enjoying myself a little too much and expecting everyone else to know where i was coming from and to share my warped sense of humour. i'm glad it was picked up on. i feel humbled. and thats' no bad thing. peace, y'all. x
Quote from: Digital Larry on July 12, 2013, 06:00:24 PM
Quote from: armdnrdy on July 12, 2013, 05:45:44 PM
This is not a blog where people log on to rant/bark about something/anything!
You do need to at least consider that Something/Anything is one of Todd Rundgren's most creative endeavors and has some heavy guitar, e.g. "Black Maria". ;D
I've taken it under consideration and I'll get back to you with a reply. A response to my use of the phrase "Something/Anything" could take some time, possibly as long as a year. (kind of like the government) ;D
Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?
There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.
http://fuzzcentral.ssguitar.com/schematics/fz1.gif
That one starts off with a no-bias emitter follower! The transistor starts to conduct when the input voltage swings start to pull some current through the cap. But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.
That one uses base leakage to get a bias voltage across the base resistor
The circuit here has no resistor at the 2nd devices base so the way it work is the signal from the first stage is cap coupled to the base and if it is large enough one way it turns the device on.
It is two gain stages in series
grounded emitter with a C to B bias feedback resistor is not the most predictable way to bias a circuit
Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?
There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.
http://fuzzcentral.ssguitar.com/schematics/fz1.gif
That one starts off with a no-bias emitter follower! The transistor starts to conduct when the input voltage swings start to pull some current through the cap. But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.
Larry,
That FZ-1 is a positive GND, 3 x PNP circuit, so each stage
is biased. This one is negative GND with 2 x NPN and Q2 has no resistors going to its base at all. Therefore, no base current, so Q2C locks up at +9V. In addition, the 1M from Q1C to Q1B is surely providing negative feedback, thus limiting the gain achievable.
I can see how this would work if there was another 1M, for example, from Q2C to Q2B but...
Quote from: Gus on July 12, 2013, 06:35:10 PM
Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?
There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.
http://fuzzcentral.ssguitar.com/schematics/fz1.gif
That one starts off with a no-bias emitter follower! The transistor starts to conduct when the input voltage swings start to pull some current through the cap. But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.
That one uses base leakage to get a bias voltage across the base resistor
The circuit here has no resistor at the 2nd devices base so the way it work is the signal from the first stage is cap coupled to the base and if it is large enough one way it turns the device on.
It is two gain stages in series
grounded emitter with a C to B bias feedback resistor is not the most predictable way to bias a circuit
Quote from: R O Tiree on July 12, 2013, 06:39:57 PM
Quote from: Digital Larry on July 12, 2013, 03:07:44 PM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
@Ryan - Any insight into how Q2 is being biased?
There is certainly precedent for a "no-DC-bias" circuit, witness the Maestro FZ-1.
http://fuzzcentral.ssguitar.com/schematics/fz1.gif
That one starts off with a no-bias emitter follower! The transistor starts to conduct when the input voltage swings start to pull some current through the cap. But I would expect this to have a gated fuzz sound rather than a smooth overdrive that plays nicely with volume knob tricks.
Larry,
That FZ-1 is a positive GND, 3 x PNP circuit, so each stage is biased. This one is negative GND with 2 x NPN and Q2 has no resistors going to its base at all. Therefore, no base current, so Q2C locks up at +9V. In addition, the 1M from Q1C to Q1B is surely providing negative feedback, thus limiting the gain achievable.
I can see how this would work if there was another 1M, for example, from Q2C to Q2B but...
Larry's correct and Gus and R.O. are incorrect here.
The Striker circuit, and the FZ-1, bias in roughly the same manner. That is, leakage between collector and base will cause the transistor to conduct at least partially. In an NPN it is leakage from collector toward the base (conventional current flow) and with PNP it is leakage from base toward collector, but it's the same thing. The FZ-1 has some extra resistance in parallel with the base-emitter junction but the principle is exactly the same.
Got it. Thanks, earthtonesaudio. Still wondering why Ryan didn't answer the question (four times) though...
I gotta cut down on my (ab)use of parentheses as well...
Quote from: R O Tiree on July 12, 2013, 08:54:56 PM
Got it. Thanks, earthtonesaudio. Still wondering why Ryan didn't answer the question (four times) though...
Fair enough. :)
As the last several posts show, there are some folks who understand the way the electrons dance in these circuits (I'm not one of them :-[). Ryan is new enough that we don't have a good idea of his understanding of the theory. There are quite a few of us who come up with circuits without understanding all of the theory behind them. It could be that he isn't responding to some questions because he doesn't have answers, or it could be that he doesn't like your avatar ;). Whatever the case, it would help if he responded in some way, and if the answer is "I don't know", that's an acceptable answer. I can imagine that it's daunting to present your ideas to gurus and try to answer their questions. There's nothing wrong with gurus asking questions, I'm just saying that some folks may not know how to respond.
Something I've noticed, though, is that there are a bunch of folks who regularly have good, friendly conversations on the forum. However, there are a few (very few) of the same friendly, helpful people who get snippy or sarcastic very easily (I've never done that ::)). And then it doesn't help when the target of the insult fires zingers back (I've never done that either ::)). But as Aron said recently, this is supposed to be fun. There's no need to insult or return insults.
Anyway, I hope that someone gets some benefit from my rant. :)
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
That's not jimi's point, comedian. His point is that, once you have digitised and compressed an analogue signal to turn in into a lossy mp3, you can never get a truly accurate analogue signal from it. Similarly, if you remove all bar a vanishingly small amount of any part of the frequency spectrum from a signal, you can never get it all back into balance without compromises. Like noise, for example. You alluded to that yourself, when you raised the question of its practicality.
@Ryan - Any insight into how Q2 is being biased? I realise that it's only the third or fourth time that I have mentioned it, but I'm not alone in wanting to know.
oh, well that I would agree with. I didn't catch him talking about lossy digitization. A sine wave has an infinite number of variance, which is a lot more information than just 10 1's and 0's.
It partially has to do with DC current flowing to/in more than 1 area at once. We normally have a bias resistor in place, i have found no need for this because, the current reverse actually takes place within that capacitor, #1: the DC current has already been filtered by our gain stage capacitor, the current flowing out of our cap is now an AC generated SIGNAL. Without the presence of a dominating positive DC voltage, the current gracefully flows to its desired place where needed, if we were talking about a DC signal here then yes a bias resistor would be ideal. DC current flowing from one transistors collector into another transistors base is the perfect example of current shifting, the said DC current is attracted by the opposite in polarity sending said current flowing in the wrong direction. The AC entering Q2's base acts like a completely separate voltage from that of the collector or emitter, it has no polarity, it cannot be redirected by other current flow. there is no current shift because there is no attraction. rendering the bias resistor...........useless. try it. There is absolutely 0% difference with it in the circuit and without it in the circuit. ;)
it`s a DETECTOR!
it's a DETECTOR!
Kir... Kirrr.... Kirchhofff!!!
<'scuse me>
http://en.wikipedia.org/wiki/Kirchhoff's_circuit_laws
I'll modify my previous response a bit. The FZ-1 schematic shows a resistor from base to ground. If there is DC current flowing from collector to base, and there were no resistor from base to ground, then all collector-base current would be coming out of the emitter. In the case with a resistor, some of this current will go through that resistor instead and the transistor will be closer to being turned off. So it's more of an un-bias resistor in the FZ-1's case.
And I see that the AC176 has a nominal spec of Icbo of 35 uA.
http://pdf1.alldatasheet.net/datasheet-pdf/view/131458/ETC1/AC176.html
And here's a helpful article talking about leakage.
http://www.tpub.com/celec/48.htm
I'll be honest, I learned about transistors in engineering school 30 years ago, but I have reviewed many of my textbooks in the past few years and things which are common practice for stompbox circuits are NOT described, other than as situations to be avoided. Sending audio through a grounded emitter stage is certainly one of those because of the (naive) assumption that we don't want to distort the signal! :D :icon_biggrin: ::) So trying to understand how these circuits work bends my brain in a good way. I learned something here because I've never done a design where leakage was an important part of the intent.
My only conclusion here is that circuits which depend a lot on the characteristics of the transistor should try to identify what those things are because lots of times, especially with Germanium transistors, somebody's going to try to use a different one.
Quote from: earthtonesaudio on July 12, 2013, 07:58:32 PMLarry's correct and Gus and R.O. are incorrect here.
The Striker circuit, and the FZ-1, bias in roughly the same manner. That is, leakage between collector and base will cause the transistor to conduct at least partially. In an NPN it is leakage from collector toward the base (conventional current flow) and with PNP it is leakage from base toward collector, but it's the same thing. The FZ-1 has some extra resistance in parallel with the base-emitter junction but the principle is exactly the same.
Having slept on your reply, am I right in saying that, when I suggested a 1M from Q2C to Q2B at Reply#46, that would mimic the leakage from C to B in these old NPN trannies? High-ish leakage, you can get away without it under certain circumstances, low leakage, you'll probably need it?
Quote from: R O Tiree on July 13, 2013, 05:29:48 AM
Having slept on your reply, am I right in saying that, when I suggested a 1M from Q2C to Q2B at Reply#46, that would mimic the leakage from C to B in these old NPN trannies? High-ish leakage, you can get away without it under certain circumstances, low leakage, you'll probably need it?
Leakage is nearly-constant current source at micro-micro levels. It provides current, but at an incremental impedance that usually way higher than 1M. This was the substance of the Millenium Bypass, and my recently-posted idea to better-fake germaniums by adding a reverse biased germanium diode from collector to base of a silicon transistor to up the leakage to germanium levels.
You may need the current to bias, but its AC impedance is much higher than a resistor.
well played ryan!!! :icon_twisted:
make everybody figure it out, then explain yourself in a way even a dolt like me seems to understand!! :icon_eek:
i'm gonna sit right here....
(http://thumbs.dreamstime.com/z/office-worker-eating-popcorn-15568428.jpg)
:icon_mrgreen:
but still... WE NEED VIDEO (or it never happened!!) ;D
>I have reviewed many of my textbooks in the past few years and things which are common practice for stompbox circuits are NOT described, other than as situations to be avoided.
Good point. I've seen R.G. and PRR post similar comments. Many of our stompbox circuits take advantage of distortion by creating "forbidden" circuits.
Quote from: Thecomedian on July 13, 2013, 03:26:52 AM
Quote from: R O Tiree on July 12, 2013, 06:22:58 AM
That's not jimi's point, comedian. His point is that, once you have digitised and compressed an analogue signal to turn in into a lossy mp3, you can never get a truly accurate analogue signal from it. Similarly, if you remove all bar a vanishingly small amount of any part of the frequency spectrum from a signal, you can never get it all back into balance without compromises. Like noise, for example. You alluded to that yourself, when you raised the question of its practicality.
@Ryan - Any insight into how Q2 is being biased? I realise that it's only the third or fourth time that I have mentioned it, but I'm not alone in wanting to know.
oh, well that I would agree with. I didn't catch him talking about lossy digitization. A sine wave has an infinite number of variance, which is a lot more information than just 10 1's and 0's.
i was talking about throwing away frequencies in general. it would seem to my uneducated posterior, in theory and in limited practice, that once you attenuate frequencies to ground potential, they, and the area they once occupied would be gone, even from a sine wave. if you're merely "turning them down" with an adjustable device on a recorded (not live) media, yah, turn it back up and no harm no foul.
but in this case, the shape of the sine wave would seem to be altered permanently.... you would need something to restore them to reconstruct the original sine wave, right? "de-amplifying" on a LIVE instrument tho, i think may be different. with recorded media, yah, the original sine wave is gonna remain intact no matter how much you molest it with filtering or amplification or modulation. it can be restored immediately and at any time.
but wouldn't a live signal, being transient, be forever altered? i mean, obviously you can play the same thing the same way thru the same stuff again to get there, i guess... but once something is gone, to me, it would seem to be really hard to replace later. i mean, how can you boost stuff that you've already cut? wouldn't it be better not to cut it in the first place?
man i'm tieing myself into a granny here, i know it..
i've found sometimes that larger freekin' output caps can sound great. but i would think whatever attenuation occurred prior in the circuit could only be amplified...not restored.
so whatever shape or *distortion* of the original sine had been twisted into can be made louder.. but is it actually replacing the signal lost by storing the charge in the larger capacitor? or just amplifying the curve of the audio signal more?
i guess maybe i should fire up my venerable heathkit scope (my ancient tektronics one hit the floor and died...sad tragic story, snif..) and look..
sorry to hijack this thread, i don't think my understanding is up to snuff..
but yes, i was talking like mp3, where loss is VERY noticeable and pretty much impossible to repair.
Quote from: pinkjimiphoton on July 13, 2013, 10:22:26 AM
i was talking about throwing away frequencies in general. it would seem to my uneducated posterior, in theory and in limited practice, that once you attenuate frequencies to ground potential, they, and the area they once occupied would be gone, even from a sine wave. if you're merely "turning them down" with an adjustable device on a recorded (not live) media, yah, turn it back up and no harm no foul.
If you have a series capacitor going into a shunt resistor, that forms a "high-pass" filter. It's said that it "rolls off" frequencies below the corner frequency, e.g. 6dB/octave. But that doesn't mean it makes them disappear completely. If you have a resonant notch filter, like shall we say in a Varitone, you could achieve infinite attenuation if all of the components had zero resistance. In the real world this does not happen and so the notch might be deep, but not truly infinite. Also that deepest part of the notch is at just one frequency and anything to either side is not attenuated as much. But if you notch something out and then boost it to try to recover, you will be boosting a bunch of noise, as someone already mentioned. And while you can conceivably recover the amplitude of a signal via a boost following a cut, each step will introduce phase shift (frequency dependent time delay) that cannot be cancelled out. This is the "no free lunch" clause in action. Fortunately our ears are not very sensitive to phase shift, unless for example you mix the shifted signal back in with the dry signal which causes cancellations in amplitude.
About 10 years ago I studied the details of MP2 and MP3 compression as I was writing a program to transfer WAV backing tracks into a Korg PXR4 digital recorder (http://www.zzounds.com/item--KORPXR4). So I know a little about it but not a lot.
First your audio gets digitized and sliced into frames which are 1/75th of a second long. Then it goes through a frequency analysis stage called an FFT, so you have a bunch of numbers representing the spectrum of that 1/75th of a second of audio. Then the "perceptual encoder" goes through and says "hey there's a signal at -25 dB at 440 Hz, but it's right next to a signal at -3 dB at 469 Hz, therefore nobody's ever going to hear the 440, so let's throw it away". That's right, 440 Hz doesn't get encoded at all for that 1/75th of a second. This is in essence a computer program making decisions about what you can or cannot hear and completely throwing away the bits it doesn't think you'll be able to notice missing. Or, care enough to complain to iTunes that "Bon Jovi and Barbra Streisand cover the songs of Slayer" has something messed up and you want your money back.
On playback the audio signal is reconstructed frame by frame using frequency and phase information that was stored in the compressed file.
Later on, if you decide "ah, I never really liked 469 Hz anyway" and take pains to remove it, your 440 Hz isn't going to rise out of the muck because it simply isn't there (at least for the 1/75th of a second in question). The lower your bitrate, the more aggressive (and audible) these decisions to throw away parts of the spectrum become.
Thank ya Jimi! ;)
Quote from: R.G. on July 13, 2013, 09:17:37 AMLeakage is nearly-constant current source at micro-micro levels. It provides current, but at an incremental impedance that usually way higher than 1M. This was the substance of the Millenium Bypass, and my recently-posted idea to better-fake germaniums by adding a reverse biased germanium diode from collector to base of a silicon transistor to up the leakage to germanium levels.
You may need the current to bias, but its AC impedance is much higher than a resistor.
Now that makes perfect sense. Thanks, RG.
Some Schottkys have appreciable leakage as well... BAT46, perhaps? From the datasheet, at the voltages and temperatures we're typically going to be looking at, about 10-15µA? Might need 2 in parallel to get to the typical 30-odd µA quoted above by DigitalLarry.Edit: made a mistake on the exponents values. Ignore that bit about Schottkys