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Rob
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 Posted: Sun Aug 13, 2006 3:45 pm Post subject:
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| elektro80 wrote: | Trivia:
Most tube compressors are not tube compressors. |
To enlighten a bit on this, tetrode and pentode tubes can be used as VCA circuits 'out of the box': by applying a varying voltage to one of the grids the gain of the tube can be controlled. This effect can be used to design a compressor where it is indeed the tube that does the compression. Early broadcast compressors used this method to keep the modulation level of the AM signal relatively constant over a longer period of time. But this is technology from the fifties and the tubes need to run at quite high voltages. In the sixties it became common to use light dependent resistors (LDR's) to control the gain (due to their 'slowness') and in the seventies transistorized VCA elements were used as well (more precision and variable compression speed).
Tubes can be run at relatively low voltages like 60 Volts to be used in the small desktop boxes that are common today. But at these low voltages the tubes perform much poorer as with e.g. 300 Volts. A compressor using pentodes at 60 Volt would probably not make for a compressor with the sound and compression characteristics expected from a modern compressor. So, a modern tube compressor that is not bulky with heavy weight power transformers and sucks over 100W supply power can safely be considered highly suspicious as a 'tube compressor'.
Btw, it is the heating filament (running only at 6 Volts) that makes a tube glow, there is no need for an extra yellow LED to make it glow. A tube can even glow blue, which is a sign the tube will soon be departed.
When properly heated a tube can even be run at 12 Volt, though it will produce lots of distortion at such a voltage and be highly inefficient as a gain element. In such a case distortion can be the only goal of the tube and the compressor is most probably another part of the circuit.
So, real tube compressors using a certain property of vacuum tubes are definitely a possibility. But they will run at quite high voltages, don't expect such circuits to be powered from a simple 12V power adaptor.
Traditionally compressors had a different purpose as today, they were simply used to keep signals constant over longer periods of time, needed for mastering for vinyl and broadcast purposes. They were more like AGC circuits (AGC=automatic gain control). Today's compressors are also heavily used to 'shape transients', enhance reverb, side-chaining, ducking, etc. This requires a much broader control range for the time it takes before the compression kicks in and releases. E.g. I once worked in a little radio studio where a compressor with fixed 'attack/release' times of two minutes was used on the overall mix before it was fed to the AM transmitter. Not really the type of compressor one would use to give a percussive hit some extra punch. 
/Rob |
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elektro80
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| Posted: Sun Aug 13, 2006 4:19 pm Post subject:
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That´s right , Rob.
So what I tried to say was that most and probably all "affordable" tube compressors are not really traditional tube compressors.
The inexpensive budget "tube compressors" are mostly very basic analog solid state VCA designs ( or digital devices ( and still quite simple designs as such ) with a tube distortion circuit strapped on. Most of the designs I have seen have a crude mixer for blending the distortion with the "clean" signal ( -the output from the compressor - ) and many use at least one filter to clean up the distorted signal so you still have some sort of solid bass and some cleanish highs left.
Hmm, and i have seen some of these "tube compressors" using a solid state or a digital circuit for actually adding that magical tube warmth. The tubes are instead simply being used in the last gain stage.
That said, proper "vintage" tube compressors are an aquired taste. Some designs are highly musical and quite fun. Others are simply decent compressors. Personally I think there is no such thing as a magical warmth machine that will save any recording no matter what.
Anyways, I am not trying to police the warm zone here. I think it is a good idea to have a lot of outboard around just in case. Even disgustingly cheap gear can be useful. I am not saying that the budget "tube compressors" are bad for you. I am trying to explain in simple terms what they really are.
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elektro80
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| Posted: Sun Aug 13, 2006 4:26 pm Post subject:
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| Rob wrote: | | Btw, it is the heating filament (running only at 6 Volts) that makes a tube glow, there is no need for an extra yellow LED to make it glow. A tube can even glow blue, which is a sign the tube will soon be departed. |
I posted a pix of such a cute little analog led, the magical warmth maker, somewhere around here a year ago. I think Kassen got it too. 
It says a lot about how far some vendors will take the cosmetics.
Anyways, my own term for these devices are "Chanels". ( the perfume.. you know.. )
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DrJustice
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| Posted: Sun Aug 13, 2006 4:35 pm Post subject:
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Rob, I though we were done with this 
You do present a somewhat strange view of a few things here, and you are making several incorrect assertions and confusing the matter and our readers thoroughly. It would take me hours to try and put all of it right. Since it is becoming obvious that it wont matter what I say, I wont waste the effort, except for a few brief points on some of the more serious errors:
| Quote: | | ...as even a digitally anti-aliased signal is still a stream of samples that chained together forms a nice 'staircase' signal,... |
The "staircase view" is a very common misconception. Mathematically, samples does not form a staircase. Each sample is an infinitely narrow pulse. In a multirate system (one where several sample rates are at play, i.e. because of up and down -sampling) the correct representation of a sample as a pulse is paramount.
| Quote: | | Don't forget that in a stream of samples in the digital domain the sample rate itself is part of the frequency content of the signal. |
This is not the case. I'm not quite sure what you are thinking here.
| Quote: | | So, for generating sound digitally the sampling theorem might be expanded by the notion that to get really pure results the sample rate should theoretically be infinitely high. |
No. The sampling theorem holds. Remember that all systems are bandlimited even if the required and/or realized bandwidth may be high.
| Quote: | | So, while for recording audio the sample theorem is straightforward, for generating audio it simply isn't. It is nothing to argue about, it is just a fact of life. |
This is plainly wrong - sorry Rob.
Since you do talk about real reconstruction filters not being ideal (this is correct), I'll have to quote myself:
| Quote: | | ...the failure of physical implementations to be ideal... |
"Everybody" knows this. Note that this has only a minimal (non audible) impact in any decent sampled system. Real implementations use a sample rate that is a bit higher than twice the bandwidth to introduce a margin such that the problematic transition band of the anti-aliasing (recording/input) or reconstruction filter (playback/input) doesn't ruin the day.
As I said, I don't want to create any controversy. However your incorrect and highly confusing assertions are pretty damned controversial with respect to science and demands some sort of reply (you also say a few things that are correct). Regrettably I don't have the energy to put it all right; there is so much more that could be said. I'm now utterly sorry to have started this - I know from experience that in situations like this, the scientific facts has a tendency to be drowned out. Someday I may learn to keep my mouth shut 
Quote frankly, I find this particular exchange extremely unfunny.
DJ
P.S. Again, sorry for the OT.
P.P.S. If anybody is interested Iæm sure we could start a new thread to discuss signal processing, or even a subforum for discussion of it. It is clearly needed.
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elektro80
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| Posted: Sun Aug 13, 2006 4:40 pm Post subject:
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| Rob wrote: | Traditionally compressors had a different purpose as today, they were simply used to keep signals constant over longer periods of time, needed for mastering for vinyl and broadcast purposes. They were more like AGC circuits (AGC=automatic gain control). Today's compressors are also heavily used to 'shape transients', enhance reverb, side-chaining, ducking, etc. This requires a much broader control range for the time it takes before the compression kicks in and releases. E.g. I once worked in a little radio studio where a compressor with fixed 'attack/release' times of two minutes was used on the overall mix before it was fed to the AM transmitter. Not really the type of compressor one would use to give a percussive hit some extra punch.
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Good points.
( post edited - there is no god in the gaps anymore.. )
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paul e.
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| Posted: Sun Aug 13, 2006 5:12 pm Post subject:
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| DrJustice wrote: |
As I said, I don't want to create any controversy. However your incorrect and highly confusing assertions are pretty damned controversial with respect to science and demands some sort of reply (you also say a few things that are correct).
Quote frankly, I find this particular exchange extremely unfunny.
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keep in mind..music..and the perception of this..is not strictly scientific or mathematically provable...
theories can hold true on paper...but in practice...things are not as they always appear
somehow i think Rob was getting at some of these aspects and describing a 'phenomena' more than hard facts..
or maybe hinting toward the idea that no science perfectly describes something fully...
chaos theory?
... he seemed to suggest there was no 'pure' math on all of this anyway
it might not be as objective as it appears
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paul e.
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| Posted: Sun Aug 13, 2006 5:14 pm Post subject:
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| elektro80 wrote: |
God points. |
God has awarded Rob with 2 points?
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DrJustice
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| Posted: Sun Aug 13, 2006 5:21 pm Post subject:
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| paul e. wrote: | | theories can hold true on paper...but in practice...things are not as they always appear |
In this case both the theory and practice is very well known.
I really don't have the sap left to give a lengthy answer; I fear It will be futile anyway.
Peace, DJ out.
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paul e.
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| Posted: Sun Aug 13, 2006 5:23 pm Post subject:
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| DrJustice wrote: | | paul e. wrote: | | theories can hold true on paper...but in practice...things are not as they always appear |
In this case both the theory and practice is very well known.
-- |
not quite....it's a pursuit...not all wrapped up and sorted out
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elektro80
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| Posted: Sun Aug 13, 2006 5:27 pm Post subject:
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Paul, the sampling theorem still holds.
As for the "generating audio" issue, I am guessing we are looking at a discussion of digital synthesis and not at a digital recording / playback system.
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elektro80
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| Posted: Sun Aug 13, 2006 5:29 pm Post subject:
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| paul e. wrote: | | elektro80 wrote: |
God points. |
God has awarded Rob with 2 points? |
he has ? 
Well, you know.. this reminds me of the Euro Song Contest
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paul e.
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| Posted: Sun Aug 13, 2006 5:33 pm Post subject:
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| elektro80 wrote: | Paul, the sampling theorem still holds.
. |
i think kassen said elsewhere something interesting about this
it does ...i know..but..yeah...anyway...it's over my head anyway hehe
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Kassen
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| Posted: Sun Aug 13, 2006 6:02 pm Post subject:
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| paul e. wrote: |
... he seemed to suggest there was no 'pure' math on all of this anyway
it might not be as objective as it appears |
Well, there is perfectly good math on DSP, both on theoretically ideal processes and on the practice in real world chips with real world in and output formats.
How the real world stuff sounds with regard to pleasing tones is indeed not objective at all; very minor errors from a mathematical perspective might sound very unpleasant to some ears.
the math works (albeit in very hard ways), what does not work so well is proving some process sounds "right" on just mathematical grounds. there are also big gaps between the actual numbers and perspectiveson the numbers that people can inutitively relate to.
Let's not get caught up in those devides.
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paul e.
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| Posted: Sun Aug 13, 2006 6:08 pm Post subject:
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| Kassen wrote: | | paul e. wrote: |
... he seemed to suggest there was no 'pure' math on all of this anyway
it might not be as objective as it appears |
Well, there is perfectly good math on DSP, both on theoretically ideal processes and on the practice in real world chips with real world in and output formats.
How the real world stuff sounds with regard to pleasing tones is indeed not objective at all; very minor errors from a mathematical perspective might sound very unpleasant to some ears. |
right..what you said..yup....
but what do you mean by "Real" and "World"?
hehe no no no..just kidding hehe
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elektro80
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| Posted: Sun Aug 13, 2006 6:13 pm Post subject:
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Paul.. .. that is the modern understanding of the real world.. the duality.. real and world at the same time.
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paul e.
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| Posted: Sun Aug 13, 2006 6:14 pm Post subject:
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| elektro80 wrote: | | Paul.. .. that is the modern understanding of the real world.. the duality.. real and world at the same time. |
The Ministry of Philosophy issued this....and i missed the memo !?
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elektro80
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| Posted: Sun Aug 13, 2006 6:19 pm Post subject:
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| paul e. wrote: | | elektro80 wrote: | | Paul.. .. that is the modern understanding of the real world.. the duality.. real and world at the same time. |
The Ministry of Philosophy issued this....and i missed the memo !? |
Any black choppers coming your way?
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| Posted: Sun Aug 13, 2006 6:25 pm Post subject:
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Uh... re the "tube compressors".. Behringer also has a "tube EQ".
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SecretAsianMan
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| Posted: Sun Aug 13, 2006 6:53 pm Post subject:
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Very, very good information. Wow. I am amazed that this knowledge is being transferred to me for free. I think I need to click that "donate" button on the front page.
We've learned that "tube compressor" is a deceptive term and that such units typically do not use a tube to actually compress audio. However, the original poster and others have reported a success in using such units to process the G2's output. Thus, whether the units work as advertised is, in some ways, irrelevant; they still can have a positive musical effect.
Now, let me steer this thread back into practicality. I will pose a more direct question: If I were to buy only one piece of outboard equipment for my G2X, one piece to warm, tilt, or otherwise add cojones to the sound, what would that one unit be?
See, I'm a hobbyist, and I don't have the funds to keep a fleet of outboard stuff specifically suited for this or that sound. And I'm definitely on the fence about the sound of my G2X, like the original poster. |
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elektro80
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| Posted: Sun Aug 13, 2006 7:02 pm Post subject:
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| SecretAsianMan wrote: | | Now, let me steer this thread back into practicality. I will pose a more direct question: If I were to buy only one piece of outboard equipment for my G2X, one piece to warm, tilt, or otherwise add cojones to the sound, what would that one unit be? |
I need to think about that one. I don´t have the G2 myself, so I am a bit unsure about what to suggest.
How do you record your G2?
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Rob
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| Posted: Sun Aug 13, 2006 7:13 pm Post subject:
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| DrJustice wrote: | Rob, I though we were done with this
You do present a somewhat strange view of a few things here, and you are making several incorrect assertions and confusing the matter and our readers thoroughly. It would take me hours to try and put all of it right. Since it is becoming obvious that it wont matter what I say, I wont waste the effort, except for a few brief points on some of the more serious errors:
| Quote: | | ...as even a digitally anti-aliased signal is still a stream of samples that chained together forms a nice 'staircase' signal,... |
The "staircase view" is a very common misconception. Mathematically, samples does not form a staircase. Each sample is an infinitely narrow pulse. In a multirate system (one where several sample rates are at play, i.e. because of up and down -sampling) the correct representation of a sample as a pulse is paramount.
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1) According to the sampling theorem the measured value should be the average of a duration of 1/samplerate seconds
2) The measurement itself could probably be done in an infinitely short time, which is a technical matter
3) But even then the value is pending until the next measured value becomes available
Edit: This was a badly formulated sentence, probably due to the haste in which the reply was written. The meaning I had in mind here is that after a meaurement is done the measured value is the one to use until another measurement has completed. The word pending should have been the words considered valid. The word pending slipped into the sentence because I had in mind that what happens in between two measurements is undecided until the next measurement becomes avalailable, and taking into account that the actual act of the previous measurement is completed and final and there is no uncertainty about that actual measured value, but there is uncertainty about what happens in the analogue signal under measurement until the current measurement has completed. I did not mean to say that I had in mind to constantly adjust the previous measurement over the time until the next measurement becomes available. I thought it clear that everybody would understand that a measurement by an AD converter is a final act when completed, despite the fact that the analogue signal under measurement is still continuously changing.
4) So, the pulses are not infinitely narrow but have a finite duration of 1/samplerate seconds
5) When representing the stream of samples for the times they are pending graphically on a timescale you will notice something that looks jaggy
Edit: again the word pending should be replaced with considered valid
6) Any pulse with an infinitely narrow width that is not of infinite 'height' has infinitely little 'energy', and so will not be able to excite any real world reconstruction filter into anything other than silence.
Edit: a realworld pulse is mentioned here, e.g. a DA converter converts the digital numbers into currents. Whenever the number changes the current will change and the delta of this change will excite the 'ideal analogue filter' to recreate the continuous analogue waveform. The delta of change in current can be considered and actual pulse where the energy depends on the delta and the duration before a new change occurs. The duration is finite and is quite important to define the released energy. In this way it acts like a vector that pushes the output voltage from one level to another over a given amount of time along the vector.
A one padded with zeroes in a list in the digital domain, which these days is also named a pulse, is not able to produce any effect on an actual analogue voltage without a DA converter, which is in essence an anlog electronic circuit and not part of the digital domain. Such a one padded with zeroes is a virtual pulse, but in fact is more like a potential. The reason why it is attributed with an infinitely narrow with is not because it has an actual infinitely narrow witdh, in fact nothing concrete can be said about the width as it is only a number in a list. But when it acts like a pulse the energy it can release is defined by the number ánd the samplerate. But as the samplerate cannot be deduced from the single number stored in the list, or the fact that it is padded with zeroes, the infilitely narrow width is just an assumption. Basically a redundant assumption, as when the samplerate becomes known by the programmer it becomes clear how much actual potential energy can be released during one sample period and for this equasion a width of one sample period is taken.
This infinitely narrow width thing reminds me of something from my childhood. Someone, I guess an uncle as I don't remember who it was, had given me two small metal block with perfectly flat polished surfaces and made from some light alloy. When put on a table on top of each other and lifting the top one the bottom one would stick because of the vacuum. Of course I first had to rub them thoroughly on my trousers to clean them to get a good vacuum. This never ceased to fascinate me. And I remember very well that I always wondered about the space in between the two blocks, realising it was a perfect two dimensional plane without thickness. And so I wondered if that plane really existed or not, or better if a plane that actually lacks the third dimension because of infinite thinness can actually be perceived in our threedimensional real world (I do not mean a surface here but an actual 'stand-alone' 2D plane). I could never decide on a yes or a no. The virtual world in the digital domain is much more forgiving here, any set of nine numbers can simply be called a plane in a 3D space. Still, its only nine numbers in a list and nothing else.
My apologies for the confusion created by my bad choice of words. I hope my edits set it right.
| DrJustice wrote: |
| Quote: | | Don't forget that in a stream of samples in the digital domain the sample rate itself is part of the frequency content of the signal. |
This is not the case. I'm not quite sure what you are thinking here.
| Quote: | | So, for generating sound digitally the sampling theorem might be expanded by the notion that to get really pure results the sample rate should theoretically be infinitely high. |
No. The sampling theorem holds. Remember that all systems are bandlimited even if the required and/or realized bandwidth may be high.
|
Yes, but this does not necessarily apply to whatever is represented by the samples. E.g. multiplying several digitally represented waves of different frequencies can mathematically produce partials higher as the sample rate of the digital representation. Because it is waves that are multiplied, waves which just happen to be represented in a limited digital form. Whatever is added to the waves by the limits of the digital representation adds into the wave equasions as partials (including the samplerate, as if not there would not be such a thing as aliasing at all). It is only the 'resampling' of the mathematical results to the samplerate that will reflect these partials back to frequencies lower as the samplerate. And this is the crux of the matter that Tim referred to. This just puts the idea of bandwidth in a whole different light, not? In fact, the bandwidth is defined by the processing algorithm and not necessarily by the samplerate.
| DrJustice wrote: |
| Quote: | | So, while for recording audio the sample theorem is straightforward, for generating audio it simply isn't. It is nothing to argue about, it is just a fact of life. |
This is plainly wrong - sorry Rob.
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If you still say so.
| DrJustice wrote: |
Since you do talk about real reconstruction filters not being ideal (this is correct), I'll have to quote myself:
| Quote: | | ...the failure of physical implementations to be ideal... |
"Everybody" knows this. Note that this has only a minimal (non audible) impact in any decent sampled system. Real implementations use a sample rate that is a bit higher than twice the bandwidth to introduce a margin such that the problematic transition band of the anti-aliasing (recording/input) or reconstruction filter (playback/input) doesn't ruin the day.
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Uh, there is probably nothing to anti-alias at the (recording/input) stage before the AD circuitry, but one might want to prevent partials above the Nyquist frequency to be filtered away to not ruin the day. And the input filter/S&H combination should also make sure that the measurements accurately represent the average over that 1/samplerate time period.
| DrJustice wrote: |
As I said, I don't want to create any controversy. However your incorrect and highly confusing assertions are pretty damned controversial with respect to science and demands some sort of reply (you also say a few things that are correct). Regrettably I don't have the energy to put it all right; there is so much more that could be said. I'm now utterly sorry to have started this - I know from experience that in situations like this, the scientific facts has a tendency to be drowned out. Someday I may learn to keep my mouth shut
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There may be more science out there that you are aware of. A real scientist keeps an open mind, many real cute scientific theorems have later proved to be inadequate when the cases they were applied to became more complex. The digital domain is quite a bizarre world, as its not really about the bits themselves but about what the bits represent. The technical processing works on the bits only, but what is represented by the bits should be considered with the math behind what is represented and not by the math of bits. So, if waveforms are represented the math on waveforms applies, and that math is probably only practically bandlimited by the time lapsed since the big bang.
| DrJustice wrote: |
Quote frankly, I find this particular exchange extremely unfunny.
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Why? We only exchange arguments and points of view, don't we? What's wrong with that? Everyone is free to draw their own conclusions, pretty cool I think.
Last edited by Rob on Mon Aug 14, 2006 5:47 pm; edited 2 times in total |
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SecretAsianMan
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| Posted: Sun Aug 13, 2006 7:27 pm Post subject:
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| elektro80 wrote: | | How do you record your G2? |
Into a PC running Ableton Live. But I'd like a hardware unit that could be used sans computer. |
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Rob
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| Posted: Sun Aug 13, 2006 7:31 pm Post subject:
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| SecretAsianMan wrote: | Now, let me steer this thread back into practicality. I will pose a more direct question: If I were to buy only one piece of outboard equipment for my G2X, one piece to warm, tilt, or otherwise add cojones to the sound, what would that one unit be?
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My answer to that is pretty disappointing: the unit that has the effect that yóu like most.
It is very subjective and might depend a lot on the genre of music you're making. What really works for someone might not work for you at all, and vice versa.
If you listen really carefully and as objectively as possible to your recordings and mixes you might be able to deduce the sort of thing that might further improve your tracks.
Still, there are persistent rumours about some obscure device that is only known to a small and closed elite society that is sworn to total secrecy. |
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Kassen
Janitor
Joined: Jul 06, 2004
Posts: 7678
Location: The Hague, NL
G2 patch files: 3
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| Posted: Sun Aug 13, 2006 7:35 pm Post subject:
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| Rob wrote: | | And the input filter/S&H combination should also make sure that the measurements accurately represent the average over that 1/samplerate time period. |
Hmmm I'm not at all sure that bit is right yet you mention it twice. Do you have a source for this? I have a naging suspicion that this would come down to input-waveform-dependant sampling clock jitter but I might be wrong?
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Kassen |
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Afro88
Joined: Jun 20, 2004
Posts: 701
Location: Brisbane, Australia
Audio files: 12
G2 patch files: 79
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| Posted: Sun Aug 13, 2006 7:49 pm Post subject:
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| SecretAsianMan wrote: | | Very, very good information. Wow. I am amazed that this knowledge is being transferred to me for free. |
I completely agree. Even if things get a little heated, there is some great information contained in these posts. Thanks to everyone who has participated.
| SecretAsianMan wrote: | | Into a PC running Ableton Live. But I'd like a hardware unit that could be used sans computer. |
I'm really hanging out for the Focusrite Liquidmix processor. It might not be sans computer, but at least the processing is. Dynamic convolution modelled vintage compressors and eq's, 32 channels worth, for under $1000. Apparently it sounds pretty damn nice!
Also hanging out for this obscure device Rob mentions |
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