Does analog move more air. . . ?

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When I think of all the trouble I go to in live work to make things sound "the same but louder". What a waste of time--all I need is to run things through a tape recorder to make it better than live!

You see flutter is soul because it wiggles. If you watch Elvis on digital TV, his hips don't move at all. It's a bit creepy.
 
Have you actually read your "great link"?

Actually it IS great. It gives details of all the things wrong with analogue recording that contribute to the "analogue warmth" that some people like the sound of.

Nowhere does it say that analogue is more accurate than digital--because it's not. The warmth comes from a huge mix of fortuitous flaws. As I said about 20 pages ago, these flaws can produce an effect that is pleasing to the ear (just like grain in a black and white photo can look good) but none of that means analogue is more accurate.

By all means say that you like the sound of vinyl recordings. So do I. Say that you perceive a "certain something" in analogue that you miss in digital and that's hard to argue with too. Even say you like the tactile experience of recording with spinning reels and bouncing meters. Nobody here can argue with that.

But saying that analogue--even with a well maintained and expensive reel to reel--is a more accurate representation of the original signal is simply not true and weakens any argument.

Of course I read it. The problem with the digital is the A/D and D/A converters doing what they do, "Converting". Is that really "Recording"?

VP
 
If you run the signal to you mixer, you can send it to the digital recorder from one aux and send it to the other input on the scope from another aux.

The DA really doesn't reconstruct anything.

If you record a bass guitar hitting a low E and filter out everything above 50hz, you will end up with a sine wave at 41 Hz. That's how it works, you aren't constructing a sinewave, it's there the whole time.

Just like with the analog recorders, you are modulating the bias frequency by the frequency of the sound you are recording. When you filter out the bias frequency, you are left with just the sound.

I can figure out how to do the experiment. What I want is to see input and output signals simultaneously, impossible with a digital system.

The A/D D/A converters do construct the points between the sample.

Sure the digital can "Convert" a 41.5hz signal with accuracy, it is the high frequencies it makes a mess of.

Digital doesnt record "Just Like" Analog recorders, I am quite sure of that.

VP
 
By that standard, anything involving microphones and amplifiers is not "recording" since changing the sound pressure waves to electricity is also a conversion. A varying electrical signal is no more sound than a series of 0s and 1s.
 
Actually it is not; according to VP recordings sound better than live concerts, because the tape recorders have soul.

I tried to make a point that the lowest fundamental tone of that organ is 16Hz, which can be recorded accurately on a digital recording but not on tape. In either case it doesn't really matter because no playback system that anyone is likely to own can reproduce the sound of a 32' pipe, but that point fell on deaf but golden ears.

So we have learned that tape recordings sound better than real life and digital recordings sound worse, because VP said so.

Apparently you are into a different kind of music. I record many types of music but a live symphony is not one. I am inspired by the great recorded albums of the past. "The Dark Side Of The Moon" for example. I think the completed album would sound better than the actual seperate tracks that make up the album. Being in those tracking rooms at the time might not give me "Goosebumps".

VP
 
By that standard, anything involving microphones and amplifiers is not "recording" since changing the sound pressure waves to electricity is also a conversion. A varying electrical signal is no more sound than a series of 0s and 1s.

Not an Analog signal, no way! Apparently you dont understand Electricity at all.
Analog signal - Wikipedia, the free encyclopedia
Quote taken from the above link "An analog signal has a theoretically infinite resolution"
VP
 
Dude, we've been over this many times--nothing in the physical world has infinite resolution. No analog amplifier has ever had infinite resolution.

Why don't you use that scope of yours to measure the bandwidth of, say, the 12AX7 stage in a tube amp? Is it in the terahertz range? Gigahertz? Well? How infinite is it? I'll give you a hint and tell you it partially depends on the circuit gain and source impedance of your test signal because of Miller capacitance, but the tube has its own limit as well.

Can you accept that as a law of physics that no amplifier has infinite bandwidth, which is required for infinite resolution?

Can you further accept that because of thermal noise (that pesky Johnson again) that infinite dynamic range is impossible because you cannot cool a circuit to absolute zero? And even if you did get cold enough to make a difference in thermal noise, you'd still have noise in the active components of your circuit?

So we can clearly understand that no analog circuit--never mind a recording, I am just talking about amplifiers--can come remotely close to infinite resolution. Jim Williams tries pretty hard; he uses just about the fastest and lowest noise audio circuits of anybody I know of. Don't ask him what he thinks about tape though . . . but I think he claims his circuits at about 150dB dynamic range and 200kHz bandwidth at 80dB gain or so. Impressive, but not infinite.

Next, let's get back to talking about how the bias signal in a tape recorder serves as a bandwidth limit--actually half the bandwidth of the bias frequency, otherwise you would get intermodulation distortion into the audio bandwidth.

You were going to measure that, remember?
 
Dude, we've been over this many times--nothing in the physical world has infinite resolution. No analog amplifier has ever had infinite resolution.

Why don't you use that scope of yours to measure the bandwidth of, say, the 12AX7 stage in a tube amp? Is it in the terahertz range? Gigahertz? Well? How infinite is it? I'll give you a hint and tell you it partially depends on the circuit gain and source impedance of your test signal because of Miller capacitance, but the tube has its own limit as well.

Can you accept that as a law of physics that no amplifier has infinite bandwidth, which is required for infinite resolution?

Can you further accept that because of thermal noise (that pesky Johnson again) that infinite dynamic range is impossible because you cannot cool a circuit to absolute zero? And even if you did get cold enough to make a difference in thermal noise, you'd still have noise in the active components of your circuit?

So we can clearly understand that no analog circuit--never mind a recording, I am just talking about amplifiers--can come remotely close to infinite resolution. Jim Williams tries pretty hard; he uses just about the fastest and lowest noise audio circuits of anybody I know of. Don't ask him what he thinks about tape though . . . but I think he claims his circuits at about 150dB dynamic range and 200kHz bandwidth at 80dB gain or so. Impressive, but not infinite.

Next, let's get back to talking about how the bias signal in a tape recorder serves as a bandwidth limit--actually half the bandwidth of the bias frequency, otherwise you would get intermodulation distortion into the audio bandwidth.

You were going to measure that, remember?

Sorry for you to have wasted all that typing, but if you reread the quote again: "An analog signal has a theoretically infinite resolution"

VP
 
Apparently you are into a different kind of music. I record many types of music but a live symphony is not one. I am inspired by the great recorded albums of the past. "The Dark Side Of The Moon" for example. I think the completed album would sound better than the actual seperate tracks that make up the album. Being in those tracking rooms at the time might not give me "Goosebumps".

VP

DSOTM is kind of uneven. Yeah, I used to own it on vinyl, that was required during the 700 years it was on the charts. Not bad if you do some drugs. I liked "Time" and "Brain Damage" and that track where the chick screams a lot, we used to joke that was each member of Floyd taking a turn, but whatever . . .

I listen to lots of kinds of music, prefer jazz, gospel, and some classical. I listened to lots of classical about 20 years ago and got burned out a bit, but I still like Bach and some of the modern stuff like Xenakis, Cage, and some others. I am mostly done listening to rock music because it isn't as creative as it used to be. I like to listen to sounds that are very outside, mostly because it's something I might not have already heard, but then there are a lot of modernist types that are trying a bit too hard to be different. That's why I like Cage's philosophy the best:

 
Sorry for you to have wasted all that typing, but if you reread the quote again: "An analog signal has a theoretically infinite resolution"

VP

I understand that. And a digitally-generated signal can also have a theoretically infinite resolution. But then we have to go and try to listen to those theoretically perfect sounds, and all kinds of bad things happen due to physics.
 
No, and a theoretical analog signal is not a real signal, it's a theory. You seem oddly opposed to some theories (like Nyquist-Johnson) but perfectly happy with other theories . . . but all of those theories are based on the same math.
 
No, and a theoretical analog signal is not a real signal, it's a theory. You seem oddly opposed to some theories (like Nyquist-Johnson) but perfectly happy with other theories . . . but all of those theories are based on the same math.

It is amazing how that "Theory" sounds so good to so many people.

VP
 
I can figure out how to do the experiment. What I want is to see input and output signals simultaneously, impossible with a digital system.
Try a "Y" cable. Or are you complaining about the 4ms delay between the input and the output? I'm just not sure what the problem you are having is.

The A/D D/A converters do construct the points between the sample.

Sure the digital can "Convert" a 41.5hz signal with accuracy, it is the high frequencies it makes a mess of.
You misunderstood what I was saying. The bass guitar thing was an analogy to help explain the effect of filtering and how, once you filter out frequencies higher than a fundamental, you are left with a sine wave. The sine wave was always there, there was no reconstruction needed.

Digital doesn't record "Just Like" Analog recorders, I am quite sure of that.
Again, that was an extension of the bass guitar explanation and didn't say that digital recorders record like analog recorders. It was pointing out how, once the high frequencies are filtered out, you are left with something that was always there to begin with.
 
It is amazing how that "Theory" sounds so good to so many people.

VP

Not really, a lot of people like Justin Bieber, the main difference there is none of them will admit it :)

Good sound often does not equate to good circuit performance. Ignoring recording entirely we could look at the entire field of guitar electronics, which is dedicated to the principle that the signal of an unloaded pickup sounds so terrible that it needs to be quickly bandwidth-limited and distorted.
 
Can someone please clarify what everyone is debating at this point?

A couple of things worth mentioning:

Simply running a digital signal 'through tape' is only going to impart artifacts. If you're attempting to conduct a test using a digital signal to begin with, the analog is set up for failure because it will simply impart artifacts ('flaws').

This is very different from recording an analog source to a tape machine. If an analog source is recorded to tape, then transferred to digital, the digital will then impart it's own flaws and artifacts.

So the only fair test would be to use an analog signal and record it simultaneously to both the digital and analog recorders and compare the results.

That said, this test would ultimately provide no useful information because 'accurate' is always going to be a subjective term because we're dealing with human hearing ... it can mean many different things. I mentioned on the first page of this thread I agree digital sounds more accurate on a superficial level, but a vinyl record dubbed from a master tape (without digital delay) sounds more 'present'. I believe this presence is the thing that digital can't quite 'recreate', if you will.

There is no perfect medium ... every medium has it's own flaws. I get the sense that some people view digital as 'perfect' and analog as 'flawed' and it's these comments that I take issue with ... especially considering this in an analog forum.
 
Yeah, it's kind of making a ruler with a ruler, eh? But as with any scientific experiment we only need to state our error of measurement. Do we need to machine a precision part, or build a house? Because we need a lot less precision to build the house.

My position is that recording audio signals is kinda like building a house--it doesn't require a lot of precision in terms of needing lots of bandwidth. Either system is limited to less than 100kHz bandwidth by the constraints of its method of operation.

Note that if we were happy with limited dynamic range--say the unweighted 60dB typical of a tape recording--we could choose from plenty of digital converters that operate well into the mHz range. These are what digital o-scopes are made with. But that is useless bandwidth for audio, so we don't do that--we'd rather have the dynamic range (although feel free to discuss taking PCM4222's 6/6mHz output and writing your own algo to generate said dynamic range/bandwidth; it could be done).

Anyway, if we generate a sine wave using an analog oscillator circuit then we still need a measure of quality of that sine wave . . . so we need a ruler. Most people in audioland ultimately refer back to something like Audio Precision for that, but once a given bit of kit is calibrated we can use it to measure other stuff given that calibrated bit of kit's limit of resolution. So long as that resolution sufficiently exceeds the unit under test, we are fine.

So if we use a digital signal generator to create a sine wave and a digital converter to measure that sine wave, we are limited to the combined resolution of both. That's why the first step is always the loop test, we need to know the limit of measurement resolution. If we then place the unit under test inside that loop and we get the same result, then we merely know that the unit under test measures better than the loop (at least 3dB or so better, but we don't know how much better).

On the other hand if it measures worse then we know it is worse. Once a digital signal leaves digiland and passes out a DAC, it is now a continuous, infinite (at least as VP believes) signal. It isn't going to be a pure, perfect sine wave anymore . . . it will be a combination of many pure, perfect sine waves of varying frequency and amplitude (Fourier, again), which are the intended signal plus distortions (but in a good converter, very small distortions)! But it's analog, and any receiving analog device doesn't magically know that it originated as a digital signal.

It's like if you are cutting a 2x4 with a ruler you made in reference to a tape measure and as a result your ruler only has +/-1/8" accuracy; you can't get 1/16" tolerance out of it, but if your cut is 1" off you can tell that.
 
Oh, and no digital signal is perfect! At least once it leaves a DAC! Every digital signal is limited in bandwidth and in dynamic range by its sample rate and bit depth. Also, since filter algos are limited by DSP resources they are imperfect approximations of the otherwise perfect math that is required of the filter. So we get a bit of in-band attenuation of the sample rate is not >>2x bandwidth, and we get some aliasing products because we do not have infinite attenuation of out-of-band signal.

On top of that we have the analog circuit errors imposed on our signal: noise, distortion due to amplifier bandwidth limitation, and jitter in our timing circuit.

So those are the main imperfections of digital recording. The good news is we can measure all of them (which I did on this thread for a particular converter), and they are really small in a good bit of kit, well underneath the noise floor of a comparable tape recorder.

That is to say if you took at analog noise generator circuit after the output of a good DAC and mixed them such that the resulting noise completely swamped the distortion products, you'd still have more dynamic range than a tape recorder has and you wouldn't be able to hear any of the distortion (I think it's very hard to hear anyway, again from a *good* converter).

The aliasing and jitter products from my main converters are at -130dBFS; add a noise signal 10dB louder (a bit more than enough), that's -120dBFS, or integrated about 80dB dynamic range.

The analog THD is a bit higher, but that's sauce for the gander . . .
 
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