[ev13wt]

I have a few of those original Sheffield direct-to-disc records. They are flat-out spectacular! Some of the best recordings ever made. But I digitized them, and burned them to CD, even AAC files. Hate to break it...still every bit as spectacular, can't possibly tell the digits from the analog original. All the original glory, "feel of the room", pops, ticks, surface noise, all of it is preserved perfectly at 16/44.1.

I also have this, also spectacular in many of the same ways, except it's in 5.1! Down-sampling it to 16/44.1 didn't change a single thing. Recording to analog tape...um...that made it worse, took me right out of the "feel of the room". Sorry.

Aren't we having anecdotal fun now!

Nice one

I can relate. Done both. I still record digital playlists to my reel to reel for fun.
There is nothing like watching those reels spin!

 

[castleofargh]

I am not saying you will get these 3uS impulses from any real world audio source, agreed. And yes, you'd need pretty fast and well-damped electronics to produce them cleanly. But these days, that's not too hard. A BUF634 would probably do a reasonable job in high current mode, see fig 18 on the data sheet.

A microphone fed with an acoustic impulse cannot ever give you a sinc x function. There is no pre-ringing. Pre-ringing needs a through time delay in the filter, and an infinite one if you are going to get a true sinc x function out. A digital filter can only approximate it.

record and reconstruct a signal that we don't hear and won't find in music seems like a totally irrelevant thing to do and it's hard to see how it would make music sound any different.

agreed on analog and post ringing. almost everything in real life applies some variation of a low pass on the signal. the frequency and the attenuation level changes but that's all. air, transducers, a cable, the ear... a digital signal needs band limiting to work but it's not like band limiting didn't exist in analog media or in life. so I don't get why it would be important to remove stuff we can't hear that are being removed/attenuated everywhere anyway. as for ringing, again it's obvious that we'll have some for those various reasons so just hating on ringing seems like an unrealistic expectation about audio.
now if your big issue is pre ringing, we have pretty amazing digital stuff and DAC filters that can simulate analog filter really well(perhaps with less noise/disto, every simulation has its limit ^_^). so that makes digital vs analog a lot less relevant IMO. aside from how digital can do it all better but it doesn't matter as we still have to go analog somewhere and often came from analog.

the way I see it, anything happening at frequencies and amplitudes above what a mic records is irrelevant if not simply undesirable. and anything happening above what my hear can notice is irrelevant because it will not get past it. band limiting can easily be and usually is implemented above those limits so I don't get what the issue really is at an audible level. if 44.1khz was such an issue because it's too close from home, how come we can totally find DACs where we will fail a 44.1 vs highres blind test? such a thing shouldn't be possible if those extra freqs or the ultrasonic ringing meant something to our ears. and when the filter rolls off the audible range or when we get crazy aliasing because the designer didn't get digital audio theorem, when those stuff become audible, they're easy to get in blind test, so it's obviously a threshold thing. we can hear it, then past a given frequency/amplitude, we can't and it doesn't matter.
be it theoretical or practical, I came to think that ringing outside my audible range has little to do with me and what I hear. if band limiting was creating massive ringing at 3khz, I wouldn't say those stuff. and if music had frequent 125dB 23khz signals, maybe I would have another approach. and indeed if my favorite bands were making Dirac impulses all day long, I would then agree with the weirdo link you gave. but all those are science fiction scenarios as far as I know, and not relevant to music and the average human ear.

 

[castleofargh]

can I just say how nice it is to argue stuff without having to explain what a sound wave is, what is Ohm's law, or how to think in a rational way.

quasi-modo ON:
if you guys feel that there is more bring to the damping topic, let me know and I'll ask mother to move our off topic lounge somewhere else. I'm waiting on some cable for my mic(I have one... somewhere, but where????), and plan to test poorly a few things with adding resistors to the output in REW. I do that all the time with IEMs, but I never really tried to look into it much with fullsize dynamic headphones, and most of all I never tried values beyond 100ohm. I tend to think that a 10ohm amp output is way too much(again, IEM world).
I can already see all the reasons why my results will be irrelevant (not accurate enough with my super lowfi rig, resistors too cheap that change significantly when getting heated by a strong signal, the human operator...). but I'll try next week when I get my cable (or before if I find the one I have...), and let you know about how I failed.

 

[pinnahertz]

There is nothing like watching those reels spin!

Just building the tangent....Well....my ultimate in watching reels spin was working for an FM station that used an automation system (IGM-500) comprised of 3 reel decks which could handle 14" reels...that's not a typo! We got music from a service on 14" reels. Watching the system play one track from one deck, then dead-roll the next in advance for a perfect segue...now That's watching reels spin! We quit using the 14" reels when we switched to a different music provider because of quality issues, and ended up with 4 decks, two of the old Scully 270s with 14" reel capacity, and two newer decks. Found a VERY old pic of the actual beast...the 14" reel decks have the supply reel below the takeup, and make 10" reels look very small. When high-winding 14" reels the things would spin up and sound like they would saw your hand off, and indeed would! I once calculated the top speed of the reel flange at around 90mph. All 4 decks in the photo are play only. The big ones are Scully 270, the small ones are cheapo ITC 750. That's a 48 "tray" tape cartridge player in the upper left, 48 heads, 48 pinch rollers, 4 very long capstans. Odd beast, the advantage was full random access of any tray at any time, even all at once.

The system had a 4 hour "walk-away" time with 10" reels, 6 hours with the 14" reels. There were different categories of music on each real, a sequencer controlled the "format", which was 14 minute music blocks.

I have to say, watching and listening to that thing was a rare treat. That's how we automated radio in the early 1970s. Nothing to watch today, it's all in a PC.

 

[gregorio]

But don't think you are getting exactly the original analogue signal back.

Why not? What in the audible range am I not going to get back? I'm certainly going to get back significantly more of the original analogue signal if I record it with digital than with analogue, even the best high speed tape!

There is a loss of air, energy and life with the digital which sounds sterile; quite obvious. All subjective terms.

I presume by air you're talking about high frequency noise. With digital there is NO loss of anything, with analogue there is the addition of hiss, high freq noise and yes, it is "quite obvious". What you subjectively like then is added noise which wasn't in the "original signal", which is fine if your preference is for noisy, lower fidelity analogue reproduction. However, you can't then say that digital is loosing "energy" or "life" (because it's not loosing anything) and you can't say your preference for analogue is not due to it's distortions/deficiencies!

I read somewhere that an Abbey Rd engineer said digital does not capture the acoustic feel of a room well.

Which engineer? I've worked on numerous occasions at Abbey Road, spoken at length with many of the engineers, one of my ex-students is an engineer there and I've never heard one of them state that analogue captures acoustics better than digital. Quite the opposite in fact, Abbey Road were an early adopter of digital and were the first studio to take delivery of the first Neve digital desk.

You could argue these effects are too small to be of consequence, but they are there.

What do you mean "you could argue"? It's outside the range of human hearing, low amplitude and has never been reliably demonstrated to be audible with music, so how could you rationally argue anything else?

G

 

[71 dB]

I am not saying you will get these 3uS impulses from any real world audio source, agreed. And yes, you'd need pretty fast and well-damped electronics to produce them cleanly. But these days, that's not too hard. A BUF634 would probably do a reasonable job in high current mode, see fig 18 on the data sheet.

3 µs pulses don't belong to music. I'm not an expert, but I guess you get such EMPs when a nuclear bomb goes off.

A microphone fed with an acoustic impulse cannot ever give you a sinc x function. There is no pre-ringing. Pre-ringing needs a through time delay in the filter, and an infinite one if you are going to get a true sinc x function out. A digital filter can only approximate it.

My mistake, of course not pre-ringing since microphones are pretty much minimum phase systems, not linear phase, but you have post-ringing nevertheless.

Yes, for a sine wave. And that's precisely the problem.

It's not a problem as long as our signal doesn't contain frequencies above nyquiest frequency.

Yes, agreed. Magnetic hysteresis for one. But it is timewise continuous and causal. FIR digital is neither; filters are arbitrarily curtailed in the time domain. You could argue these effects are too small to be of consequence, but they are there.

Finite FIR filter is windowed version of the infinite sinc. I don't see how that breaks causality. The spectrum is the product of the window spectrum and sinc spectrum. But if think digital is so inferior you can stay with your analog tapes.

 

[SimonPac]

There's a lot to tackle this afternoon! I agree if people want to persist with this maybe it should move to another thread. Basically you guys seem to think digital is essentially perfect and I don't. Objectively and subjectively. I may change my mind with more experience but I doubt it. I am not trying to force my opinion on you. Obviously you are died in the wool, digital is more or less perfect, people. I don't want to ruin your enjoyment, I just don't agree. If you think AAC is perfect, or that high bit depth or sample rate is irrelevant, again, I simply don't agree. They're all very good though.

No tone controls in the tape demo? It was tape, right? Um...EQ? Calibrated to...um...exactly which test tape? SO little understanding of tape, the variables...etc.

The point here is that the ADC-DAC version was fed from the self-same master tape copy, already equ'ed to nominally flat in the tape machine and then out to the ADC. I agree there is only potential for things to get worse, but worse they did seem to get.

Finite FIR filter is windowed version of the infinite sinc. I don't see how that breaks causality. The spectrum is the product of the window spectrum and sinc spectrum. But if think digital is so inferior you can stay with your analog tapes.

Yes a time windowed FIR will be causal. But it is an approximation as you'd concede. Chord make a great deal of the FIR tap length issue and rightly so IMO. My favourite Nyquist filter algorithms (very limited test set) are the PMD ones. I actually seem to prefer fast roll-off to some of the 'clever' stuff. But my point though is 1) that the sync x function is not paralleled in any real world acoustic effect or even practical analogue filter. Here are or step responses:

http://www.analog.com/media/en/training-seminars/design-handbooks/Basic-Linear-Design/Chapter8.pdf -see p25 and 31, 32 and others later.

No pre-ring on any of them. Later on there is an analogue anti-alias filter design presented for CD btw. Note this particular discussion regarding time symmetric impulse response is an issue with filter implementation as normally (read just about always) done in digital audio, not digital audio per se. 2) IMO digital improves as you go beyond 16 bit (I guess you think HDCD was a waste of time. Microsoft didn't, but customers, on the whole, did). and 44.1/48 ks/sec. And even if you are a little way below the noise floor (an analogue problem..), which would be an already excellent -96dB for 16-bit.

I actually don't have any analogue tapes! I don't even have any vinyl any more! Just CD, SACD, DVDA and downloads! I have Wolfson 24-bit and BB PCM R2R ladder based dacs. I prefer the (older) BB in my set up on CD. I like the ESS9018 and the new high end BB multibit/delta-sigma chip. The latter can give really good results

It's not a problem as long as our signal doesn't contain frequencies above nyquiest frequency.

It's not a problem if you want to play infinitely long sine waves, and that's why digital measures good on traditional parameters. Probably you have played with a spectrum analyser and adjusted the averaging number or time window. It's the same principle. Your sine wave spectral peak is ill-defined if you don't average enough cycles. That's not just SNR, it's maths. Well the converse is also true. Time and frequency are conjugate variables. Unless your waveform is being infinitely repeated, frequency domain limiting (admittedly of any form) results in temporal imprecision or distortion. This is just a byproduct of Fourier transformation. End of story. Music is not a repeating waveform.

So another point. Yes, analogue systems, or digital systems representing analogue parameters, all introduce time domain imprecision/distortion by means of bandpass effects. But the analogue forms are simple functions with everyday parallels, just like our mechanical electrical equivalents we were discussing with headphones. They are causal in time and infinite continuous in time. Digital ones are not. They are artificial in both respects. This is done because it is the easiest way to make a near-phase and -frequency linear anti-alias filter. In phase and frequency, they are indeed excellent. But actually, these parameters, carried over from analogue, are not really a good measure of time domain performance and hence of music waveform accuracy.

One thing I feel would resolve this for both of us would be a nulling scenario on music to look for the error. Feed both signals, time aligned and gain matched, into a really good differential amplifier. But again, this is actually really hard to do in practice because of time alignment and again, the filtering!

 

[SimonPac]

Another anecdote! And two layers of unsubstantiated reference at that.

Which engineer? I've worked on numerous occasions at Abbey Road, spoken at length with many of the engineers, one of my ex-students is an engineer there and I've never heard one of them state that analogue captures acoustics better than digital. Quite the opposite in fact, Abbey Road were an early adopter of digital and were the first studio to take delivery of the first Neve digital desk.

I'm looking for this article in a small stack of UK Hi-Fi magazines I buy sometimes in the airport. If my memory is correct, it was something to do with someone cutting something recorded there direct to disc for vinyl. So perhaps it was done there but the opinion was not of an Abbey Rd engineer. I'll let you know if I find it.

 

[SimonPac]

I'm looking for this article in a small stack of UK Hi-Fi magazines I buy sometimes in the airport. If my memory is correct, it was something to do with someone cutting something recorded there direct to disc for vinyl. So perhaps it was done there but the opinion was not of an Abbey Rd engineer. I'll let you know if I find it.

Found it. Hi-Fi News (probably anathema to you guys) Sept 2015 page 22. Mumford and Sons are among those who have gone for Direct Metal Mastering under Alex Wharton. There were several staff there associated with cutting vinyl. Quote, page 27. 'And there's something about cutting straight to disc that captures the sound of the room like no other format. Tape isn't quite there, digital's not there really'.

 

[pinnahertz]

Found it. Hi-Fi News (probably anathema to you guys) Sept 2015 page 22. Mumford and Sons are among those who have gone for Direct Metal Mastering under Alex Wharton. There were several staff there associated with cutting vinyl. Quote, page 27. 'And there's something about cutting straight to disc that captures the sound of the room like no other format. Tape isn't quite there, digital's not there really'.

You can find someone with an opinion to support anything. I sat in on part of an analog tape panel at Axpona a couple of years back and heard, with my own ears, the moderator declare tape to be the definitive recording medium beating vinyl and digital. Somewhere just after that I left the room.

Opinions are not statistics, science or evidence of anything other than an opinion. If you collect enough of them under controlled conditions you might get a statistical fallout of a trend of something or other, but you'll need more than one, or two, or 10. There have been lots of opinions published. They're published because something about them makes them interesting, not because they represent reality.

 

[pinnahertz]

There's a lot to tackle this afternoon! I agree if people want to persist with this maybe it should move to another thread. Basically you guys seem to think digital is essentially perfect and I don't. Objectively and subjectively. I may change my mind with more experience but I doubt it. I am not trying to force my opinion on you. Obviously you are died in the wool, digital is more or less perfect, people. I don't want to ruin your enjoyment, I just don't agree. If you think AAC is perfect, or that high bit depth or sample rate is irrelevant, again, I simply don't agree. They're all very good though.

Nobody here would ever say AAC is "perfect". Good enough, perhaps. Transparent, perhaps. Perfect, no. High bit depth/rate, that's something quite different.

The fact that you're proclaiming opinion vs fact is fairly obvious.

The point here is that the ADC-DAC version was fed from the self-same master tape copy, already equ'ed to nominally flat in the tape machine and then out to the ADC. I agree there is only potential for things to get worse, but worse they did seem to get.

How do you know? Was it being done simultaneously before you? Was the comparison double-blind? Because, if it's not, you have one cubic carload of expectation bias there.

I actually don't have any analogue tapes! I don't even have any vinyl any more! Just CD, SACD, DVDA and downloads!

Yet you've developed enough opinion on analogue media to proclaim it superior without actually having the hands-on experience, and in opposition to those who have had it in their professional career decades. Even opinions need backup at some point.

So another point. Yes, analogue systems, or digital systems representing analogue parameters, all introduce time domain imprecision/distortion by means of bandpass effects. But the analogue forms are simple functions with everyday parallels, just like our mechanical electrical equivalents we were discussing with headphones.

Show us the everyday parallel of a multi-pole analog filter like that of the composite of the many band limiting effects in an analog recorder (other than an actual multi-pole analog filter).

They are causal in time and infinite continuous in time. Digital ones are not. They are artificial in both respects. This is done because it is the easiest way to make a near-phase and -frequency linear anti-alias filter. In phase and frequency, they are indeed excellent. But actually, these parameters, carried over from analogue, are not really a good measure of time domain performance and hence of music waveform accuracy.

Are you familiar with the concept of "spectral contamination", as advanced by Deane Jensen ("Spectral Contamination Measurement", Deane Jensen and Gary Sokolich, AES November 1988)? The test signal is 100 closely spaced high frequency sine waves. It was (and is) a fairly good indicator of waveform damage in band-limited systems by measure the results of intermodulation of the test signals. However, some of the worst offenders are analog recording systems. The waveform damage in them is rather severe, and orders of magnitude above today's digital systems. There appeared to be a relationship between intermodulation of the test signals with system bandwidth.

One thing I feel would resolve this for both of us would be a nulling scenario on music to look for the error. Feed both signals, time aligned and gain matched, into a really good differential amplifier. But again, this is actually really hard to do in practice because of time alignment and again, the filtering!

...but would you then argue that a 70dB null was not adequate?

 

[SimonPac]

pinnahertz I'm rather aware this could go on forever and we've both presented our present ideas and opinions. I respect your experience and opinions and will bear them in mind. I'll see if I find the time to listen to one of my few 192-24 files at various down samples and do it blind and see if I can guess right. Thanks everyone for developing the discussion.

IMD is fine on digital with as many simultaneous sine waves as you like provided they are allowed to run long enough. It's a linear superposition. It is the stopping and starting that's the problem.

 

[SimonPac]

How do you know? Was it being done simultaneously before you? Was the comparison double-blind? Because, if it's not, you have one cubic carload of expectation bias there.

On the master tape comparison at the audio show this is my recollection. The source was a first gen copy from a an analogue all the way master and onto 1/2 inch 15 ips tape. As far as I recall, it was two rigs, same amp design (they were selling it too), one pair of phones, I think there was a dac light or something else showing 192-24. I guess it could have been tampered with in the file by sample duplication, empty lsbs or too much dither. I think he let me guess which was which and I guessed right. Of course all this is subject to memory but I think I remembered the essentials....I was extremely interested and wanted to know it was a genuine comparison.

 

[castleofargh]

I believe people often lose themselves into the idea that analog is real sound and digital isn't, so of course analog must be better. at least intuitively it looks solid. then we look at the storage side of things and analog is something we can't leave behind fast enough. the various time inaccuracies depending on the media, the added noise at each new transfer, it's hard to be a full analog advocate IMO.

as for increasing sample rate of digital files, let's assume somehow 44.1khz isn't enough for various possible reasons (young ears, special music with high energy in the high freqs, DAC filter rolls off a good deal within the audible range, something does a crappy resampling...) then what is enough to forget those potential troubles ? 48khz? 96? 9999999999999999999hz? I jokingly talked about radio waves and infrared frequencies the other day, but some highres advocate don't seem to have any limit at all. more is always better always audible, always needed. seems very unrealistic.

 

[SimonPac]

I believe people often lose themselves into the idea that analog is real sound and digital isn't, so of course

....

Yes but the opposite is true too. Most people you ask say 'digital just is better...everyone knows that now.....'.

All continuous time analogue storage systems have major issues depending on how far you push them. Digital ones also have significant issues when used to represent and reconstruct continuous time analogue signals. I've discussed the type of problem rather than the extent of it so far. That's a matter of opinion. If you went straight to digital for mix, and stay at high bit depth and adequate rate, it seems crazy to go back to analogue. If you started in analogue, I personally would stay in analogue if it's all done to the state of the art or close to it. That is, ignoring cost, convenience etc. Which is why I'm digital, practically speaking!

You're absolutely right in saying I have little practical experience in recording.