How To: record/encode the highest quality digital audio from pristine tape and vinyl source
Sep 3, 2013 at 1:00 AM Thread Starter Post #1 of 11

duxbellorum

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I happen to have access to a friend's extensive collection of audiophile quality 7.5in/sec tape, 15 in/sec tape, white lable vinyl, etc... He has given me an open invitation to convert some of his collection to a high resolution digital format that could properly reproduce the sound quality. This translates to something like 32 bits per sample and 192k samples per second, essentially comparable to an industry standard studio master.
 
He has equipment that can provide near perfect audio source, but I have nothing to just plug that output into and start recording.
 
So I would like to learn more about how to do this. What hardware is involved? What software works well? What do I need to do this right? I'm not looking for a fast resolution, but the best answer, ideally one that doesn't have me spending huge piles of money on the project.
 
The end result I'm going for is a piece of software and a single analog to digital device that I can attach between my laptop and his system, the complete combination of which will yield near perfect replication of his media, just in immortal digital form.
 
Sep 3, 2013 at 6:05 AM Post #2 of 11
There are no 32/192 ADCs in existence, the highest rate you can do is 24 bit, 192KHz, but, and this is significant, no "affordable" ADC ever comes close to true 24 bit performance, even though you get 24 bit data from it. They are all noise limited to more like 20 bits or less.

A couple of things about what's really necessary though. 192KHz is unnecessary for digitizing any analog format. There is no useful information in analog above about 25KHz, and arguably below even that. The noise floor for analog tape requires less than 16 bits to faithfully reproduce. The noise floor of even the best vinyl is barely 12 bits equivalent. 24 bit is also unnecessary, but since we have the capability, albeit really only 20 bits, no reason not to do it. 24/96 is still way overkill, 24/48 is completely adequate, but you can do whatever makes you feel good. The more important question to ask might be how the resulting files will be played, and will there be any post processing.

For recording software, Adobe Audition is about as good as it gets without getting into ProTools. Audacity (free) is also very good.

I could suggest hardware if you shoot me a budget figure beyond "not huge piles of money". Keep in mind what you asked for was "near perfection for cheap"...the two go in opposite directions, you know.

You also said your friends gear can provide perfect playback. I would need convincing of that, as perfect playback of vinyl and tape requires precision calibration off the entire analog chain, which is non-trivial, and requires test equipment and special tools. Most of what's wrong with analog playback is the lack of calibration.
 
Sep 3, 2013 at 8:02 AM Post #3 of 11
His system is a bit daunting...speakers from Quad, 5000$ mysterious super DAC, Revox to handle the tape, a tonearm he built and patented himself on what I presume is a great turn table to handle the vinyl, etc...I can't recite the manufacturers other than Quad and Revox off the top of my head, but from everything I have heard out of it, and the limited things I understand about audio, it is about as good as it gets. I mean he's an actual audiophile, that's what you'd expect.
 
As for 24-96 vs 24-44, There is a lot of information at between 20-40khz that I personally think is involved with really good sound (Listen to an mp3 that cuts things off at 19KHz and then listen to the same song in 24-96, and there is a big difference in the body of the sound.) Certainly, sound in the 18-25khz envelope is going to sound better at 96 thousand samples just because the resolution is twice as fine. I'm taking a look at an Audacity generated spectrogram of a 24-96 flac of System of a Down's "Bounce" done from vinyl, and there is a surprising amount of information between 20KHz and 40Khz that does not exist in the 24-44 flac.

As for what's stored in 15 inch/sec reel to real, there is a whole lot more information than cannot be contained in 24-96 (general consensus is that 15ips > 24-192 by a fair margin), let alone 24-44...these tapes store from 5Hz to beyond 35 KHz with incredible resolution. I'd be comfortable putting vinyl onto 24-96, but 15 ips and possibly 7.5 ips call for better sampling rates and, ideally, more bits per sample (I'm aware that most DAC and ADC don't really process more than 20bits per sample, probably not something I can address, but this does seem like a rather artificial ceiling with all of our other technology getting so fast)
 
Sounds like 24(20) bit at 192,000 samples is what I want to aim for. The big question I now need to answer is what to look for in an ADC to do this.
 
Sep 3, 2013 at 9:42 AM Post #4 of 11
Quote:
Originally Posted by duxbellorum /img/forum/go_quote.gif
 
(Listen to an mp3 that cuts things off at 19KHz and then listen to the same song in 24-96, and there is a big difference in the body of the sound.)

 
You do not know, however, if that is actually because of the 19 kHz cutoff, the MP3 compression, or the 24-96 version being mastered differently. If you are interested, you could post some 96/24 format samples (30 seconds maximum length for file size and copyright reasons) on the Sound Science forum, and downsampled/filtered/quantized versions of it can be created for testing how much loss of information is needed for the sound quality to be audibly worse.
 
Quote:
Originally Posted by duxbellorum /img/forum/go_quote.gif
 
Certainly, sound in the 18-25khz envelope is going to sound better at 96 thousand samples just because the resolution is twice as fine.

 
That is not true, in fact. You only need somewhat higher than 50 kHz sample rate to capture 25 kHz without significant degradation. In theory, exactly 50 kHz is enough for anything below 25 kHz, but some extra bandwidth (10-20%) is useful in practice so that a lowpass filter with an extremely fast roll-off is not needed.
 
Quote:
Originally Posted by duxbellorum /img/forum/go_quote.gif
 
I'm aware that most DAC and ADC don't really process more than 20bits per sample, probably not something I can address, but this does seem like a rather artificial ceiling with all of our other technology getting so fast

 
The limiting factor is not the digital "resolution", but simple analog noise (hiss). At room temperature, passing a 2.0 Vrms Red Book level signal through a 4.7 kΩ resistor will by itself already degrade the signal to noise ratio to not better than 126 dB A-weighted (~21 bits) due to thermal noise.
 
Quote:
Originally Posted by duxbellorum /img/forum/go_quote.gif
 
As for what's stored in 15 inch/sec reel to real, there is a whole lot more information than cannot be contained in 24-96 (general consensus is that 15ips > 24-192 by a fair margin), let alone 24-44...these tapes store from 5Hz to beyond 35 KHz with incredible resolution. I'd be comfortable putting vinyl onto 24-96, but 15 ips and possibly 7.5 ips call for better sampling rates and, ideally, more bits per sample

 
You only need more bits per sample if the analog input signal has sufficient dynamic range. If it does not, the ADC noise will get "swamped" by the noise from the source (you calculate overall noise level by summing power - assuming uncorrelated random noise - so, with a source noise floor of -70 dBFS, and an ADC noise floor of -80 dBFS (something that even onboard codecs can achieve), you get a total noise floor that is still below -69.5 dBFS. With dithering, noise from digital quantization is, for practical purposes, not really different from analog white noise, so it is not correct that a noisy analog signal has "infinite" resolution compared to anything digital.
 
In other words, it is mostly the playback and analog equipment before the ADC that makes the difference. For the recording, you need some decent USB pro audio interface for your laptop, but nothing exotic or very expensive.
 
Sep 3, 2013 at 12:07 PM Post #5 of 11
Yes, everyone looks at an Audacity spectrum and concludes that just because there's spectral content that it's audio and contributes to the audible result.  But they are ignoring the physics of the analog medium completely.  
 
Here's my Analog "Did You Know?", and we'll start with tape.
 
The statement "these tapes store from 5Hz to beyond 35 KHz with incredible resolution." clearly comes from someone who's never set up and calibrated a tape recorder.  There are physical limits to the high frequency response dictated by the magnetic wavelength on the tape moving at 15ips and the size of the gap in the head.  Once a wavelength approaches the gap, you begin a rather radical and unequalizable HF rolloff.  The degree and rate of change is actually quite similar to an analog anti-aliasing filter, replete with in-band group delay.  Then there's bias self-erasure, the effect that AC bias has that as you increase bias you decrease distortion but increase the effect of bias erasing the high end.  As you hit the optimum point where distortion is optimized, you have to try to equalize back the extreme top end.  But since bias erasure doesn't follow the identical curve of the record equalizer, you end up with less than flat response, with the extreme top again rolling off fairly quickly. The record head has gap loss as well, and so does the play head.  All combined, you have an analog multi-pole low pass filter with pretty much terrible transient response.  Now, unless the record and play side of the deck were calibrated, there's no optimizing this after the fact. And, unless whoever recorded the tapes in the first place included repro cal tones on the head or tail, there's no matching the play characteristic to the tape.  
 
Then we have another mechanical issue: head alignment.  The gap of the play head must be exactly and perfectly parallel to the record head gap, or there will be a severe loss of high frequencies and a phase shift between left and right.  The adjustment is called head azimuth. In some cases it may be possible to best-guess this position using the program material on the tape itself, but since you have not idea how many generations of tape that went through, it could be wrong.  The other thing about head azimuth alignment that people don't know is how record bias affects the apparent azimuth.  During calibration you physically adjust the play head to a standard test tape.  But during record, most people will align the record head to the play head by recording test tones...an then go and adjust bias and EQ.  The problem here is that bias and EQ affect high frequency phase a lot, so if you don't get both channels biased and equalized, at least approximately first, there's no getting record azimuth right either.  
 
Lets move on to that 5Hz at 15ips statement.  Here's the problem: the ability of a recorder to record and reproduce low frequencies is inversely proportional to the speed of tape travel.  Low frequency response of ANY tape recorder at 7.5ips will be much better than 15ips, which will be good to actually about 50Hz before we see low frequency fade.  7.5ips will go to 30Hz, but 20Hz is well into roll off.  5Hz response at 15ips will be down an incredible amount, seriously enough to ignore as a contributing part of response.  This is one reason why we saw so little recording done at 30ips...the LF response was horrible!  
 
Nonlinearity...now there's a big one.  Magnetic tape is basically incapable of recording any audio at all without huge amounts of distortion.  This is because tape won't accept magnetization in a huge dead-band around zero. How big that dead-zone is is defined by the figure "coercivity".  We get around that by slamming bias to the tape head so that all audio is bumped way beyond that dead band, up where tape begins to behave more linearly.  But linear is relative and there is a maximum amount that we can magnetize tape too before it saturates.  Tape saturation causes distortion, quite badly, THD and IMD, both of which produce harmonics.  Because of the tape record equalization characteristic, tape saturates at high frequencies way before mid band frequencies, and that means highs will distort quickly and produce harmonics, dominantly odd-order.  The third harmonic of 8KHz is 24Khz...sound familiar?  All tape machines do it, and multiple tape generations (master > final mix > copies, etc) increase these harmonics.  But it's not audio, and not desirable, and doesn't add to the experience at all.
 
I'm a little time-limited today, so I have to stop here.  Next time I get a chance, I'll finish up with tape (no, we're not done!), and move to vinyl.  Short story: vinyl has even more ability to produce high frequency distortion!
 
One final thing, I never said 24/44 was acceptable for your project.  I said 24/48 minimum.  But more important, and still not addressed, how will the files be used?  IF you record at 24/192, you'll end up down sampling at some point unless you'll always and forever have your 24/192 DAC in your back pocket.  There are issues...if someone would like to hit that, I'll just deal with analog stuff.
 
 
 
Sep 3, 2013 at 5:12 PM Post #6 of 11
In other words, it is mostly the playback and analog equipment before the ADC that makes the difference. For the recording, you need some decent USB pro audio interface for your laptop, but nothing exotic or very expensive.

 
Thanks for the advice, leads into the question:
 
Does the E-MU 0404 fit the bill?
 
Are the ADCs listed in this thread overkill?:
 
http://www.head-fi.org/t/606173/comparison-adc-interface-for-hifi-recording-1000-usb-pre2-adl-esprit-rme-babyface-adl-gt40-other-analogue-digital-usb-convertsers-experiencies-comparisons-reviews
 
Sep 4, 2013 at 6:03 AM Post #7 of 11
   
Thanks for the advice, leads into the question:
 
Does the E-MU 0404 fit the bill?
 
Are the ADCs listed in this thread overkill?:
 
http://www.head-fi.org/t/606173/comparison-adc-interface-for-hifi-recording-1000-usb-pre2-adl-esprit-rme-babyface-adl-gt40-other-analogue-digital-usb-convertsers-experiencies-comparisons-reviews

They are all overkill for your project, yet all fall miserably short of theoretical performance.  LIke the E-MU at -.5dB @37KHz when running at 96KHz sampling rate? (from the Sound on Sound review and tests) What's up with that?  Should have made it to at least 45KHz.  
 
Might also look at this:
http://us.focusrite.com/usb-audio-interfaces/forte/specifications
 
There are reports of driver issues, but that should have been solved by now.  Also overkill, but with better specs than anything above, possibly cheaper.
 
For the theoretically perfect performance you long for: http://www.stagetec.com/en/audio-technology-products/standalone-converter.html   
 No, it won't do 192KHz, but it's the only box out there with true 24bit dynamic range (albeit done with the trick of cascading multiple ADCs). 
 
Perhaps there's no interest in the rest of the analog story, and the real reason why you see ultrasonic spectrum from tape without it actually being audio.  Or why 30KHz response from analog tape is impossible.  Perhaps there's no interest in why the ultrasonics found on records also aren't audio.  And why none of it need be digitized.  Or why 192KHz files are a liability. 
 
Well, I thought I might finish the story anyway, but if nobody cares, why bother?  So, you'd probably better pay up for the best, quietest, fastest ADC/DAC you can find, or you'll forever regret not digitizing all of that critical info.  I'd just save up for the Stagetec, do the job, then rent it out or sell it off on eBay. 
 
Sep 4, 2013 at 5:48 PM Post #9 of 11
Part 2 ... for insomniacs everywhere
 
Summing up some of the afore-mentioned issues with analog tape:
 
Limited high frequency response because of the physical size of the head gap and magnetic wavelength on tape
 
Limited HF response due to record EQ, and bias self-erasure
 
Limited HF response, particularly in playback on a different machine than recorded the tape, by relative play head misalignment, particularly azimuth. 
 
Limited LF response, inversely related to tape speed.  Higher tape speed = less LF extension.  There is no recording 5hz at any speed. 
 
Let me expand on that last one just a bit.  The process of recording (magnetizing) and play (the inducement of a current in a coil of wire by a fluctuating magnetic field is an AC coupled system.  That means you can't record DC, and the output of the system decreases with falling frequency.  The slower the magnetic flux change, the less output you get out of a tape head.  Translate this to the magnetic wavelength on a moving tape, and you'll start to see why LF response falls apart with high tape speed.  The magnetic wavelengths on tape are far too long to induce much current in the head.  You can equalize that response back to a point, but there is point of diminishing returns beyond which you can't recover LF response.  There is a huge problem with calibration here too.  LF test tones on test tapes have been historically inaccurate.  One studio client of mine used one test tape to equalize the general response, another from a different manufacturer to adjust LF eg.  There's also the "fringe effect", where a recorded track that is wider than the track width of the play head causes elevated (and inaccurate) LF response (called a "head bump") that is partially predictable, but counter intuitive.  And that's just the play side.
 
So when I say there's no recording 5Hz, you can almost quadruple that frequency and be just as right.  There's no recording 20Hz accurately either at 15ips, certainly no chance at 30ips. Oddly, 7.5ips can do a pretty good job at 20Hz!
 
Another point on tape, track format.  I've worked with everything from full-track mono to 24 track 2", and everything in-between.  8 track 1/4" was a LOT of fun.  Track width affects signal to noise, LF response, head alignment tolerance, and a few other things.  Ideally, you'd be playing the tape on a machine that exactly matches the track format it was recorded on.  In the consumer world, this can be an issue.  We had 4 standards for 1/4" tape, 1/2 track mono, 1/4 track mono, 1/2 track stereo and 1/4 track stereo. None are really compatible with each other…you have to do at least something just to get usable playback.  And usable isn't optimal (except playing 1/4 track mono on a 1/4 track stereo deck, that works fine).
 
Dynamic range?  Well, it's dependent on tape formulation, and not all are created equal, track width, and speed.  Higher speeds move tape hiss up higher to a less audible range.  Wider tracks allow more magnetism to be presented to the play head, so lower noise.  Higher output tapes have more ability to record at high levels with less distortion, but what happens to all tapes and all machines is, the higher the level, the more distortion, with IMD happening quite quickly.  The maximum record level on tape is frequency dependent. So if we insist on fairly clean and undistorted peaks, the typical dynamic range of tape is around 70dB (about 12 bits).  If we add Dolby A noise reduction, we can push that to 80dB (about 13-14 bits).  Dolby SR stretched tapes dynamic range to right around 16 bits, but was a professional-only system (as was Dolby A). 1/4 track stereo tapes have a dynamic range of around 60dB (10 bits). 
 
So, we've got limited HF response, limited LF response…what are we seeing on that Audacity spectrum?  In the LF region, it's basically noise, but will look like modulated audio.  LF noise in analog tape is caused by several things, one being the "calendaring" or precision finishing of the tape surface which relates to intimate head contact.  Slight variations there will cause a bit of LF noise.  More significant is the effects of DC magnetism caused by magnetized machine components or DC currents through any of the heads.  Remember I said you can't record DC?  You can't, but trying causes a very low frequency groveling noise, which can also be modulated by audio.  That stuff has the potential for sub-sonic content, and that's what's being seen on the spectrum at the LF end.  There's a whole discussion on demagnetizing tape machines to be had, just not today.  Short story, people who do it all the time probably make it worse than people who never do it at all.
 
Spectrum over 25KHz is almost certainly distortion products caused by nonlinearities of the tape itself, the heads, and the electronics.  The third harmonic of 8KHz is 24KHz, recall.  Someone will argue that audio can have ultrasonic harmonics, and that's true, but we can't really record or play them, certainly not above the limits of head gap and bias erasure, without doing something non-standard.  Interesting thing about non-standard, if we changed the standardized parameters we are now locked into with magnetic tape recording, we actually could improve performance in all areas, including frequency response.  But we literally have to break ALL the rules to do it.
 
Vinyl
 
The entire vinyl/LP system end to end can be considered a flat, wide band medium so long as we realize that "flat" and "wide band" are highly level dependent, and ultimately have physical limitations imposed by the size and shape of the cutter stylus, resulting groove, and play stylus.   At the high frequency end, there is a physical limit imposed by the contact surface area of the stylus on the groove wall.  Tiny wavelengths eventually are smaller than the contact area, and simply get bounced right over (actually creating frequency down-folding imaging).  The higher the frequency we try to deal with, the lower the maximum modulation that can be handled by the system.  So, those ultrasonic 35KHz signals we think the system can handle must actually be very low in level, or the results of distortion and not real audio at all.  An example would be the FM carrier used in CD-4 quadraphonic discs, which was centered at 30KHz, and contained modulation products up to 48KHz.  A special stylus was used, and the level of that carrier was quite low.  It also didn't matter too much that the recovery of the carrier was laced with distortion, because it was frequency modulated and hit a demodulator that was relatively insensitive to a distorted carrier. 
 
So what are we seeing on that spectrum analysis?  Signals that come off a phono cartridge are in fact full of ultrasonic stuff.  Mistracking, where a stylus does not remain in full contact with a groove wall, produces a lot of ultrasonic as the stylus repeatedly slams into the groove wall then gets thrown away from it.  Those signals will be modulated, like audio, and in the ultrasonic range, but it's not really audio itself, rather a form of distortion product.  Again, high frequency distortion, odd order, places the third harmonic of everything above 8KHz well into the ultrasonic range.  And here's the point: Actual harmonic content in audio is there too, but since most LPs were mastered on analog tape, those harmonics have already been reduced or eliminated before mastering to disc.  What we are seeing is the result of issues in the mechanical system, not audio.
 
Moving to LF response of the LP.  High level low frequency information has a maximum limit defined by the ability of a stylus to stay in a lateral groove that has a lot of physical excursion.  Compliance is a factor, as is the mass of the tone arm, but sub-sonic bass is actually challenging to recover from disc.  For one thing, the disc itself is full of recorded rumble, and even the best turntables also have quite a bit of it.  Adding high levels of subsonic bass to that rumble can, and does, clip preamplifiers.  The maximum bass level is also a function of pan position.  You can record quite a bit of level into a lateral groove, but not as much in a vertical groove.  The vertical limit is defined by the maximum depth before the cutter hits the aluminum substrate, and the minimum depth before the groove becomes too shallow to hold a stylus.  Left or Right only signals are effectively a combination of vertical and lateral groove modulation, so they have those limits imposed.  It's one reason loud records have bass and kick drums mixed center. 
 
So can we get a bit of extreme LF out of a record?  Sure, but how did it get there?  Not from tape!  There were a precious few direct-to-disc recordings made over the years where a live stereo mix was fed directly to a lathe, and those have some pretty amazing signals on them. But the "mask" of vinyl is essentially the same "mask" as analog tape, plus some new physical limits. 
 
Vinyl has a signal to noise ratio of 50 to 70 dB depending on the original tape master, the type of vinyl used in pressing, and the condition of the record, and method of measurement.  50dB is equivalent to just under 9 bits. 
 
So, to effectively capture everything on analog tape and vinyl, we need to exceed those medium's capabilities.  16 bits at 48KHz accomplishes that more than adequately.  24 bits (remember, it's not REAL 24 bits, it's a REAL 18 - 20 bits!) is substantial overkill, but if there's a desire for post processing, de-clicking, noise reduction, eq etc., then it's nice to have a few extra bits to play DSP games with.  All high-frequency needs are easily met by 48KHz sampling rate, but if we wanted to give in to paranoia, the next rate up will approximately double that limit, so 88KHz would be far more than we need, but the next common step.  And it's not as common as 96KHz, so that's more likely what would be used.  24/96 would be the choice for the paranoid, so long as you ignore the ultimate "release" format, probably down-sampled so it can be universally played.  Resampling never improves audio, not in either the up or down direction.  It would be best to digitize in a format that could actually be used. 
 
How paranoid is digitizing a 9 - 13 bit equivalent signal to 24 bits?  How paranoid is digitizing ultrasonic distortion products and noise?  That's a deep-seated psychological question out of my area of expertise.  There's no "engineering" reason to do it though.
 
Sep 6, 2013 at 9:58 AM Post #11 of 11
Thanks for taking the time to read it!  I'm not really trying to burst any bubbles, but I think the relative newcomers to analog audio might benefit from the perspective of someone who worked with it for 40 years.  There are reasons we ended up with digital audio, and most of them are analog problems that digital improved on or eliminated.  There are a few more myths of course, like the real source of the "vinyl sound" (clue: it's not the vinyl), etc.  But that'll be enough derailing a thread for now.
 

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