Q: I love vinyl and want to capture that ‘analog sound’ — will I get that if I record my synthesizer to a cassette recorder?

A: The characteristic ‘sound’ (complex of deviations from accurate signal, aka, distortion) imposed by the cassette format is very different from that created by vinyl transcription.

While a good grooved disc can do a pretty good job with regard to fidelity (accuracy), at least within certain parameters — and, no doubt there are those who seek out those characteristics, preferring them over objectively less distorted, higher fidelity transcriptions — the state of the art vinyl record’s main divergences from fidelity are: random noise in the form of surface noise, scratches, dust, etc, as well as the basic, always-running sound of stylus-in-groove (drop the needle in a ‘blank groove’ with the volume at an ‘active listening’ level and see what you hear), as well as various forms of intermodulation distortion produced by both record wear and mechanical imperfections or design flaws in pickups, motor noise and or bass feedback rumble (the stereo becomes a resonant system with bass frequencies being transmitted from speakers through floors and cabinetry back to the pickup, which is why high end turntables put so much effort into anti-vibration/shock mounting), and finally, while the fidelity at the outside of the groove can be high and frequencies quite extended, by the time you get to end of a record side, the stylus-in-groove speed is much slower, higher frequency response has diminished greatly and because the diameter of the inner groove is so much tighter, dynamic levels must be watched carefully or various forms of distortion will become problematic. Oh, and don’t forget lack of stereo separation, with separation between channels typically between 25 to 40 dB. And, in lesser turntables, throw in fast and slow speed variations: wow and flutter.

While the grooved record was the best mainstream delivery medium of the analog era, pre-recorded reel tapes offered some arguable advantages but were fraught with tradeoffs: they were expensive and consumers balked at threading tape machines (??? — but they did!), and good decks were plenty expensive. (Add to that good tape was far from cheap.) And they were far from immune to their own pernicious form of noise: hiss. Still, with wide enough tracks and high enough speed, they could carry a decent signal. (Tape/capstan flutter was still an issue, but record scratches weren’t. Unfortunately, record scratches were ‘replaced’ by the tendency of magnetic tape to shed the ‘active’ oxide layer — where the magnetic signal is stored — from the backing, letting your signal flake off slowly — and sometimes not so slowly.)

The cassette was introduced first as a portable format — voice and tape letters were the big selling point. Through the 60s, different attempts to get tape into cars had supplanted the goofy lover’s lane phonographs that some folks mounted under their dashes. (You couldn’t drive with them running. The needle would skip to hell and back.) ‘Endless’ tape carts like those used in broadcasting were employed, leading to commercial car decks like the 4 and 8 track models that dominated the end of the 60s. Around that time, a move was made to bring ‘high fidelity’ to the cassette. Sony was one of the first makers to bring out a ‘hi fi’ cassette deck. It cost $600 back in ’69 (equivalent to around $3800 today). And it sounded AWFUL. The flutter made it sound like it was running through a cheap guitar chorus pedal just about. There was almost no high frequency response.

Cassettes definitely got much, much better, but they were always limited in their frequency response and accuracy (at both ends of the frequency spectrum), slathered in hiss, and beset by lingering speed issues in the form of flutter, which was particularly bad because of the low speed of the tape and the tiny circumference of the capstans — meaning that even micro-tiny imperfections in the mechanism could be ‘magnified’ into relatively noticeable speed problems.

Now, of course, we know that many ‘lo fi’ enthusiasts like cassettes for many of those perceived negatives: they’re looking for ways of ‘mangling’ the sound and imparting a distant-in-time-or-space sort of vibe. And that’s fine, particularly for those aware of the issues. They grew up with clean, relatively high accuracy signals and they’re bored with them. I get it. Just as, hopefully, they get it that I grew up trying to squeeze accurate signal out of analog media like vinyl records and tape and, so, have been delighted to finally have that fidelity delivered by modern recording systems and media.

Now… as someone who recorded a lot of analog synths to a lot of analog tape in the 1980s, I have to say that I, personally, prefer to get — as a baseline — more or less the sound that came out of the synth. That really didn’t happen until I got my first DAT machine around 1990. Me, I did not like the sound of the synths I was working on to get mangled by the recording medium. But, that’s me. Not you. Not anyone else. Everyone must ultimately find their own way.

At Gearslutz, the mega-popular recording technology/gear lust site, a member asked…

Originally Posted by deadby100cuts
Ok so I got a pair of Roland ds-90’s from a buddy for 100 bucks. Ive been mixing through my pa system so its GREAT having monitors lol.Anyway, I just got them hooked up and they sound GREAT.I used some instrument cables that I had been using to plug my interface up to my pa. They sounded great, no problems at all. Fast forward an house after I get back from buying some short speaker cables. I hook it up, and I’m now getting a REALLY annoying high pitched noise from the speakers. I use to get this from my Pa system as well but It could be managed by keeping the pa itself on a lower volume. However with the new monitors I don’t know how to get rid of it. Its not super noticeable when a song is playing but the second the song gets softer or ends, its right there in your face. Its SUPER annoying and it didn’t do this when I used the instrument cable. Its not a solid pitch, it changes slightly. it doesn’t matter what settings I change on the interface it still has that annoying sound. Its driving me crazy.I recorded the sound ( I hope) here is a clip Vocaroo | Voice message[1]


Ok, so I grabbed my long like, 20ft speaker cable from my pa system and dragged the speaker into several other rooms so I would know its on a different circuit, thats where it gets weird. If I plug the speaker into the wall in another room BOTH speakers will emit a lower sustained kind of tone. Even though one is in the other room, and is still plugged into the other outlet, so the only way they are “connected” is the fact they are both plugged into the same interface. Keep in mind they are powered separately.

So I don’t THINK its a ground issue, but I’m not sure whats going on. My whole computer setup is a little strange, wires everywhere. If it interference from something, like the power cables, then I’m not sure how to get around it. Because I mean, the only place I can put my tower is right in between the speakers and the monitor is in front of the tower.

[I answered…]
Could that ‘REALLY annoying high pitched noise’ you’re getting from your powered (amplified/active) speakers (and also got from your PA when turned up loud) be the broad spectrum noise often referred to as ‘hiss’? That’s the term often used to refer to such self-noise from amplifiers.

In conventional standalone amplifiers, there is typically a master volume control that controls the output level of the amplifier as it goes to passive speakers. There will always be some noise floor, even with that master volume all the way down — but on most conventional amplifiers, as you turn the output level up, the self-noise rises with the volume. If you have music going, you don’t notice it because the music has ALSO gotten louder — but if there’s no music, the self-noise (hiss) from the amp will likely be very noticeable at the top of its loudness range.

Now, imagine you had to run that amp all the way up (for some arbitrary reason) but then control the volume from an external source like a CD player with a built in volume control. The amp is always running at full bore, so the self-noise is always high, but when you adjust the volume from the CD player down to a comfortable level, the level of the hiss from the amp stays just as loud as it was.

This is analogous to how (almost all) powered speakers work. (Some do have individual volume controls and some variable or switched input trims, which, if designed right, should allow some lowering of the gain the amp is supplying, lowering self noise.)

When you turn down the output level from your computer you control the level of signal going into the input of the amplified speaker — but not the amplifier’s gain. Lower the volume from your computer/DAW and the overall volume goes down — but not the self-noise from the amplifier/speaker.

These powered speakers look like pretty powerful amp/speaker combos judging from the Roland writeup (DS-90A :: Products :: Roland) 60/30 biamped — the ‘equivalent’ in some broad sense, of a conventional 180 watt stereo amp. (And pretty good frequency specs, too, at 48Hz to 20kHz (+/-3dB), at least on paper.)

And that means they’re probably louder than you’re used to, and have higher self-noise than you’re used to.

If the speakers have no volume control or trim (there’d likely be knobs, switches, or small, inset nylon screw shafts on the back — but check the manual, which I didn’t download [not familiar with these units, myself]) then I’m afraid it’s more or less something you’re going to have to get used to.

Now, cables. You should NOT run dedicated speaker cables between your source (DAW/computer/whatever) and the LINE in put of the powered monitors — such cables are for running between the output of an amplifier and passive speakers! You won’t damage anything (but your signal) but they are designed for the impedance and power handling necessary for carrying powered signal between the output of the amp and a speaker’s load.

Signal leads (balanced mic cables or unbalanced ‘instrument’ or patch cables) should be used between the LINE source and the amplifier input of the powered speakers.

Now, finally the low pitched noise you got from running the long cable into the other room where the speakers were plugged in?

That is almost certainly a so-called ground loop. You created a near-perfect setup to introduce ground problems by interconnecting two audio devices that (I’d bet a cyber-dollar) are connected into power outlets that are running to different circuit breaker or fuse panels. That introduces two paths to ground. Electricity has a voodoo like attraction to ground (we won’t get into the physics) so electrical power can, in effect, be drawn through the circuit in unintended ways, causing 50 or 60 Hz modulation of the signal from ‘power contamination.’

The solution: it’s always best to have interconnected video and audio devices supplied from the same panel circuit if at all possible. To be on the safe side, many smaller installations (as in houses that may have funky wiring — and many do) will supply power to all interconnected units from a single point.

Hope that helps.

Music is math, yeah, sure… but can math reproduce analog audio signals?

A bit of background: I am, by academic training, a failed poet. On the basis of that training, I became a hippie musician/songwriter. I stand second to few in my embrace of willfully undisciplined, fly-by-the-seat-of-intuition artsy-fartsyness. But, somehow, in all that liberal education, I also picked up an appreciation of logic and the Scientific Method. Two very different hats.

When I’m making music, I’m all about vague and indefinable stuff. I act like I believe the artistic muses are real entities (for me, it’s a very useful model; in this realm, wearing my ‘artist’s hat,’ I don’t much care about the literal truth, as long as my perhaps ‘fanciful’ concept helps me describe/predict actual experience). I play it by feelings and hunches and seemingly wild shots in the dark.

But when I’m dealing with gear and technology, clearly that artiste’s beret is a bad fit. And that’s where I bust out the ol’ logic and clear thinking.

OK, that’s out of the way.

Recently, in the social media arena revolving around studio recording, a community member posted his thoughts about how mathematics is inadequate for describing and dealing with analog audio electrical signals. Not unusual in that milieu, he seemed to be largely ignorant of how integral mathematics is to the design and building of analog audio gear. Worse, he seemed to have little understanding that mathematics is a key component to how we describe and understand the entirety and complexity of all of our universe, from the microcosm to the macrocosm.

People who don’t understand the underlying mathematics tend to equate the individual samples of digital audio with the individual frames of film — which ‘fools the brain’ with a succession of images, the faster the film (to a certain point), the smoother the action appears. But film never creates an actual truly continuously moving image… it’s a series of images as presented to the eye.

Digital audio, on the other hand, does produce a continuous wave. Where’s the ‘missing data’ ‘between the samples’? It’s above the agreed upon band limit. If we want to be able to serve up a signal accurate up to, say, 30 kHz (to ‘entertain’ our cats and dogs) we can use a sample rate that affords us that ability, a minimum of something over 60 kHz, typically with a frequency band limit margin that allows upper frequency bandwidth that accommodates the efficiency of the antialias filtering applied.

There is no ‘fooling the ear’ with such a signal — the ear is not creating the ‘illusion’ of continuous analog signal coming out the speakers — the analog signal arriving at the speakers is, indeed, just as continuous as the original signal. If the process was performed properly and accurately, the results should be a precise duplicate — up to the filter boundary. If we set that filter boundary above the threshold of the listener’s hearing, we have a system capable of reproducing an electrical analog audio signal with far greater precision than any previous audio transcription system.

(In addition to conflating digital audio processes with film, it seems some tend to mix digital transcription with the very, very different, and very complex issues revolving aroundlossy perceptual encoding, as with mp3, Vorbis, AAC, etc. Those systems very definitely are designed to ‘fool’ the human audio perception system by eliminating data predicted by the governing algorithms to be ‘unnecessary’ to recreate a perceived semblance of the original signal. The more data that is thrown out, the greater the chance the changes will be perceived. Lower rates can be almost painfully obvious, but once we get to around 1/5 to 1/4 retention of data (320 kbps is ~ 1/4.4 of the data bandwidth of CD-A), the ability to differentiate the lossy format from original is found to be quite rare, even among trained listeners. But it IS, indeed, ‘fooling’ the ear. Unlike non-lossy PCM audio.)

With regard to the upper limits of human hearing, there is roughly a century of testing of human hearing that has gone into the current scientifically accepted understanding of the upper limits of human hearing. There is a large body of direct perception testing (can one hear a given tone by itself at a given level) as well as indirect (testing of program material against the same material with narrower frequency band limits applied to see if the difference between signals can be perceived).

The scientists who have been studying our hearing during the modern era have set the nominal limit of human audio perception at 20 kHz, although by adulthood, most humans are considerably under that, while the very young may perceive somewhat above that threshold. However I’m not aware of any accepted work offering solid evidence of ability to perceive over 22 kHz. There have been what seemed to be some ‘tantalizing’ outlier experimental findings, but the circumstances of those findings have been subject to much questioning and not a little criticism, when they haven’t been rejected outright by professional review. 

The overwhelming mountain of data so far collected suggests the nominally accepted limits are reasonable and realistic. If someone wants to overturn that paradigm, it behooves them to build a more persuasive body of evidence supporting their claims. Until then, the reasonable assumption flows from currently held data.