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Last Manufacturer of Pro Analog Audio Tape Closes
goosman writes "Quantegy, the last manufacturer of professional reel-to-reel analog audio tape in the world has closed their plant in Opelika, AL leaving a reported 250 workers without jobs, according to the Opelika-Auburn News. Emtec (the former BASF, which used to be AGFA) was the last European manufacturer and ceased manufacuring in 2002. An audio account of the closing can be heard at NPR."
Sigs cause cancer.
Cassette tapes aren't pro tape.
Or did they buy the audio division when Ampex went to "Ampex Data Systems"? If I am to believe the article, then there would be no further sources of 2" reels. There are a lot of 24 track studios out there that still use this tech.
BBH
The article says they're just closed for restructuring. This is vague, but it may not mean they are closed down permanently.
I work at the BBC World Service, broadcasting in (approx) 42 different languages around the world - and we still use analogue tape for about 80% of our programmes! We are slowly being digitised, but believe it or not, analogue tape is great to work with, quick to edit, and extremely reliable, both for playback and archiving... I'm no luddite, but as someone who has to deal with on-air disasters, I know that tape recorders don't crash.... Our latest digital system runs on windows 2000... Say no more.
No. You're wrong. There's a certain compression that highly-driven tape produces, which is much more complex than some lowpass filter.
The finest consumer tape deck ever produced, the Pioneer RT-909, had a frequency response to 30kHz. Studio decks that record at 15 inches-per-second have response clear out to 40kHz and beyond. A CD has response to only 22.05kHz, and even studio digital equipment has a hard time working up to 48kHz.
An actual 24-bit system has a theoretical Dynamic range of around 140dB but you'll be hard pressed to get better than 80dB with most gear. With analog recording there are at least two well-known foolproof methods to improve dynamic range and SNR: get a bigger tape, and run the tape faster. The dynamic range and SNR on 2", 32ips tape is amazing.
And of course tape can be driven to +9dB recording levels in some cases, but a digital system will clip hard at 0dB.
Digital is definitely the future but right digital recording has its problems. Next time you go to the record store notice how many High Resolution DVD-Audio recordings are being mastered from tapes.
Hello sir, I hate to break it to you, but the front-end analog electronics and jittering timebase on the Audigy limit it to dynamic ranges of around 80dB and SNR of around 60dB, giving about the same performance as a good 20-year-old cassette deck.
For one thing, if you send too much signal into an analog tape, you get a nice sounding tape compression, whereas if you send too much signal into a ADC, you get really horrible sounding digital clipping.
That's why you use high-resolution ADCs and run them at a safe margin less than full scale. Then, when you load the file into your mixer, you take the arctangent of each sample to get soft clipping.
This is multi-track tape. So you need 32 cell phones and 16 crinkly plastic bags.
You are not a beautiful or unique snowflake -- but you could be if you got off your ass.
When I was at MIT (circa 1980) there was a recording studio down the hall from TMRC in building 20 (it was across the hall from an old anechoeic chamber...but I digress). The was pretty much abandoned and used by a small group of students for recording punk demos. The actual studio was isolated from the control room completely...the studio was on springs to completely prevent sound from bleeding through to the control room. The recorder in the control room was an old Ampex rack-mounted 2" 4-track machine...yes, FOUR-track. Recording at 15ips on 2" of tape yielded some incredible sound quality...think about later machines that squeezed 24 tracks of material on the same 2" of tape. In 1981 someone from Ampex contacted us and gave the group a new 1/4" 4-track so they could get the 2" 4-track for their museum. Seeing as how Ampex changed hands since then I wonder what ever happened to those vintage machines.
People used to buy those old Ampex machines just to get the tube pre-amp electronics...nice warm sound, pleasant distortion (when you wanted it), and no harsh digital clipping.
"We make our world significant by the courage of our questions and by the depth of our answers." Carl Sagan
Analog is better for Pro-audio because tracks are ADDED together. The high frequencies really get mangled when you add a bunch of digital representations together. Isn't that audiophile 101? Poser.
"And of course tape can be driven to +9dB recording levels in some cases, but a digital system will clip hard at 0dB."
That's is what it's all about: use.
Theoretically you can absolutely duplicate an analog recording sound in a way that no human can tell the difference. You can at least duplicate it in a way that recreates it when you transfer it to a CD later; this is necessarily true seeing as they're both are just accessible bits.
The thing is, how hard is it to do this, at least, right now? Ridiculously hard. There's no DirectX plugin yet, and it's a long, long, long, long way off.
Though analog can be expensive and tape is finnicky in it's own way, if you're looking for an "analog sound"(as you see written on oh so many digital devices) you'll find that a computer + sound card + pro tools is much more finnicky and/or expensive.
Not to mention that it's not nearly the same to work in front of a computer as it is to work in front of a reel to reel. Being 'virtualized' in the screen is a much different experience than merely being plugged in to a reel to reel. For an artistic process(I hope you're making art, otherwise I'll probably have a minimal interest in your music) these things are important for what you want to do. Even if digital offered a total simulation of analog, it's still just a simulation. Although a lot of Slahdot readers most likely apply a sort of functionalism to a lot that they do, artists are not all total functionalists, and for good reason. The non-functional difference between a simulation and the real thing can still be quite important. Even if just in the process. I don't like quoting it, but "the medium is the message" can be very true for many works of art.
...but you'll never beat that infinitely variable and infinitely dividable quality of analog.
Um...if memory serves, magnetizable objects are made up of a bunch of little regions, called "magnetic domains," that are like little magnets. (A magnet just has the magnetic fields of the domains all lined up rather than in random directions.) These domains aren't infinitely small, just as photographic film suffers from grain. (Hence the cranking up of tape speed for higher fidelity, increasing the domains passed over per unit time.)
Also, it's not clear to me that an analog medium can be written to or read with infinite accuracy--but I hope someone more knowledgeable than I am can expound on that.
Much more happens in a tape recorder than just the addition of noise. There is something called warping, that is inaccuracies in the speed the tape passes the playback/recording heads. There is crosstalk, that is mixing of audio across several channels. There is also the fact that the signal is being recorded with a bias to bring it into the most linear part of the tape. And then there's the most noticable effect: Compression. When you over-record a tape, you get compression, that is reduction in audio levels compared to the original levels. A digital recorder will just clip, sounding horrible. A tape recorder will do it more gently. Modern musicians and technicians are very fond of what's described as vintage sound. I need just mention the UREI LA-2A compressor, an opto-electric tube compressor, used on numerous recordings. I dare bet anyone who has ever listened to recorded music has heard the handywork of that machine, or it's digital emulations. I love the sound of warping, especially when it comes from record players. It's what I call a becoming untunedness or unstability in pitch. Creates a warm fuzzy feeling inside of me, at least. Tube equipment is popular for grunching things up a bit, the newest Korg synthesizer model, a purely digital machine in all other aspects, has a tube stage for adding an edge to the sound. Guitarists unanamously agree that tube amplifiers give the best sound... I can go on and on... But it's a part of musicians' culture, basically.
Almost every oscillator out there uses a quartz crystal connected to a 74HCU04 inverter and a couple of ceramic capacitors. Such a thing is an oscillator, and it has very good long-term accuracy, but it has atrocious jitter, on the order of tens of nanoseconds. A good implementation would have the quartz crystal, a common-base amplifier with PNP transistors, and its own regulated power supply, with the squaring of the signal being taken care of by a comparator or, in a pinch, an inverter. A good implementation can have jitter below 1ps RMS, or looked at another way, -120dBc phase noise 100Hz from the carrier.
If you are going to publish at 44.1 you should probably record at 88.2 or 176.4, but asynchronous sample rate conversion has pretty good performance even for arbitrary conversion ratios. Look at the datasheet for AD1896, it can convert 96 to 44.1 with THD+N way down to -120dB or better. You could probably do better than that converting in software slower than realtime.
- Irony describes a result that is the opposite of what would commonly be expected under the circumstances.
- From that definition, you can see that there must be a common expectation in the first place. If an event happens that is merely coincidental or unrelated to the circumstances, it is "unlikely" or maybe "unfortunate" but not ironic. Even if something is coincidental in a regrettable, cynical, extreme, or unusual way, that does not make it ironic.
- Example 1: Rain on your wedding day -- regrettable, but your wedding day has nothing to do with the weather. Not ironic.
- Example 2: Running off with the best man on your wedding day. Ironic.
- If an event is appropriate given the circumstances, it is "fitting" or "apropos," not ironic. Even if something is fitting in a clever or unusual way, it cannot be ironic. In fact, apropos and ironic are more or less antonyms.
- Example 1: A traffic jam when you're already late -- something that just makes a bad situation worse is appropriate to the circumstance. Not ironic.
- Example 2: A traffic jam on a newly-opened expressway. Ironic.
So technically, I must say that no, the event you mentioned is not ironic but is better described as...[ ] extremely unfortunate
[ ] weirdly coincidental
[X] amusingly apropos
[ ] oddly fitting
[ ] poetic justice
and I hope you find this post useful.
Guest wrote:
No, you take the arctangent of each sample to simulate soft clipping. They are not the same.
Soft clipping is a generic word for an end result. Using an arctangent waveshape on a waveform is a means to this end; it will "get" you soft clipping, though it's only a rough "simulation" of what a valve amp or tape deck does. Simulating a particular flavor of soft clipping takes a lot more computing power, but if you just want to avoid harshly clipping your audio's peaks, arctangent is a nice-sounding way to do it.
Timing is also much more important with audio than with video. People, for whatever reason, are not sensitive to timing jitter in a video signal, but are easily able to hear phase noise in a digital recording. Video uses faster clocks than audio, but their clocks are not as good (and don't need to be).
Yep. And sampling @96kHz gives you response up to 48kHz (in theory, though in principal it the last parts suffer from intonation problems if there's any jitter at all to your sampling or reproduction clock.
Frequency distribution is a nice sharp spike at n Hz...
| |
| |
|_______|__
n
So we'll sample at 2n Hz
_ _
/ \ / \ - forgive the ascii art and
\_/ \_/ pretend that was a sine wave
_|__ __|__ - sampled at 2n Hz
| |
We'll even pretend that the studio gear sampled it perfectly (no clock jitter) since that gear
is likely pretty damned good. So our digital signal is +1,-1,+1,-1 just like it should be
But now we play it back on a cheapo walkman that doesn't have a perfect clock, so what it synthesizes is
_|_ ___|_ _
| |
and after filtering, the analog signal it produces now looks like this
_ _
/ | _/ | - again, forgive the ascii art,
|_/ |_/ but clearly it has steeper
sections and shallower ones
so it's no longer a pure tone
So the frequency distribution now looks like
|
| |
|_____|_|_|
n
it has some frequency content to both sides of the 'real' signal (how much and how far depends on the amount of jitter present). Obviously, the signals very close to the nyquist limit suffer most from this - the lower pitches get to average the wave-shape out over multiple samples, so they will not spread out as much in the freuency domain if a point is a little off in time. But this is why the nyquist limit is not the whole story. Along with the fact that no filter is a completely sharp dropoff, this is why CD's lowpass filter to <20kHz, not at the 22050Hz Nyquist limit).
Sampling beyond 96kHz is not (yet, anyway) mainstream gear. So I think the grandparen't claim that digital equipment works to 48kHz, but has a hard time as that limit is approached is pretty fair - that's the theoretical limit (for prosumer-grade stuff), and in practice it will have trouble near the edge.
The Matrix is going down for reboot now! Stopping reality: OK. The system is halted.
And what ears will you use for those frequencies above 20KHz? Most adults in Western societies are lucky to hear to 15KHz. Get your hearing tested sometime, and hit some books too, I wish you knew the difference between a sample rate and a signal frequency.
The big change was the advent of delta-sigma D/A converters. Old D/A converters didn't work very well because they tried to directly convert digital PCM sound. Well this meant that to get 16-bit sound (CD quality) you had to have a converter that could produce 65,536 discreet voltage states. When you ocnsider this was over a range of two volts or so, you see why this was a huge problem. It made a very harsh sound that people didn't like.
Well the answer to the problem came from high power vairable speed electric motors. With those you also need to regulate the voltage and it turns out to be a bitch to design a stepped system that does so properly, cheaply, and efficiently. So what they did is use a different kind of control, called pulse wave modulation, PWM. What you do is take a high frequency digital (square) wave that alternates between maximum and minimum voltage, and just vary the duty cycle to give you the level of power you want. The more pwoer you want in a given direction, the more pulses that direction you have. Turns out to be real efficient, and easy to make. Only downside is it makes the device whine at the frequency of the wave.
Well, this could be applied to audio as well. Simply vary the rate of pulse to control the output signal. The whining is solved by using a wave of sufficiently high frequency that it exceeds human hearing and speaker capabilities (usually in the MHz range). This proves to work extremely well, and eliminate the problems with digital sound. All corrent D/A converters I'm aware of use this method. You'll see it adverised at 1-bit DAC sometimes.
Some systems, like SDSD, forgoe the conversion process and store PWM directly.
I am not sure about Emtec but BASF is not former AGFA!!! Badische Anilin und Soda Fabrik (BASF) is German chemical and plastics manufacturing company originally founded in 1865. Agfagevaert Gruppe, Dutch Agfa-gevaert Groep, is German and Belgian corporate group established in 1964 in the merger of Agfa AG of Leverkusen, W.Ger., and Gevaert Photo-Producten NV of Mortsel, Belg. The merger established twin operating companies, one German (Agfa-Gevaert AG) and one Belgian (Gevaert-Agfa NV, which in 1971 became Agfa-Gevaert NV). Controlling interest in the group was purchased by Bayer AG in 1981.
(Or am I missing something?)
I used to use quantegy (quantigy? formerly Ampex) tapes in my ADAT machine, a digital 8-track recorder that records 42 minutes of 8 channel, 48 khz digital audio on what is basically an analog VHS tape. Of coure, ADAT tapes aren't the same as reel-to-reel tapes- the packaging is different. I suspect that division will still be running for quite a while, as digital ADAT tapes tend to have better compatibility across machines than analog reel-to-reel. Still I have a hard time believing that not a single studio is going to record anything (analog) on (analog) tape anymore. Not because I don't think harddisk recording hasn't caught up with analog technology, but because the natural compression of tape gives quite a pleasant harmonic distortion to the sound recorded on it. Also, harddisks crash and burned media gets unreadable. For longer-term audio storage, tape is still the medium of choice. Given this, what's the alternative to reel-to-reel tape?
Visit http://ringbreak.dnd.utwente.nl/~mrjb/growingbettersoftware to download your free copy of the book
I've recorded in digital and "old school" tape studios.And they both kicked ass.How?The smart digital studio guys run the old tube preamps and eq's.That way you get that nice analog warmth going in,and the ease of digital editing afterwards.It's the best of both worlds.Now let us pray that those few plants making tubes don't go out of business,or all us musicians are REALLY screwed.
ACs don't waste your time replying, your posts are never seen by me.
Looping and recursion are fundamentally the same thing...
Aho's book uses the same joke for "loop" in the index.
Perhaps, I should have said the joke was more of an example of iteration than recursion. Certainly, the repetition of the joke is.