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Building a 70s system??

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Re: Building a 70s system??

Postby Hugh Robjohns » Wed May 27, 2020 5:38 pm

MOF wrote:I can’t remember the last time we rolled test tone and bars when I’ve been working with a crew, although maybe bars and internal tone are recorded by the cameraman/woman.

There's no point in recording calibration signals if there's nothing to calibrate... and in an all-digital signal chain there is nothing to calibrate, by definition! The digital value ascribed to each sample is passed down the digital chain without error or alteration.

I'm all for the maintenance of technical skills, but they need to be informed skills, too! The 'old ways' are necessarily the appropriate ways, let alone the best ways for modern workflows.

In analogue days it was necessary to record bars and tone at source because there was a distinct possibility of video and audio signal level alignment errors between the record and playback machines, and down the analogue signal paths and transmission lines.

But with digital recorders and digital connections generally, whatever digital value is ascribed to each sample will absolutely be passed down the digital chain without error or alteration... hence no source alignment signals are required. The system either works transparently or it's broken... but there are no alignment drifts to worry about or correct.

Having said that, if you are providing an analogue sound feed to a digital recorder there is very obviously a requirement to calibrate your signal level with the recorder's A-D converter. And, of course line-up tone is the ideal tool for that purpose, to make sure that when you think you're sending your nominal reference level -- 0dBu / PPM4, or whatever -- the digital recorder sees that as -18dBFS (or whatever nominal calibration you're working with).

But -- and this is important -- that tone calibration process is an entirely local one between your analogue output and the digital recorder's analogue input. There is no point whatsoever in actually recording the tone as it would convey no useful information and serve no useful purpose to anyone else.

The entire purpose of performing your local alignment is to abide by the assumed knowledge that your audio signals comply with the defined standards which will then be maintained absolutely through the rest of the digital chain.

So, tone, calibrate, start work. No need for recording tones, and definitely no need for recording colour bars... and no need for aligning convergences or colourimitries, or frequency responses, or any of that other stuff we had to do in the days of analogue tube cameras and analogue lines! It's a different world with different technologies and so different operational practices are required.
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Re: Building a 70s system??

Postby MOF » Wed May 27, 2020 6:21 pm

Thanks Hugh, sorry I could have saved you all that explanation, I did already know, although I’m fairly sure I’ve seen digital cameras that have bars and tone capabilities or maybe it’s further back than I like to think it was. :D
Edit, thinking about it it will be for down the line live use even though it won’t be necessary to actually line up (I can’t think that any links are analogue these days), just to show the vision mixer that there’s a camera waiting to feed pictures.
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Re: Building a 70s system??

Postby Hugh Robjohns » Wed May 27, 2020 8:04 pm

MOF wrote:Thanks Hugh, sorry I could have saved you all that explanation, I did already know...

Good to know, but you're not the only one reading the forums and I'm sure others might have wondered and not known.

...although I’m fairly sure I’ve seen digital cameras that have bars and tone capabilities...

A handy facility if you're likely to need to connect a digital camera to an analogue line, but these days really just to establish and maintain an identity signal to show when a line is busy and has a valid source.... As you say. :-)
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Re: Building a 70s system??

Postby Eddy Deegan » Thu May 28, 2020 12:56 am

Hugh Robjohns wrote:
Also, I suspect I'll find what I'm really nostalically yearning for are the ears that I had back in the 70's.

That and all the things you got up to back in the 70s... :oops: :silent:

In my case the late '80s/early '90s but the point is well taken and in no way lost on me!
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Re: Building a 70s system??

Postby Arpangel » Thu May 28, 2020 7:37 am

What is this thing called digital? I still don’t understand it. I can picture magnetised oxide on a tape, fluctuating voltages on magnets in a tape head responding to the tape, amplified, that finally come out of a cone that’s attached to a coil and magnet, but digital? WTF?
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Re: Building a 70s system??

Postby Eddy Deegan » Thu May 28, 2020 11:27 am

Arpangel wrote:What is this thing called digital? I still don’t understand it. I can picture magnetised oxide on a tape, fluctuating voltages on magnets in a tape head responding to the tape, amplified, that finally come out of a cone that’s attached to a coil and magnet, but digital? WTF?

Imagine using an oscilliscope to measure the signal over time.

If we were to freeze time at any point, the oscilliscope would be showing the signal strength at that moment. If silence has a value of 0 and the maximum signal strength possible to store on tape is 1000 then you can make a note of signal strength as represented on that imaginary scale (so if the dot was half way up the oscilliscope screen, the value would be 500 in the above case).

Now imagine making a note of the number 44,100 times/sec as the signal is being played, and imagine the range of possible numbers extended so that silence is 0 and the maximum signal strength is 65,535.

Storing that stream of numbers is your digital representation of the continuous signal created by playing the tape. Stuff may have occurred between each point we made a note of a number, and the point we made a note of the next one, but it doesn't matter because we've decided that 44,100 times a second results in an accurate enough record of what happened for our purposes.

Now use that series of numbers as an input to a device that generates an output voltage. The output voltage is refreshed 44,100 times a second, based on mapping the values from 0 - 65,535 to the desired range.

Assuming you send the numbers into that device at the same speed you recorded them at, if you hook a speaker to that output voltage you'll hear something remarkably similar to the original tape signal. Close enough, in fact, that a lot of people won't really notice the difference - hence deemed good enough for CDs.

Now repeat the entire exercise but 96,000 or 192,000 times a second, and with the range of numbers going from 0 - 9,223,372,036,854,775,807. This would be a 24-bit system at 96k or 192k.

In all cases, the result is that an analogue voltage is generated for a speaker to move in response to. The difference between analogue and digital is the nature of the source from which that voltage is derived.

As you now have the series of numbers, you can do fancy maths on them and replay them into the device that generates the output voltage again. Depending on the fancy maths, any possible audio process can be achieved in theory. The trick is coming up with the maths :-)
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Re: Building a 70s system??

Postby Sam Spoons » Thu May 28, 2020 11:33 am

Alternatively you could just call it 'magic' and be done with it :D
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Re: Building a 70s system??

Postby Hugh Robjohns » Thu May 28, 2020 11:54 am

Eddy Deegan wrote:...if you hook a speaker to that output voltage you'll hear something remarkably similar to the original tape signal. Close enough, in fact, that a lot of people won't really notice the difference - hence deemed good enough for CDs.

It's not just 'remarkably similar' or 'close enough'... it is exactly the same as the original signal (provided you accept the requirement to band-limit the original signal in the first place).

In so many ways, digital audio is actually nothing more than a wide-bandwidth AM radio system... but without the broadcast transmission bit!

:D :ugeek:
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Re: Building a 70s system??

Postby The Elf » Thu May 28, 2020 10:10 pm

Hugh Robjohns wrote:
Eddy Deegan wrote:...if you hook a speaker to that output voltage you'll hear something remarkably similar to the original tape signal. Close enough, in fact, that a lot of people won't really notice the difference - hence deemed good enough for CDs.

It's not just 'remarkably similar' or 'close enough'... it is exactly the same as the original signal (provided you accept the requirement to band-limit the original signal in the first place).

In so many ways, digital audio is actually nothing more than a wide-bandwidth AM radio system... but without the broadcast transmission bit!

:D :ugeek:
Hugh, Hugh, Hugh... Tut tut!

Of course it's not the same - it's 'cold' and 'digital'.

Tch! Schoolboy error! ;) :D :lol:
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Re: Building a 70s system??

Postby MOF » Thu May 28, 2020 10:18 pm

Of course it's not the same - it's 'cold' and 'digital'.

You need to pass it through analogue tape and valve amplifiers first, plugins if you don't have the real things, as a second best.
The ultimate would be to pay for it to be cut to vinyl and then play it through a valve amp.:lol:
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Re: Building a 70s system??

Postby Eddy Deegan » Fri May 29, 2020 3:14 am

Hugh Robjohns wrote:
Eddy Deegan wrote:...if you hook a speaker to that output voltage you'll hear something remarkably similar to the original tape signal. Close enough, in fact, that a lot of people won't really notice the difference - hence deemed good enough for CDs.

It's not just 'remarkably similar' or 'close enough'... it is exactly the same as the original signal (provided you accept the requirement to band-limit the original signal in the first place).

In the case of tape, I'm assuming that the limited frequency reproduction capabilities of the physical medium result in that 'exactly the same' output. I'm not sure I fully understand what 'band-limit' means in this context though.

My assumption (I'm making a few!) is that the 'resolution' of a tape would be the smallest coherent unit of time that can be represented meaningfully on the grains of oxide as they pass over the read head, which I would intuit as being higher than audible frequency (and would vary depending on the ips?) although in a practical sense that might be purely academic.

To put it another way If the input signal wasn't tape, but the output of a sawtooth oscillator processed by a sweeping analogue filter then I would imagine that a lot of information is lost between samples at 44.1kHz although that information is probably of no real consequence. However, the waveform recreated by a DAC would not be exactly the same as the input was even if you couldn't hear the difference. I'd applied the same logic to tape although perhaps the physical constraints of tape make that distinction moot.

I'm sure I'm missing something here, but I am interested in understanding more. Would you mind elaborating on this a bit Hugh?
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Re: Building a 70s system??

Postby innerchord » Fri May 29, 2020 3:43 am

That's why I hate CDs. I carefully crafted a whole host of 22kHz to 26kHz filter sweeps and effects for one of my pieces, but damn CDs don't allow it to be heard. If we'd all been using SACDs by now, it'd be fine and we'd all be able to hear it. Perhaps I should release it anyway and call it "Lost Between The Samples"? Come to think of it, I never could get the mix finished because my dog was quite ill at the time.
:crazy: :wtf:

Oh yes, I remember the 70's!
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Re: Building a 70s system??

Postby Arpangel » Fri May 29, 2020 5:56 am

Sam Spoons wrote:Alternatively you could just call it 'magic' and be done with it :D

Thanks Sam, I’ll go for that...

:D
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Re: Building a 70s system??

Postby Kwackman » Fri May 29, 2020 9:21 am

Eddy Deegan wrote:My assumption (I'm making a few!) is that the 'resolution' of a tape would be the smallest coherent unit of time that can be represented meaningfully on the grains of oxide as they pass over the read head

Wouldn't the head be the limiting factor of a tape based system, not the tape?
The air gap that the read head will be a specific width and therefore would "smudge" all the grains of oxide in that area into an average value?
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Re: Building a 70s system??

Postby Hugh Robjohns » Fri May 29, 2020 10:30 am

Eddy Deegan wrote:In the case of tape, I'm assuming that the limited frequency reproduction capabilities of the physical medium result in that 'exactly the same' output. I'm not sure I fully understand what 'band-limit' means in this context though.

In reality, everything is 'band-limited', meaning it all has a usable working frequency range (or bandwidth) and cannot work with signals outside that range. This is true of microphones, tape recorders, mixing consoles, loudspeakers, ears and so on...

However, for all of these things, the transition from the 'pass-band' -- the region where it works as intended -- to the 'stop-band' where it doesn't work at all, is usually relatively gentle (6 or 12dB/Octave, typically) -- at least initially -- and so may extend over several octaves.

In contrast, the transition for a basic 44.1k digital system has to be the infamous 'brick-wall' filter where it goes from working perfectly at 20kHz, to being completely deaf at 22.05kHz (in theory, at least).

So one of the compromises that has to be accepted when using digital is that very hard limit on the frequency range that can be accommodated -- hence, band-limited.

Of course, if you choose to run the digital system at 96kHz its band limits will be far beyond the natural limits of (almost) any microphone / mixer / loudspeaker / ear anyway, so we won't notice...

My assumption (I'm making a few!) is that the 'resolution' of a tape would be the smallest coherent unit of time that can be represented meaningfully on the grains of oxide as they pass over the read head

The size and proximity (density per unit length, in other words) of the magnetic domains determines the signal/noise ratio and equates broadly to the wordlength of a digital system. The width of the gap in the replay head defines the upper frequency bandwidth limit (known as the 'extinction frequency, actually).

If... the output of a sawtooth oscillator [is] processed by a sweeping analogue filter then I would imagine that a lot of information is lost between samples at 44.1kHz although that information is probably of no real consequence.

No, nothing is lost at all... and this is a fundamental aspect of sampling theory that eludes many people -- and understandably so as it is a difficult concept only really provable with complex maths. Although it is easy to demonstrate in practice.

For something to happen between samples, that something must inherently involve frequencies above half the sample rate -- but we've already accepted in our contract to
go digital that those frequencies aren't allowed in to the system in the first place by virtue of the band-limiting policy that is part and parcel of working with any sampled system.

If those very high frequencies were allowed in (usually because the input filtering was omitted), the reconstruction process gets confused and they get translated to much lower frequencies -- a process we call aliasing. It's that tinkly distorted effect you hear if you and record a voice message into one of those electronic greeting cards and then play it back!

However, the waveform recreated by a DAC would not be exactly the same as the input was even if you couldn't hear the difference. I'd applied the same logic to tape although perhaps the physical constraints of tape make that distinction moot.

Perhaps the more obvious example to consider is a pure square wave which, as we all know, contains odd harmonics that go on forever (and certainly way above 20kHz). So if you have an analogue synth that generates a perfect square wave (none can, of course, but many will get very close), and you look at that output signal on a 'scope it looks lovely and square wave-ish. Then you record it into a digital system, play it back and look at it on the 'scope again, and it comes out looking all wriggling and much less square-wave-ish.

And all the digital-haters shout out, there you are... it's missed stuff and it's not accurate!

But actually, all they're seeing is the effect of the analogue brick-wall filters at the input and output of the digital system -- the filters that restricts the bandwidth to slightly less than half the sample rate. If you remove the higher frequency harmonics from a square wave you get a wriggly square wave! Basic physics 101.

Of course, if you use a digital systems that samples at a much higher rate, many more of the odd harmonics will be included and the replayed square wave will look much more like the square wave generated by the analogue synth. But it won't sound any different because we can't hear any of the harmonics above 20kHz anyway.

But you can easily prove that the digital system handles everything up to its prescribed frequency limit with sine waves. Feed in a 20kHz sine wave, and you'll get a 20kHz sine wave out. (if you're trying to do this with 'scope's on computer screens be very wary of their own display sampling limitations, though!

I'm sure I'm missing something here, but I am interested in understanding more. Would you mind elaborating on this a bit Hugh?

The very best video demonstration of all this that I've found is by Monty on Xiph.org. it should be compulsory viewing!

https://ftp.osuosl.org/pub/xiph/video/Digital_Show_and_Tell-360p.webm

...And Monty actually demonstrates the square wave thing about 17 minutes into that video! :D

And there's my own effort at a written explanation here:

https://www.soundonsound.com/techniques/digital-problems-practical-solutions
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