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Analog vs Digital Emulations - A culinary comparison
Old 17 hours ago | Show parent
  #361
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Quote:
Originally Posted by vernier ➡️
Beyond Meat or Filet Mignon ...Which do you want?
Dude, eventually they'll be able to grow filet mignon in a lab. They're not quite there yet. It would be great for the environment.

So, there will be a third choice.
Old 16 hours ago
  #362
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🎧 5 years
Quote:
Originally Posted by Padano ➡️
Hi everyone. I’ve been following this forum forever and remember posting here back in the days.
Anyway I was following another thread about the ethereal fight between analog and digital emulation and I think I came up with a good analogy.

“You are a Chef at a restaurant and you are making a meal for your customers (the mix). You can either use more expensive fresh ingredients (analog gear) or a bunch of dehydrated powders and chemical additives (plugins).
Like modern plug-ins, some food additives can improve the results in better ways than old school methods, I’ll give you that.
But if you are telling me that the emulations sounds the same as the original hardware, it’s like saying a dish made with garlic powder has the same taste than one made with fresh garlic.
Just because you don’t hear the difference, doesn’t mean it’s not there.”

What you guys think about this? Does this at least clarify the ethereal dilemma?
TRANSLATION:

Hi guys, I got kicked off this forum a long time ago, but every now and then I come back and start a new account to put up asinine posts and have the biggest laugh ever as the page counts increase and everybody argues amongst themselves.

WHEN WILL YOU GUYS LEARN???
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Old 13 hours ago | Show parent
  #363
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🎧 15 years
Quote:
Originally Posted by monkeyxx ➡️
What sort of gear, tools, software do you use for generative composition? It's an area I'm interested in but under informed about. I have a friend that does it in Ableton Live, we might start a band of some sort, so I'd like to know more, if it's easy enough to speak about.
Go to loopop’s youtube page and search for generative - he does an excellent breakdown.

I use guitar pedals, hardware synths and a lot of plugins (filters, and delay lines) to create feedback loops which I then make new sounds out of. Without going into detail, it’s hard to generalize. I’m heading out for a short vacation, but will describe the process of something I’m working on now when I get back.
Old 12 hours ago | Show parent
  #364
How is it being said digital is a continuous waveform capture .....

It’s binary code derived from sampling the amplitude of voltages at the encoder no?

(Sampling ....as in an intermittent capture of values taken at a rate/frequency that is chosen to exceed our ability to perceive the missing parts of the “analog ”, ie, Nyquist )

I thought that was the essence of digital. It is an intermittent representation of continuos waveforms
Old 12 hours ago | Show parent
  #365
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🎧 5 years
"Getting back to what I thought that poster was saying- analog never had a chance to go in this direction when digital took hold. And there are lots of ways analog/continuous waveform capture method could have been utilized, even with digital (non-audio) elements." Yes, that is what I was getting at. Trying to imagine how that tech might have/could evolve.
Old 11 hours ago | Show parent
  #366
Gear Maniac
 
🎧 15 years
Quote:
Originally Posted by ElmoHope ➡️
Dude, eventually they'll be able to grow filet mignon in a lab. They're not quite there yet. It would be great for the environment.

So, there will be a third choice.
Soylent Green will be that choice.

They're already serving Soylent Yellow...those fake-meat burgers.
Old 11 hours ago | Show parent
  #367
Quote:
Originally Posted by Johnny Wagon ➡️
One of the really cool aspects of Blade Runner was that they showed how analog tech might have progressed. It seems with the advent of digital recording and playback that development of purely analog alternatives stopped completely(?). Let' imagine a system that uses both technologies to bring you affordable continuous waveform capture(analog) on a digitally controlled system. Maybe instead of tape, it uses a platter like a mechanical hard drive and the play and record heads "float" so there is no physical contact(again like a hard drive). Digital timing and location info is recorded along side to facilitate alignment with other platters and provide random access to recorded material whos playback could be arranged and controlled through your DAW. Similar to a sequencer I guess. You would be able to control speed and even direction of platter. Want more tracks? Just add more platters.
Without getting into which format is superior, it will give you an option for CWR..lol(Continuous Waveform Recording) with random access if PCM(Pulse Code Modulation) doesn't do it for you. I'm imagining similar if not better specs to a late model Studer 2" 24 track.
CWR, i love it....

we could be waaaaay better re: audio standards with the tech we have today
Old 9 hours ago | Show parent
  #368
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It's 09:27 and I'm already completely smashed out of my skull.
Old 7 hours ago | Show parent
  #369
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18 Reviews written
🎧 10 years
Quote:
Originally Posted by ghostman ➡️
How is it being said digital is a continuous waveform capture .....

It’s binary code derived from sampling the amplitude of voltages at the encoder no?

(Sampling ....as in an intermittent capture of values taken at a rate/frequency that is chosen to exceed our ability to perceive the missing parts of the “analog ”, ie, Nyquist )

I thought that was the essence of digital. It is an intermittent representation of continuos waveforms
Digital has bandwidth limitations beyond the range of human hearing. Guess what, so does analog recording.

So neither is, hm, 'perfection itself.'

But the fact remains that the modern digital recorders are orders of magnitude more 'perfect' than an analog recorder. Regarding the word that was defined earlier as 'fidelity.'

But analog 'sounds good' so now we're really spinning. Round and round as it starts all over again, as it opens wide with a filter on the lens.

If I can answer your post more specifically, what comes out of a DAC is indeed a continuous waveform, watch the [EDIT: mont montgomery] video when you get a chance!

Last edited by monkeyxx; 5 hours ago at 02:05 PM.. Reason: wild goose chase
Old 6 hours ago | Show parent
  #370
Quote:
Originally Posted by monkeyxx ➡️
Digital has bandwidth limitations beyond the range of human hearing. Guess what, so does analog recording.

So neither is, hm, 'perfection itself.'

But the fact remains that the modern digital recorders are orders of magnitude more 'perfect' than an analog recorder. Regarding the word that was defined earlier as 'fidelity.'

But analog 'sounds good' so now we're really spinning. Round and round as it starts all over again, as it opens wide with a filter on the lens.

If I can answer your post more specifically, what comes out of a DAC is indeed a continuous waveform, watch the Monty Hall video when you get a chance!
I hear you bro

Yeah I’d never venture either digital or analog are. “Perfect “

Neither Analog or digital are the thing itself .

I love both . I use both . IMO they compliment each other .

Re : continuous waveforms

A dac does indeed create a continuous waveform

Binary code to dac leading to voltage and phase leading to electromagnetic field , movement of magnets in speaker cone leading to movement of air

But the dac is not the wave capture part

The adc is the capture stage
And a adc does the reverse

Waveform to code via intermittent samples of voltage amplitude at a specified frequency

Hence the bandwidth you speak of

And in analog those voltages would magnetise tapes and cause a restructuring of the orientation of molecules via the electromagnetic field applied to the tape

Also imperfect

Both have their uses to this day IMO


That’s my understanding of all this . Thanks for the link

I’ll go and watch the vid now and see if I’ve just talked a load of tosh lol
Old 6 hours ago | Show parent
  #371
Quote:
Originally Posted by monkeyxx ➡️
Digital has bandwidth limitations beyond the range of human hearing. Guess what, so does analog recording.

So neither is, hm, 'perfection itself.'

But the fact remains that the modern digital recorders are orders of magnitude more 'perfect' than an analog recorder. Regarding the word that was defined earlier as 'fidelity.'

But analog 'sounds good' so now we're really spinning. Round and round as it starts all over again, as it opens wide with a filter on the lens.

If I can answer your post more specifically, what comes out of a DAC is indeed a continuous waveform, watch the Monty Hall video when you get a chance!
Lol !
I don’t know if I watched the right vid

I watched a vid about the probability of picking a goat or a car

Which I’m not sure how probabilities relate to our dac/adc convo

But what I would say about the monty hall problem as this vid explained it

They didn’t factor in randomness

In order to see the trend of a probability evidenced in an experiment you need to have a sufficiently large sample size to reduce the inevitable influence of random results

Ie . For one person making one guess

The probability of the problem presented is almost irrelevant

If you have the option to make a hundred guesses then the probability becomes more influential

So I’d stick with my first guess if I had one guess

If I had a hundred I’d go with the probability strategy

Did I even watch the right vid lol
Old 6 hours ago | Show parent
  #372
Gear Guru
 
monkeyxx's Avatar
 
18 Reviews written
🎧 10 years
Quote:
Originally Posted by ghostman ➡️
I hear you bro

I’ll go and watch the vid now and see if I’ve just talked a load of tosh lol
It all makes good sense! You have a good understanding.

In delta sigma conversion, decimation filters are used in the AD stage, and reconstruction filters in the DA stage, to filter out frequencies above half the sampling rate, the Nyquist frequency.

What goes in, and what comes out, are 'very accurate' representations of every audible frequency within that bandwidth. "Information" between samples is also captured, due to the "rules" of physics / audio. For example, an intersample peak that can clip your DAC by exceeding full scale. Amplitude, frequency, time information, and so on, are all 'there'. There is only "one possible waveform" that matches the data, which happens to be the same waveform that was input, minus what was filtered out.

Analog recorders typically have loads of nonlinear circuits and components, so what comes out is a beautifully distorted version of what you put in, with a scrambled phase response, lumpy frequency response, all sorts of noise and harmonic distortion, and so on. We likey.
Old 6 hours ago
  #373
Gear Guru
 
monkeyxx's Avatar
 
18 Reviews written
🎧 10 years
Sorry I sent you on a wild tangent, it's Mont Mongomery, oops X-D

https://xiph.org/video/vid2.shtml
Old 5 hours ago | Show parent
  #374
Quote:
Originally Posted by monkeyxx ➡️
It all makes good sense! You have a good understanding.

In delta sigma conversion, decimation filters are used in the AD stage, and reconstruction filters in the DA stage, to filter out frequencies above half the sampling rate, the Nyquist frequency.

What goes in, and what comes out, are 'very accurate' representations of every audible frequency within that bandwidth. "Information" between samples is also captured, due to the "rules" of physics / audio. For example, an intersample peak that can clip your DAC by exceeding full scale. Amplitude, frequency, time information, and so on, are all 'there'. There is only "one possible waveform" that matches the data, which happens to be the same waveform that was input, minus what was filtered out.

Analog recorders typically have loads of nonlinear circuits and components, so what comes out is a beautifully distorted version of what you put in, with a scrambled phase response, lumpy frequency response, all sorts of noise and harmonic distortion, and so on. We likey.
...fascinating stuff.

im a nerd for the field of electromagnetics, physics and audio engineering and how music is both truly science and art.....art that couldn't exist without the science . i'll learn as much as i can on anything to do with this stuff so yeah....nice one

"the only one possible waveform " that matches the data is an interesting concept.

Andrew Schepps gave a brilliant lecture about the effect of audio quality in streaming music suggesting that our current sampling rates of digital audio for streaming are causing listening stress on a global scale as our brains interpolate the missing info in the music (after all......all perceptions of audio are psycho-acoustic and dependent on the brain, our cns and the ear)


https://www.youtube.com/watch?v=SXbH-yzGNfg

He talks about the analogy of how if you begin to remove vowels from words in a text our brains can still decipher the words.....up to a very impressive point...and then it just looks like a bunch of random letters. He suggests this is what we are doing with music everyday and its exhausting us . I think he has a very very important point tbh. Have a watch if you get time

re: tape, there's a nice little trick i saw shawn everett do on MWTM.

To get some more control over all that lumpy frequency response and distortion ...if you like to warm anything up with tape....

Sometimes, I use actual tape cassette recorders sometimes to send soft synths to.....

you cut the synth to tape.....
record back into the daw

create an eq profile of your original soft synth in the daw pre-tape.

Use a match eq across your tape-bounced synth.

Match the new eq curve from your taped synth to bring its hz-response to that of the original soft synth....save that curve as a preset.

Thats the eq / harmonic response of your tape cassette recorder saved in the daw.


Now....do another pass cutting the soft synth to tape, but with the new eq curve already on there...

...record that back to the daw and now you get a taped copy with its original pre-tape curve intact.

Very geeky but more fun than throwing on a kramer / j37 waves tape plug in.

and....

re:

Quote:
Originally Posted by monkeyxx ➡️
Sorry I sent you on a wild tangent, it's Mont Mongomery, oops X-D

https://xiph.org/video/vid2.shtml
...ha . ok well ill return the favour.

what door are you picking?

https://www.youtube.com/watch?v=mhlc7peGlGg
Old 5 hours ago | Show parent
  #375
Gear Guru
 
UnderTow's Avatar
 
🎧 15 years
Quote:
Originally Posted by ghostman ➡️
Andrew Schepps gave an interesting lecture suggesting that our current sampling rates of digital audio for streaming are causing listening stress on a global scale as our brains interpolate the missing info in the music (after all......all perceptions of audio are psycho-acoustic and dependent on the brain, our cns and the ear)
Are you sure he wasn't talking about lossy compression? There is no missing info due to sampling rates. (Assuming you are using 44.1KHz or above).

Alistair
Old 5 hours ago | Show parent
  #376
Quote:
Originally Posted by UnderTow ➡️
Are you sure he wasn't talking about lossy compression? There is no missing info due to sampling rates. (Assuming you are using 44.1KHz or above).

Alistair
....apologies, yes in the lecture he's speaking specifically on lossy codecs.

the whole lecture is great

he begins on codecs at 20mins if you're interested

re:

"There is no missing info due to sampling rates. (Assuming you are using 44.1KHz or above)."

is there somewhere i can read up on a bit more information to this.

I was interested in your post saying that digital is continuous .....i defo dont have this understanding...so it seems there's something to learn
Old 5 hours ago | Show parent
  #377
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signalpudding's Avatar
 
Quote:
Originally Posted by ghostman ➡️
I was interested in your post saying that digital is continuous .....i defo dont have this understanding...so it seems there's something to learn
https://en.wikipedia.org/wiki/Nyquis...mpling_theorem

"The Nyquist–Shannon sampling theorem is a theorem in the field of signal processing which serves as a fundamental bridge between continuous-time signals and discrete-time signals. It establishes a sufficient condition for a sample rate that permits a discrete sequence of samples to capture all the information from a continuous-time signal of finite bandwidth."
Old 5 hours ago | Show parent
  #378
Quote:
Originally Posted by signalpudding ➡️
https://en.wikipedia.org/wiki/Nyquis...mpling_theorem

"The Nyquist–Shannon sampling theorem is a theorem in the field of signal processing which serves as a fundamental bridge between continuous-time signals and discrete-time signals. It establishes a sufficient condition for a sample rate that permits a discrete sequence of samples to capture all the information from a continuous-time signal of finite bandwidth."
great thanks.....

i can see this is gonna be a couple of days of study for me....im looking at the fourier transform right now.

First impression:,

is this is a theory that rests on agreeing with the hypothesis that the waveform (in complex music) being sampled has a finite bandwitth, which seems to me one must therefore immediately discount any idea that hz above and below the human ear have no response on us.

im open to it though... ..study time

Good stuff, thanks
Old 5 hours ago | Show parent
  #379
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Quote:
Originally Posted by ghostman ➡️
great thanks.....

i can see this gonna be a couple of days of study for me....im looking at the fourier transform right now.

First impression:,

is this is a theory that rests on agreeing with the hypothesis that the waveform (in complex music) being sampled has a finite bandwitth, which seems to me one must therefore immediately discount any idea that hz above and below the human ear have no response on us.

im open to it though...

Good stuff, thanks
Yes, the theory only works on band limited signals.

It's important to note that every signal we use in audio is band limited and always has been. That includes every analog recording format ever, every bit of analog signal processing ever, and every microphone and speaker ever. Also, most importantly, human hearing.

Even if they weren't there has not been a single bit of evidence to show that sounds we can't hear have any effect on us outside of things like weaponizing ultrasonic frequencies. Frequencies outside of our hearing range can definitely have a use in digital signal processing, but that's a totally different subject than capture and reproduction.
Old 4 hours ago | Show parent
  #380
Gear Guru
 
UnderTow's Avatar
 
🎧 15 years
Quote:
Originally Posted by ghostman ➡️
....apologies, yes in the lecture he's speaking specifically on lossy codecs.

the whole lecture is great

he begins on codecs at 20mins if you're interested
I think I've seen it.

Quote:
"There is no missing info due to sampling rates. (Assuming you are using 44.1KHz or above)."

is there somewhere i can read up on a bit more information to this.

I was interested in your post saying that digital is continuous .....i defo dont have this understanding...so it seems there's something to learn
The Monty Montgomery video that monkeyxx posted is one of the clearest videos on the topic: https://xiph.org/video/vid2.shtml

Quote:
is this is a theory that rests on agreeing with the hypothesis that the waveform (in complex music) being sampled has a finite bandwitth, which seems to me one must therefore immediately discount any idea that hz above and below the human ear have no response on us.
There is no lower bound. Sampling goes down to 0 Hz. But there is indeed an upper bound. The signal must be bandwidth limited (A low-pass filter basically). And we choose the sampling rate and filters so that this filter sits above the highest frequency we can hear. So it isn't so much about the music itself having a limited bandwidth. It is that we actively limit the bandwidth with a filter just above the audible range so that it fulfils the Nyquist requirements.

So yes, we are removing frequencies above a certain point but that point is chosen to be above and beyond what our hearing can perceive. So we are not removing sound. Just inaudible stuff.

Alistair
Old 4 hours ago | Show parent
  #381
Quote:
Originally Posted by signalpudding ➡️
Yes, the theory only works on band limited signals.

It's important to note that every signal we use in audio is band limited and always has been. That includes every analog recording format ever, every bit of analog signal processing ever, and every microphone and speaker ever. Also, most importantly, human hearing.

There has not been a single bit of evidence to show that sounds we can't hear have any effect on us outside of things like weaponizing ultrasonic frequencies. Frequencies outside of our hearing range can definitely have a use in digital signal processing, but that's a totally different subject than capture and reproduction.
Thanks bro I get you . This is all very interesting and the transform theory is new to me

I’m gonna study all this tonight when I get back home

I studied animal behaviour at uni and the influence of external / environmental factors on animals is of personal interest

(Hence my question about limited bandwidth)
Old 4 hours ago | Show parent
  #382
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signalpudding's Avatar
 
Quote:
Originally Posted by ghostman ➡️
Thanks bro I get you . This is all very interesting and the transform theory is new to me

I’m gonna study all this tonight when I get back home

I studied animal behaviour at uni and the influence of external / environmental factors on animals is of personal interest

(Hence my question about limited bandwidth)
If you can show definitively that ultrasonic frequencies in the environment are affecting us physically some way you'll make a lot of audiophiles very happy, a lot of conspiracy theorists very happy, and I'd be happy to get in line with that new info. Haha. So far though the science just hasn't found anything. But I'm always open to new knowledge.
Old 4 hours ago | Show parent
  #383
Quote:
Originally Posted by signalpudding ➡️
If you can show definitively that ultrasonic frequencies in the environment are affecting us physically some way you'll make a lot of audiophiles very happy, a lot of conspiracy theorists very happy, and I'd be happy to get in line with that new info. Haha. So far though the science just hasn't found anything. But I'm always open to new knowledge.
...lol, youre a mind reader!

dont get me started on the Schumann Resonance, Electromagnetic fields and 430 hz vs 440 hz bro
Old 4 hours ago | Show parent
  #384
Here for the gear
 
Quote:
Originally Posted by UnderTow ➡️
I think I've seen it.



The Monty Montgomery video that monkeyxx posted is one of the clearest videos on the topic: https://xiph.org/video/vid2.shtml



There is no lower bound. Sampling goes down to 0 Hz. But there is indeed an upper bound. The signal must be bandwidth limited (A low-pass filter basically). And we choose the sampling rate and filters so that this filter sits above the highest frequency we can hear. So it isn't so much about the music itself having a limited bandwidth. It is that we actively limit the bandwidth with a filter just above the audible range so that it fulfils the Nyquist requirements.

So yes, we are removing frequencies above a certain point but that point is chosen to be above and beyond what our hearing can perceive. So we are not removing sound. Just inaudible stuff.

Alistair
What he said. You have a filter on the input to band-limit ('anti-alias') you sample at 44.1k or above so you get more than double Nyquist at the theoretical limit of human hearing and 'enough' to take into account we can't build a perfect brick-wall filter. The sampled signal is continuous. There are no gaps between samples. Likewise on the output. Even without a reconstruction filter the signal is continuous, there are no gaps between samples. What you do have is every sample being squared off ,i.e. lots of HF 'ringing' hence the need for a reconstruction filter on the output (think of a capacitor), it charges as the wave rise, hits the peak, then discharges until the next sample rises to its peak - it 'joins the dots' at the peaks but the waveform is continuous whether the smoothing is there or not.
Old 4 hours ago | Show parent
  #385
Quote:
Originally Posted by UnderTow ➡️
I think I've seen it.



The Monty Montgomery video that monkeyxx posted is one of the clearest videos on the topic: https://xiph.org/video/vid2.shtml



There is no lower bound. Sampling goes down to 0 Hz. But there is indeed an upper bound. The signal must be bandwidth limited (A low-pass filter basically). And we choose the sampling rate and filters so that this filter sits above the highest frequency we can hear. So it isn't so much about the music itself having a limited bandwidth. It is that we actively limit the bandwidth with a filter just above the audible range so that it fulfils the Nyquist requirements.

So yes, we are removing frequencies above a certain point but that point is chosen to be above and beyond what our hearing can perceive. So we are not removing sound. Just inaudible stuff.

Alistair
.thanks man, your original post about digital being continuous really caught my attention and has opened up a whole new area of learning for me today.....who knew.
Old 4 hours ago | Show parent
  #386
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Sharp11's Avatar
 
🎧 15 years
Quote:
Originally Posted by monkeyxx ➡️
What goes in, and what comes out, are 'very accurate' representations of every audible frequency within that bandwidth. "Information" between samples is also captured, due to the "rules" of physics / audio. For example, an intersample peak that can clip your DAC by exceeding full scale. Amplitude, frequency, time information, and so on, are all 'there'. There is only "one possible waveform" that matches the data, which happens to be the same waveform that was input, minus what was filtered out.
This is the part many people just cannot wrap their heads around - that physics follows certain rules that are predictable and repeatable - they believe, instead, that there’s some kind of “magic” happening inside an analog electronic circuit - especially if it’s wrapped up into a box with lights, switches and vu meters.

The science of audio and brain perception are well understood, and have been for many years - pixelated pictures were reproduced for newspapers in the late 1800s - no reconstruction filters were used - except the human brain. Same with moving pictures in film - the brain “connects” the sequential still images (eventually settling on 24 per second). In hi Rez audio sampling, the space between the samples is filled in (the reconstruction filter) by the computer calculating the only possible choice for the “missing” route from one sample to the next which is represented by a line to complete the wave form - your brain sees the “original continuous waveform”.

Your brain only cares about seeing a waveform, not how it was created.

It was and still is an ingenious system - you’d think audio people would fall all over themselves, but instead, they say stupid things like “it’s cold and sterile”. Ughh - the importance of binary code and how it’s changed the world, in a myriad of ways, cannot be emphasized enough. Not all of it for the better, of course, but that’s outside the scope of this discussion.
Old 1 hour ago | Show parent
  #387
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28 Reviews written
🎧 10 years
Quote:
Originally Posted by ghostman ➡️
.
Andrew Schepps gave a brilliant lecture about the effect of audio quality in streaming music suggesting that our current sampling rates of digital audio for streaming are causing listening stress on a global scale as our brains interpolate the missing info in the music (after all......all perceptions of audio are psycho-acoustic and dependent on the brain, our cns and the ear)



People just fundamentally do not understand how digital audio works.

but I need to go take a nap because my brain is tired from "interpolating" all the missing stuff.
Old 44 minutes ago | Show parent
  #388
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🎧 10 years
Quote:
Originally Posted by monkeyxx ➡️
Digital has bandwidth limitations beyond the range of human hearing. Guess what, so does analog recording.

So neither is, hm, 'perfection itself.'

But the fact remains that the modern digital recorders are orders of magnitude more 'perfect' than an analog recorder. Regarding the word that was defined earlier as 'fidelity.'

But analog 'sounds good' so now we're really spinning. Round and round as it starts all over again, as it opens wide with a filter on the lens.

If I can answer your post more specifically, what comes out of a DAC is indeed a continuous waveform, watch the [EDIT: mont montgomery] video when you get a chance!
The problem with the word 'fidelity' is that it does not equal 'sound quality'.

Also the issues with measuring "quality" or "fidelity" with data doesn't include what is most important- the human perception of the sound.

The example I use is- a DAT recorder will wildly outperform an Ampex 350 in terms of specs. Yet I think more (most?) people would indicate the Ampex as having better "sound quality" in a listening test. And the Ampex is coveted today, and has been in use for 60 years. Whereas DATs came and went. Why? Because digital will always eat its own tail, as it always has.

So while digital may be more "perfect" on paper, why do we want that? Answer is we don't want perfection, we want convenience, more features with less cost, and we are vulnerable to marketing. Not because we want accuracy of sound (fidelity) or sound quality. One man's opinion.
Old 24 minutes ago | Show parent
  #389
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signalpudding's Avatar
 
Quote:
Originally Posted by donnylang ➡️
The problem with the word 'fidelity' is that it does not equal 'sound quality'.

Also the issues with measuring "quality" or "fidelity" with data doesn't include what is most important- the human perception of the sound.

The example I use is- a DAT recorder will wildly outperform an Ampex 350 in terms of specs. Yet I think more (most?) people would indicate the Ampex as having better "sound quality" in a listening test. And the Ampex is coveted today, and has been in use for 60 years. Whereas DATs came and went. Why? Because digital will always eat its own tail, as it always has.

So while digital may be more "perfect" on paper, why do we want that? Answer is we don't want perfection, we want convenience, more features with less cost, and we are vulnerable to marketing. Not because we want accuracy of sound (fidelity) or sound quality. One man's opinion.
I have a micro cassette recorder that sounds pretty terrible. Does that mean I can say all tape based recorders sound bad as well? Things have moved on a lot since DAT and whatever sound quality problems those had are not inherent to digital audio.

Tape adds something to the sounds you're recording and that's totally cool. I've recorded to tape many times and if I ever do another album with rock instrumentation I'd use it again. That's because I like what it adds to the sound.

Most of the time I do want accuracy of the sounds I'm recording and so do lots of other people. It's not all convenience, features, cost, or "marketing" as you said. It's awesome that we have a recording format that can capture things accurately. I don't know why some people who like the sound of tape and analog have such a problem with that and continue insisting that digital audio is wrong in some way.
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