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Does analog gear really sound "better", or is it just a learned response?
Old 4th July 2020
  #4711
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[QUOTE=Space1999;14837757]A compressor is not made of a single piece of copper wire from input to output. There are components, there will be latency. You can look up information about calculating the time constant for the 5 time periods to charge or discharge a capacitor for instance.

A compressor can’t run on AC current. It must be converted to DC. You are going to need capacitors to do that. That is about as basic of an explanation as I can give you here as to why circuit design necessitates delaying signal. But it is just the tip of the iceberg.

Pat[/QUOTE

No, there is no latency. Capacitors and inductors cause phase shift, this is basic electronics.
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Old 4th July 2020
  #4712
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Wolf LeProducer's Avatar
 

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Originally Posted by shobud View Post
this is basic electronics.
Absolutely, that's what I said

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Old 4th July 2020
  #4713
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Video isn’t 60 FPS... more like 30 as a standard 60 fields per second is a standard with 2 fields per frame....there are also a couple of different “standards.”
FWIW....
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Old 4th July 2020
  #4714
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Quote:
Originally Posted by shobud View Post

No, there is no latency. Capacitors and inductors cause phase shift, this is basic electronics.

Agreed.

I think there are lot of terminology problems with the concept of “phase” in audio circuits verses time shifts. IMO, this is because the concept of phase in an AC signal was developed as part of AC power transmission and motor theory, as a time when audio amplifiers didn’t exist.

My 10c:

The classic concept of phase allows the change in timing between two identical waveforms to be expressed as an angle, with 360 degrees representing one full cycle of the waveform. By example, in 3-phase transmission, the phases are 120 degrees apart from each other. The two Live lines delivered to a US household are 180 degrees apart. This concept only works is you have a consistently repeating waveform such as the output of an alternator, however. If the waveform is not constantly repeating, the concept of phase angle is useless because it is referenced to the fixed period of the waveform.

With any other sort of waveform (like music) there is really no such thing as “phase.” (ducks in anticipation of hail of missiles). There are only timing differences between two otherwise identical non-repeating signals. Instead of phase difference, we should really refer to time difference.

A lot of discussion also confuses inverted polarity with phase. Back in the 70s, we would refer to a speaker being “out of phase” to another if the connections were reversed. Again, this is not a “phase” difference between two signals because true “phase” differences result from timing differences. You can never cause an inversion of an audio signal relative to another by changing its timing (unless the signal is a constantly repeating waveform).

Now, in real life the matters less than the proverbial hill of beans. When someone says “I reverse the phase of the bottom mic on the snare” we all understand what they are talking about, and it’s not delaying one signal. It would likely to be more confusing say “that distant mic is causing comb filtering because it is out of time” rather than “that distant mic is out of phase.”

but I still get a bit irritated by, to me, misuse of the term "phase"
Old 4th July 2020
  #4715
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Quote:
Originally Posted by norfolk martin View Post
I repeated the test with other possible variables removed


I put the snare on track 1 of a new project.

I then cloned it to track 2

I put the Arturia 140 plate as a channel effect on track 2 set to 100% dry.

When I inverted one channel , they almost nulled to zero, but not quite.

To see if the pl;ugin was changing the character or the timing of the signal. I then panned them panned them hard left and hard right, and mixed dowm

The attached is the waveform as displayed on audition.

Although the waves are not quite identical, they are complexity time aligned.

This proves only one thing - that it is possible to do latency compensation effectively.

Any echo sound is on the original track
I'm glad you're getting such good results and those waveforms do look very time aligned. Maybe try the experiment a little differently. Try having your track 1 snare, instead of copying and creating a track 2, create a bus. Send the signal to say buss 1, put an eq on the bus engage it, then follow with a compressor engaged, then a reverb set to 100% dry signal. For the time being, keep any delay compensation out of the loop, play audio and a/b by kicking the bus in and out of the signal flow. Do you hear a difference other than volume? If not, then you've got a great system and good, fast converters. If you do hear a difference, try delay compensation and see if will give you perfectly aligned audio.
Old 4th July 2020
  #4716
Deleted 10089a2
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Quote:
Originally Posted by norfolk martin View Post
Agreed.

I think there are lot of terminology problems with the concept of “phase” in audio circuits verses time shifts. IMO, this is because the concept of phase in an AC signal was developed as part of AC power transmission and motor theory, as a time when audio amplifiers didn’t exist.

My 10c:

The classic concept of phase allows the change in timing between two identical waveforms to be expressed as an angle, with 360 degrees representing one full cycle of the waveform. By example, in 3-phase transmission, the phases are 120 degrees apart from each other. The two Live lines delivered to a US household are 180 degrees apart. This concept only works is you have a consistently repeating waveform such as the output of an alternator, however. If the waveform is not constantly repeating, the concept of phase angle is useless because it is referenced to the fixed period of the waveform.

With any other sort of waveform (like music) there is really no such thing as “phase.” (ducks in anticipation of hail of missiles). There are only timing differences between two otherwise identical non-repeating signals. Instead of phase difference, we should really refer to time difference.

A lot of discussion also confuses inverted polarity with phase. Back in the 70s, we would refer to a speaker being “out of phase” to another if the connections were reversed. Again, this is not a “phase” difference between two signals because true “phase” differences result from timing differences. You can never cause an inversion of an audio signal relative to another by changing its timing (unless the signal is a constantly repeating waveform).

Now, in real life the matters less than the proverbial hill of beans. When someone says “I reverse the phase of the bottom mic on the snare” we all understand what they are talking about, and it’s not delaying one signal. It would likely to be more confusing say “that distant mic is causing comb filtering because it is out of time” rather than “that distant mic is out of phase.”

but I still get a bit irritated by, to me, misuse of the term "phase"
Nay and thrice nay I say unto thee ( I am going to make my reply in Elizabethan English to make it less bone crushingly dull) . Verily music is clusters of waves of different frequencies, yet they may be expressed as a function off a sine wave even though perfect they are not. The sound itself be naught but a spherical pressure wave, with different densities of air molecules therin - when said spherical pressure wave reaches yon eardrum vibrated in sync it is (sorry I'm turning into Yoda now) triggering the haircell most apt within thy cochlea. Should a stereo or twin mono source be out of time with its two halves this does indeed create phase cancellation as said air pressure from each speaker tries to expand and retract in the same patch of air at the same time at the same frequency. Verily I do decry thou art bogged down in semantics and even your electrical circuits could be described as timing differences - although indescribably minute ones. I beseech thee to reply to me in the tone of an irritable Scottish bus driver.
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Old 4th July 2020
  #4717
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Quote:
Originally Posted by voodoo4u View Post
I'm glad you're getting such good results and those waveforms do look very time aligned. Maybe try the experiment a little differently. Try having your track 1 snare, instead of copying and creating a track 2, create a bus. Send the signal to say buss 1, put an eq on the bus engage it, then follow with a compressor engaged, then a reverb set to 100% dry signal. For the time being, keep any delay compensation out of the loop, play audio and a/b by kicking the bus in and out of the signal flow. Do you hear a difference other than volume? If not, then you've got a great system and good, fast converters. If you do hear a difference, try delay compensation and see if will give you perfectly aligned audio.
I'll give it a try .
Old 4th July 2020
  #4718
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Quote:
Originally Posted by Deleted 10089a2 View Post
I beseech thee to reply to me in the tone of an irritable Scottish bus driver.
Ohhh I don’t do Scots well. I’m from East Anglia. I’ll give my native accent a try.

Well boy, oi don’t know ‘bout that.

I reckon your raesonin’ is a bit on the sosh, an maybe even threepence short of a shillin'.

Yew may very well be right ‘bout them discrete soine waves combinin’ to make one o’ them there complex waves, but I reckon that if you try an’ calculate a phase difference expressed in degrees out a’ one of them, it’ll be buggerd up f'sure.

A phase angle has to be relative to the timing a’ one wave y’see. You'cun calculate one of ‘em, for a complex wave, but only if it repeats itself, loike my granddad us’t to do.

If it don't repeat, you got no 360 degree t’ start with. You can make up a number, but it’d be as much use as a broken winded dickie.

Mind you, oi don’t know how its done in foreign countries like London, and oi don’t want to know neither. When folks go west, we never see ‘em agen. Best to stay with what y’know.


Phase angles is for ordinary "lectric what come out a' the wall.
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Old 4th July 2020
  #4719
Deleted 10089a2
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Quote:
Originally Posted by norfolk martin View Post
Ohhh I don’t do Scots well. I’m from East Anglia. I’ll give my native accent a try.

Well boy, oi don’t know ‘bout that.

I reckon your raesonin’ is a bit on the sosh, an maybe even threepence short of a shillin'.

Yew may very well be right ‘bout them discrete soine waves combinin’ to make one o’ them there complex waves, but I reckon that if you try an’ calculate a phase difference expressed in degrees out a’ one of them, it’ll be buggerd up f'sure.

A phase angle has to be relative to the timing a’ one wave y’see. You'cun calculate one of ‘em, for a complex wave, but only if it repeats itself, loike my granddad us’t to do.

If it don't repeat, you got no 360 degree t’ start with. You can make up a number, but it’d be as much use as a broken winded dickie.

Mind you, oi don’t know how its done in foreign countries like London, and oi don’t want to know neither. When folks go west, we never see ‘em agen. Best to stay with what y’know.


Phase angles is for ordinary "lectric what come out a' the wall.
Marvelous Seeing as you did not honour me with a style I will become a cockney mafia boss.

You see wot your fukin problem is that you don't think you can rotate a ****ing complex wave around an axis. It don't have to be a perfect sine wave or nuffin, just each point in the same wave complex or not just needs to rotated by the same amount. So for example mate, if I were to get yer missus, and there's no doubt shes a complex one phrrrr (sorry) and the create a mirror image of 'er - then she would cancel herself out complex or not, and that would be one hundred and aytaaaaayyyy (turning into a darts commentator) . Please reply in the style of Gollum from Lord of the rings (and don't forget to give me or the next person an alternative accent/character!!)
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Old 4th July 2020
  #4720
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zerocrossing's Avatar
Quote:
Originally Posted by voodoo4u View Post
I think what a lot of digital cheerleaders on this thread seem to either completely ignore or at least fail to mention is that recorded digital audio by itself may out spec recording to tape in a lab. In practical application, ie a recording studio, we're often having to add processing and plugins which then adds the time element into the mix (bad pun). IMHO it's latency that creates the biggest challenge to getting the best result with digital recordings not digital recording itself. Converters, plug-ins all add latency which creates a time smear that analog (tape or outboard gear) doesn't have to contend with.

The more intensive the processing power required by a plugin, the more it drags the signal back in time. Add five or six plugins per track over 24 or 30 tracks and it really adds up. Summed, it makes a large audible difference.
“Time smear.” Classic.
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Old 5th July 2020
  #4721
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Quote:
Originally Posted by zerocrossing View Post
“Time smear.” Classic.
Funny thing is, in most home studios there's tons of time smear, nulls, peaks, standing waves - enough to write a book about (or several thousand as it turns out. lol)

This is not because of the inferior processing of daws and "time calculations", it's because of the physics of acoustics.

Pro studios aren't perfect either, but trying to get things as close to "right" is the goal, even if never completely reached.

Why do I mention this? Because I can just imagine someone sitting at a daw obsessing over a few sample bits out of alignment and decrying digital recording as "inferior" to the old methods, while all this audio muck is happening around him.
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Old 5th July 2020
  #4722
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Quote:
Originally Posted by Sharp11 View Post
Funny thing is, in most home studios there's tons of time smear, nulls, peaks, standing waves - enough to write a book about (or several thousand as it turns out. lol)

This is not because of the inferior processing of daws and "time calculations", it's because of the physics of acoustics.

Pro studios aren't perfect either, but trying to get things as close to "right" is the goal, even if never completely reached.

Why do I mention this? Because I can just imagine someone sitting at a daw obsessing over a few sample bits out of alignment and decrying digital recording as "inferior" to the old methods, while all this audio muck is happening around him.
My mirth was more due to the poster’s idea that his problem is a universal problem. Have I dealt with plugins that introduce latency beyond my set buffer size? You bet. Is there sh!tty software? Boy-. Sh!tty digital hardware? Yes, indeed. But to blame a technology on a poor representation of that technology is lazy and sloppy thinking. When I was a teenager my best friend’s family had a cat. I basically never saw it, but his house smelled of cat piss. My girlfriend and I were there once to pick him up and in the short time we were there the cat pissed in her handbag. I thought, “cats are sh!tty pets.” I had no idea that they’d never neutered a male cat. I had a myopic experience. I was young and ignorant. A few years later I met a super sweet cat (sleeping it off on a friend’s couch) and I thought, “huh, their cat doesn’t piss all over the house and look how friendly it is sitting on my chest licking my hair.” A few years after that I got my first pet cat and I’ve had cats ever since. They’re awesome.

So fix your miss behaving software, or interface, or whatever, but to hurl out definitive statements about digital recording... I was working for a woman in the early 80s and the entire project was going into a Sony digital reel-to-reel. That album still sounds great to me, and the technology has made huge strides since then. Actually, the artist now does a lot (all?) of her work in software.

The truth is, you can make a crappy recording with analog gear, or digital gear, or software, or a combination of all. As you mentioned, all recording is rife with compromise and issues. If you know what you’re doing, you can avoid the worst of it. If your music is good, a lot will be forgiven. Even time smear. Trust me, whenever I use my portal gun to open a portal to another reality, I always get messed up by it.
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Old 5th July 2020
  #4723
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Quote:
Originally Posted by zerocrossing View Post
My mirth was more due to the poster’s idea that his problem is a universal problem. Have I dealt with plugins that introduce latency beyond my set buffer size? You bet. Is there sh!tty software? Boy-. Sh!tty digital hardware? Yes, indeed. But to blame a technology on a poor representation of that technology is lazy and sloppy thinking. When I was a teenager my best friend’s family had a cat. I basically never saw it, but his house smelled of cat piss. My girlfriend and I were there once to pick him up and in the short time we were there the cat pissed in her handbag. I thought, “cats are sh!tty pets.” I had no idea that they’d never neutered a male cat. I had a myopic experience. I was young and ignorant. A few years later I met a super sweet cat (sleeping it off on a friend’s couch) and I thought, “huh, their cat doesn’t piss all over the house and look how friendly it is sitting on my chest licking my hair.” A few years after that I got my first pet cat and I’ve had cats ever since. They’re awesome.

So fix your miss behaving software, or interface, or whatever, but to hurl out definitive statements about digital recording... I was working for a woman in the early 80s and the entire project was going into a Sony digital reel-to-reel. That album still sounds great to me, and the technology has made huge strides since then. Actually, the artist now does a lot (all?) of her work in software.

The truth is, you can make a crappy recording with analog gear, or digital gear, or software, or a combination of all. As you mentioned, all recording is rife with compromise and issues. If you know what you’re doing, you can avoid the worst of it. If your music is good, a lot will be forgiven. Even time smear. Trust me, whenever I use my portal gun to open a portal to another reality, I always get messed up by it.
You've either not read what I've written or you've misunderstood it. This is a universal problem with computer based audio recording. I've been recording on workstations since 1989. Through by company I've owned Soundtools, Sonic Solutions, Sadie, ProTools, Sequoia, and Logic. In the last 31 years I've not only witnessed the evolution of digital recording, I've experienced it. All DAWs have to deal with latency. Some handle it poorly and some handle it so well, it's almost completely a non issue. The only lazy or sloppy thinking I see here is your fuzzy understanding on how DAWs really function under the hood.

I'll be the first to admit, because of a lot of brilliant people in R&D in both the computing world and in digital audio, things have improved dramatically. Pretty much to the point where most people don't have to think about it.

You've got it backward. I'm not blaming a technology on a poor representation of that technology, I'm praising the people who've taken a technology that has an inherent flaw and developed it to the point where on most systems today, that flaw goes almost completely unnoticed by people like you.

Keep in mind the topic of this thread "Does analog gear really sound better..."
That topic can't really be explored without pointing out the strengths and flaws in both approaches and this is the biggest fundamental flaw in "computer based audio recording".

That's why your Sony example doesn't really apply to what I've been talking about. Different architecture, different principle, different fundamental flaw. They were tape based systems (DAT, ADAT etc.) and had no issue with latency. Drop outs? Yes.

This isn't about what you can make a crappy recording on, it's about does analog gear sound better? IMHO the answer is "it depends". Both have their strengths and weaknesses.
Old 5th July 2020
  #4724
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WiZKiD's Avatar
 

Quote:
Originally Posted by voodoo4u View Post
You've either not read what I've written or you've misunderstood it. This is a universal problem with computer based audio recording. I've been recording on workstations since 1989. Through by company I've owned Soundtools, Sonic Solutions, Sadie, ProTools, Sequoia, and Logic. In the last 31 years I've not only witnessed the evolution of digital recording, I've experienced it. All DAWs have to deal with latency. Some handle it poorly and some handle it so well, it's almost completely a non issue. The only lazy or sloppy thinking I see here is your fuzzy understanding on how DAWs really function under the hood.

I'll be the first to admit, because of a lot of brilliant people in R&D in both the computing world and in digital audio, things have improved dramatically. Pretty much to the point where most people don't have to think about it.

You've got it backward. I'm not blaming a technology on a poor representation of that technology, I'm praising the people who've taken a technology that has an inherent flaw and developed it to the point where on most systems today, that flaw goes almost completely unnoticed by people like you.

Keep in mind the topic of this thread "Does analog gear really sound better..."
That topic can't really be explored without pointing out the strengths and flaws in both approaches and this is the biggest fundamental flaw in "computer based audio recording".

That's why your Sony example doesn't really apply to what I've been talking about. Different architecture, different principle, different fundamental flaw. They were tape based systems (DAT, ADAT etc.) and had no issue with latency. Drop outs? Yes.

This isn't about what you can make a crappy recording on, it's about does analog gear sound better? IMHO the answer is "it depends". Both have their strengths and weaknesses.
I think you bring up some valid points here. Some may not take into consideration (not saying anyone here) that Pro Tools in its early incarnation was not even considered close to an end to end *professional* solution for recording/mixing/mastering. At this point its beyond safe to say those days are long gone, maybe starting with the moment Charles Dye mixed Livin La Vida Loca. Not the first ITB mix, but the first #1 . So since about ten years after that this is at the forefront of the conversation.

Now that we are are here in this age of analog/hybrid/ITB mixing all creating hit records, it is worth considering this is the last frontier of analog. It really is. I personally mix hybrid, with a healthy assortment of analog and will probably due so for the foreseeable future, unless there's an advancement around the corner that changes everything. Then I'll adapt. Its also entirely possible that I may be in the minority soon, or even now in terms of even using analog gear to mix commercial music.

I think it is certainly valid of you to consider these inherent technical flaws of the two formats. To those who deal with this on a daily basis (and have for many years), the conversation actually just got very interesting...

Last edited by WiZKiD; 5th July 2020 at 05:19 AM..
Old 5th July 2020
  #4725
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Quote:
Originally Posted by Sharp11 View Post
Well, after 20 years I’ve mostly retired my Avalon VTp 747 - it was a fine tube mastering optical compressor, eq, spectral compressor and widener in its day, but I’ve gone back and forth with it and Ozone 9 over the past few months and I have to say Ozone gets the nod. There isn’t a difference in the sound and ozone offers so much more it’s hard to know where to begin - from its tool kit of analysis plugins to vintage and modern eq to its spectral compression and widening - then there’s the ability to output in different formats. It sounds just as great and offers more, and despite your claim about latency, the hardware Avalon accrues quite a bit.

That’s a real world assessment from a guy who does this every day and can “afford” what he feels is necessary to get the best results.

My closet is full of hardware, but I do take stuff out at least a couple of times a year just to keep the gear alive (I’m not good at selling things). I do use guitar pedals and hardware synths - it’s a throwback to what I’m used to after 40 plus years in the business.

Anyway, my subjective two cents
Maybe a more modern piece of Mastering hardware? The Bettermaker Mastering Limiter plus Ozone... Now we're talking Kobe in the lane.

Last edited by WiZKiD; 11th July 2020 at 12:03 AM..
Old 5th July 2020
  #4726
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Quote:
Originally Posted by norfolk martin View Post
The classic concept of phase allows the change in timing between two identical waveforms to be expressed as an angle, with 360 degrees representing one full cycle of the waveform.
And that waveform has to be cyclical - ok for oscillators but not for continually changing waveforms.

However, phase is a good term for a delay that is a function of wavelength. A delay implies a constant whereas phase is a timing delay that changes with frequency/wavelength. But then, musical notes have high enough frequencies that in a snap shot they could be considered to have a constant, cycling waveform and two identical waveforms reaching two different points in space ... therefore phase would be appropriate. Same in circuits.

One use where it's confusing is with speaker crossover design - people talk about phase changes that depend on frequency. Well, surely that is a constant delay?
Old 5th July 2020
  #4727
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Originally Posted by art felton View Post
As much care needs to be exerted in applying each step of DSP as is put into applying each step of analog processing if you are to achieve optimum performance. Whether it is designers of gear or users of gear it comes down to care. I shift tracks around on a daily basis to make up for latency issues. Each DAW or plugin may behave differently.
Different converters have different round trips too, when I run stuff through my burl I have to nudge it back a few samples after I print.
Old 5th July 2020
  #4728
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Quote:
Originally Posted by nat8808 View Post
And that waveform has to be cyclical - ok for oscillators but not for continually changing waveforms.

However, phase is a good term for a delay that is a function of wavelength. A delay implies a constant whereas phase is a timing delay that changes with frequency/wavelength. But then, musical notes have high enough frequencies that in a snap shot they could be considered to have a constant, cycling waveform and two identical waveforms reaching two different points in space ... therefore phase would be appropriate. Same in circuits.

One use where it's confusing is with speaker crossover design - people talk about phase changes that depend on frequency. Well, surely that is a constant delay?

That is a good way to look at the question, nat . A delay certainly can be described as a a phase change when it is related to a certain wavelength, i.e "a shift of 90 degrees at 1KHz" Reminds me of those hideous ac circuit classes in which the instructor would draw an ugly looking network of capacitors,indicators and resistors, and as us to calculate the ac results (expressed as both a phase angle and in the x + i format.) Urrgh! hated it then and haven't grown any fonder is the ensuing 40 years or so!


I personally find it easier to think in terms of delay rather than phase because my basic concept of phase was established in the "AC power and Transmission" class I took at the same time.
Old 5th July 2020
  #4729
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[QUOTE=Gusss;14838754 . Please reply in the style of Gollum from Lord of the rings (and don't forget to give me or the next person an alternative accent/character!!)[/QUOTE]

I'll work on it ..........
Old 5th July 2020
  #4730
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Quote:
Originally Posted by norfolk martin View Post
I'll work on it ..........
This may prove difficult although I don’t know your acting abilities. I follow your posts and from what I’ve seen you’re anything but a Sméagol. I also share your assessment of delay vs phase. I rarely encounter perfectly out of phase instances in practice, outside of things like snare top/bottom. Most issues I encounter are some inherent delay or latency but I rarely find two identical wavelengths perfectly out of phase, even with multiple mics. Obviously, this is room dependent. If you’ve got reflections returning to the mic (or ear from speaker) slightly out of phase this can also cause issues. That said, I was taught *in practice* that phase is more related to compression/rarefaction than delay. Although they are also related. Second cousins maybe?

In most cases, I spend more time purposely messing things up with a Dim D or H910 than fixing phase issues.

Last edited by WiZKiD; 5th July 2020 at 04:05 PM..
Old 5th July 2020
  #4731
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Quote:
Originally Posted by voodoo4u View Post

Keep in mind the topic of this thread "Does analog gear really sound better..."
That topic can't really be explored without pointing out the strengths and flaws in both approaches and this is the biggest fundamental flaw in "computer based audio recording".
Are we to consider all the flaws of analog processes when comparing the objective fidelity of digital processes to analog? In the spirit of the OP's title of the thread?

let's lay out some of the obvious: A full chain of analog recording - starting with multiple passes of multi track tape and its attendant wear, then mastering that result to tape another generation down (with an RIAA curve and a 60hz cutoff at the bottom end), then cutting a master - third generation, then printing vinyl from the master - fourth generation (wearing down the master while printing vinyl copies), packaging those copies so people can play them on fundamentally flawed electro - mechanical devices (turntables with cartridges) which generate heat on the surface of the vinyl, grinding down and wearing out the vinyl with each pass?

Then there are all the potential (and unfortunately realized) speed variations in all four generations of those processes, just try and play your guitar and piano along with some of your favorite recordings and find the pitch.

Phase issues, incidentally, are inherent in most every recording environment, see my comments on room acoustics. These types of problems affect what people hear and allow into their recording way more than latency in recording - just read thread after thread about it. Because my ear is attuned to it, I can hear phase problems In many of my favorite recordings, especially late 50s into the 60s jazz releases, where comb filtering rules. Set up a subwoofer and you’re chasing down phase - that’s why almost every decent sub has a phase knob on the back panel.

The long and short of it is analog represents many conversions and generations of copies; many motors, wheels and idler arms conspiring to create wear, heat and speed variations, in other words, latency - latency you can hear as wow and flutter.

Digital equipment has no moving parts and if chasing down the last spec of latency in a plugin or monitoring chain is the worst of the process, I’m just fine with it.

Last edited by Sharp11; 5th July 2020 at 04:20 PM..
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Old 5th July 2020
  #4732
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robert82's Avatar
It's like this parable of Jesus:

And why beholdest thou the mote that is in thy brother's digital eye, but considerest not the beam that is in thine own analog eye?
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Old 5th July 2020
  #4733
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Quote:
Originally Posted by Sharp11 View Post
Are we to consider all the flaws of analog processes when comparing the objective fidelity of digital processes to analog? In the spirit of the OP's title of the thread?
Yes, of course we are. After 150 some odd pages of this thread, it's starting to become the defacto, let's get all the differences on the table, comparison between digital and analog recording methods and to leave out the weaknesses of either is not truthful or helpful to the discussion. All your points about the flaws of the analog process are in my opinion, honest and perfectly valid.

Analog recording had a huge number of technical flaws in recording, manufacturing and reproduction process, but it had one strength that some people still can't get over even now and that's that for most, it's still pleasing to the ears. I've pointed out one major flaw in DAWs that I think is responsible for my often not liking the end result of mixing completely ITB.

Some people have laughed at my use of the term "smearing" the audio when latency creates little micro shifts in timing between tracks in a DAW, but I stand by it and to me, it's most obvious in live recorded drums that are then processed with a lot of plugins the shift between the close mic'd snare over and under and what leaks into the hat mic or the overheads ( I know part of this can be attributed to acoustical phasing re: a time shift) but most of it is in how the DAW handles latency. I'm sure I'm not the only one who's walked through a store or a restaurant and heard muzak playing and heard a modern digital recording and thought "That snare sounds like absolute crap. It sounds like the drummer was banging on piece of F--ing cardboard. Why can't people get the snare to sound like a real snare anymore". Then I'll go back to my studio, mix drums on a Daw with plugins and get the same crappy results. Now I know why.

There are a lot of famous mix engineers who now swear by mixing entirely in the box and I listen to their mixes and think "why can't you hear how much worse your drums sound than they used to. Particularly the snare".

To me, it's not digital recording technology that's to blame, it's what we do with the audio once we get it in the box. How we treat busing, how we use summing and how we use plugins.

For any body who cares, my mixing methods have now evolved to the point where I record at 96k into the box, send each channel straight out one for one straight into a console (just like a tape machine). Use as few plugins as possible, use hardware effects, comps eq, revs etc. Use the the console, not the DAW for summing. I've got my drums back. They sound real. In fact, I feel I'm getting better results mixing this way than I ever got pure analog (tape) or purely ITB.

Last edited by voodoo4u; 5th July 2020 at 06:03 PM..
Old 5th July 2020
  #4734
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Quote:
Originally Posted by voodoo4u View Post

Analog recording had a huge number of technical flaws in recording, manufacturing and reproduction process, but it had one strength that some people still can't get over even now and that's that for most, it's still pleasing to the ears.
Whose ears? Yours? Not mine, I prefer the transparency of digital recording. There's a world of recording outside of rock music - film, tv, major sporting events that depend upon thousands of tracks of audio, with tons upon tons of plugins (eq's and comps, mostly). I've never meant anyone who has said to me "gee, I wish the London Summer Olympic Games were recorded and broadcast in analog, bummer"

Quote:
I've pointed out one major flaw in DAWs that I think is responsible for my often not liking the end result of mixing completely ITB.
This is a theory of yours, but not fact, and below you make a mixed case for it, mostly revolving around music you hear which has a mixing aesthetic you don't prefer (more on that later).

Quote:
Some people have laughed at my use of the term "smearing" the audio when latency creates little micro shifts in timing between tracks in a DAW, but I stand by it and to me, it's most obvious in live recorded drums that are then processed with a lot of plugins the shift between the close mic'd snare over and under and what leaks into the hat mic or the overheads ( I know part of this can be attributed to acoustical phasing re: a time shift) but most of it is in how the DAW handles latency.
"Most of it"? Says who? I can't find a single AES paper or measurement to demonstrate this in action - like we always say, if you can hear it, you can measure it. You even contradict yourself when you say it's mostly been dealt with (through delay compensation). If it's "mostly been dealt with" and is "almost a non-issue", the only place to look is in the recording process - due to rooms and mic placement. It's 2020, and I can assure you, the latter two items have not been completely "dealt with".

Quote:
I'm sure I'm not the only one who's walked through a store or a restaurant and heard muzak playing and heard a modern digital recording and thought "That snare sounds like absolute crap. It sounds like the drummer was banging on piece of F--ing cardboard. Why can't people get the snare to sound like a real snare anymore". Then I'll go back to my studio, mix drums on a Daw with plugins and get the same crappy results. Now I know why.
I would never walk into a store or restaurant and attribute a noticeably bad snare drum sound (being played on store speakers?) to a "microshift" timing issue from a plugin. I would, however, attribute it to someone's bad idea of how a snare drum should sound, or perhaps one of the other one thousand or so variables that a recording/mastering process must endure before a finished product is released.

Quote:
There are a lot of famous mix engineers who now swear by mixing entirely in the box and I listen to their mixes and think "why can't you hear how much worse your drums sound than they used to. Particularly the snare".
Again, this is just your opinion - have you ever listened carefully to how the drums, particularly the snare sounds on many famous and iconic rock recordings done in the late 60s, early 70s? They sound like sh8t, that's how - listen to Tower Of Power's "What Is Hip" (mono drums, two mic's), or Chicago's "Make Me Smile" or "25 to 6 to 4" - all completely, out of the box analog recordings, soup to nuts, and all sound pretty much like crap. (still love 'em though). certainly, they should sound just so "pleasing to the ears", right?

Quote:
To me, it's not digital recording technology that's to blame, it's what we do with the audio once we get it in the box. How we treat busing, how we use summing and how we use plugins.
Now you're onto something.

Quote:
For any body who cares, my mixing methods have now evolved to the point where I record at 96k into the box, send each channel straight out one for one straight into a console (just like a tape machine). Use as few plugins as possible, use hardware effects, comps eq, revs etc. Use the the console, not the DAW for summing. I've got my drums back. They sound real. In fact, I feel I'm getting better results mixing this way than I ever got pure analog (tape) or purely ITB.
You were onto something, then you fell victim to two common logical fallacies.

In your first example of the restaurant and major mix engineer's itb mixes, you're right on top of the fallacy that affects us all: "Confirmation Bias"; you've ignored any other of dozens of reasons for why the snare sounds bad (to you) and made a decision it has to be "time smear" - it could simply, in your mind, be nothing else.

As for your own mixes, you're engaging in "Anecdotal"; we have no way of knowing, unless you presented us with the exact same mix done a different way, what's happening in your mixing process. It may be you just enjoy and feel more comfortable working in the way you've established. Which is fine, but attempting to create an entire alternate theory about the inferiority of plugins as the "main reason" for mixes sound "bad" while ignoring all other more obvious possibilities and variables is really a stretch.

Having said all that, I, too, dislike the mixes of a lot of modern pop music, however, as a working composer of television library music and documentary films, and as a teacher, I do have to recreate or at the very least, steal some ideas from these guys so I've listened (and continue to listen) to a lot of their work. I don't think their mixes turn me off because of small timing shifts in the plugins they use (or large chains of plugins), what I know I dislike about this style is the over-the-top nature of the mixes and the fact that the mix is usually hiding a mediocre song. It's like cinema Verite full time, without a break and without a cool movie underneath the technique. I've recorded many bands and snare drums and full drum sets that sound just wonderful, and processed them with plugins. And I've fixed many a recording with phase issues due to operator error. I don't know why you can't do it well without outboard gear, but I suspect you could if you wanted to.
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Old 5th July 2020
  #4735
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voodoo4u's Avatar
Quote:
Originally Posted by Sharp11 View Post
Whose ears? Yours? Not mine, I prefer the transparency of digital recording. There's a world of recording outside of rock music - film, tv, major sporting events that depend upon thousands of tracks of audio, with tons upon tons of plugins (eq's and comps, mostly). I've never meant anyone who has said to me "gee, I wish the London Summer Olympic Games were recorded and broadcast in analog, bummer"
It is an axiom. In general, analog recordings of the past are considered by many to have a characteristic of warmth. This is a generalization and fully acknowledged as such. Jeez man, get some perspective.

Quote:
Originally Posted by Sharp11 View Post
This is a theory of yours, but not fact, and below you make a mixed case for it, mostly revolving around music you hear which has a mixing aesthetic you don't prefer (more on that later).
No, it's not a theory of mine, it's a subjective opinion based on my own experiences. That's why I used the phrase "I think"


Quote:
Originally Posted by Sharp11 View Post
"Most of it"? Says who? I can't find a single AES paper or measurement to demonstrate this in action - like we always say, if you can hear it, you can measure it. You even contradict yourself when you say it's mostly been dealt with (through delay compensation). If it's "mostly been dealt with" and is "almost a non-issue", the only place to look is in the recording process - due to rooms and mic placement. It's 2020, and I can assure you, the latter two items have not been completely "dealt with".
Are you honestly implying that there isn't a single AES paper or measurement that deals with the topic of latency within DAWs? So latency compensation was developed just out of the blue for no reason whatsoever?


Quote:
Originally Posted by Sharp11 View Post
I would never walk into a store or restaurant and attribute a noticeably bad snare drum sound (being played on store speakers?) to a "microshift" timing issue from a plugin. I would, however, attribute it to someone's bad idea of how a snare drum should sound, or perhaps one of the other one thousand or so variables that a recording/mastering process must endure before a finished product is released.
Did you even bother to listen to the sample I posted earlier of the snare? I used to be a mastering engineer and I would hear this kind of effect on people's projects all the time. This kind of latency would be missed by the recording engineers often. Warning: this phrase has contained within subjective experience.



[
Quote:
Originally Posted by Sharp11 View Post
Again, this is just your opinion - have you ever listened carefully to how the drums, particularly the snare sounds on many famous and iconic rock recordings done in the late 60s, early 70s? They sound like sh8t, that's how - listen to Tower Of Power's "What Is Hip" (mono drums, two mic's), or Chicago's "Make Me Smile" or "25 to 6 to 4" - all completely, out of the box analog recordings, soup to nuts, and all sound pretty much like crap. (still love 'em though). certainly, they should sound just so "pleasing to the ears", right?
Straw man argument. Because I notice latency on the snare of digital commercial recordings, I should notice how badly some snares sounded back in 1973? What kind of poor correlation is that?





Quote:
Originally Posted by Sharp11 View Post
You were onto something, then you fell victim to two common logical fallacies.

In your first example of the restaurant and major mix engineer's itb mixes, you're right on top of the fallacy that affects us all: "Confirmation Bias"; you've ignored any other of dozens of reasons for why the snare sounds bad (to you) and made a decision it has to be "time smear" - it could simply, in your mind, be nothing else.

As for your own mixes, you're engaging in "Anecdotal"; we have no way of knowing, unless you presented us with the exact same mix done a different way, what's happening in your mixing process. It may be you just enjoy and feel more comfortable working in the way you've established. Which is fine, but attempting to create an entire alternate theory about the inferiority of plugins as the "main reason" for mixes sound "bad" while ignoring all other more obvious possibilities and variables is really a stretch.

Having said all that, I, too, dislike the mixes of a lot of modern pop music, however, as a working composer of television library music and documentary films, and as a teacher, I do have to recreate or at the very least, steal some ideas from these guys so I've listened (and continue to listen) to a lot of their work. I don't think their mixes turn me off because of small timing shifts in the plugins they use (or large chains of plugins), what I know I dislike about this style is the over-the-top nature of the mixes and the fact that the mix is usually hiding a mediocre song. It's like cinema Verite full time, without a break and without a cool movie underneath the technique. I've recorded many bands and snare drums and full drum sets that sound just wonderful, and processed them with plugins. And I've fixed many a recording with phase issues due to operator error. I don't know why you can't do it well without outboard gear, but I suspect you could if you wanted to.
You've criticized me for sharing an opinion which I can back up with examples and then shared a paragraph with nothing but your own opinions.

You and I have sparred on this topic on other threads and frankly, I for one have gotten tired of your bragging about your four Emmy's which you've mentioned on at least three occasions that I've read. As well as your teaching position and all the work you do in film (mentioned once again at least three times that I've read). This IMHO does not give you or your opinions any more credibility than anyone else on this topic. It's just another opinion of the many opinions on this thread. If you want to impress, provide samples, written reports, studies, papers, anything to back up your case other than telling us how well you're doing and that your opinion should be the last word. And that digital recording technology has reached the height of audio perfection.
Old 5th July 2020
  #4736
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Quote:
Originally Posted by voodoo4u View Post
It is an axiom. In general, analog recordings of the past are considered by many to have a characteristic of warmth. This is a generalization and fully acknowledged as such. Jeez man, get some perspective.



No, it's not a theory of mine, it's a subjective opinion based on my own experiences. That's why I used the phrase "I think"




Are you honestly implying that there isn't a single AES paper or measurement that deals with the topic of latency within DAWs? So latency compensation was developed just out of the blue for no reason whatsoever?




Did you even bother to listen to the sample I posted earlier of the snare? I used to be a mastering engineer and I would hear this kind of effect on people's projects all the time. This kind of latency would be missed by the recording engineers often. Warning: this phrase has contained within subjective experience.



[

Straw man argument. Because I notice latency on the snare of digital commercial recordings, I should notice how badly some snares sounded back in 1973? What kind of poor correlation is that?







You've criticized me for sharing an opinion which I can back up with examples and then shared a paragraph with nothing but your own opinions.

You and I have sparred on this topic on other threads and frankly, I for one have gotten tired of your bragging about your four Emmy's which you've mentioned on at least three occasions that I've read. As well as your teaching position and all the work you do in film (mentioned once again at least three times that I've read). This IMHO does not give you or your opinions any more credibility than anyone else on this topic. It's just another opinion of the many opinions on this thread. If you want to impress, provide samples, written reports, studies, papers, anything to back up your case other than telling us how well you're doing and that your opinion should be the last word. And that digital recording technology has reached the height of audio perfection.
My career has nothing to do with this, except in your mind - so you’re back to ad hominem attacks and discursive writing in lieu of solid discourse. I apologize to me for wasting my time.
Old 5th July 2020
  #4737
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OK since I brought the subject up, the burden of proof is on me. Here are 5 samples of snare drum. The first is a control sample. Snare with nothing on it. The second is one plugin on aux buss. The third is 2 plugins the fourth is 3 etc. It was done in Logic 10.48. The plugins were EQ, comp, Lex rev and the fourth Waves Plate. And this is just 1 aux. I've compensated for level.

Can this be corrected for? Yes, if you're aware of it. If you have different instruments with different plugins on a session, it's not always this easy to hear that time is shifting on you and your tracks are getting smeared in time and manual compensation can be time consuming and tedious.

Oh, and I apologize to no one.
Attached Files

Snare pure.wav (923.1 KB, 463 views)

Snare 1 Plugin.wav (923.1 KB, 461 views)

Snare 2 Plugins.wav (923.1 KB, 446 views)

Snare 3 Plugins.wav (923.1 KB, 431 views)

Snare 4 Plugins.wav (923.1 KB, 446 views)

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Old 6th July 2020
  #4738
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Sharp11's Avatar
 

Quote:
Originally Posted by voodoo4u View Post
OK since I brought the subject up, the burden of proof is on me. Here are 5 samples of snare drum. The first is a control sample. Snare with nothing on it. The second is one plugin on aux buss. The third is 2 plugins the fourth is 3 etc. It was done in Logic 10.48. The plugins were EQ, comp, Lex rev and the fourth Waves Plate. And this is just 1 aux. I've compensated for level.

Can this be corrected for? Yes, if you're aware of it. If you have different instruments with different plugins on a session, it's not always this easy to hear that time is shifting on you and your tracks are getting smeared in time and manual compensation can be time consuming and tedious.

Oh, and I apologize to no one.
Delay compensation engaged?
Old 6th July 2020
  #4739
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voodoo4u's Avatar
Quote:
Originally Posted by Sharp11 View Post
Delay compensation engaged?
That's not the point. It's an attribute of recording on A DAW. Now we're getting into a circular argument. Did you read the bit where I said it can be compensated for? How about the part where I compared it analog tape's biggest flaw, signal to noise, and I compared latency compensation to Dolby noise reduction? How about where I said it's like the elephant in the room with digital audio? Something no one wants to admit or they're unaware of it.

Besides to engage delay compensation is to sacrifice something else along the way. Otherwise it would just be a permanent feature buried in the system and wouldn't have a choice to engage it or not. It would just be permanently on.
Old 6th July 2020
  #4740
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zerocrossing's Avatar
Quote:
Originally Posted by voodoo4u View Post
You've either not read what I've written or you've misunderstood it. This is a universal problem with computer based audio recording. I've been recording on workstations since 1989. Through by company I've owned Soundtools, Sonic Solutions, Sadie, ProTools, Sequoia, and Logic. In the last 31 years I've not only witnessed the evolution of digital recording, I've experienced it. All DAWs have to deal with latency. Some handle it poorly and some handle it so well, it's almost completely a non issue. The only lazy or sloppy thinking I see here is your fuzzy understanding on how DAWs really function under the hood.

I'll be the first to admit, because of a lot of brilliant people in R&D in both the computing world and in digital audio, things have improved dramatically. Pretty much to the point where most people don't have to think about it.

You've got it backward. I'm not blaming a technology on a poor representation of that technology, I'm praising the people who've taken a technology that has an inherent flaw and developed it to the point where on most systems today, that flaw goes almost completely unnoticed by people like you.

Keep in mind the topic of this thread "Does analog gear really sound better..."
That topic can't really be explored without pointing out the strengths and flaws in both approaches and this is the biggest fundamental flaw in "computer based audio recording".

That's why your Sony example doesn't really apply to what I've been talking about. Different architecture, different principle, different fundamental flaw. They were tape based systems (DAT, ADAT etc.) and had no issue with latency. Drop outs? Yes.

This isn't about what you can make a crappy recording on, it's about does analog gear sound better? IMHO the answer is "it depends". Both have their strengths and weaknesses.
Well, why not go on a rant about tape cross talk and bleed? Go on, now! Get!
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