...Gear crash is coming..

I still own the Vinyl I grew up with. For me it’s about the artwork, liner notes, and nostalgia. Sound quality doesn’t come into it. My first two CDs in the 80s was Rush ‘Grace Under Pressure’ and Saga ‘Heads or Tales’. My dad brought home a Sears CD player that he fixed as an electronics repair man (customer didn’t want to pay). My brother and I couldn’t believe our ears how great those CDs sounded over those cheap Fisher speakers. We certainly didn’t miss the vinyl hiss, pops, and lack of high end. We did miss studying the lyrics, gear lists, and artwork of the original vinyl.
 
Yeah, it's gone a step further with streaming, where you get virtually no information or reading matter with your music.
I used to look at CD covers and read the credits, liner notes etc. Vinyl LPs obviously have it in spades.
 
Chris Whitten said:
Yeah, it's gone a step further with streaming, where you get virtually no information or reading matter with your music.
I used to look at CD covers and read the credits, liner notes etc. Vinyl LPs obviously have it in spades.
Click to expand...

yeah...it is so easy to include this stuff, I wish they would do it. Put a little tab in the corner of the artwork that is included, and then link it to all the information....
 
I don't expect any minds were changed with all the content posted.
 
Chris Whitten said:
Quote:
The average hearing range in someone with perfect hearing is 20 Hz to 20 kHz. This means that from a hearing perspective, there’s no reason to record at 96 kHz or even 48 kHz rather than 44.1 kHz.
In other words, 96 kHz doesn’t do any better than 48 or 44.1 kHz at accurately reproducing a 20 kHz tone.


The main benefit of recording at 96 kHz is that it gives you added versatility when processing audio.

High sample rates, like 96 kHz, are beneficial if you plan to stretch and slow down your audio later.

For example, if you’ve recorded a sound source at 96 kHz and then slow it down by half, you’ll still have audible content up to 24 kHz in the half-speed version.

If you instead record that same sound source at 48 kHz and do the same thing, you’ll only have content up to 12 kHz.

For this reason, you may even see sound designers using sample rates of up to 192 kHz because of the flexibility this offers in processing the recording after the fact.

High sample rates also have the advantage of lower latency. Latency is a delay between when a sound is recorded and when it’s played back. The higher the sampling rate, the lower the latency.

96 kHz can also be beneficial when working with plugins during mixing.

Saturation, exciter, and compression create distortion that adds high frequencies to your mix.

If you’re working at 44.1 kHz or 48 kHz and apply these effects to sounds like cymbals or synths with a lot of content in the highest frequencies, you end up with aliasing.

This occurs because the plugins add frequencies above the Nyquist limit (i.e. frequencies higher than your sample rate can accurately handle), leading to unwanted distortion.

producerhive.com

48kHz vs 96kHz: Differences, Which To Use (And WHY)

Maybe you always use the default setting of 44.1 kHz?
producerhive.com producerhive.com
Click to expand...
What myths did I spread and what disinformation? It was not my opinion it is the actual science behind A to D processing.

This :
"High sample rates also have the advantage of lower latency. Latency is a delay between when a sound is recorded and when it’s played back. The higher the sampling rate, the lower the latency." is absolutely incorrect. (the bolded parts)

Also: The fundamental frequency "clang" generated by most types of cymbals is actually in the midrange (under 6 kHz, sometimes as low as 500 Hz). so even the lower sample rates are plenty capable of covering that.
The playable and most audible range of an in-tune electric guitar spans frequencies from 80Hz to 7kHz.
A standard 14 inch snare drum can usually be tuned sound great at a fundamental frequency of 170 Hz and also tiger up at 200 Hz too. Thinner and lighter drumheads can be tuned to vibrate at higher frequencies, which is an acoustics principle that applies to guitar strings too.

It is not that the sample rate can't handle it (the sample takes all the available frequencies, it just doesn't get as much detail if the number of samples taken is lower. The correct sentence is the higher number of samples = higher clarity but also equals more latency (because your computer has to work harder to take more samples per second). That is why you need to use ASIO instead of your Windows drivers because ASIO is designed to optimize resources to reduce that latency, and also offers you some control over the amount of samples you wish to take.

It is ok if you don't understand or something is confusing to you, but to say that I am expressing misinformation... why don't you ask me to clarify instead of saying that?

I work with frequencies of all kinds for a living so I believe I know something about how they work... I don't question your knowledge of studio recording and the industry in general, nor do I make it a point to attack you or anyone here. I just try to illustrate where my ramblings are coming from. And for that I quoted some known science (just so that I don't use my own words but those of the people who came up with that knowledge). Is it possible that some of that science is wrong?? yes but not likely.

I agree with you that if you slow down or invert a file you will do much better with a higher sample rate due to the increased clarity.

I hope you got the point this time.
 
  • Roll Eyes
Reactions: DHD
This is misinformation:

doggyd69b said:
Digital is not anywhere near as good as analog.
audio signal from a guitar to an amp and to a mic is all analog. The computer has to convert that to digital, in order for that signal to have the highest fidelity ( as close to the analog as possible) the sound wave has to be scanned more times which requires more processing power and disk space.
Click to expand...
 
doggyd69b said:
I hope you got the point this time.
Click to expand...
I got your point all along. The premise is that analog is better than digital. Analog is compromised from beginning to end. that's actually why People like it, because it adds character, rolls off harsh highs, adds some element of distortion. It sounds nice to the human ear, but by far the more clean and accurate Audi is digital audio.
As soon as you put any microphone in the chain you are compromising the sound. that's why EQ was invented, to correct the inaccuracies of microphones.
 
The readers can recognize a recurring pattern.
 
  • Like
Reactions: jda
"I only like human interaction"
 
Last edited:
No desire to be a "Clive Davis' or what's that other guy the one who murdered..or any other Engineer
- but you can come round and record me any time you're in town" 😁
------mae west 😁
 
Last edited:
doggyd69b said:
What myths did I spread and what disinformation? It was not my opinion it is the actual science behind A to D processing.

This :
"High sample rates also have the advantage of lower latency. Latency is a delay between when a sound is recorded and when it’s played back. The higher the sampling rate, the lower the latency." is absolutely incorrect. (the bolded parts)

Also: The fundamental frequency "clang" generated by most types of cymbals is actually in the midrange (under 6 kHz, sometimes as low as 500 Hz). so even the lower sample rates are plenty capable of covering that.
The playable and most audible range of an in-tune electric guitar spans frequencies from 80Hz to 7kHz.
A standard 14 inch snare drum can usually be tuned sound great at a fundamental frequency of 170 Hz and also tiger up at 200 Hz too. Thinner and lighter drumheads can be tuned to vibrate at higher frequencies, which is an acoustics principle that applies to guitar strings too.

It is not that the sample rate can't handle it (the sample takes all the available frequencies, it just doesn't get as much detail if the number of samples taken is lower. The correct sentence is the higher number of samples = higher clarity but also equals more latency (because your computer has to work harder to take more samples per second). That is why you need to use ASIO instead of your Windows drivers because ASIO is designed to optimize resources to reduce that latency, and also offers you some control over the amount of samples you wish to take.

It is ok if you don't understand or something is confusing to you, but to say that I am expressing misinformation... why don't you ask me to clarify instead of saying that?

I work with frequencies of all kinds for a living so I believe I know something about how they work... I don't question your knowledge of studio recording and the industry in general, nor do I make it a point to attack you or anyone here. I just try to illustrate where my ramblings are coming from. And for that I quoted some known science (just so that I don't use my own words but those of the people who came up with that knowledge). Is it possible that some of that science is wrong?? yes but not likely.

I agree with you that if you slow down or invert a file you will do much better with a higher sample rate due to the increased clarity.

I hope you got the point this time.
Click to expand...
Admin's point is quit posting analog vs digital in this thread start a new thread Off Topic.
 
Z26R-# said:
Price reductions starting to show upon Reverb.Nothing that you'd jump at though . Between 5 to 26%
Click to expand...
I had never searched this way, but once it was mentioned I found the 'Price Drop' search category on Reverb.

Looked like stuff I would never buy, or stuff that dropped from "WTF, you must be high!??!!!!" to "That's still way too high."
 
You must log in or register to reply here.
Back
Top