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A thread for asking the things you should know by now but don't
Old 12th May 2019
  #6091
What are the RCA outputs on the Focusrite Scarlet 2i4 designed for? Running to mixers? Running to speakers? There's 4x RCA output jacks.
Old 12th May 2019
  #6092
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Richard Crowley's Avatar
 

Quote:
Originally Posted by hello people View Post
What are the RCA outputs on the Focusrite Scarlet 2i4 designed for? Running to mixers? Running to speakers? There's 4x RCA output jacks.
According to the English User Manual (p. 15):

Quote:
17. Line outputs 1 and 2 - 2 x ¼” (6.35 mm) TRS jack sockets; +4 dBV output level (variable), electronically balanced. Either ¼” TRS (balanced connection) or TS (unbalanced connection) jack plugs can be used.
18. Line outputs 3 and 4 – 2 x phono (RCA) sockets; unbalanced outputs, -10 dBV level (fixed).
19. Line outputs 1 and 2 – 2 x phono (RCA) sockets; unbalanced outputs, -10dBV level (variable). These sockets carry the same signal as [17] at all times.
The manual shows connecting channels 1-2 (either the RCA or TRS) to your primary powered monitor speakers. Channels 3-4 are fixed-level (not controlled by the 2.4 front panel knob) and could be used for whatever other purpose you may find convenient.

Ref:https://customer.focusrite.com/sites...user-guide.pdf
Old 12th May 2019
  #6093
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Quote:
Originally Posted by hello people View Post
What are the RCA outputs on the Focusrite Scarlet 2i4 designed for? Running to mixers? Running to speakers? There's 4x RCA output jacks.
This might be what you'd be wanting.
https://images-na.ssl-images-amazon....1RsnLPGQMS.pdf
Old 12th May 2019
  #6094
Quote:
Originally Posted by Richard Crowley View Post
According to the English User Manual (p. 15):


The manual shows connecting channels 1-2 (either the RCA or TRS) to your primary powered monitor speakers. Channels 3-4 are fixed-level (not controlled by the 2.4 front panel knob) and could be used for whatever other purpose you may find convenient.

Ref:https://customer.focusrite.com/sites...user-guide.pdf
Thanks fellas...I know...RTFM!

Those RCA outputs 1 & 2 are unbalanced variable -10dBV...I guess this means that the front rotary monitor knob can be used to make adjustments to that level?

RCA outputs 3 & 4 are fixed -10dBV.

The TRS balanced outputs 1 & 2 carry the same signal as the RCA 1 & 2 out but are +4dBV variable.

So I guess my question is...what level is typical for running signal into outboard compressors?

What is a 'line' level?
Old 12th May 2019
  #6095
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Richard Crowley's Avatar
 

The front knob controls both the 1-2 TRS outputs, and the 1-2 RCA outputs.

The TRS outputs are "professional" balanced line level +4dBm, and the RCA outputs are unbalanced "consumer" -10dB line level.

Most pro gear is designed to operate on +4dBm professional line level.
But most modern gear will also operate on -10dBm consumer line level.
Probably not something to spend a lot of time worrying about unless you run into a problem.

https://en.wikipedia.org/wiki/Line_level
Old 12th May 2019
  #6096
Gear Head
 

Quote:
Originally Posted by terrible.dee View Post
1. You are not seeing the forest for the trees. The principal is this: RECORD WHAT YOU WANT TO HEAR

"EQ and Compression" are not a phase though one must pass on their way to the top of the mountain, they do not exist for their own sake.

They are tools to help you get THE SOUND YOU WANT, like a guitar amplifier, or a certain type of room.

Why record with amps? You can go back and re-amp to your taste if you just record direct. Why go through the trouble to record in a certain room? Why not keep it dry and use a reverb plugin?

You can certainly do or not do any of these things depending on why you are doing it in the first place, but if you are recording something, and you have a "vision" as to what the thing you are recording is supposed to be, then you record THAT THING.

You get THAT THING by whatever means necessary, nothing else matters.

If you are concerned about limiting your options down the line, then I suggest it's because you have no vision for what it is you are recording.

Focus on having a vision, not on "This tool, that tool and when"

Unless, the tools are what you are into, in which case......I don't know, I don't relate.
Ok, so disclaimer first, I'm asking these questions from the perspective of a musician who writes and records his own music. Everything. So about that vision... I have it. But that may mean we talk past each other if that's not what you do, so I'm trying to keep that in mind. Alright, moving on.

I get your point, but you kind of contradict yourself. EQ and compression are tools, correct, which means you shouldn't necessary use them on the way in. So my question stands as to why you should have a channel strip or use them going in. Of course you should have a vision of what it should sound like, but my question still stands of whether or not EQ and compression should be used to achieve that vision going in, or not. I say NOT, because you can't undo it later if your vision changes, (or the artist's vision changes if you're recording someone else) and you can always send the track back out to that same hardware (question 2) if you want that particular piece of hardware (or plugin). I'm not fetishizing EQ and compression. The exact opposite, actually. I don't get the point of using them going in. Not useful, in my opinion. You do make a very good point about rooms, which cannot be done any other way. But even Joe Barresi uses re-amping, cab and speaker sims, plugins, etc., and he's widely regarded as one of the best at getting guitar sounds in the world.

Sometimes I think it's a "this is the way it's been done" argument. People in the past did it this way because they tried to match what they heard in their head. Great. But I bet they would have done re-amping, etc. and gone a different way if it were easier, quicker, and more flexible. I completely agree with committing to a sound IF that is the EXACT sound you are after. It's often not, because it needs to sit within a mix. A guitar or other track is useless, no matter how well it fits your vision for that sound, if it doesn't fit the mix. You'll change that sound anyways. Plus, committing doesn't have to happen at any particular point until it sounds right. You don't commit just because you're tracking... Oh God, we can't change it now or we aren't adults... Limiting mindset.

So really, all I'm getting from you is that it should be our goal to develop our ears to the point of knowing exactly what we want, for each instrument, before tracking... And EQ and Comp (and channel strips) are useful when we get to that point IF you need to use them going in. This is sound advice, of course.

But if you have the right room, mic, placement, performance, etc. etc. etc., you may not need EQ and Comp anyways... That's smarter recording, right... To not need them? So to me, "by whatever means necessary" is better served not using EQ and compression going in. You get that sound in context with the whole mix at the source, then use EQ/Comp/whatever later if you need it. That to me, is seeing the forest AND the trees, and that's what good engineers do. The vision of the whole mix, the whole album. You don't need anything in particular going in to achieve that vision.

Quote:
Originally Posted by terrible.dee View Post
Jesus Christ: "..And if the blind lead the blind will they both not end up in the ditch?"
Someone's an angry elf...
Old 12th May 2019
  #6097
Gear Guru
Quote:
Originally Posted by mortalsphere View Post
Ok, so disclaimer first, I'm asking these questions from the perspective of a musician who writes and records his own music. Everything. So about that vision... I have it. But that may mean we talk past each other if that's not what you do, so I'm trying to keep that in mind. Alright, moving on.

I get your point, but you kind of contradict yourself. EQ and compression are tools, correct, which means you shouldn't necessary use them on the way in. So my question stands as to why you should have a channel strip or use them going in. Of course you should have a vision of what it should sound like, but my question still stands of whether or not EQ and compression should be used to achieve that vision going in, or not. I say NOT, because you can't undo it later if your vision changes, (or the artist's vision changes if you're recording someone else) and you can always send the track back out to that same hardware (question 2) if you want that particular piece of hardware (or plugin). I'm not fetishizing EQ and compression. The exact opposite, actually. I don't get the point of using them going in. Not useful, in my opinion. You do make a very good point about rooms, which cannot be done any other way. But even Joe Barresi uses re-amping, cab and speaker sims, plugins, etc., and he's widely regarded as one of the best at getting guitar sounds in the world.

Sometimes I think it's a "this is the way it's been done" argument. People in the past did it this way because they tried to match what they heard in their head. Great. But I bet they would have done re-amping, etc. and gone a different way if it were easier, quicker, and more flexible. I completely agree with committing to a sound IF that is the EXACT sound you are after. It's often not, because it needs to sit within a mix. A guitar or other track is useless, no matter how well it fits your vision for that sound, if it doesn't fit the mix. You'll change that sound anyways. Plus, committing doesn't have to happen at any particular point until it sounds right. You don't commit just because you're tracking... Oh God, we can't change it now or we aren't adults... Limiting mindset.

So really, all I'm getting from you is that it should be our goal to develop our ears to the point of knowing exactly what we want, for each instrument, before tracking... And EQ and Comp (and channel strips) are useful when we get to that point IF you need to use them going in. This is sound advice, of course.

But if you have the right room, mic, placement, performance, etc. etc. etc., you may not need EQ and Comp anyways... That's smarter recording, right... To not need them? So to me, "by whatever means necessary" is better served not using EQ and compression going in. You get that sound in context with the whole mix at the source, then use EQ/Comp/whatever later if you need it. That to me, is seeing the forest AND the trees, and that's what good engineers do. The vision of the whole mix, the whole album. You don't need anything in particular going in to achieve that vision.



Someone's an angry elf...
I would say that personally I like to record with compression and not a lot. Obviously that gets printed but it allows me to sing/play into it, and I find it pleasing. I monitor reverb for the same reason but don’t print it. Eq I don’t need but again is something I’d monitor if I did.

Committing when tracking is really a personal choice. Obviously better to leave everything to post, but I only have one outbound comp so use it that way. I also have a tube pre, and gain stage for color.

If I was recording other people I’d try for maximum flexibility. For my stuff I have a pretty good idea of what I like as a palette. Saves time and is inspiring....
Old 16th May 2019
  #6098
Gear Head
 

Quote:
Originally Posted by ardis View Post
I would say that personally I like to record with compression and not a lot. Obviously that gets printed but it allows me to sing/play into it, and I find it pleasing. I monitor reverb for the same reason but don’t print it. Eq I don’t need but again is something I’d monitor if I did.

Committing when tracking is really a personal choice. Obviously better to leave everything to post, but I only have one outbound comp so use it that way. I also have a tube pre, and gain stage for color.

If I was recording other people I’d try for maximum flexibility. For my stuff I have a pretty good idea of what I like as a palette. Saves time and is inspiring....
That's good to hear. I can definitely understand recording with comp if you feel more inspiration through it, and also monitoring things like reverb that can definitely inspire and encourage a good performance. I've done that also.

When you are recording yourself, it all comes down to personal taste and workflow, I suppose!
Old 16th May 2019
  #6099
Gear Guru
Quote:
Originally Posted by mortalsphere View Post
That's good to hear. I can definitely understand recording with comp if you feel more inspiration through it, and also monitoring things like reverb that can definitely inspire and encourage a good performance. I've done that also.

When you are recording yourself, it all comes down to personal taste and workflow, I suppose!
Funny just perusing the Sheps thread and he uses compression going in. Again light because you'll have to live with it!. Verb, etc I definitely monitor and don't print.......
Old 29th May 2019
  #6100
Lives for gear
 

It's been years since I've read about this so hopefully someone can refresh my memory. I remember reading somewhere years ago that when a digital fader is not at unity, that it's... 'not making use of all the bits' or something. I forget. Something like that. I think it was because I was using my master digital fader in my DAW as the actual volume control since my interface didn't have a master monitor controller. I had to turn down my master fader to like -3billion decibels to get a proper listening level and hope that I didn't accidentally click it and make it jump to the top and give me hearing damage. Once I got a separate monitor controller though, the master fader stayed a unity.

Wouldn't the problem be that essentially my bit depth would be much, much less, and that the digital noise floor would be much more prevalent since the audio wouldn't be pushed up over it by very much?

I'm asking primarily because these days I'm thinking of using fader automation in my DAW when mixing instead of riding my console's analog faders. I think I'd essentially be introducing the same problem I'm talking about here with bit depth and noise floors (that's how using digital faders works, right?), except now it's across multiple audio tracks and not a master fader. But I'd only turn up/down the individual digital faders very slightly when needed and not -3billion so it's probably nothing to worry about.

Now I'm wondering what the equivalent effect would be for analog faders, as far as noise that they can introduce at various levels. Hmm...
Old 29th May 2019
  #6101
Quote:
Originally Posted by HSLand View Post
It's been years since I've read about this so hopefully someone can refresh my memory. I remember reading somewhere years ago that when a digital fader is not at unity, that it's... 'not making use of all the bits' or something. I forget. Something like that. I think it was because I was using my master digital fader in my DAW as the actual volume control since my interface didn't have a master monitor controller. I had to turn down my master fader to like -3billion decibels to get a proper listening level and hope that I didn't accidentally click it and make it jump to the top and give me hearing damage. Once I got a separate monitor controller though, the master fader stayed a unity.

Wouldn't the problem be that essentially my bit depth would be much, much less, and that the digital noise floor would be much more prevalent since the audio wouldn't be pushed up over it by very much?

I'm asking primarily because these days I'm thinking of using fader automation in my DAW when mixing instead of riding my console's analog faders. I think I'd essentially be introducing the same problem I'm talking about here with bit depth and noise floors (that's how using digital faders works, right?), except now it's across multiple audio tracks and not a master fader. But I'd only turn up/down the individual digital faders very slightly when needed and not -3billion so it's probably nothing to worry about.

Now I'm wondering what the equivalent effect would be for analog faders, as far as noise that they can introduce at various levels. Hmm...
A typical DAW today operates internally with at least 32 bit float precision which corresponds to something like 1000 dB headroom. So riding faders has absolutely no detrimental impact on the noise floor in your DAW.
Old 29th May 2019
  #6102
Quote:
Originally Posted by HSLand View Post
It's been years since I've read about this so hopefully someone can refresh my memory. I remember reading somewhere years ago that when a digital fader is not at unity, that it's... 'not making use of all the bits' or something. I forget. Something like that. I think it was because I was using my master digital fader in my DAW as the actual volume control since my interface didn't have a master monitor controller. I had to turn down my master fader to like -3billion decibels to get a proper listening level and hope that I didn't accidentally click it and make it jump to the top and give me hearing damage. Once I got a separate monitor controller though, the master fader stayed a unity.

Wouldn't the problem be that essentially my bit depth would be much, much less, and that the digital noise floor would be much more prevalent since the audio wouldn't be pushed up over it by very much?

I'm asking primarily because these days I'm thinking of using fader automation in my DAW when mixing instead of riding my console's analog faders. I think I'd essentially be introducing the same problem I'm talking about here with bit depth and noise floors (that's how using digital faders works, right?), except now it's across multiple audio tracks and not a master fader. But I'd only turn up/down the individual digital faders very slightly when needed and not -3billion so it's probably nothing to worry about.

Now I'm wondering what the equivalent effect would be for analog faders, as far as noise that they can introduce at various levels. Hmm...
A typical DAW today operates internally with at least 32 bit float precision which corresponds to something like 1000 dB headroom. Riding faders has no practical impact on the noise floor in your DAW.
Old 31st May 2019
  #6103
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I want to record my bass using a DI into my mixer, but I also want the sound of a dirty Sansamp combined with it. I plan to buss these over to a group and blend as necessary. But I'm thinking that since running into the Sansamp, even though it's analog, it might introduce an ever so slightly delay which might cause phase issues when summed with the DI signal. The trick to fixing it is in the mix by zooming in and nudging. BUT! What if I want to get it perfect on the board itself in the first place? I'm not sure it'll even create that small delay but it wouldn't surprise me. So I wonder how people fixed this back in the analog days, and how they even knew and measured if there was a phase issue.
Old 31st May 2019
  #6104
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Piedpiper's Avatar
no appreciable delay in analogue.
Old 31st May 2019
  #6105
Lives for gear
 

Quote:
Originally Posted by HSLand View Post
I want to record my bass using a DI into my mixer, but I also want the sound of a dirty Sansamp combined with it. I plan to buss these over to a group and blend as necessary. But I'm thinking that since running into the Sansamp, even though it's analog, it might introduce an ever so slightly delay which might cause phase issues when summed with the DI signal. The trick to fixing it is in the mix by zooming in and nudging. BUT! What if I want to get it perfect on the board itself in the first place? I'm not sure it'll even create that small delay but it wouldn't surprise me. So I wonder how people fixed this back in the analog days, and how they even knew and measured if there was a phase issue.
Back in the day ... taking a Bass guitar Direct as well as Mic'ing up the cabinet, we used a Digital Delay [Fx box], and manually dial in a delay setting on the Direct.

Another example, 'spot' mics in a large ensemble recording [Big Band, Orchestra], we would bring out the tape measure and calculate the delay time to the Main Mics, and apply delay to the 'spots' .
Old 31st May 2019
  #6106
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12ax7's Avatar
 

Quote:
Originally Posted by RJHollins View Post
Back in the day ... taking a Bass guitar Direct as well as Mic'ing up the cabinet, we used a Digital Delay [Fx box], and manually dial in a delay setting on the Direct.

Another example, 'spot' mics in a large ensemble recording [Big Band, Orchestra], we would bring out the tape measure and calculate the delay time to the Main Mics, and apply delay to the 'spots' .
While it is true that this technique works very well as described for bass, in other situations it can sometimes spell trouble:

Sometimes, he closer you get the instruments "in phase" using these manipulations, the more you get the ROOM out of phase!

This can often "screw the pooch" (and create more harm than good).

I'm not saying that this technique is useless; I'm just saying that it often kills the beauty of the ambiance (while returning comparatively little benefit).

...As always, YMMV...
.
Old 1st June 2019
  #6107
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Why do some analog gear are “transformerless” ?
Old 1st June 2019
  #6108
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Richard Crowley's Avatar
 

Quote:
Originally Posted by planck View Post
Why do some analog gear are “transformerless” ?
Back many years ago before transistors and integrated circuits, we had only tubes ("valves") for audio circuits. Tubes are rather high impedance, and it took transformers to interface them to the "Real World" So most audio gear had input transformers and output transformers. (And sometimes even more transformers internally). So essentially ALL audio gear had transformers.

But we got transistors and integrated circuits. And some standards for audio gear interfacing changed, so it is quite common to find both consumer and pro audio gear without input or output transformers. Because we can implement the same kind of input and output interfacing to other gear directly with transistors or integrated circuits and transformers are not needed.

But even here in the 21st century we still have audio transformers for two major reasons:

1) Some people like the "sound" of tube gear. And the "sound" of tube gear is mostly the sound of those transformers. So many people "harvest" or "recycle" transformers from old discarded/redundant gear. They like putting random transformers in the audio path to see what kinds of different "sounds" they can get out of them. And there is still some audio gear that copies those old designs that had the transformers. Alas, many of the companies that made those transformers are gone out of business. So some models of old transformers are valuable like gold.

2) In many cases, it is valuable to ISOLATE audio circuits. Like if you have a problem with a ground-loop causing hum, a common remedy is to use an isolation transformer. Or if you need to take a high-impedance, unbalanced instrument signal (as from an electric guitar) and convert it to low-impedance balanced (to look like a microphone). Or if you want to "split" a microphone signal to two or more destinations (house, monitor, recording, etc.)

Transformers introduce distortion into the audio path. They have limits to their low frequency response, and limits to their high-frequency response, and limits to their voltage/power level. And they add harmonic distortion and intermodulation distortion. But some people LIKE that kind of distortion.

AND transformers are large, heavy, and expensive compared to the solid-state modern equivalent. So that is why most modern audio gear is "transformerless".
Old 3rd June 2019
  #6109
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norfolk martin's Avatar
 

Quote:
Originally Posted by 12ax7 View Post
While it is true that this technique works very well as described for bass, in other situations it can sometimes spell trouble:

Sometimes, he closer you get the instruments "in phase" using these manipulations, the more you get the ROOM out of phase!

This can often "screw the pooch" (and create more harm than good).

I'm not saying that this technique is useless; I'm just saying that it often kills the beauty of the ambiance (while returning comparatively little benefit).

...As always, YMMV...
.
I agree. Unless its an acoustic bass, the mic is usually right on the cab, and the time difference between DI and Mic is going to be fractions of a millisecond (1" = approx 0.09 mS) This will have very little effect at bass frequencies, but might be enough to upset some HF ambiance.

The further away the mic gets, the more corrections is needed, and the worse the effect on the spill.
Old 8th June 2019
  #6110
Gear Head
 
IGotWorms's Avatar
 

I know that console users in the 60's relied heavily on the vu meter calibrated to -18 for the individual tracks. What was the meter on the master bus calibrated to? It must have been higher or else the master would be constantly slammed. Any factual answers?
Old 8th June 2019
  #6111
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12ax7's Avatar
 

Quote:
Originally Posted by IGotWorms View Post
I know that console users in the 60's relied heavily on the vu meter calibrated to -18 for the individual tracks. What was the meter on the master bus calibrated to? It must have been higher or else the master would be constantly slammed. Any factual answers?
On every board I ever worked on, the VU meters on the master bus were calibrated to the same levels as the other VU meters.

...But while we're talking about this:

VU is NOT directly referenceable to peak metering!:

Even though both meters are calibrated in dB, Peak Meters are inteded to measure levels instantaneously, while VU meters read sort of an "average" over time.

...So a VU meter can get hit with a pretty hard-ass peak and barely move at all, while a steady-state tone will read very high in comparison.

This keeps us from being able to do a direct reference between VU and Peak readings on real-world program material.
.
Old 9th June 2019
  #6112
Lives for gear
 

... and why many high end consoles had a 'plasma' PEAK meter along with a VU meter.

The Console Meter was the 'master house meter'. Tones sent from the Console [with its meter] was used to cal all the Outboards and Decks.
Old 4 weeks ago
  #6113
Gear Head
 
IGotWorms's Avatar
 

Was he listening to himself through a monitor?

Old 4 weeks ago
  #6114
Quote:
Originally Posted by IGotWorms View Post
Was he listening to himself through a monitor?

When this video was recorded he'd been singing live with orchestras for at least 25 years. Without monitors. My guess is he nailed it whatever sound system he was faced with.

Note to self: go practice vocal chops
Old 4 weeks ago
  #6115
Lives for gear
 
12ax7's Avatar
 

Quote:
Originally Posted by IGotWorms View Post
Was he listening to himself through a monitor?

No.

He was actually there when it was being recorded.

...And he wasn't deaf, so he could hear himself.

Believe it or not, there was music (and vocals) a long time before people started recording it.

...Hell, Beethoven was farking DEAF for cripe's sake, and even he didn't use any stinking monitors)!
.
Old 4 weeks ago
  #6116
Gear Addict
 
Blackdog128's Avatar
Quote:
Originally Posted by thedberg View Post
When this video was recorded he'd been singing live with orchestras for at least 25 years. Without monitors. My guess is he nailed it whatever sound system he was faced with.
...
So great to see & hear that fantastic piece of history! It brought tears to these old eyes (and the orchestra ended on the dominant!).
Old 4 weeks ago
  #6117
Lives for gear
 
12ax7's Avatar
 

Quote:
Originally Posted by IGotWorms View Post
When using a compressor for mastering how should impedance be observed? I think the interface is sending out a low impedance signal, and receiving a low impedance line level signal. Is there no effect if I leave it set at a high impedance?
Well, (whether for mastering or any other purpose) it really depends upon the topology of the compressor in question (as well as that of any device to which it is connected):

Most "modern" (solid state) gear is quite forgiving about impedance...

...But back in the olden days (when I was young and attractive, and dinosaurs roamed the Earth), "Impedance Matching" was a BIG DEAL (and can still be relevant even today when using older gear).

So if you REALLY want to understand this (yeah, sounds like a great way to spend a Sunday afternoon), you'll have to grok the difference between "constant voltage" and "constant current".

While most folks probably NEVER will want to know this much about it, here's a good place to start for anyone who really DOES want to do so:
Termination, Impedance Matching, Maximum Power Transfer Theorem in Audio Systems
(The trick is to just skip over the parts you don't quite grok upon first reading; later you can come back to these part when you have questions.)
.

Last edited by 12ax7; 4 weeks ago at 09:00 PM..
Old 3 weeks ago
  #6118
Gear Head
 
IGotWorms's Avatar
 

How do you deal with compressors that have few markings for gain reduction levels? How do you make an album have consistency if different songs have more or less reduction and makeup?
Old 3 weeks ago
  #6119
Gear Guru
 

Quote:
Originally Posted by IGotWorms View Post
How do you deal with compressors that have few markings for gain reduction levels? How do you make an album have consistency if different songs have more or less reduction and makeup?
Setting the markings the "same" is no guarantee of 'consistency' anyway. Compressors react differently to different incoming signals. Unless your album consists of the same song 12 times, you are going to have to use your ears.

or more conveniently, use a mastering engineer's ears.
Old 3 weeks ago
  #6120
Lives for gear
 
12ax7's Avatar
 

Quote:
Originally Posted by IGotWorms View Post
How do you deal with compressors that have few markings for gain reduction levels? How do you make an album have consistency if different songs have more or less reduction and makeup?
Compressors don't have "markings for gain reduction levels":

They have "markings" for things like "attack time" "release time" "threshold" "ratio" (and whatnot).

Gain reduction happens as a result of these settings (and the program material inputted), and the result is thereafter shown by some sort of METERING; (NOT by "markings").

...But even if one was using a compressor with none of these things available, one would still "deal with it" in much the same way as one would "deal with it" when it DOES have them:

By LISTENING!

It's like asking "How do you know which way to turn the steering wheel without a roadmap?"
.
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