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The Reason Most ITB mixes don’t Sound as good as Analog mixes (restored) Effects Pedals, Units & Accessories
Old 6th September 2009
  #61
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fossaree's Avatar
intriguing , but it makes sense ...
Old 6th September 2009
  #62
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fossaree's Avatar
jsl , you know what? i'm mixing something right know , and gonna pay some special attention about setting LPF ... be back later .
Old 6th September 2009
  #63
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drBill's Avatar
Quote:
Originally Posted by jslevin View Post
This is pretty intriguing, never heard of this before. Anything published on this subject by the digital gurus?
It wouldn't be a secret then, would it??
Old 6th September 2009
  #64
The high frequency thing seems to match my (admittedly limited but pretty heavily exploratory) experience. To get a sound that matches the kind of mixes I like, it seems like I'm cranking down the highs on lots of tracks. Not necessarily a low pass, sometimes it's just removing some of the presence frequencies, and leaving some of the air. Sometimes it's a low pass.

Particularly if you want to create any sort of sense of depth in the mix, you can't have things that are supposed to be behind the speakers all bright and sparkly. If all that high end is captured and it never gets lost, no matter how many times you 'run the bits over the record head', they have to be removed manually sometimes.

Obviously compression does this job to some extent as well, since in some cases like drums the highest frequency content is often in that initial attack that will be reduced by any fast attack compression.
Old 6th September 2009
  #65
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drBill's Avatar
Quote:
Originally Posted by Dean Roddey View Post
The high frequency thing seems to match my (admittedly limited but pretty heavily exploratory) experience. To get a sound that matches the kind of mixes I like, it seems like I'm cranking down the highs on lots of tracks. Not necessarily a low pass, sometimes it's just removing some of the presence frequencies, and leaving some of the air. Sometimes it's a low pass.

Particularly if you want to create any sort of sense of depth in the mix, you can't have things that are supposed to be behind the speakers all bright and sparkly. If all that high end is captured and it never gets lost, no many how many times you 'run the bits over the record head', they have to be removed manually sometimes.

Obviously compression does this job to some extent as well, since in some cases like drums the highest frequency content is often in that initial attack that will be reduced by any fast attack compression.
Ok then, now were starting to get somewhere. heh heh
Old 6th September 2009
  #66
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Quote:
Originally Posted by Skip Burrows View Post
Sure I can tell you but then I have to Kill you...
Ha Ha
Ha--I deleted b/c I thought you gave it away already with the URS post which wasn't up whilst I was writing...hmmm...Well, then, keep your secrets!
Old 6th September 2009
  #67
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Hi Skid,
this is exactly how I have been mixing 100% of all time, delivering gold/platinum-selling records mixed ITB. It's so obvious, isn't it?
I now have a G-Series (console, not plug-in) as well and the great thing about feeding it with Converters (versus Tape in the old days) is that we can hit that console at the "gain staging sweet spot" using plugs to prepare the signal in a way that makes console mixing a breeze!

Isn't it a wonderful world?

Cheers,
Marc

Quote:
Originally Posted by Skip Burrows View Post
You folks will think I am smoking crack but here is what I will normally do. I put the trim plug in first followed by either the SSL channel or the URS strip Pro. Then with the trim plugin up on the encoders Of My Icon D-dontrol. I put the fader at around 0. Then I adjust the trim plugin until the track fits into the mix fairly well. Now just EQ and compress to taste and do fine adjusts with the fader. This always works well and I never have a headroom issue. Plus all my fader are at the sweet spot. I know I am smoking Crack.
Very true, but keep in mind it requires 24bit recording. I still remember the early 2000s when 24bit was quite new.
24bit give you 144dB of dynamic, more than the dynamic any converter can "display". Loosing 20dB on the way in doesn't mean ****. Your recordings resolution can still be higher than your converters dynamic.

Quote:
Originally Posted by zboy2854 View Post
You could do that as well. One of the other most insidious myths that has been propogated is the myth about "maximizing the bits" in digital. Track at lower levels and you'll be just fine.
Old 6th September 2009
  #68
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h4nc0's Avatar
Thanks for this thread! heh
Old 6th September 2009
  #69
Gear Head
 

Great thread.

One thing that gets under my skin is that meters on DAWs are peak meters rather than averaging ones like VU meters. I always seem to get zoomed in a hybrid situation on whether I am reading peak or average levels and I hate guessing at what a VU meter would look like when I'm reading a DAW style peak meter. I would think it would not be impossible to have DAW meters be averaging and allow a custom configuration of the 0VU point?

Drass.
Old 6th September 2009
  #70
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Tom H's Avatar
 

Quote:
Originally Posted by hankdrummer View Post
can someone please make a one sentence conclusion of this post ?

thanks !
There is this magical trim plugin; it will make your mixes sound more analog!

Old 6th September 2009
  #71
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Quote:
Originally Posted by Skip Burrows View Post
The Reason Most ITB mixes don’t Sound as good as Analog mixes. This is a repost from another thread. Hope you find it usefull.

Ok, I'm going to try and give you An ITB education, as my over 24 years has taught me. Here is what I try and teach to students. I'll try and keep the math to a minimum.

First, I own a high end analog setup's Via an SSL 4K with 1/2' 2 Track YADA YADA, ICON with Killer OB FX And classic Compressors, YADA YADA/ Hybrid Setup Via AWS 900w/ 24 Channels Of Xlogic Killer OTB FX and Comps YADA YADA. Point is not to impress, or brag in any way, but to let you know everyday I work on a verity of systems. This has led me to The Following conclusion.

To learn to mix ITB coming from an analog world you must revisit what Voltage reference Analog consoles work at, and make appropriate adjustments to translate this to work ITB.

The first thing we must ask is simply what is 0VU. What does it mean to us. Lets use an SSL G+ as our point of reference mainly because I work on those every day. If we put a signal into the line input of the SSL so the channel meter reads 0vu, that also, is referenced as +4 or 1.23 volts. A kick ass SSL will go out to about +24DB, so we have approximately 20 DB of headroom above the 0 VU point on the meter before the signal goes to crap.

Now let take a common situation. A Client hands you a Protools session and you spread it out over the SSL console. Like most people today every track is recorded as hot as hell. Most pro Eng's will use proper gain staging and get the now slammed meters reading around 0VU or 1.23 volts. By lowering the line trim we now have a good level into the desk so we can Compress/Gate/EQ the Signal without it overloading the processing. Sounds simple right? Remember that all outboard equipment was designed to work around the 0VU/+4/ 1.23 Volt reference. So by putting the incoming signal at around this reference, your rack equipment will work better as well.

Why use a +4 reference? Well remember that the 1.23 volt reference came from the tube days where 1.23 volts was enough voltage over the plate noise that you still had a good signal to noise ratio, but still left room above 1.23 volts to allow for normal audio operations.

Now to ITB. Lets pretend we have the same setup as we did on the SSL. Client hands you a session that’s recorded hot as hell. Now most folks mixing ITB don't understand reference levels when relating it to Digital. To have the same amount of "headroom" as we do on the SSL we must create a reference of 0VU or 1.23 volts at -20 from 0DBFS or the top of the Digital scale.

So if you simply place the good old trim plugin as the very first plugin, you now have the ability to adjust your tracks to our Mixing (+4/1.23 volt) reference IE -20. Just like you did on the SSL. You have have the same amount of headroom. Now with your tracks properly gain staged, you can add EQ/dynamic plugins and not run out of headroom. You can also insert hardware and they will operate much better as they are operating at the level they were designed to operate at.

Plugins use the same reference at real equipment. Never try and drive them to the top of the Digital scale. Don't try and make your mix look like a master. You don't do that on an analog console, so why do we do it ITB?

The answer is simple. DAW meters suck Butt. There should be a meter mode in all DAW's that makes the meter at 3/4 scale equal -20 at 1.23 volts. Just like the old VU. This way, novices will quit corn-holeing their levels.

Something to think about. The noise floor of an analog desk is about -75 DB from our +4 reference. Our equivalent "problem level" below our -20 reference in digital is well over 100 DB. So please don't let people tell you analog has more "headroom" than digital. This is simply not true. Headroom is only relative to your noise floor below your reference. Remember if the volume is to low, turn up the darn speaker volume.

Running a Digital mix right to the top of the scale is like running your SSL mix buss where the VU meters are slammed all the way to the right and you are constantly hitting it at +25. No one will get a good sounding running the desk like that. You won’t get a good sounding mix in digital either.


So what does all this mean? Put simply, proper gain staging is essential to both analog and digital mixing. You just need to correlate the references between the two. Once you figure this out, I'll Guarantee your mixes will start to sound open and wide, just like the good old analog days.

Everyone should read this post and read it again until the message sinks in :-) I can't count the number of times I have posted exactly this stuff.

I helped to design much of the celebrated analogue consoles mentioned, much of the digital stuff - and more recently several of the most highly regarded plug-ins. And believe me, this is the single biggest issue affecting the quality of your digital work.

The biggest 'own goal' of the whole of the digital industry from the outset has been the total disregard for the notion of operating levels and headroom.

Do not accept it - if your production environment dictates that you must provide limited, maximised and level-blasted material, just do it at the end :-)
Old 6th September 2009
  #72
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Quote:
Originally Posted by Drass View Post
Great thread.

One thing that gets under my skin is that meters on DAWs are peak meters rather than averaging ones like VU meters. I always seem to get zoomed in a hybrid situation on whether I am reading peak or average levels and I hate guessing at what a VU meter would look like when I'm reading a DAW style peak meter. I would think it would not be impossible to have DAW meters be averaging and allow a custom configuration of the 0VU point?

Drass.
This is because in a system without headroom where there is no overload margin you cannot have an averaging meter - as it will not show you the clipped overs! It was only possible to have these in analogue as the whole recording chain + the tape machine had a valid overload level region that did not hard clip the signal.

This is all part of the culture of what's wrong with digital that people are using and many have used exclusively- they have never actually experienced a system that was properly scaled.

And you're right - that there was never any technical reason why it needed to be this way (except the way digital grew up and naivety). The OXF-R3 console had an internal headroom value that could be changed at start-up and reported operating level to all internal processes. It was typically set at around 24dB (i.e. -24dB below clipping), similar to the analogue world - for very good reasons :-)
Old 6th September 2009
  #73
Post #1 = A great candidate for a "Sticky"
Old 6th September 2009
  #74
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So what does this say about all these so called mastering engineers doing it in the box......???
Old 6th September 2009
  #75
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Quote:
Originally Posted by Dean Roddey View Post
The high frequency thing seems to match my (admittedly limited but pretty heavily exploratory) experience. To get a sound that matches the kind of mixes I like, it seems like I'm cranking down the highs on lots of tracks. Not necessarily a low pass, sometimes it's just removing some of the presence frequencies, and leaving some of the air. Sometimes it's a low pass.

Particularly if you want to create any sort of sense of depth in the mix, you can't have things that are supposed to be behind the speakers all bright and sparkly. If all that high end is captured and it never gets lost, no many how many times you 'run the bits over the record head', they have to be removed manually sometimes.

Obviously compression does this job to some extent as well, since in some cases like drums the highest frequency content is often in that initial attack that will be reduced by any fast attack compression.
Sorry to do 3 posts in a row - but getting carried away in a quiet moment in an empty house :-)

The point you make here about freq content versus mix depth is exactly correct and very very important. :-)

In the natural world, stuff that is further away tends to have less HF content because of distance effects.

So getting depth into a production made from close mic'ed or direct sources does involve judicious EQ in the mixing process - remembering that the term 'equaliser' was actually invented by the early film industry for a device that equalised for distance effects. That was it's very purpose :-)

In the analogue world, losses here and there in the signal chain could often have the effect of providing a degree of this HF loss during production, multiple passes of tape, bouncing down etc etc. could sometimes effectively put the 'backing into the background'..

But in the lossless 'what you hear is what you get - always and forever' digital world no such effect happens by itself. So it's necessary for people to be fully aware of the art of mixing and apply it themselves :-)

One of the historical issues (now much less important as W/S get more powerful processing) is that to get this kind of stuff right you need to have EQ simultaneously running on many tracks while doing the mix as a whole - so you can tweak them against each other to get the correct depth impression.

Whilst simultaneous EQ on all channels was the norm in analogue console - in the early processing strapped digital world such a luxury was not available. So this fundamental art of mixing was either lost or never learnt..
Old 6th September 2009
  #76
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Quote:
Originally Posted by jslevin View Post
Skip -- great post!

Quote:
Originally Posted by drBill View Post
Many years ago when talking with an engineer (the slide rule kind, not the faderjocky kind) who designs the stuff we use, he asked me if I would ever do a 2" analog mix on a Neve without using HPF's. I said "no, they are essential to cleaning out the mud and getting a tight and open sounding mix". He said "Good. In the same respect, you shouldn't be doing an ITB mix without using LPF's either". Same problem, opposite end of the spectrum. 180 degree problem. The buildup on a super accurate HF's, what you put in is what you get out, digital playback system is equally destructive to getting a smooth top end as LF mud is to getting a clean mix on the bottom.

As to how to do that, what to use where and where to set your LPF's, I'll leave that to your own experimenting. I can't deliver it all in a nice neat package now, can I???
This is pretty intriguing, never heard of this before. Anything published on this subject by the digital gurus? Anyone else have any experience with this?

JSL
Ok sorry for the 4th post in a row - I'm on a roll here :-)

The business of LF control and tightness is another extremely important subject - but it has several dimensions. I hope I get them in the right order....

You are absolutely right that rolling off extreme LF and 'mud' does significantly increase tightness and impact of bass parts and large percussion like kick drums etc. This is another art of mixing that people are re-learning and is stock in trade for experienced engineers :-)

However - harking back to the headroom issue (or lack of it in digital), rolling of LF increases the signal peak level most of the time - because it changes the wave shape (by differentiation)! Changing waveforms can create larger peak levels - even if you have done so by 'losing something'! So doing this on a digital system with everything flat out - and avoiding overs - will actually reduce the level and presence of the instruments involved, and even the whole mix, if you're aiming at high volumes and modulation!

Ok - so now if we go back to the completely analogue signal path 2 things are very important:

1. It had headroom, so rolling of LF deliberately with EQ did not cause clipping. Tape had large headroom values at LF and would tolerate this well - as would the analogue play-out systems in people's homes. Your 'tightened sound' made it all the way out largely unscathed..

2. ALL analogue systems at every level and every instance - and fundamentally at the very basis of the format removed DC. DC and extreme LF (below 5Hz or so) could not pass through the system at all.

So you can view an analogue signal chain as a successive series of high pass filters, the more stuff in the signal path - the steeper the extreme LF roll-off forcibly becomes.

So what this means in practice is that under these conditions the analogue system itself can actually cause a tightening effect on the sound - just by passing it around through various analogue units!

And the increased peak levels this causes do not show up on VU meters - and can pass through the system without clipping. You don't really even know it's happened - but it sounds different :-)

BTW - this effect was not lost or ignored to the more savvy and experienced designers of analogue stuff ;-)


OK - now if we go back to a digital system; apart from the ADC and DAC nothing else in the processing chain naturally filters LF, unless it is done deliberately like in compressors and the like, where the corner freq will be so low (fractions of a Hz) that it does not count. What goes in tends to come out unchanged apart from the intended effect.

This means there is effectively no LF filtering - and so there is no intrinsic LF tightening effect by simply passing stuff in and out of processors.

So to get this you have to do it with an EQ by rolling off the extreme LF manually (the Oxford LF filter on the steeper settings is great at this)..

However to maintain the impact of this and avoid overs and clipping you will need to lower the levels throughout your mix to accommodate the extra peak level - and make sure your final mix maximiser/limiter 'does the right thing' in pushing the density and gain of the program without simply clipping off your tightened sound and turning it back into mush again.. The work I have been doing has been heavily involved in preserving these subtle effects - if they are actually still there in the program, while remaining compatible with current loudness trends - and the search for improvement goes on still..

So yet again operating at lower peak levels - and knowing what you are doing when mixing, rather than leaving it to little-understood side effects of your gear - can indeed give you a fantastic advantage and save you a fortune :-)

You really can have analogue style LF 'tightness' from a digital system - if you know what you're doing :-)
Old 6th September 2009
  #77
Gear Maniac
 

A little question: turning down the audio clips' volume in the timeline is the same as using a trim plugin?
Old 6th September 2009
  #78
Quote:
Originally Posted by Pawel View Post
A little question: turning down the audio clips' volume in the timeline is the same as using a trim plugin?
Depends on the Trim plug. If it has a higher internal bit resolution than your DAW's mixer, it could provide a more hi-fi means of changing the gain.

The other thing to keep in mind is that it changes the level going into the subsequent plugins, which is a very important consideration.
Old 6th September 2009
  #79
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Quote:
Originally Posted by zboy2854 View Post
This thread should absolutely be a sticky. Skip and Paul just dropped so much knowledge on this thread that its importance for those who want to mix ITB cannot be overstated.
Should be a mandatory class, not an elective or optional course. This is not trivial, nor is it simple or easy. Great thread - and Paul, thanks for the contributions and insight.

cdlt
Old 6th September 2009
  #80
Gear Maniac
 

Great thread! Another vote for a sticky!

I've actually started mixing this way about 6-12 months ago and it has made a major improvement when I'm just mixing itb. Keep my levels peaking max at around -6 dbfs, and usually hpf and lpf everything!
Old 6th September 2009
  #81
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Skip Burrows's Avatar
 

Quote:
Originally Posted by Paul Frindle View Post
This is because in a system without headroom where there is no overload margin you cannot have an averaging meter - as it will not show you the clipped overs! It was only possible to have these in analogue as the whole recording chain + the tape machine had a valid overload level region that did not hard clip the signal.

This is all part of the culture of what's wrong with digital that people are using and many have used exclusively- they have never actually experienced a system that was properly scaled.

And you're right - that there was never any technical reason why it needed to be this way (except the way digital grew up and naivety). The OXF-R3 console had an internal headroom value that could be changed at start-up and reported operating level to all internal processes. It was typically set at around 24dB (i.e. -24dB below clipping), similar to the analogue world - for very good reasons :-)
Hey Paul, It’s an real honor, In my pervious Post I was attempting (Perhaps not very well) to describe a meter that would help stop young people from screwing up digital audio. I 1000% agree with you that your meter must show when you run out of headroom. As you better than any human on the planet know, people seem to have this ill conceived notion that you must use up every bit of headroom right from the first stage of AD conversion. My meter I have been trying to work with digidesign on, would be a default mode(changeable by a preference) that would have the -20/ 1.23/+4 reference at about ¾ the scale of the meter, then the remaining 20 db of headroom would still work and show a clip. I have done extensive research with students and my research has shown that this type of design would “Help” stop them from slamming the level to the top all the time. There is something physiological about seeing level go up to the top that helps stop them. So by that rational, if your -20 ref was much closer to the top of the meter I believe that folks will be more hesitant to beat the crap of their levels. I could be wrong industry wide however, I would like to at least see the option. I’d love to hear any further thoughts from you on this important matter. Ohh, by the way IMVHO you new plugin design is the most forward thinking product in audio I’ve seen in the past 20 years. Everyone should purchase at least 2 of them!!!!. Once again it’s an honor.
Old 6th September 2009
  #82
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Melgueil's Avatar
 

We are on a roll...

Skip - is there such a 3rd party meter available ? I think Waves has a Metering plug in
Waves Dorrough Meters ?

I grew up on analog, and those reflexive habits are hard to shake (didn't help that we passed through 16 bit land on the way to 24 bit - the old "use every bit" song that prevailed back in the day).

Other than the aforementoned, manually counting backwards - what is the easist workaround ? A plug in such as above would be worth a look I might think, though I am not very famiiar with it....

cdlt
Old 6th September 2009
  #83
Quote:
Originally Posted by MikeyMike View Post
So what does this say about all these so called mastering engineers doing it in the box......???
Skip and Paul's post's are discussing headroom issue's in the digital domain and how it should be related to headroom issue's in the analog domain... for the most part, in the mixing and tracking environment.

O dbfs (full scale) and how it relates to O dbvu (volume units).

In a well designed analog console there is headroom above Odbvu to work with (sometimes in the excess of 20db). In the digital domain there is plenty of headroom below 0 dbfs to work with (in excess of 20 db).

Mastering engineers deal with headroom issues the same way as anyone else, but under slightly different circumstances since their need is to utilize the headroom to acheive the cleanist and "most of the time", most transparent result in each and every stage through which the audio is processed whether it's itb or otb.
Old 6th September 2009
  #84
Gear Nut
 

Quote:
Originally Posted by Skip Burrows View Post
Rick, A HUGE difference. The fader only will adjust the level feeding the mix buss or master fader summing. The trim plugin as the first plugin will adjust the reference level into the following processing. IE EQ/ Comp. Hope that helps.
Very good point !

I'll take note of this,


Sly
Old 6th September 2009
  #85
D K
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This post from Skip and Paul's subsequent contributions is pure gold

Skip brought this from the other thread and this is exactly what i was searching for when I commented in it...


This is when GS is at it's finest!!

Class is in - Thank you so much guys

DEFINTELY NEEDS TO BE A STICKY
Old 6th September 2009
  #86
Gear Maniac
 

Quote:
Originally Posted by Skip Burrows View Post
You folks will think I am smoking crack but here is what I will normally do. I put the trim plug in first followed by either the SSL channel or the URS strip Pro. Then with the trim plugin up on the encoders Of My Icon D-dontrol. I put the fader at around 0. Then I adjust the trim plugin until the track fits into the mix fairly well. Now just EQ and compress to taste and do fine adjusts with the fader. This always works well and I never have a headroom issue. Plus all my fader are at the sweet spot. I know I am smoking Crack.
May I ask why you don't use the input trim of the SSL Channel instead of the dedicated trim plugin?
Old 6th September 2009
  #87
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Skip Burrows's Avatar
 

Quote:
Originally Posted by Pawel View Post
May I ask why you don't use the input trim of the SSL Channel instead of the dedicated trim plugin?
You could, however I simply got in the habit of using the trim plugin. Sometimes I use a plugin that doesn't have an input control. IE the Oxford Dynamics. Love that plugin however it doesn't have an input control. So I simply got in the habit of using the trim. I do wish however (Digi are you listening) that the trim plugin had a dither section on it. It would really help in the low level detail. I will always prefer a little noise over quantization distortion every day.
Old 6th September 2009
  #88
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Skip Burrows's Avatar
 

Quote:
Originally Posted by Melgueil View Post
We are on a roll...

Skip - is there such a 3rd party meter available ? I think Waves has a Metering plug in
Waves Dorrough Meters ?

I grew up on analog, and those reflexive habits are hard to shake (didn't help that we passed through 16 bit land on the way to 24 bit - the old "use every bit" song that prevailed back in the day).

Other than the aforementoned, manually counting backwards - what is the easist workaround ? A plug in such as above would be worth a look I might think, though I am not very famiiar with it....

cdlt
No cant use a plugin meter. Only good for the final output of the mix. Think of the signal entering a console. You need to see what is going at every point of processing. So the DAW should have this style of meter I am talking about. Remember, Even if you use 5 plugins in a row for a vocal. The Nominal level should be around 1.24 volts or +4. Give or take 8 DB. A plugin meter would not allow you to see this.
Old 6th September 2009
  #89
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Bob Olhsson's Avatar
 

Quote:
Originally Posted by Pawel View Post
May I ask why you don't use the input trim of the SSL Channel instead of the dedicated trim plugin?
The trim on the plug-in is emulating the noise and distortion of the real thing. What you want to do is hit the plug-in with an average level that would be comparable to that of a tape machine. This would be -18 to -20 average and -10 to -6 peak.
Old 6th September 2009
  #90
Here for the gear
 

what about bits?

Now want to throw this in: what about bit and resolution loss? Like in a system like protools where it does a hard 24 bit quantize on the output of plugins, wouldn't putting something like a -10 trim plugin on every channel effectively remove 2 or more bits from every channel you're mixing? Am I missing something here?
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