How to get more volume?

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ashcat_lt said:
It neither knows nor cares. Nor does your DAW, for that matter.

To simplify just a bit to aid clarity:

The ADC divides the input voltage by its rail voltage, which gives it a number between 0 and 1*. This is the value that it sends to the computer, which gets stored on your hard drive and manipulated in your DAW. Your DAW then does that 20 log x thing to that number and shows you the answer on the meter.

Send the file to me. My DAW sees the same number between 0 and 1, does that 20 log x thing and shows me the same thing on my meter that you saw on yours.

Now I send it to an output. My DAC multiplies that number between 0 and 1 by it's rail, and the actual output voltage will only be the same as what you put into if the rail on my DAC is accidentally the same as that on your ADC.

Master a song to 0dbfs and load it on your phone. Your phone will try to get as loud as it can, not as loud as your computer can.

It works on the other end also, though. 1/17 is quite a bit bigger than 1/35.

Which is to say (the part of whatshisname's post that made sense) that if I tell you that I record to -10dbfs in my DAW, it doesn't tell you anything about the actual voltage I'm running on the analog path in unless I also include data about my converter.


My converters (Fostex 2424LV, Tascam US1641, PodStudioUX1) don't have analog meters. The 2424LV has digital meters which read exactly the same as my DAW, but I know from experience that it runs a little hot.


*Anything bigger than 1 is clipped off somewhere along the line, of course. Also, of course, there are rounding errors introduced - the noise that I was talking to the "number guys" about.


Edit - Actually, now that I've looked back at my converter specs, let me get even more specific. If I take any line level source, and split it to both the Fostex and the Tascam, the Fostex will always be 4db hotter in DAW.

Edit again - Wait, let me get even more specific since I think you'll get a kick out of this story. ;)

The Fostex machine feeds my studio computer. This is where I "prototype" my live rig for my most recent show. We rehearsed through it, tweaked things, played again, tweaked a little more and then decided it was fine and time to test it on the live machine. Saved the entire Reaper file to my USB stick, stuck it in the laptop, and opened it. All the same plugins live on both machines. Everything else the same. Unplugged pedalboard from Fostex, plugged into the Tascam and it immediately felt just a tad anemic and weak. I couldn't get it to hit the amp or work the edge of the overdrive the way I had gotten used to. I never really memorized the exact difference between the two, and didn't bother to go look it up at the time. I just turned up the amp sim's In knob until it did what I wanted it to do. Turned out to be right around 4db.
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Hey ashcat_lt, thanks for chiming in and explaining why what I wrote always was correct. This thread will be a learning lesson for some X million engineers out there... :yawn:

Here is what PrismSound write in their Orpheus manual:

Analog Line Inputs

Electronically balanced, with fully-balanced analog signal path
Input sensitivity:
Switchable ‘+4dBu’ (0dBFS=+18dBu) or ‘-10dBV’ (0dBFS=+6dBu)

Analog Line Outputs

Electronically balanced, with fully-balanced analog signal path
Output amplitude:
Switchable ‘+4dBu’ (0dBFS=+18dBu) or ‘-10dBV’ (0dBFS=+6dBu)
 
ashcat_lt said:
.... I tell you that I record to -10dbfs in my DAW, it doesn't tell you anything about the actual voltage I'm running on the analog path in unless I also include data about my converter..
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Mmm...we are agreeing....but maybe also looking at it in different ways.
That's what I'm saying too....the DAW has NO idea what dBFS is "correct" relative to any converter. We can choose any reference we want. It has nothing to do with specific converter headroom.
The listener's playback DAC will always dictate how thge dBFS is read....so using an ADC converter with a higher headroom spec, doesn't control listener DAC specs or playback.

We've kinda gotten off on a tangent about the digital metering, which as already mentioned, is all over the place since there is no direct conversion standard from dBu to dBFS...which is what MusicWater was trying to do and suggesting that there was some "secret" about using converters with more headroom as a way to boost your level and/or get more audio quality on the digital side....which isn't the case.
The analog headroom only matters on the analog side.

AFA the digital metering...IMO, it really just doesn't matter if you are seeing -10dBFS and I'm seeing -12dBFS for the same file because we are using different converters.
There is nothing special about using a +24 dBu converter VS a +18dBu converter. It's purely a mater of picking one over the other, and I believe in the US, the AES adopted standard is +24 dBu for 0 dBFS....in most European countries, it's +18 dBu.
Do American made masters sound better than ones done in Europe....??? :D

I still don't see the point MusicWater was trying to make about some converter headroom "secret".....?...and he's not bothered to explain, other than to quote converter specs. :)
 
MusicWater said:
Hey ashcat_lt, thanks for chiming in and explaining why what I wrote always was correct.
Click to expand...
It was honestly more to help miro (and others who might not have quite got it) to understand than to defend you specifically. When you start talking shit like...
This thread will be a learning lesson for some X million engineers out there... :yawn:
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You've lost even me. It's not some big secret. It's not a conspiracy, and it is definitely not what's keeping any of us from winning a Grammy.

Drop the sensationalist bullshit that implies that you have the keys to ultimate enlightenment (you can't, cause they're chained to my belt ;) ) and stick to facts and you will do much better around here.

Here is what PrismSound write in their Orpheus manual:

Analog Line Inputs

Electronically balanced, with fully-balanced analog signal path
Input sensitivity:
Switchable ‘+4dBu’ (0dBFS=+18dBu) or ‘-10dBV’ (0dBFS=+6dBu)

Analog Line Outputs

Electronically balanced, with fully-balanced analog signal path
Output amplitude:
Switchable ‘+4dBu’ (0dBFS=+18dBu) or ‘-10dBV’ (0dBFS=+6dBu)
Click to expand...
Wouldn't it be nice if they all just came out and said it this clearly? ecc83 was talking about the one that is "fully adjustable". It's nice, but not completely necessary as long as you know what you're working with. As he also mentioned, these values actually tell us something about how much the analog signal is attenuated before hitting the ADC chip itself. Which gets us back to the point that the pots and faders on your analog gear can make any converter "fully adjustable" in practice. I suppose that it is best for noise reasons to attenuate as late in the game as possible, but...

ecc83 - I knew you knew. I was just warning folks not to be too cavalier about it. If you've got really low levels in DAW then you're probably compromising your S/N somewhere along the way and that while we don't really need to "use all the bits", it's kind of a shame to waste too many of them, especially if we might need them later.

It is sort of amusing that a thread about minimizing the dynamic range of a recording has turned somewhat toward optimizing the dynamic range of individual tracks...
 
ashcat_lt said:
It was honestly more to help miro (and others who might not have quite got it) to understand than to defend you specifically.
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The only part I had doubts about was how a DAW would digitally meter the same signal from two different converters...which I said earlier. I don't say you are wrong...I just said I never bothered to compare, and I don't have different converters to use....all three of mine are identical, so it was/is an insignificant thing for me.

Everything else was centered around this false notion of using converters with more headroom as some sort of "secret" weapon.
I think when you use multiple converters with different specs (as you do)....then there might be some concern to match levels/etc purely for balanced playing field, if you use one for some tracks and another for other tracks in the same project....otherwise, once it's digital, it just doesn’t matter.

Ayone that purely sets their levels by the dBFS readings, might fall into the trap of thinking a converter with more headroom is better/louder...which is what I think MusicWater is doing.
 
ashcat_lt said:
If you've got really low levels in DAW then you're probably compromising your S/N somewhere along the way and that while we don't really need to "use all the bits", it's kind of a shame to waste too many of them, especially if we might need them later.
Click to expand...

This is why I think (and what I do) that metering the analog signal is the key. That's where the S/N is considered and set.
Metering the digital signal is purely there to provide some backward reference to the analog side....of course, in dBFS it is a movng target and can be set per user.
 
K good. Most of us are on the same page... I love how this thread started about one thing, petered out into a bunch of fucking around, and is now on to something that really doesn't have anything to do with it.

There is maybe something to be said about having a converter that you know will handle anything your analog gear can throw at it, but again we have knobs for that.

Nobody commented on my thing about clipping the mix to avoid clipping the converters...
 
miroslav said:
This is why I think (and what I do) that metering the analog signal is the key. That's where the S/N is considered and set.
Metering the digital signal is purely there to provide some backward reference to the analog side....of course, in dBFS it is a movng target and can be set per user.
Click to expand...
Ugg we're playing ninja tag!

In my situation, once it gets in the computer it doesn't come back out except to go to my playback system. Analog metering is important in tracking. The DAW's meter at that point is just there to make sure you've got everything plugged in correctly. Once tracking is done, though, the DAW meter is all that matters. The meters on my playback system again just tell me that everything is connected properly.

Edit - No, that's a lie. In my most common work I don't really have analog meters at all. (Shrug). In situations where I do have more analog stages along the way, though, and for the many people that do use a lot of outboard gear, analog metering is the key.
 
Don't get me wrong...I'm not saying the digital meters are not useful or have no purpose.....it was mostly about the playback level aspect I was referring to.
AFA your audio quality ("am I overloading the ADC?")....that's on the analog side, so again, metering there is key.
I know some folks have no other meters but the digital dBFS....which doesn't always paint a clear picture of what's really happening to your analog/voltage signal ...and why we have so many dBFS analog references and confusion about "what's the right dBFS number to use?"...when there really isn't any one number....and why my -10dBFS might be your -12 dBFS.
That's the main reason I think analog metering is more valuable. If you get that stuff right....it really all falls into place on the digital side.


And yeah....this thread has wandered all over the place....but I think it does touch back on the OP, "How to get more volume".
 
Vigo said:
So im at this state where i want the mix to sound good but be loud enough too, my problem is that i feel like its squashed when i raise it too much I added a compressor so i could work it out a little bit but i dont want to ruin the signal so what other options could I have? im a total noob at mastering im guessing it'll be mostly dynamic working?
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The best way you can understand this is that you want a high product of A) signal-to-background noise * B) signal-to-noise * C) signal-to-ear, set at each stage: 1) production, 2) recording, 3) mixing, 4) mastering, so that you end up with the right mix of frequencies. Study these aspects, what gear to use and how to work with them at each stage. When you have covered the product of these (A,B,C,1,2,3,4) you will no longer need louder mixes. And remember my converter signal capacity tip too. ;)
 
To understand what happens to a digital signal when you pull the output fader down (do you see where I'm going?) You need to understand tracking vs output. If you track too hot and get digital distortion (hash) pulling down the faders will not reduce the digital numbers by any factor at all, giving you back a clean signal. The question is, I think, perceived loudness and I'm pretty sure I already answered that question on this thread.
Rod Norman
Engineer

MrWrenchey said:
I'm no mastering expert, but I'm fairly certain that's not true, and would depend on where your faders are during mix down as opposed to tracking level.

If you track too hot, just pull all the faders down and your digital headroom is back. Simple as that.
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Rod Norman said:
... If you track too hot and get digital distortion (hash) pulling down the faders will not reduce the digital numbers by any factor at all, giving you back a clean signal. ..
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I presume(?) if I were to rewrite that it would be more along the lines of 'pulling down the faders will not get you back to the cleaner signal.. lost, or, that could have been. ;)
 
Is this thread still searching for more volume...? :D



MusicWater said:
And remember my converter signal capacity tip....
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So now you want to invent your own term for "headroom". :facepalm:
There is no such thing as "converter signal capacity". A converter doesn't store anything.

Oh...what "tip" did you give exactly besides reading off some specs and noting that there can be different specs for different converters. Like...what was your actual "tip"...?
Oh yeah...."the ones that get it (those at the top) kind of keep silent about it because they know this is where the money/quality is".

The thing that people need to know and do is to calibrate their metering and follow proper gain-staging, and they will have proper levels and good volume without unwanted distortion or clipping.

That's nothing new, not a secret and not something that "(those at the top) kinda keep silent about".
If anything, most pros talk about it all the time and comment that newbs need to focus on that, especially in the ITB world where "dBFS" becomes confusing because it's a moving target, because there is no single standard, and because 32-bit floating point math allows excessiveness and improper gain staging without warning.
It's easier on the analog side, where you are somewhat forced to follow proper gain staging, and where level standards exist.

Anyway, more volume or better quality doesn't just come automatically from converters with a few dBu more headroom....and I've never seen anyone talk about more converter headrom, like it was some secret weapon.
Like I said, US is generally +24dBu = 0 dBFS, and most European countries use +18 dBu = 0dBFS....so do American productions automatically sound louder/better because of that...? :)
 
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mixsit said:
I presume(?) if I were to rewrite that it would be more along the lines of 'pulling down the faders will not get you back to the cleaner signal.. lost, or, that could have been. ;)
Click to expand...
Who's to say what the heck he thought he was talking about? Turning down faders will not "undistort" something that was clipped in conversion or distorted in the analog realm before it got to the digital realm.

Most modern DAWs today actually can mix at levels way beyond 0dbfs without adding distortion of their own (not counting any plugins which might have internal limits). If you've got a bunch of clean tracks mixing together and hitting the mix buss beyond 0dbfs, you can usually just turn down the master fader to avoid any distortion in rendering or at the DAC.

Course, now that we've optimized our s/n ratio and preserved all of our transients in a pristine state by staying away from the analog rails of our system (which necessarily keeps us safe in digital), we find out that the average level is just too low for real world listening, and WAY too low to "compete" with modern big-budget mixes. So we distort the mix. ;)
 
Hey! What happened? We're actually back to the original topic? Amazing. How do you get more volume. Record at proper levels. Mix at proper levels. Get, or learn ho to do, professional mastering. Never been any secret. MM is forever telling us that a mastering professional can actually get more volume from well balanced lower signal than from pushed to the max at all points tracks. Anybody listening?
 
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