Hitting the digits

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Golden

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as for slapping an analog master to digital... be it a computer or external device...

I'd rather not have to use a plugin to raise the overall level of the song, what are my options?

Hit the digits of say a masterlink as "hot as possible" in the digital domain?

would this achieve a nice overall level?

Or is the plugin/compressor/limiter inevitable?
 
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what I meant to say (and didn't) was this... I know how to raise the overall level with a compressor and a limiter in the computer...with plugins...

what I want to do is capture my 2-track stereo mix from 1/4" as accurate as possible, without having to raise the overall volume with anything...

say, blast the mix into a masterlink... adjust the level on the masterlink into the red and blam! a finished master file... with nice overall level...

possible? or am I spaced?
 
I'm asking because I want to print the 16bit master straight from the 1/4" to digital... I don't want to use any processing once the file is made... cause processing a 16bit file is not cool.
 
Can you go in at 24-bit/96 kHz? I wouldn't worry too much about some subtle processing at a high bit depth and sample rate with a decent quality plug-in. Even at 16/44.1 you probably couldn't tell the difference. You don't want to clip the inputs or the converters by going in too hot, I would say its better to go in at a reasonable level and use software to max it out. A little bit of light digital limiting isn't gonna kill you, just don't overdo it.
 
You don't have a problem, so don't create one.

Send your analog master in at a reasonable level, with enough headroom to let the meters bounce, and don't add a thing. Get it clean to digital, keep the dynamics and listen a bit.

Then master from there. You can go with a light 'over-easy' compressor to round things off and keep a good level.
If you are into volume madness, slam the dog out of it with a brick limiter.

But the key thing is: keep the love you put on tape, then make your decisions. I think you will like it better.

Just my 2 cents,
C.
 
slapping an analog master to digital...

It depends on what type of music you’re recording… the dynamics.

Generally speaking, nothing really changes whether mastering to analog or digital. You just need a good outboard compressor/limiter. If going to the trouble of recording with analog first you don’t want to hack things up with a lot of plug-ins. One can’t tamper too much with the sound in the digital realm and expect to preserve the original analog goodness… if that’s what you’re after.

The FMR RNC is one of the best Hardware compressors around for the money. But there’s a long list of hardware comps I would use before resorting to a plug-in.

Using an outboard compressor will allow you to reach a little closer to 0 dBFS, but you defiantly don’t want to join the loudness war… no one is winning that at the moment. It’s a dead heat with everyone sounding equally dreadful.

How much you can push the level depends on the system you’re using. Best practices for the Masterlink may not be best for something else. Different manufacturers vary as to how close the meter actually represents the true level, so you can experiment. For example -14 dBFS might be -14dBFS or there might be a safety margin added (that you don’t see) to make the system more idiot proof.

About word length and sampling rate… my general rule: If it’s going to end up on standard CD you should go from analog to digital at 16/44 and stay with it, unless you plan to do significant editing in the digital realm, which I don’t recommend… so 16/44 is the best way to enter the digital realm for the analog purist.

Our music ultimately has to end up on a digital format to be heard, but the process by which one gets it there makes all the difference.

:)
 
I went to John Golden to get a record mastered over the summer. I don't know what kind of music you record, but if it has a wide dynamic range then then his methodology for going from tape to digital will probably work for you. Just record your stereo mix from tape to the masterlink at 16/44 (for the reason beck explained) at a reasonable level- make sure that there are no overs whatsoever, digital clipping sounds awful and even getting right up next to zero can be dangerous (I'm sure beck can explain this better than I can). And just out of personal preference, don't compress unless you really have to- remember that if you ever do get radio airplay there will be several compressor/limiters between your track and the broadcast signal, so it'll come out exactly just as loud as everyone else's anyway.
 
you have noticed that digital levels are hotter than analog levels. One reason for this is that most "mastered" CD's attempt to utilize all of the available headroom and many times to operate entirely within the range denoted as "headroom". My mixdown deck probably clips at +22 db so it has more headroom than my digital converters...yet the average level is still low for normal material, and music doesn't come close to clipping the converters.

So you have two options. The first is to record at normal line level at 24/96 (or 24/44), and then normalize, dither and covert to 16/44. This should ensure that your volume level isn't above optimal operating conditions for your converter, especially if you don't have the best converters. Just because the converters aren't clipping doesn't mean they're at their most linear response. With 24 bits, you could burn some db's and still be in the clear.

The second option is to insert a piece of gear with some good clean gain (or colored if you prefer that) between the 2-track and the ADC, and capture at 16/44 already dithered. This is what I do...eq and compress between the 2-track and the converters, monitoring through the converters. That is the key, for me. When mixing, I monitor the repro head of the 2-track, and when mastering, I montor the digital signal. This allows me to eq and listen to exactly what the final product is. This to me, is the downfall of anyone trying to master for mp3's, because I do not know of a way to listen "through" an mp3 coverter, as you are tweaking. So my chain goes 2-track -> eq -> compressor (light with good gain) -> ADC @ 16/44.

Both options are commonly used AFAIK.
 
I've been bringing the signal in 16bit, and then raising the volume with a plugin limiter... nothing getting squashed...

as I A/B'd the 2-track to the CD on my stereo system, there was little difference. The tape sounded a tad better.

Improved converters should match that up better...
 
I wouldn't bother to use a limiter plugin if you're not doing any limiting. You could just normalize it. I would use more than 16 bits if you are going to do any volume changes and use more than 44.1 khz if you are going to do any processing.
 
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