Mastering Software

No I said +30 and I meant it. There are any number of reasons why you might end up rendering a file that wants to peak above 0dbfs even if while listening you're not clipping your converters. I did post pretty absurd numbers to make a point. More commonly what might happen is your mix peaks just barely above 0. If you render to floating point it's not the end of the world, your ME has the full transient to work with and can decide whether to turn it down or squash it down or clip it off as necessary. If you render to fixed point, the damage is done and the ME is forced to try to mask or repair.

I'm not actually advocating against proper gain staging. I AM saying that - especially with floating point - the actual numbers aren't nearly as important as people seem to make it. More importantly, I'm saying that if you've already got a good sounding mix that's peaking right around 0dbfs, the idea that you should turn it down just to "leave room" for your ME is asinine, and in fact harmful when you render to floating point. We can argue how much it matters (and at 24bit it probably doesn't matter much until you get down toward -30 or so), but if the actual point is to leave your ME all of the available dynamic range to work with, then attenuating the signal closer to the noise floor is working at cross purposes.
 
32 or 64 floating point is only relevant within DAW right? 24-bit recording exported and sent to mastering house cannot be over 0dBFS. That would be past the point of digital clipping. I would hate for those who do not understand this to get the wrong idea.

I feel there is a miscommunication here...

Please be clear. :)

Oh and have a great new year! :D
 
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Reaper has the capacity to render to floating point, and to open floating point files. If your ME can't handle them, probably find a new ME. :)

Edit - but again, if you're rendering to fixed-bit, you have even less good reason to "leave headroom" by mixing to peak significantly lower than 0dbfs, because you are compromising DR/SNR.
 
One of the benefits of digital audio is dynamic range and signal to noise ratio. With a good digital system, the most significant "noise" in the system should be background or room noise that gets captured, not self noise from the system. If you were to apply 100 layers of TPDF dither to a 16 bit audio file the resulting self noise of the dither would be much less than tape machine hiss, and tape hiss never stopped anyone from making a great record.

Gain staging for tape is more finicky than digital, and yes, s/n ratio is a concern there. Digital is a lot more forgiving that way, and unlike tape there is no sweet spot. There is no advantage to running out of headroom for the sake of using all the bits. Clipping is a much bigger problem than s/n ratio. It might be different if you track at -40 LUFS or something, but I don't think anyone is saying that.

Also, as target loudness seems to be becoming sane again in some places, reaching -9 LUFS for a CD that nobody will buy is no longer necessary. If broadcast services will attenuate your files to somewhere between -14 and -22 LUFS, you have even less reason to think you need to be anywhere near clipping.
 
I'm absolutely sure that I said something that can be boiled down to "In fixed point it doesn't matter much as long as you're not clipping, and in floating point it doesn't matter at all." In fact, the whole point I've been trying to make is that as long as there's not too much noise and there's not any unintentional distortion, it ultimately doesn't matter at all. If your file meets those criteria, and your Mastering Engineer says any god damn thing about your absolute levels, they're a dick and you should move on.

I'm not actually saying you should push your peaks all the way to 0. I'm saying you shouldn't NOT do so just for the sake of the next step in the process. Floating point files allows us to not have to worry about it at all.

Now we're going down the rabbit hole.

I've got my monitors calibrated such that when the knob is at 12 oclock, nothing in the analog chain is even close to distorting and an RMS level of around -15dbFS creates about 85dbSPL at my ears. It's not as loud as I want it to be, but it's actually just about right for a standard level. I turn it down sometimes. I crank it up sometimes. But I know where I'm at and it's about right when I turn the knob "home".

If I have a mix with a 15db crest factor, and the peaks are hitting -10dbFS, it's going to be pretty damn quiet at my listening position. So, like, am I supposed to turn up my speakers 10db just so that my ME can have some room to work? That's absurd. What happens when something actually does hit 0dbFS?!? These things happens sometimes. You hit the wrong button or unmute something unheard, or turn the wrong knob on some plugin. BANG!!!!!

Then remember that in my case (this thread is about "Mastering Software") "Mastering Engineer" is literally spelled "ME" and then what? I recalibrate my monitors when I move to the mastering stage? Nope.

So then I'm mixing, and at certain point in the process I drop some master bus processing on it. I know I'm going to add some glue and some squish and some saturation at the master stage, and I want to get some idea of how that's going to work, and then adjust this mix to best "survive" that treatment. But what actually happens is going to depend on the rest of the material that's going onto the album, so when I go to render, I'll bypass that whole chain. And since I've already got it up close to the actual honest to goodness FS levels I'm shooting for at the end, well some of the peaks are probably going to go over 0. So I render to 32 bit floating point. Then the adjustments at mastering are dbs rather than 10s of dbs.
 
I´m new in recordings world, but for me things going well using this steps:

- > noise reduction > 60hz hum removal > Eq'ing > Multi band Compression > Stereo Enhancer > some cuts > limiter > play loud as hell
 
I´m new in recordings world, but for me things going well using this steps:

- > noise reduction > 60hz hum removal > Eq'ing > Multi band Compression > Stereo Enhancer > some cuts > limiter > play loud as hell

If it works for you and your project, then enjoy. :)

That seems to me like a bunch of crap masking a poor tone to begin with. I would never use a chain like that myself. Sorry but that is just not the ideal...
 
I´m new in recordings world, but for me things going well using this steps:

- > noise reduction > 60hz hum removal > Eq'ing > Multi band Compression > Stereo Enhancer > some cuts > limiter > play loud as hell

As is always said here: if you get a good recording, you wont' need a lot of work to make it sound good. First figure out why you've got to reduce noise and 60 cycle hum - you shouldn't be recording these at all.
 
I'd be interested in how you master your song and what equipment or software you use. I'm not real good with a lot of this stuff so if you'd let me know how easy or detailed it is to use that would be great. I'm guessing this question has been asked many times so if there is a file or folder that will provide some answers I'll use I.

Thanks,
Don.....

Hi Don,

Sequoia by Magix is a great DAW for mastering. It let's just enter ISRC codes (so artists can get payed if their songs gets played) and can create DDP images!
 
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