I bought a volume control unit a while back because I did some reading on bit loss since I didn't have my soundcard's level set to 0.0 db, rather I had it -5 db or whatever to control volume. I had no other way to control my monitor's sound output. With the cound control unit, I can keep my soundcard at 0.0 db, avoiding bit loss.

However my question is, does the whole keeping at 0.0 db to avoid bit loss apply to hardware AND the software? In other words, what if I have my master output in my sequencer at say -4.5 db while my soundcard's level is still 0.0 db, am I ok? Should I try and keep my sequencer's output at 0.0 db always?

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i don't think there is any way to attenuate volume, in the digital domain, without bit loss. therefore, in order to have no loss from volume changes, you would have to have every fader and plugin not affecting gain (obviously impossible).

i wouldn't worry about it dude, unless you're talking about -40db or something

It's virtually impossible to get any degradation using floating point. The "bit loss" term generally applies to 16- or 24-bit fixed-point digital transmission protocols like S/PDIF or ADAT.

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I'm not sure I understand the problem exactly. Can't you turn the sequencer's main volume down so you're not overloading your monitors, then when you're ready to export the track, just put the volume back up so it peaks at -0.1 dB or so? (to answer your question - in the end you want your track to be around -0.1, -0.0. Whether you do that or you send it to a mastering studio, in the end it should be up there. It's not going to make much difference at the end of the day, if you have a completely clean mix which doesn't clip and peaks at -0.1 or one which peaks at -3.0. A mastering studio will be able to use either. Compared to many other production techniques, this one is not a major concern. Worry about a lot of other things before you worry about this.)

I do most of my mixing at extremely low levels, just by having an extra processor on the master channel to drop the level (and also to give me the option to listen/ mix in mono). When it comes time to export, I just remove that processor.

You got it the wrong way round, the soundcard volume is in the analogue domain, so you won't get bit loss there. You get bit loss from the level controls in your DAW software. But even at 16 bit, its not a big deal, don't obsess.

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quote:

Originally posted by kitphillips You got it the wrong way round, the soundcard volume is in the analogue domain, so you won't get bit loss there. You get bit loss from the level controls in your DAW software. But even at 16 bit, its not a big deal, don't obsess.

Some soundcards have digital master volume. And no, you can't get bitloss in a DAW with a 32-bit floating point engine.

when you can't actually hear a difference, nobody gives a crap.

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Thanks for all the info everyone its very insightful. One question though since you mention the 32-bit floating point engine. Does this mean I should be writing my tracks in 32bit float? Since you mentioned this, I looked and I see now in Cubase that is an option under project setup. I've always had this set to 24bit. Does it matter than my soundcard, the Audiophile 2496, is only capable of 24bit, 96khz?

I guess I'm a bit uncertain then what I should be writing my tracks in. I currently write them in 24bit, 48khz. I realize everything gets broken down eventually to 16bit, 44khz so what would be ideal for my situation? 32bit float, 44khz, 32bit float, 48khz, etc?

I appreciate all the help.

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quote:

Originally posted by Pjotr G when you can't actually hear a difference, nobody gives a crap.

Actually the whole reason I came on here and asked about this is because I can hear a difference and I'm trying to determine what caused this difference because many factors were involved in me getting the 2 conflicting outcomes, confusing the hell out of me. I will see what answers come for my response above this post and then explain the details of my problem (if anyone is interested of course). Again, I'm thankful for all your help.

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I used to always run at 32bit/96khz, which I always thought sounded better but, then I never liked the way it sounded when I dithered down to 16bit/44khz. So nowadays I usually run at 32bit/44khz and then render down,and it actually sounds better to my ears but, maybe I was doing something wrong when I ran at the higher rates. The only time I record at the higher rates anymore is if I am recording a vocal, so I can capture all nuances.

before i used the digital fader volume on my echo audiofire, then i bought a passive volume control by sm pro audio and set master out to +4 and the fader -4 so the output is 0dB (dont know if the sm pro audio is balanced with jacks), i recieved alot of dynamics at low volums. also i stopped rendering at 48kHz 24bit and use instead normal 44.1 16bit as i had problems when converting to mp3.

quote:

Originally posted by palm before i used the digital fader volume on my echo audiofire, then i bought a passive volume control by sm pro audio and set master out to +4 and the fader -4 so the output is 0dB (dont know if the sm pro audio is balanced with jacks), i recieved alot of dynamics at low volums. also i stopped rendering at 48kHz 24bit and use instead normal 44.1 16bit as i had problems when converting to mp3.

Interesting you say this because that is what I noticed too and part of the reason for my questions. When I render out the project at 24bit 48khz, I noticed the resulting MP3 when I converted it sounded way shittier than if I rendered out of Cubase at 16bit, 44khz and turned it into an MP3. Sorry if this sounds dumb but why is this?

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