What kind of outboard compressor do you have? What kind of plugin do you have? What kind of converters do you have?

There are crappy outboard compressors and crappy plugins. There are good outbord compressors and good plugins. If you run your audio through an outboard compressor you're running it through D/A & A/D - not a good idea if you're using a soundblaster card - but an OK way to work if you have good converters.

Does that clear things up?

a little

yeah a little bit, but like I heard that compressing vocals, and kick and snare and over heads befor eyou put them in work best. What about like samson scom4?

There is some headroom to be gained if you're recording in 16 bit. Record at 24 bit and compress in software. I have no experience with the Samson compressor, but I have to believe that at $150 for 4 channels, it's no gem.

The other thing about compressing in software is you can always change it or try another compressor. Once you've compressed in hardware there's no going back.

Speaking as a "learner" in this field, I've found it beneficial to learn what a compressor does first by using plugins. There are many decent plugin comps out there, and blockfish is a good free one. Once I had played around with plugin comps, and understood their function and what sounds could be achieved using a compressor, I bit the bullet and bought an outboard compressor. Specifically, an ART Pro VLA, which seems to be pretty well regarded for low-end stuff.
The main thing with outboard compression is to not over-do it, or you'll be stuck with it. I tend to use the Pro VLA on drums, where I am trying to impart a certain sound with fairly heavy compression, and vocal overdubs (with much lighter compression) mainly to catch any peaks when the singer decides to blast one right into the mic. Andy

Some of us are so old, we still love to make it "sound right" while going in. I'm not really afraid of what I can't undo, if I don't overdo, which I always do, because I love to do what I do. It's only a problem if it doesn't sound good. So you have to ask yourself.....do I feel lucky today?

One of the other reasons why I like to track vocals with some compression is because most vocalists are lousy and it helps to make them sound a little less lousy/smoother in their delivery. It's also something that the vocalists will generally appreciate in their headphones. Not that you can't deliver that to their headphones without tracking the compression. You can. But why bother? In all probability, in the mix, you might want to crunch some more? Especially since equalization before-and-after compression, sounds different.

24-bit is for amateurs.
Ms. Remy Ann David

not to hijack a thread, and i know it's an offhand joke. but why is 24 bit for amateurs? i don't get it. :?

It's not. I just generally don't have too many problems working within a 96 DB window. Do you?

Daring and reckless
Ms. Remy Ann David

RemyRAD wrote: It's not. I just generally don't have too many problems working within a 96 DB window.

You know what you're doing. The OP clearly does not. Just trying to give him a little more breathing room when tracking very dynamic sources. I have nothing against 16 bit recordings, but a few extra bits never hurt anyone.

You can never be too rich, too thin, or have too much headroom.

I'm fat according to my Doctor, poor according to my gear salesman, but I'm recording in 24 bits.

As far as the original question goes, I'm with Karl that if you know what you are doing it really doesn't matter. If you are learning, a plugin is a much better teaching tool.

Take a few measures of kick, a few measures of snare, a single vocal phrase, recorded uncompressed. Loop them individually. Pu on a plugin compressor and play with the parameters while looping the sounds. Threshold, ratio, attack, release. learn what sounds best, learn what "pumping" sounds like.

Sure, 24 bits offers up to 140 DB of internal "digital dynamic range", which does provide a greater cushion and more processing headroom's. But what most "green enthusiast's" don't understand and are unrealistic about is, that there is virtually no audio equipment that has a 140 DB dynamic range. Obviously, people think that they should see a noise floor that is down at -140 DB and they don't understand why their preamps only deliver a noise specification, generally no lower than -50 to -85DB, when they view their expanded metering, in software. That's because, the equipment that we are using is fundamentally identical to what we have been using for the past 40 to 60 years and their specifications don't equal the 24-bit resolution specification.

I think people would become better engineers when learning how to become engineers when working in a more confined environment such as 16 bits and its 96 DB of available internal processing. Remember how excited we got when we got the 10 DB noise reduction from Dolby A (along with the perceptual definition loss). And then the almost incredible noise reduction from the second leading contender DBX with their 30 DB's of noise reduction (along with the perceptual frequency response aberrations). We didn't use any of that stuff at our studios. We ran barefoot at 15 and 30 IPS which yielded a workable dynamic range of only about 65 DB and we all still made great recordings.

I mean, how is anybody supposed to learn how to drive a car if you have six lanes of open road without any other traffic to negotiate? You'de never have to learn to use a lane change indicator nor stay in your own lane. That's not very smart now is it?

Very smart and talented
Ms. Remy Ann David

the more bit quantization one has, the more possible values per sample. is that not the most important aspect when sampling a continuous time analog signal with tons of values?

it may not have the dynamic range but it definently has the possibility for the frequency bandwidth

Some people realize that the higher sample rate recordings have an extended high-frequency bandwidth. The higher the sample rate, the higher the frequency response.

While most humans cannot physically distinguish sounds above 15 to 20kHz, we have come to believe there is a perceptual difference if the high-frequency content is less restricted. Therein lies the biggest argument for higher sample rates such as 88.2, 96 and 192kHz. When you get up to that sample rate the high-frequency anti-alias rolloff filters do not have to be as aggressive or brick wall like. That contributes more to the otherwise undesirable quality of Digital when you have to use brick wall filtering at 20kHz, to prevent aliasing, when utilizing 44.1kHz sampling and at 22kHz when utilizing 48kHz sampling. Double the frequency response and change your brick wall filtering to something more gentle like 12 DB per octave and your master will sound better.

But that's still pulse code modulation and it sucks. Direct Stream Digital or "DSD", also incorporated in the SACD (super audio CD) is an order of magnitude different! That is a single bit system that is sampled at 2.53MHz! With that system, no high-frequency limitations are required and the frequency response is flat to 100kHz! Coupled with that, it is easier to convert to any other extended resolution format and downwards. Simply because we are dealing with less complicated "digital words" that actually demand more from the equipment, because of the speed.

I'm also not sure why you are getting so bent out of shape here? You're getting a fine education for free, from professionals. Nobody owes you anything. You must take all of this in and amass a greater understanding that much of the information you seek has no bearing on recording or mixing prowess.

I want you to be happy here at Recording.org and glean what you can from the professionals, even if you believe that some of us are smart asses. We are because many of us are hard-core, hard-nosed, lifetime professionals, with egos that go with our award nominations and success level.

Hey! Sir Paul McCartney is a good guy but Heather Mills doesn't think so. Does that affect high-frequency response?

Have I made it more understandable for you?
Ms. Remy Ann David

i already understood the things you were telling me but it still doesn't help to explain your comment of why 24 is for wussies. but don't worry about it, i appreciate your advice either way.

the DSD sample rates are 2.8224 and 5.6448 (64 and 128fs

i already understood the things you were telling me but it still doesn't help to explain your comment of why 24 is for wussies. but don't worry about it, i appreciate your advice either way.

If you cant properly mix at 16 bit, then 24 bit wont make you any better!!