When mixing my projects, the overall quality sounds great. But when i bounce it out, it seems like its being over compressed and i am losing the dynamics to the mix. Everything gets really quiet and fuzzy. I have tried different bit ratios and i can't really tell a difference in them. I loose the punch of the kick drum and the snare drum and the vocals seem to duck under the mix. What am i doing wrong and how do i boost the levels of the overall mix without clipping and keep the dynamics of the track?
Hey there Kentucky, ya might want to check that chicken?
So you're telling us what's wrong. But you're not giving us any examples and you have not indicated what software or computer platform you are using? It's a it hurts when I do this question.
Obvious questions would be, what is your workflow? Do you have clip lights coming on once you've rendered your mix? What is your recording format and what is your mix down format? What types of dither might you have selected? And then again, you might be hearing what a lot of us here when it comes to digital summing? This is another reason why so many people have started working in a hybrid fashion. Some folks feel that analog summing on a hybrid system, sounds significantly different from digital summing, within your multi-track software. But really, you shouldn't be noticing the kind of differences you seem to be describing? You can still obtain a good punchy mix with digital summing. Plenty of guys do. Which would then also say if you're the only one having this problem, it must be operator error.
There are issues with transcoding from one recorded format to another. For instance, they've recorded at 96 kHz sample rate, it does not mathematically work out properly to 44.1 kHz. And this affects the sound and people's perceptions of it. So sometimes, folks of run up against these mathematically incongruent transcode's to different formats such as 16-bit, 44.1 kHz. Which is generally the accepted delivery format. While there are those are offering higher definition downloads at say, 24-bit, 96 kHz. But you won't find that on a CD, ever. So anytime something has to be transcoded, that's a translation. And whenever there is a translation, something is inevitably lost. So someone wanted something that was higher in resolution from me, I would make sure it was mathematically divisible such as running at 24-bit, 88.2 kHz. And certainly not 96 unless it was for television. In which case you do use 48 kHz/96 kHz. So this comes down to preproduction before you start recording to know what the end delivery format is intended for as the primary source.
So really if you could supply some more information, we would be better equipped to help you.
Mx. Remy Ann David
Well i haven't been using dither. I'm recording in 32-bit and bouncing to 24-bit at 44.1kHz. That was the factory default on logic. the computer is an imac 16gig ram, with os 10.5. My clip is typically 0db. and i believe i've been normalizing the bounce.
Maybe i'm hearing things but i just feel like the mix thins out. I could be over analyzing myself.
If you're going to convert from 32-bit to 24-bit, you're going to need to add some dither. I'm not completely up to speed with Macintosh or its associated hardware and software? And perhaps you're hearing that truncation without dither? Which would, yes, make it rather grainy, granular sounding. And then a wretched crunch on anything that trails off, fades out.
And you still neglected to tell us who's computer audio interface you are using? So you still haven't provided proper information. It's not just the computer. It's not just the software. It has something to do with the computer audio interface.
Tommy can ya hear me?
Mx. Remy Ann David
You're not recording in 32 bit. 32 bit refers to "floating point processing". You're either recording in 24 bit word length or 16 bits. And when you say you're "bouncing it out", what do you mean by that? Bouncing it out to what? or, where? And external device? Are you burning your final mix to CD, then listening to the CD and you don't like what you hear? Or are you compressing to an mp3 file and don't like what you hear?
I think that recording at 32 bit float is actually screwing him up?
I think part of the problem here is that when you are a successful and wealthy engineer, of course you want to record at 24-bit at some high sample rate. And then your analog to digital process also has to be professional. And your computer needs to be a specifically designed computer audio workstation and not a general-purpose consumer home computer. And if you expect your computer to perform flawlessly, it can't be designed to do anything else than that. So when you're also using it to play computer games and surf the Internet, you might find your computer won't work well for digital audio multi-track productions? Because it really isn't set up for that purpose in mind. This is another reason why most of us have and use more than one computer. I have three purpose built desktop machines that are configured for strictly audio, video and/or audio and video production. Then there is the laptops. One was strictly designed for ProTools and the other is my general-purpose HP laptop. Again it's configured specifically for audio and video work and not for Internet surfing even though I can. This is where I will start the virus software running and not allow the computer to automatically start it. And other stuff like that that requires a re-tweak of the operating system, memory allocation, variable instead of fixed RAM cache, etc.. And if you don't do all of the above, you're always going to have problems.
So why not just try recording at 16-bit, 44.1 kHz? Most average computers are more adept at handling that than do some of the higher resolution formats. So the best way to rule out these problems, is to make another recording and set the software to 16-bit, 44.1 kHz. Then see if you can get a clean recording? Stop screwing around with a higher definition blah blah since it's still going to be delivered at 16-bit, 44.1 kHz. That is a perfectly adequate digital recording format. 96 DB of total dynamic range. More than any analog recorder ever delivered. And if ya can't make a good recording at that 16-bit, 44.1 kHz, format, you have no idea what you're doing.
The real only advantage working at 24/32 bit depths is simply in the digital dynamic range. And not the analog dynamic range, of which, even the finest equipment, rarely can produce more than 110 DB of dynamic range. So 24 and 32-bit provides for a lot of false impressions about resolution. Resolution really comes from the sampling rate. So a recording at 16-bit, 88.2 kHz will actually sound better than a recording of 24-bit, at 44.1 kHz. And that digital dynamic range, it gives people a false sense of their equipments working dynamic range. It's still only 110 DB with the best stuff.
Now people will tell you you need this 24/32-bit higher resolution for effects processing. What a bunch of malarkey. It's only for folks that don't know how to tweak their levels properly before adding crazy gobbledygook processing. I mean if you don't know when to turn up or turn down your volume controls, what kind of an audio engineer could you be? Answer: a clueless one.
Top-of-the-line analog to digital conversion is also something of a luxury. And those aren't cheap. You can expect to pay upwards of $1500-$3500 for really fine analog to digital input converters. Whereas many of the inexpensive KTM IC chip converters do quite a fine job. Crystal, BURR BROWN and others make similar fine analog to digital converter chips which you will find in most of the average proconsumer equipment. So I rather like the sound of my Alesis HD 24 XR fed from the Neve & API stuff I use. And so for around $1500, you get a great 24 channel front end that also offers redundant backup capabilities, simultaneously.
While the IDE drives that the HD 24 uses are all but gone, one does not need to constantly replace or replenish the hard drives in a HD 24. You simply use HD 24 TOOLS, to transfer the FST formatted HD 24 hard drives, through your computer to an external USB, fat 32 or NTFS formatted, external USB 2.0 drives. And it's that in which you hand to your client not the FST IDE drive in the HD 24. So a few IDE hard drives, should last you years of use with the HD 24. So I always get a good chuckle when people start talking about not being able to find IDE drives for their HD 24. Go figure? That's just called stupid. I mean your client wouldn't even be able to rip those tracks from the FST formatted IDE drives, anyhow. Not unless they had the right software and/or proper hardware for a simple file transfer purpose. And you don't want to hand your clients something like that. You want to give them something they can begin working with immediately which will be that external hard drive.
I'm not monkeying around here. This is for real.
Mx. Remy Ann David
Coyote... was I really that confusing?
Maybe I strayed too far?
I like to stray on top of the world.
Mx. Remy Ann David
I'm workin with what i got here remy. lol. Its a 2008 model mac. with a broken led. huge crack runnin all the way around and across diagonally through the screen. the interface is the m-audio profire 2626. there are no externals. i'm bouncing to mp3. the cd's usually sound alright. I'll try the 16 bit. lol. am i missing anything?
There's nothing wrong with a 2008 model Mac, nothing at all wrong. Broken LED? Which LED? And why should that matter? I mean I like to see the little disk access light flashing myself but I could just smoke a cigarette instead? Yeah, a big crack through your screen ain't good. But you can still plug in another screen? Don't have any old CRTs laying around? I know it looks sort of retro...
The M-Audio stuff is good. Nothing wrong with that 2626. In fact I had my eye on one of those. But here's the problem that you describe. " I'm bouncing to MP3. The CDs usually sound alright." First, why the heck are you bouncing to MP3? You first have to balance to fully uncompressed, same file format in which he recorded in such as 16 bit, 44.1 kHz.AIF, 24-bit, 32-bit float, 48/88.2/96/192 kHz uncompressed! From those master mix down files should be the only place where you convert those files too small, highly compressed, compressed 10:1 at 128 kb per second, 16-bit, 44.1 kHz. Which is what we call a lossy format. And this causes all sorts of peculiar audible artifacting. From things that sound like they are bubbling under the water to other things that are chorused and flanged that weren't before. And nothing comes out sounding like what the original was because it has had 80% of its guts ripped out of it. And you get the scraps of what's left over known as MP3's. And that's the only place where you're making the mistake. You need to do all of your productions fully uncompressed in the native file format of Macintosh which is .AIF/.AIFF. Now those files would then be cut to a CD. You can even cut CDs from MP3's but why if you have the master track/tracks? Like you said it only sounded OK because you're still playing back hacked up MP3's that is then converted back to uncompressed after its butchering has already been done to it.
This is sort of like being penny wise and dollar foolish. And mostly comes from inexperience. Hard disk space is pretty cheap these days. So the only reason to release MP3's is for convenient and quick downloading and streaming purposes. It's not intended to sound like an uncompressed master. You're hearing the trade-offs. You're hearing what a lot of people bitch about MP3's, about. And how disruptive it is to the sound, which it is. Does the average Schmo really care? The answer to that seems obvious. Absolutely not. Most people think seeing their video movie fill the entire 46 inch, $1000 LCD display to be high definition video. It's not. It's standard definition video being displayed on a high-definition screen and most people don't know the freaking difference. Only the sports puppies notice the difference when they're looking for that elusive hockey puck. And that's what drove high-definition television, sports.
So basically, that's what you're missing. I think once you've just not bounced down to MP3's everything will recover by leaps and bounds in what you're hearing. You bounced down to uncompressed.
Now basically, a CD was only designed to cut the CD from .wav file format. And that's true. Though many audio CD cutting programs will instantly and on the fly convert those Macintosh .AIF uncompressed files, in real time to .wav as it burns the CD. No other conversion need be done. And so the uncompressed .AIF file is an uncompressed .wav file already. This still gives you the ability to create for your own enjoyment, a compilation CD, filled with over 10 hours of MP3 files/songs, that can be played back in many car/automotive CD players. Imagine driving from DC to Jacksonville Florida and never hearing the same song twice or ever having to have changed your CD. So that's for convenience listening. That's for shoving a couple of songs onto a $20 memory chip with earbuds, player. But it's certainly not for master mix down purposes or channel combining and dumping. That's a big no-no.
You won't get any pudding if you don't eat your meat!
Mx. Remy Ann David
I dunno... call me silly... but it sounds like there's two different issues, one of which leads into the other, creating a simple death of dynamics spiral.
Lemme take a bit of liberty to make a coupla guesses...
Your mix room is less than 1500 cu. ft,, little to no acoustic treatment, windows or at least hard bare walls all round, and you're monitors are right up against the wall with no SBIR treatment behind them?
Because you're working in a small room, with relatively no treatment and reflective surfaces all around, you have a tendancy to turn out mixes with the exact opposite EQ of your room's response. Because the bass is too hot, and the highs are burning your ears because of the excessive reflections, you're cutting bass and overcompressing the mids and highs.
I would get a copy of Room EQ Wizard and get a response curve done first, and foremost. Then plan on at least putting in some bass trapping in the wall::wall corners and the ceiling::wall corners. Build a coupla' 2" thick x 24" x 48" absorption panels and get those set behind your monitors... and maybe think about some curtains for the windows/hard surfaces in the room.
Try this free loudness maximizer VST plugin for Mac.
It's common to loose perceived loudness and punch when you compress to mp3 format and when you dither to 16 bit from your original 24 bit mix.
The plugin I posted is ok, it's not my favorite...but my favorites cost money. FYI, my fave is Voxengo's Elephant and you can demo their Mac version for free if you're interested: http://www.voxengo.com/product/elephant/
Also, do a little research on your own and see what other loudness/maximizer plugs you can download free. Are are more out there than you can shake a stick at.