July 20, 2013

The Depths Of My Bits And The Rates Of My Samples...

by Chris Randall

Once I got past this dude's beard and the ImageLine logo that they splashed on someone else's video (the original isn't embeddable), I quickly realized that this is the single best explanation of bit depth, sample rate, and dither that anyone has bothered to make.

I've always thought that 48kHz (which puts the Nyquist frequency above 100% of the human race's hearing ability) and 24 bits (which puts the noise floor below that of most any instrument, recording method, and reproduction method) were perfectly acceptable values. This video pretty much says that's fine, although not for the reasons I thought.

Anyhow, if you're a musician that records digital audio, you owe it to yourself to spend a half hour watching this.


Page 4 of 4

Jul.26.2013 @ 4:37 PM
Well, as for the beard, maybe he does a bit of RAD (reverb algorithm development) in his spare time?


This is a great video though. I remember seeing it a couple of years ago and it did indeed confirm my prejudices and confound my beliefs. I watched it a few times since and probably need to watch it a good few times more. But I got the gist of it. I think.

Shame, most noobs, when pointed in this direction won't take the time to even consider. If 16 bits is good, well 24 bits is better. And 32 bits is best of all.

And worst of it. The bits they thought were bits, weren't really the bits they thought were bits at all. You see, it's like an information super-highway. The bit-rate is your speed (max speed) and your bit-depth is your bandwidth (lanes on the highway).

Never mind. Forget I said anything. I'm sure I got something wrong somewhere. At least I know there is a difference, and in reality it makes little difference to me. And at least I know I don't know...

Jeez, this could go on all night.

Enough for a first post, I think.


Jul.28.2013 @ 2:46 PM
Was it KMFDM that wanted or did record a record all at 22k?

I remember that same damn Neve board at Trax.... I think they spent a lot of money on that... Plastickey is the right term. Did a mix or two on it, but I was just producing, not engineering, but I remember there being some debate about why it was supposed to be so great, and others trying to justify the expense.

I have tried to stick to 24bit/44.1, but a lot of scoring gigs/commericals/games/tv etc peeps want all deliverables in 48k, as it is their standard. So depending on the project will set my SR accordingly (until i forget, and then I have to use some freeware SRC which probably does a horrible job, but at the point i'm doing that ((against the clock)) i rarely give a rats behind..)...

obviously in the past src abuse yielded some of the more interesting dsp that would leave people scratching their heads... Like the ol' trick of sampling a vocal into an emax at 22k, resampling it an octave higher, or double speed, then playing it back at it's intended time gave the best warbling aliasing due to the fact that the emax filters were pretty brutal in a good way... I'm sure the same effect could be handled in software in like 7 seconds now, but that was always good fun.

now the question I have for Chris and Adam is: with 24 bit, i like the fact that the added headroom allows me to not record as hot, and maintain decent s to n etc... and I am sick of getting tracks from colloborators etc that basically look like they are pinned to the red like you would have to do with an black face adat to avoid noise floor.... so long explanation unneccessary aside: When you code your plug ins, is it true that most plugs "react" better with lower levels feeding them.... i know use your ears etc, but with something like Rough Rider or any other compressor you would use, or EQ, is there real truth to the assertion that a lower level entering the plug gives the plug more headroom to do it's thing correctly? I have found personally that working with lower levels on channels, and making it up at the end of the 2 bus seem to give me more control, ... but since you gentleman actually MAKE and THINK about these things, just truly how important is gain staging into your products?

sorry for the long winded question.

Jul.29.2013 @ 8:28 AM
On the subject of recording at lower levels, I think the answer is a qualified maybe. Proper gain staging still applies in a digital environment. So if you're trying to hit 0dBFS on every track you're probably turning the track faders down significantly at mix time, leaving things kind of backwards from where they ideally should be. The best practice would be to record everything at as close it needs to be in the final mix. Granted, that would be the practice for making an open, dynamic recording. That is not necessarily what people are trying to do now days, so you can do whatever you want to get the sound you're trying for.

As far a plugins go, every one has the potential for handling differently, but erroring on the side of additional headroom is more likely to give you the chance to get what you want than slamming everything is. and in a 24bit system there is rarely any reason you can't leave 6 or maybe even 12dB of headroom on an individual track.

Jul.29.2013 @ 11:40 AM
Chris Randall
You can always add gain. Better to leave the headroom, in my opinion. I personally record and mix with peaks at -6. I don't squish the living daylights out of my tracks, as I don't make radio or club music that has to compete, and I like them to be a bit more naturally dynamic, but that's personal taste. (Also, in the unlikely event that I get radio play, they have their own limiters.)


Jul.29.2013 @ 7:26 PM
I prefer the open dynamic sound generally, and try and avoid overcompressing stuff on the 2 bus as much as I can possibly, with the exception being certain commercial gigs etc where you unfortuntately HAVE to to even be heard, ... I try and leave -6dB at least before going to mastering, and pay attention to RMS, something the kids these days seem to like a little a ratio as possible...:)

but thanks for your response gentleman, and I hear what you are saying re:taste Chris, and I know there were days where you had to crush the living mud out of stuff to stand out, .... I was just wondering if you find your plugins "behave" better, or in a manner more expected when feeding them "less" so to speak.... have read all sorts of dissertations about -18dBfs etc at 24 bit actually being optimum etc... and deal with a lot of mopes unfortunately who work off their laptops all the time, so tend to have everything in the red all the time so they can hear it out of their headphone jacks, which is a joke, .... cause it makes more work for me to "re-gainstage" stuff so it isn't all distorted to hell over the 2 bus, which they can't hear over headphones or care...

again, long explanation for a simple question: when you and adam design your stuff, and test it to see if it meets your desired expectations, do you find that lower levels fed yield more what you would expect or intend in your design process? (obviously regardless of the ancillary benefits of improper use of said tools that we all probably love to do:) ).

Jul.30.2013 @ 8:38 AM
Chris Randall
Well, if there's soft saturation in the plug-in it is gonna be a little crunchier the harder you hit it. And we use soft sat any time two signals are summed. As it happens, I think 100% of our plugs have at least one summing stage; Adam would have to speak to that more directly.

That said, our stuff is designed and tested at a nominal (to me, anyhow) level of -6dBfs. That's where I write the presets, and where we do most of the design work.


Jul.30.2013 @ 3:30 PM
thanks for the answer Chris! That's just what I was wondering, what your optimal preset and intended design gain staging was... appreciate it.

Aug.18.2013 @ 12:16 PM
I wanted to post this in the Man of La Mancha thread, but it is closed:

link [www.synthtopia.co...]

- djp

Page 4 of 4



Sorry, commenting is closed for this blog entry.