Hey ya'll!

So my interest in mixing and mastering is really peaking and I'm wondering what is the benefit of a AD Converter? I watched the video supplied in the forum here, but I don't really have a strong understanding of the concept of this device. Does it create more of an analogue sound? Is that it's main purpose?

 

https://en.wikipedia.org/wiki/Analog-to-digital_converter

 

Well they are a crucial part of our/anyone's signal chain when recording anything onto any digital storage medium.
At the most basic an analogue to digital converter does exactly that, Converts the incoming analogue signal into binary digital data which can then be edited and manipulated in all manner of ways that is not possible in the analogue domain, The digital to analogue converter does the exact opposite. Instead of rattle on I can highly suggest downloading these papers written by Dan Lavry of Lavry Engineering who offers some of the finest A/D and D/A converters money can buy, They're free and should give you a good solid grasp of what's what http://www.lavryengineering.com/lavry-white-papers/
It's actually getting/or got to a point where finding shoddy DA/AD converters is pretty rare in even budget audio interfaces, Not mind blowing great but definitely so much better than it was at a much better price than 10 years ago
What most people want is as neutral as possible (or "transparent" is one of those terms that is used often), So what you hear putting in is what you hear coming out, However there have been (and probably still are) some units which have their own sound or character, Universal Audio's UA292 Master or the one which came before that one IIRC is one that springs to mind instantly as a really great converter which had it's own unique sound. I got the chance to use one several times and against the old Digidesign 192 I/O converters it sounded great, alive and vibrant but that isn't really what the majority of user's want (But hell I'd of chosen to use the UA292 over those crummy Digidesign 192 I/O in a PT HD rig if they we're the two options I had available, Although Avid now let people use their own choice for interfacing with Pro Tools), They want as said reference quality with no areas of weakness, Such as stereo imaging, Jitter (awesome that it really isn't a problem these days) and no hype anywhere. This is why up to a certain price point you'll see tons of I/Os with expansion options in a dedicated all-in-one audio interface but after that it gets quite different feature wise but the conversion quality and if included mic pre-amps usually are improved but mainly you'll find units up in this area offer more line level inputs which enables to choose your own preferred signal path before going ITB, Along with Lavry, Prism, Antelope, Apogee (MAC only), Burl, Cranesong, Benchmark, Universal Audio as mentioned, Metric Halo (MAC only), Lynx, RME, Audient, Focusrite and there are loads I can't think of off the top of my head but you get the idea I hope, Have a search around the "Gearslutz" forums as you'll find everything under the sun already covered/long threads about the pros and cons plus shootouts galore
Here is a decent SOS mag article also which might help and of course searching their site will yield plenty of great information http://www.soundonsound.com/sos/jan09/articles/daconverters.htm
TapeOp magazine is also definitely worth checking out even though the physical copy of the magazine is no longer used, They've got tons of the old articles up along with good forums.
So all in all the A/D section takes your voltage information converting it into ones and zeros which are stored on a disk drive then the D/A section pickups the ones and zeros converting it back into voltage information as smoothly as possible. Depending upon how you've set the bit-depth and sampling rate will affect how much headroom, dynamic range and frequency with which the information that is available is captured (and played back). 24-bit, 44.1kHz or 48kHz works just fine for Me multi-tracking, Sometimes it can be beneficial to utilize 24-bit, 88.2kHz or 96kHz but quite honestly I've never noticed/heard any benefits of going higher than that sample rate wise *but I'm no audiophile and nor do I claim to possess a pair of "golden ears"*

Hope that is of some use to you, I'm sure they'll be more members that have decent wisdom and links to draw upon

Dean

 

Last edited: May 1, 2016

An important function of an AD converter is doing Pulse Code Modulation or PCM.
Think of that as a time controlled knife, that cuts the analogue waveform into slices.
The rate at which it does, is the samplerate, being 44100 times per second (or 48000, 96000)
These slices are measured with a certain resolution, called the bitrate, around 65000 (16bit) or 17 million (24bit) values.

This process is depicted by a clock. The quality of AD conversion and the quality of the clock signal go hand in hand.
That's the layman's explanation, now read the Wiki's..

https://en.wikipedia.org/wiki/Pulse-code_modulation

 

The converters found inside even modest audio interfaces of today are better than those used to record hit records from not so long ago. Regarding the quality of the music, well that's another story

 

Last edited: May 1, 2016

A/D converters' quality play a big role only if you record external analog (basically anything you can't connect via digital USB/Firewire/SPDIF interface) audio sources: vocals, guitars, analog synths. A combo of D/A & A/D converters is used along analog summing units. Mastering is the most complex science I know in the music field, with the least amount of buttons. I'd suggest to start reading & practicing in it. Alot.

 

Great info! I'll look into this!

 

Just quickly checking in. If you have an audio interface you already have a great A/D. If you look through external ones, they are either a package with 8 mic preamps so you just connect it via ADAT to your audio interface for additional 8 channels, or it's just a converter and you need a separate 8-channel preamp. To be honest unless you need the additional 8 channels and you know you'll use it (you record drums all the time or something), just forget about it. It won't change anything in the sound you're getting.

 

If you're looking for "analog" sound, get an emulation plugin. A classic compressor (LA-2A, 1176) or preamp/channel strip (Neve, SSL stuff are great, but unless you drive it hard into distortion, it will sound just like your audio interface - 100% transparent). Or actual hardware, but that's rarely something you can afford and/or justify in a project studio. If you insist on hardware, go DIY kits or replicas, the originals are really overpriced and incredibly costly to maintain. Anyway, for just one piece to get - grab a channel strip, that way you might end up using it for rolling off the low end and doing some conversion before the signal gets to your DAW. If you're doing electronic stuff, without recording stuff with actual mics, I'd say forget about it. MAYBE get a compressor. But really, then I'd get a plugin. If anything.

 

Get an Acqua channel strip like Pink or Amethyst,it smokes all other mentioned algo emulations,acts and sounds like the real thing.
Adds weight,dimension and high end sheen which algos have a hard time replicating.
In one word you music may still sound flat.
You can still use algos like waves,uad or slate but adding Acqua dramatically alters the sound for the better.

 

AD and DA must be clocked together, and to the main interface. When stacking lots of tracks using recordings from your AD, you run into jitter if it's not a quality AD. Jitter will build up and mask or veil your audio. There is a reason why Lynx, Burl, Lavry, Mytek, and Aurora etc... manufacture these AD and DA converters. Stick to your all-in-one Focusrite for now, until you can save up for a proper ADDA.

 

In any audio interface you have both, clocked together. They are "proper" and sound great. Top engineers in the world would kill for such great sounding converters 5-10 years ago.

 

Good point. My point was that if you have multiple pieces of digital hardware, you have to utilize your wordclock ins/outs and sync them up.

I wouldn't use just 'any' audio interface for serious production and especially mixing or mastering. Try clipping the AD on a focusrite, on the way back in, it can't handle it. Lots of these cards sound like garbage taking your signal outside the box. The DA is always a weak link. But, of course, you have to spend the money. YMMV.

 

Thanks, I think you're advice is on track with where I'm going. I just downloaded a bunch of their plugins. Will be using these for the next track.

Cheers man!

 

Incorrect. It's called bit depth.
http://tweakheadz.com/16-bit-vs-24-bit-audio/

 

Very interesting and useful, Andrew. A really good read!

 

S

SO bit depth of 24bit and keep the sample rate 44. Check.

 

Tweak is right for most part, but sometimes he's off. I'm not sure what's in your production chain, but as long as Kontakt is there somewhere, you might want to check the sample rates of your libraries. Since there's majority of 48k libs in my archive, to prevent additional SRC and thus quality degradation, I set my projects to 48k as default.
48k has some extra advantages too.

I made a thread about this ages ago: https://audiosex.pro/threads/kontakt-libraries-and-sampling-rate.10461/

Also according to Tweak, you can hear the difference between 16b and 24b, which is not correct for 99.9997% of real world cases (concerning final normalized render, not stems or freeze tracks which are usually well below 0dBFS).

 

Short version without further reading. It converts analogue information into digital steps. Imagine a wave form, (an "S" shape, a continuous curving line. This is the analogue signal, the digital version builds its curve from square steps, as it can only be converted to digital binary. The higher the sample rate and bit depth the more the resulting digital wave form looks (sounds) like the original waveform it represents.

 

Not quite. That's PCM Pulse Code Modulation, which is the most used method, but most certainly not the only one.
Direct Stream Digital uses different approach of sigma-delta modulation a sequence of single-bit values at a sampling rate of 2.8224 MHz.
https://en.wikipedia.org/wiki/Direct_Stream_Digital

Also the "square steps" are not there, it's just an approximation. Interpolation, which takes place in DA converter restores the original waveform up to the sampling rate set as maximum.
If those squares were really there, then 22k sine wave on 44.1kHz sample rate would sound like square wave, not sine. And that's not the case.