48kHz vs 44.1kHz

[qo]

Hi mhschmieder. I currently use Sonar on a cheap PC for capturing the 2-mix output. But, yes, something like Peak, or DSP Quattro, etc, on a Mac would also work, or the MasterLink.

I've only started doing this recently, so still need to do a hefty project with this method and compare it to earlier projects. But, I go outboard anyway on mixdown for various reasons, so doing SRC at the same time seems to make sense.

[grimepoch]

Wow, good description of SRC, interesting what they say is happening with DP. I'd like to try this test personally, just for the hell of it, what would be a good spectrum analyzer program/plugin to use to look at that information?

However, all that said, I think I'll be working in 44.1.

I am assuming on those peak tests, bit depth was held constant.

[giles117]

What Q) has shown is the Best way IMO.

When I still had a pair of Avalon VT-737's I processed all my mixes thiw way. Now I am looking at a Pendulum Audio MDP-1 as a front end and mix chain piece. Along with a Lavry Blue for d/a - a/d conversion.

[EMRR]

My personal thought is that the higher rates translate more importantly to greater 'speed', and this is likely more significant to depth/soundstage/etc. than the increased frequency ceiling. There are analog noise gates that claim faster performance than 96K is capable of resolving! I like the thought of working 88.2K (or 176.4K) throughout and then doing simple division down to 44.1K at the last possible moment after all processing is done. There are a number of mastering processors that upsample to higher rates for processing and then downsample the results on the way out.

In the past I stuck with 44.1K for everything, and mixed to synced 44.1K for 'poor man's automation' functionality.

Another consideration: The client takes it to another studio - use the most dummy-proof rate; this will be a personal choice based on what you've done and what you know about where it's going. Hopefully you will know this upfront.

[mhschmieder]

I guess the most dummy-proof rate then is 44.1k, since that's what you get with Quicktime and the internal drivers that Apple supplies when you don't have your MOTU interface hooked up.

Maybe I shouldn't fret so much over staying digital when using TC Electronic outboard gear. I've only put this to the test twice so far, as I stayed analog during the first two years of using DP, and haven't done much tracking since starting the priocess of going all-digital last summer.

Since we are talking S/PDIF and not AES/EBU (which anyway is capable of higher sample rates), and as I have heard that S/PDIF tends to be a bit noisy (this is according to Monster when I talked to them about prioritisng which units to assign to digital filtering on their power distribution systems), perhaps it is better after all to go round trip digital-analog-digital when using certain mid-level (vs. high-end) outboard digital gear during mixing?

Just to reiterate, some of my manuals say 48 kHz only for S/PDIF (either for incoming or outgoing, I forget which -- they aren't always bookmatched in their specs), whilst some support both 44.1 kHz and 48 kHz. When I ran this experiment for a first time a few months back, it threw a spanner into my attempt at resolving once and for all which sample rate to work at (at least as my usual default), and I've been too busy (until this past week) completing an album at a Windows/Sonar home studio to redress the issue in the intervening time.

[grimepoch]

Well I can say in my case I am really equipment locked. While my 896 goes to 96k, my 896HD goes to 192K, the ADAT in one the 896s are both 48k/24 bit max. Since I use almost all my inputs at the moment, my only choice comes down to 44.1 and 48 (24 bit is a no brainer now).

For the moment, I think I am going to stick to 44.1, it seems the best answer since I will be writing out of DP for most cases unless I am shipping out to someone else to master.

I will say this too, manufacter's frustrate the hell out of me with their digital implementations! I've got some pieces with AES/EBU, some with SPDIF Optical, some with SPDIF Coax and even others with ADAT. Most of them not even supporting multiple ones. Trying to keep my signal digital once I've converted goes through a few pieces of gear is amazing. Not to mention trying to get it all clocked. I'm almost ready to buy a word clock distributer because I have 5 pieces of gear I need synced now (and almost 2 more if I decide to push digital all the way to the stage before my monitors).

Crazy.

[mhschmieder]

I have just confirmed that, documentation to the contrary, I am able to use my TC Electronic outboard gear at 44.1 kHz and 16-bit depth via my MOTU 828 mk II into Digital Performer 4.61, as long as I set the 828 to be the clock master (I used a legacy project as the guinea pig, but never use 16-bit depth anymore).

Based on everything said here by the experts, and the resolution of this red herring regarding S/PDIF, I am now going to settle on 44.1 kHz and 24-bit depth as my standard for all non-acoustic projects (and possibly but not definitely 44.1 kHz vs. 48 kHz, and 24-bit depth for all local/independent soundtrack work, since they rarely have the budget for sophisticated time syncing using standard 48 kHz movie production tools).

If I get to the point where I'm doing pro-level acoustic recording or major production soundtracks, I'll probably have the money for the proper tools for a full start-to-finish 96 kHz recording path anyway .

[tommymandel]

I guess this comes after a consensus has been reached (to not SRC in DP), but here's my .02:

back in the mid to late 90's I used to mix down to a Sony DTC-1000 DAT recorder that only recorded at 48k. The thing sounded really good. Then a neighbor hot-rodded it by wiring a single jumper between 2 points, and the Skip-ID switch became a 44.1/48k recording toggle.

After that, I tried recording at 44.1k, and comparing. 44.1 never sounded as smooth to me, and I'd been in Pro Rock bands, and some loud ones, for almost 20 years at that point so you'd think that I wouldn't be influenced by slight high-frequency differences.

48 sounded a lot smoother, had a silkier high-end.

The only 'maybe' I'll add, is that maybe since the Nyquist filters in the Sony DAT machine were set for 48k, that was why 44.1k sounded a tad weird: it was a non-integer multiple of the filter frequency.

[EMRR]

I will say this too, manufacter's frustrate the hell out of me with their digital implementations! I've got some pieces with AES/EBU, some with SPDIF Optical, some with SPDIF Coax and even others with ADAT. Most of them not even supporting multiple ones. Trying to keep my signal digital once I've converted goes through a few pieces of gear is amazing. Not to mention trying to get it all clocked. I'm almost ready to buy a word clock distributer because I have 5 pieces of gear I need synced now (and almost 2 more if I decide to push digital all the way to the stage before my monitors).

Crazy.

For example, try making sense of when and where Alesis ever decided to put any digital connections, along with whether or not there is an Adat sync option or a word clock in/out. You will start seeing the little bean counters running around behind their displays, and it's not the acid you took years ago. Can someone tell me why the Masterlink wouldn't have an Adat optical connection? oh yeah, this is a MOTU group......one of countless manufacturers doing their own thing (man, power to the people!)

[Splinter]

...a neighbor hot-rodded it by wiring a single jumper between 2 points, and the Skip-ID switch became a 44.1/48k recording toggle.

That pretty much says it all. I wouldn't rest your conclusions on a rigged machine from 10 years ago.

[grimepoch]

Or, try working on your laptop without an audio interface installed. So far, even the people from Audio Ease than I wrote said 48k is not worth the SRC, it's better to run at 44.1k, so that is what I shall do.

[mhschmieder]

A few years ago when I was using an 896 mk I, a sales rep told me that using it at anything other than 96k was just going to slow down the processing and cause problems because it was going to have to SRC on the fly.

I remain confused on this point, in terms of whether A/D converters tend to run at ONE sample rate and convert to the selected sample rate on the fly. If this were true, then the ONLY reason for selecting lower sample rates would be to conserve disc space.

My own experiments with the 828 mk II lead me to believe that what the salesperson told me about the 896 was a bunch of nonsense. I'm going to stick with 44.1 kHz as my input A/D sample rate, and stay at that rate within DP. Except for critical projects.

[Splinter]

...a sales rep told me...

Yeah, well, how many 'sales reps' know what they're talking about? That simply can't be true.

[grimepoch]

Yeah, I wouldn't believe it either, as I have an 896, and I can say, 44.1 doesn't seem to run slower

Now, what he could be referring to is he might have heard about some of the Soundblaster Audigy cards where the DAC only runs at 48k, and every other sample rate before it us output is converted to 48k in the hardware.

(It might not be the Audigy, but it is one of the Creative cards)

Heh, my favorite salesman lie, I bought a truck, and I asked him what the 'compression' setting on the radio meant. He told me it pushed the laser closer to the CD to get better sounding audio Hahahahha.