Outboard Operandus

[Munchkin]

Hi there,

my first post here and maybe this is a dumb question, maybe I should be posting this on an Apple/Logic forum, then again maybe it's relevant to those of us building outboard gear projects for use with computer based recording set ups.

I'm recording in Logic Pro 8 on a iMac 2.4 Ghz Core 2 Duo via an Apogee Ensemble using outboard gear, a mixture of off the shelf stuff and homebrew projects, however when I come to EQ and Compress prerecorded tracks or master stuff using the outboard I'm faced with a bit of a snag. I've got my tracks sounding good through the gear, then I wanna rerecord the processed tracks back into Logic but what buffer size to use?. 128 takes away the top end, 512 takes away the bottom end, 256 is a compromise. Clearly I wanna go out at 1024 and come back in at 64 for maximum quality.

Am I missing something here, I've Googled this problem about a hundred times and can find absolutely no reference to this so I feel like the only person on the planet who sees this as a problem. I've tried rerecording at 256 and adding a bit of top and bottom eq, but it's too much trial and error for my liking. My solution at the moment is to rerecord at 128, 256, and 512 and mix them together, but it's still a compromise and takes up loads of track count. I have read some people saying that there is no difference in sound quality between the buffer sizes, this is clearly not the case to these tired old ears.

What do you pro guys or anyone else using Logic do? As I said, maybe this isn't a suitable topic of conversation for this forum, but it doesn't matter what grade of caps or flavour of iron I use if I'm losing it all in the bounce.

 

[jackies]

I don't use Logic on Mac, however, I'm pretty positive buffer size shouldn't affect the sound at all.
I do believe in difference in sound between opamps, capacitors, DAWs, even wires to an extent, but buffer size?

 

[Munchkin]

Well, it's my ears that are hearing it, I'm not measuring it on an scope or spectrum analyser, but I kinda trust my ears, although I will say it's a subtle difference, but enough to bother me.

 

[hobiesound]

i've got the feeling that you're brain is fooling yourself. You know what buffer size you're listening to and since you're expecting a change, you're hearing a change.......there was a nice AES lecture about this posted in the brewery a few months back.

 

[Rochey]

You need to compensate for the delay on playback.

 

[tv]

I have no clue about your comp. and converters configuration, but this seems like some sort of a malfunction - if it is REAL in the first place.

 

[Munchkin]

Ha Ha, well I guess that's possible, I've done tests where I've recorded an acoustic guitar and vocal then sent it out and back in at 32 and then 1024 and compared the two to hear the difference and felt pretty sure there was a quite noticeable difference. I'll do another test as it's easy enough to do.

By the way I'm not talking about latency issues here, just sound quality.

 

[tv]

>>> just sound quality
Thats what I meant as well. It is totally illogical that sound quality would be buffer-dependant. So it's either a placebo effect (but you say that the diff. could be clearly heard on re-recorded tracks), or a device failure. Perhaps drivers? As I said, I have no clue about your particular gear, so this is from the top of my head.

 

[Munchkin]

>>> just sound quality

Yeah sorry that was my reply to the previous poster Rochey. Well I've had another listening test and it is very subtle, more of a vibe thing, so possibly placebo, definitely not a converter problem.  I can still hear a difference though.

 

[shot]

This is interesting! In no way there should be difference in sound when you change the latency!

Maybe if you could try to set the latency to the biggest possible value. Then record something clicky, something with fast and sharp transient. After recording it, load it in a wave editor and zoom in to see if there's doubling of the transient. If you notice the double transient, that means that you heave "double listening". Maybe your audio interface is set to record not only the input but also the sound that is being played internally. If that's the case, that would explain the change in sound - it's a comb filtering.

 

[Munchkin]

Yeah thanks Shot, I'll try that, may take a little time. I'm beginning to think that it is just my ears fooling me, too many late nights and long sessions. Thanks all for posting I'll get back when I've done the test.

 

[abbey road d enfer]

Ha Ha, well I guess that's possible, I've done tests where I've recorded an acoustic guitar and vocal then sent it out and back in at 32 and then 1024 and compared the two to hear the difference and felt pretty sure there was a quite noticeable difference. I'll do another test as it's easy enough to do.

By the way I'm not talking about latency issues here, just sound quality.

Is there any processing on these tracks? Dynamics and time-stretching/pitch processing may sound noticeably different according to buffer size.

 

[Andy Peters]

Hi there,

my first post here and maybe this is a dumb question, maybe I should be posting this on an Apple/Logic forum, then again maybe it's relevant to those of us building outboard gear projects for use with computer based recording set ups.

I'm recording in Logic Pro 8 on a iMac 2.4 Ghz Core 2 Duo via an Apogee Ensemble using outboard gear, a mixture of off the shelf stuff and homebrew projects, however when I come to EQ and Compress prerecorded tracks or master stuff using the outboard I'm faced with a bit of a snag. I've got my tracks sounding good through the gear, then I wanna rerecord the processed tracks back into Logic but what buffer size to use?. 128 takes away the top end, 512 takes away the bottom end, 256 is a compromise. Clearly I wanna go out at 1024 and come back in at 64 for maximum quality.

Am I missing something here, I've Googled this problem about a hundred times and can find absolutely no reference to this so I feel like the only person on the planet who sees this as a problem. I've tried rerecording at 256 and adding a bit of top and bottom eq, but it's too much trial and error for my liking. My solution at the moment is to rerecord at 128, 256, and 512 and mix them together, but it's still a compromise and takes up loads of track count. I have read some people saying that there is no difference in sound quality between the buffer sizes, this is clearly not the case to these tired old ears.

What do you pro guys or anyone else using Logic do? As I said, maybe this isn't a suitable topic of conversation for this forum, but it doesn't matter what grade of caps or flavour of iron I use if I'm losing it all in the bounce.

I'm a Logic Studio 8 user, and I will tell you that buffer sizes have absolutely no such subtle effect on audio quality. Either you'd get perfect movement of data between the host and the audio device (and the converse), or you'd get audible glitching as a result of buffer underruns/overruns.

As for your listening tests, are they truly blind?

-a

 

[Munchkin]

No, no processing. Well, I've done more tests and the good news it there doesn't seem to be any difference after all - humble pie for me. I guess I must have thought there was some difference one time and then my brain re-enforced it every subsequent time (unless there actually is a difference and my brain is fooling me now - mmmm). The bad news is apart from making a bit of a fool of myself (won't be the first time and I'll get over it) I now don't trust my ears - hopefully I'll get over that too. Sorry for the red herring, many thanks for your posts and putting my mind to rest on this.

Watch for my next post on whether gold plated XLR's are sonically superior (which of course they are - he he).

Cheers.

 

[jdbakker]

i've got the feeling that you're brain is fooling yourself. You know what buffer size you're listening to and since you're expecting a change, you're hearing a change.......there was a nice AES lecture about this posted in the brewery a few months back.

This one: http://www.groupdiy.com/index.php?topic=37527.0 ?

JDB.

 

[hobiesound]

ahhh thanks so much JD "always comes over the bridge  " B

 

[jdbakker]

Munchkin: do have a look at the video linked in that thread. While you can probably trust your ears better than advice given on a random forum, our brain can be pretty good at deceiving us. Knowing when that's likely to happen and how to deal with it is quite handy.

ahhh thanks so much JD "always comes over the bridge  " B

Nah, knowing where stuff is is just a byproduct of sitting so often on the forum.

JD 'undutchable' B.

 

[Munchkin]

JD, yeah I just spent the last hour watching it, fascinating stuff, also the following "is he for real" discussion.

I think part of the problem is simply that my ears ain't what they used to be, as a younger man I could kinda multitask whilst listening to a mix, hearing and defining all components in the mix equally and at the same time hearing the mix as a whole. These days my ears/brain don't do that so well, so I naturally seem to focus in too much on certain details in a mix at the expense of other components.

 

[xmvlk]

definitely not a converter problem.  I can still hear a difference though.

Subtle difference may be caused by Sample-Rate Conversion.
There are lot of such blocks hidden in audio DSP libraries.
(Some of it have too obscure source code to perform good work...)
Maybe some SRC is initialised with buffer size - you run SRC algorithm with different parameter.
Then you get different artefacts. If You mix-up that different signals (signals recorded with different buffers), signal-to-artefact ratio gets higher.

 

[riggler]

Hey maybe there was a comb - filter effect happening if you were somehow hearing the original sound and the processed sound at the same time, the buffer size setting the delay in between them...

Do a bounce of just the processed sound only at different buffer sizes and I bet you hear no difference!