Oh no Drew I just suggested the recognized listening test...

Jonbouy

mike_mccue


 I'd narrow down the list of brands that have great drivers for your OS. That is an over riding concern to me as I want to use the device at the lowest possible latency.

That should get the number down to a very few choices.

Use a good solid middle of the road mic, maybe a AKG 414 or a U87 and have a strong female singer belt a few tunes into the thing and see if you think there is much of a difference.

Plug in a P-bass direct and see if it has any capability to sound more beautiful in the low register on one device or the other.

If you test more than just the generic ichip stuff I think those two examples of use ought to convince you that there is a spectrum of good and better.
 

best regards,
mike

Can YOU tell a difference using any recognized listening testing method?
As I already said before mics, pres and input sources will make a more significant difference than converters (even taking into account any voltage requirements) in any modern interface.
 
Latency is a major concern if you have a major concern about latency.  5-6 ms is what I operate at (48 samples at 44,100 Hz) which scales well under moderate loads, that's on what is considered a budget interface these days, offering the portability of USB to boot.
 
If latency was more of an issue to me (I am currently able track through the app. without any concerns) I'd obviously run something off a PCIe slot rather than via USB 2.
 
Theres plenty of choice out there without any fear of landing yourself a turkey.  Sure some minor and perhaps inaudible differences could be obtained like Drew said for the price of a decent guitar, amp or mic, if you are that bothered by the minutae as you should be if you are running a commercial recording enterprise.
 
For mere mortals no so worried about the surface tension of a fart most reasonably priced interfaces these day will be more than merely adequate whatever outlets you have on your wall.
 
Would you agree?
 
 

The converters I settled on have hard ware switches that allow me to actually bypass the grainy stale sounding ichip microphone preamps that most units pass your line level input through after they pad it down with a resistor network and add the brownian noise.

I think I can hear the difference... especially during the quiet parts between the really loud parts in the low and low-mid frequencies.


As drewfx1 has pointed out... any statement I make about what I think I hear is likely to be based on an erroneous understanding of what I may think I may be hearing... so I suggest you go get your own opinion.



best regards,
mike

Well, Apogee and Apple go hand in hand. Apple even advertise Apogee's GiO on their own web site, and it's easily the most recommended brand on Logic's forums.

I've never had the chance to make a comparison, but the best recording of my wife's voice I've ever heard was captured w/ Apogee converters. The mic choice (Blueberry) was obviously in it for a big part as it worked perfectly in that particular case, but the guy who recorded it (a close friend actually) swears by his Apogee. 

I trust he knows his stuff, but this doesn't mean that what he hears would actually translate into anything relevant for me. I don't own fancy microphones or expensive tube preamps, and don't need them for the moment anyway because we have access to it and to proper recording environments just for the asking if need be.

However, if it can have an impact on what I actually do here, I'd definitely take that into consideration. The set up is likely to expend anyway, so I'm also thinking long term.

drewfx1

Jonbouy

drewfx1




A few points:

In terms of things like digital amp sims (and also some synths), anytime you create distortion (or certain synth waveforms) you are creating high frequencies which will cause aliasing in digital. But for this reason in the modern world (where CPU power is abundant) any competent programmer upsamples their amp sim/synth plugins internally to avoid this problem. So it's not clear why running at higher rates is of any benefit for modern plugins (assuming whatever "quality" settings they have are set to their highest settings). If it was beneficial to upsample even more, the programmers just should have made an even higher quality setting available. Older plugins may or may not be a different story.

Hi Drew some knowledgeable, relevant good sense as usual.  Regarding the above point though did you read the article by Bootsy I linked which covers this particular subject from his view point?

 

Just read it.

There are basically 2 pieces of interest:

1. He argues that if a plugin upsamples, it has to do SRC's and starts talking about the artifacts/ringing/etc. that this introduces. While technically true, note that not only is there is no discussion of how audible this is in the real world, but he also talks about how good (more CPU intensive) offline SRC is in another paragraph. How bad does the SRC have to be for it to be audible? Or how many times do you have to repeat a better SRC before it becomes audible? What does it sound like? Is this stuff audible only for specific types of high level, high frequency test signals, or under more general circumstances?

IMHO, people who talk about artifacts without bothering to put the audibility in context are not helping anyone. In some cases it may encourage people to believe they hear stuff that isn't really audible (if it's even present). And prevents anyone from being able to make any sort of a reasoned judgment. I only care about artifacts I can hear, so don't just tell that they're there - tell me under what contexts they might be audible and what they sound like.


I do agree that computation wise, it makes sense to avoid unnecessary SRC's and the CPU costs involved, but keep in mind that if you run at a higher sample rate all of your processing is done at the higher rate, including all the linear processing that gets no benefits at a higher rate. Obviously it will also affect your disk usage and throughput as well.

Also note that for processing that produces lots of higher harmonics, just going up to 2x (i.e. 88.2 or 96k) will be insufficient, so the plugin would still have to oversample and you won't necessarily gain much CPU cost savings (you will move the plug's SRC artifacts higher though - which is a benefit if they were audible in the first place).

Whether doing repeated SRC's produces audible problems is an interesting question. It might or might not, depending on a number of factors. So in theory it might make sense to avoid this. But without doing carefully controlled double blind listening tests involving the specific SRC's in question, it's hard to say.

IOW, there may not be a single obvious best answer for all situations. 


But personally I have always argued that a DAW should allow you to set the sample rate at which individual or bins of plugins operate, as that way it could easily eliminate any redundant SRC's.


2. In terms of "filter coefficients", filter warping, IIR filters, etc., that was essentially the point I made about EQ curves changing. Minus all the melodramatics, of course. 

In terms of filter stability and the difficulty in achieving certain curves, it's again really just a question for the programmer. If you can't get what you want in a filter at a lower sampling rate, you just upsample as necessary.



So there's really a couple of distinct overall points here:

There are places it's clearly beneficial to do processing at a higher rate. Programmers can/should/will do this as necessary. Generally they sort of know what they are doing and do it. But older plugins (that were trying to save then more limited CPU) might not use a higher rate even if ideally they should.

and

What is the impact of repetitively doing SRC's at each plugin vs. running at a higher sampling rate throughout? How good is the SRC in each plug and how many times does my audio go through them? What are all the costs and benefits of doing this in different ways? Does one approach make sense universally, or does it depend on what you're doing? Are we making these decisions based on theoretical worst case scenarios (perhaps using carefully chosen high frequency test signals to bring out artifacts), or based on more typical real world audio? 

Yes your apraisal of the article pretty much concurs with mine.
 
He uses an example of bad implementation of over-sampling in a device to illustrate an apparent over-riding benefit of a higher global sample rate.  In it he clearly shows some keeness to promote the premise of the article.
 
However like you say it does illustrate where different approaches can be used to mitigate audible artifacts caused by the interaction of the DSP in relation to the audio it's working on.
 
Like I say it's nothing that normally bothers me but I have come across situations where changing the global sample rate has improved matters, and like I say even lowering it can have the same effect if it is some kind of relationship interference between audio and plug-in causing some audible nasties.
 
It I can't hear it or it's not showing up on some meter somewhere though I don't generally worry about it.  I had a synth patch once though that had some really clear bird tweeting type noise happening when some modulation was applied to it and merely changing the sample rate cured it.

Thanks for the explanation Mike.

I've read about rail voltage and design but didn't realize that's how in depth you were thinking. I was thinking more basic than that.

Which leads me to ... most everything now days is designed to run off USB 5VDC, or have a separate wall wart as well as being powered by USB. My Fast Track Ultra is that way. I can use 2 inputs without phantom power if I just use the USB port. So I'm assuming, when I do plug it in to get access to all the inputs and phantom, they are still only running on 5VDC. And IIRC, the power pack (sorry, wall wart bugs me so I'm going to stop saying it) is only 9VDC, not a huge difference anyway.

I'm curious as to which unit you ended up going with?

so I suggest you go get your own opinion.

mike_mccue

You mean you haven't understood that I've been expressing my own opinion since the threads outset?

The converters I settled on have hard ware switches that allow me to actually bypass the grainy stale sounding ichip microphone preamps that most units pass your line level input through after they pad it down with a resistor network and add the brownian noise.

 
You haven't taken into account again that since the outset I've been saying the pres are a bigger concern than the converters.
You could at least pay attention.  It makes you seem presumptuous when you don't.

I think I can hear the difference...

 
Can you quantify that in scientific terms?
Brownian noise perhaps...

Hi Jon,
 On the contrary, I had paid attention and the fact that we were in general agreement while you insisted on finding stuff to disagree with so as to justify your voicing of disagreement only makes your action seem more mean spirited.

best regards,
mike 

mike_mccue


Hi Jon,
On the contrary, I had paid attention and the fact that we were in general agreement while you insisted on finding stuff to disagree with so as to justify your voicing of disagreement only makes your action seem more mean spirited.

best regards,
mike 
 

You hijacked the thread with an unnecessary side-show with the wall wart and then had a go at Beagle for doing Beagle stuff when you got called out as the voltage consideration would have been taken into account by a REAL digital audio engineer at the design stage.  Likely an extremely talented one too.
 
Of course we are all expected to tolerate McQ doing McQ stuff, according to McQ of course.
 
So now you've just confirmed that you are indeed all 3 things you accused Beagle of at the top of the last page.
 
If you actually heard any of my criticism of your input ever you'd realise I'm not coming from a mean-spirited place at all.