Anyone have info about 192khz

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waigy

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Hey man

Recently bought a motu traveler which seems like the business so far, though I'm having problems with clicks when recording 4 tracks simultaneously at 24/176.4 or 24/192khz.
I will eventually suss what is causing the clicks, but in the meantime I'm going to have to record at 24/88.2 or 24/96khz, as the unit works fine at these settings.
I would obviously like to use as high quality as I can since I have the capability to do it, (and maybe in about 5 years time we will be playing mp3's on our mobiles at 24/192 and laughing at 44.1).
I read somewhere that if I use 176.4 rather than 192, it will be better when mixing down to 44.1khz as it's exactly 4 times the final frequency ie: 44.1x4=176.4
Anyone have good info or a web link to a site with good info about the advantages of recording at 192khz and mixing down to 44.1 (technical info, dithering e.t.c.).

Cheers man
 
Here's one article, and I think our beloved Ethan Winer adresses it somewhere on his website.

Bottom line: don't sample any higher than your final product. For pert near everyone that's a 44.1kHz audio CD. And BTW, there's no such thing as a 24/192 MP3.

:) :) :)

Have fun!
 
$ .02

man have you opened a can of worms.... as far as arguments on audio forums go (i monitor several) this one gets a lotta press... my personal take is that as far as bit depth i like having it.... even if eventually i may need to throw it away.... as to whether its "easier" for the computer to bit reduce ... sure deviding by 4 is easier that multipy by a devide by b (dont remember the #'s but its a strange enough ratio) but if its done off line its obviously moot.... and its moot as well in real time for the most part since processor speeds are SO fast that between typing ** the thing read the paper jacked off twice and had a smoke.....
 
the dividing thing is absolute bullshit.

the same thing happens in SRC whether you use a multiple or so and so or not..
 
On the subject of clicks at 192...

Most audio interfaces use a memory buffer of so many samples that the device driver uses to pass the audio data from card to memory and back. The higher the sample rate, the sooner the buffer needs refreshing. The buffers job is to spread things out over time (the computer has other things to do) so obviously, the faster the buffer needs refreshing, the more chance there is of something else getting in the way of the process.
So, try increasing the drivers buffer value (may be called latency, or DMA) in the MOTU control utility if you must work at 192Khz.
 
> Anyone have good info or a web link to a site with good info about the advantages of recording at 192khz and mixing down to 44.1 <

You already got good advice. Recording at a sample rate or bit depth higher than necessary does nothing but waste hard drive space and limit the number of tracks you can play without glitches. If you want to prove to yourself that 16 bits at 44.1 is good enough, look at it this way:

Have you ever heard a CD that sounds simply stunning? One that is far better sounding than anything you've been able to create? I'll take that as a Yes. :D

This proves beyond all doubt that standard "CD quality" is plenty adequate for excellent sound, and it also proves that you need to focus your efforts at improvement elsewhere.

--Ethan
 
Thanks very much guys this is exactly what I wanted.
In a reply to Jim Y's post.
If I have the motu buffer set at 2048, I get loads of clicks.
As I reduce the buffer size the clicks become fewer.
When I go to 512 and smaller the computer freezes.
The best I've managed is a 5 minute recording with 2 clicks (buffer at 768).
Obviously 1 click is too many clicks.
I've had good help by email from motu, but I've tried all of the suggestions with no improvement.
I will have to try an external fast hard drive, or a quality pcmcia firewire card (I'm using my laptop's hard drive and firewire port).

Cheers
 
decrease your sample rate and buffer size and I think you'll find that it will be easier to record without getting audible clicks.
 
It is a fact that very large buffers also cause clicks! I suppose it takes longer to refresh the large buffer which causes a similar timing conflict that filling a small buffer very often does!
Having said that, you really ought to be able to get click free operation with 96Khz recording and the very smallest buffer size - even possibly at 192Khz too. Something is very possibly going wrong in your machine. I use M-audio pci cards and can work with 64 samples @ 96Khz no problems if I want - though I normally have the buffer at 256 and work in 24bit/44.1Khz. The very lowest buffer does reduce the amount of free CPU power available for your recording software and effects plug-ins because the driver software's busier refreshing the buffer. I find 256 a happy enough normal buffer setting.

Whatever can be done to improve things depends on your machine and whatever else you have plugged in to it. If it is a laptop, your options are might be fairly limited - your hard drive might be too slow for one thing.
Check out the music optimizations listed at www.musicxp.net
-assuming you run WinXP of course!

Oh, and I would agree with everybody else that recording at 24bits is a more useful improvement than using the higher sampling rates. The only thing I'd say re sample-rate is that if your audio is destined for DVD soundtracks, you'd help yourself out by working at 48Khz rather than 44.1.
 
Jim Y said:
It is a fact that very large buffers also cause clicks! I suppose it takes longer to refresh the large buffer which causes a similar timing conflict that filling a small buffer very often does!

No, badly written software and/or broken hardware causes clicks. Buffers are not refreshed in a single chunk. They are pushed a piece at a time which can be no larger than the buffer size, but is limited by the USB spec to a couple hundred bytes at a time, if memory serves. (Note that I'm assuming USB because that's what 99% of people complaining about pops and clicks are using....)

Honestly, there's no excuse for pops and clicks unless your computer is heavily loaded, regardless of the buffer size you choose. If you're seeing a lot of pops and clicks, dollars to doughnuts your interface's firmware is buggy and it is lying about some of the packet sizes or similar.
 
Thanks again for the info so far.

After reading the Dan Lavry sampling theory pdf suggested by bigray I had to go and lie down.
Maybe a bit more technical than I expected.
I understood the first and last paragraphs though.
Dan seems to be saying the same as Ethan Winer, that it is pointless recording at higher than Dr. Nyquist suggested, and so I have payed a lot of money for a useless function(ie: the ability to sample at very high sample rates).
Am I right in saying that?

My original idea was to :
1. Record myself(acoustic guitar and voice) at the highest possible setting then,
2. Back up those wave files to dvd, then
3. Extrapolate or alias or antialias or Nyquistivate ar whatever it's called down to 16/44.1, then
4. Do my mixing, effects, mastering and cd burning e.t.c. using the 16/44.1 wave files.

I thought it might be a good idea to have the highest quality wave files available for the future when maybe everyone's hifi or mobile phone will be capable of playing 24/192 wave files(or mp9's as they will be known).

Another question is this :
If I record at 24/192 on my traveler then play that recording back using the traveler's outputs through an amplifier, am I hearing a truer representation of the sound of my guitar than doing the same at 24/44.1?
Or is what Dan Lavry is saying that, you get no improvement of sound at all above 44.1?
Dan and Ethan also mention using lots more cpu and hard drive resources at the higher sample rates, well I bought a 3ghz/1 gig ram laptop for that reason, so I don't have a problem using more resources or buying a faster/larger drive.

I have learned a lot so far from the info suggested here, but I haven't grasped it 100% so I apologise in advance if anyone is offended by this response.

Cheers again
 
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There will never be any point, ever, in recording higher than 44.1 kHz. Recording in 24-bit is handy for the extra headroom.
 
Well, I would disagree about no difference above 44.1. There's a very audible difference between 44.1 kHz and 48 kHz because of limitations as to how steep you can make an analog (or simple digital) low pass filter without altering the sound in undesirable ways.

If you mix down to 44.1 kHz, in theory, these differences should go away in the process, though in practice analog filters are almost never as good as a software filter during mixdown would be, so there will be a small improvement in high frequency accuracy.

Once you get above about a 30 kHz cut-off (about a 60 kHz sampling rate), there shouldn't be any audible changes within the human hearing range (22 kHz down or thereabouts). Thus, there's no real possibility of hearing any improvement beyond that point; bugs in sample rate conversion notwithstanding, there should be no perceptible difference between 88.1 kHz and 96 kHz, and certainly no difference above that.
 
Tried rolling back the firewire drivers and about 200 other different changes with no improvement.
I reckon it's either the hard drive or the onboard firewire port.
I might try an external drive and/or a pcmcia firewire card at some point, but going by the answers I've had here I just need to record at 24/44.1 or maybe 24/88.2.
The unit works fine at these resolutions.

Cheers again guys
 
That's a good call re XP SP2 and the firewire driver, but there is supposed to be a hotfix...
http://support.microsoft.com/kb/885222
... it's a bit complex to carry out as you can see, a registry edit is required in addition to running the fix.
Probably is easier to roll back to SP1 for many users.

Can't let dgatwoods comment go by.

DMA is responsible for moving the audio samples around, not the soundcard driver. The driver is only involved with buffer refreshes which really is done in a block equal to the buffer size x the number of channels. The fact of CPU usage being proportional to buffer size is universal to all drivers well written or not. It's because the data transport is done in the background by DMA that interface drivers have no idea if audio samples have gone missing, they are not sequencialy time-stamped like video frames are. You'll note that video recording software can tell you if dropped frames have occured. I know of no audio interface software that can do something similar to detect lost samples.

An audio interface, whether on USB, Firewire or PCI card has no buffers of it's own, only sufficient registers to hold and recieve one sample for every channel in or out. If it gets lost because the DMA bus master doesn't get to work in time, the samples are lost for good, hence the clicks.

Some find a utility called "pci latency tool" can help sort out DMA service problems, usually by reducing the setting for the graphics chip, most of which grab up to 255 bus cycles at a time while soundcards, USB and firewire controllers usually only need and get 32. Reducing the graphics to 128 or 64 usually improves things markedly. Anything else that uses more than 32 cycles could probably have it's setting reduced too.
Here's the tool...
http://downloads.guru3d.com/download.php?det=951
It can't be used on PCI-e based graphics, which a recent laptop may have on-board.
 
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