What sample rate do you record at?

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Thanks for the link to the White papers... some interesting reading. 

In regards to high sample rates, I listened to an AES lecture with Rupert Neve not too long ago,
and they discussed the human response to particularly high sample rates 100k+.... not sure how
scientific they get with it, but the discussion makes for some interesting listening nonetheless.

http://www.aes.org/sections/uk/meetings/a0703.html
 
I record at 24bit, 44.1kHz.  I tried 88.2kHz, but honestly didn't hear that much of a difference.  Plus my interfaces can only do up to 48kHz if you want to record more than 8 tracks at a time.
 
Freddy G said:
I'm thinking that 88.2 makes sense because the conversion algorythm should be very simple right? Just remove every other sample.....or is it not that simple? :-\

Not really. If you just drop every other sample, that would be plain decimation, and upper half of spectrum, from 22K to 44K would
alias (fold) back into audiable range. What is actually done is applying filter before decimation (sample dropping), to remove anything from
22K to 44K. If you use decent lineaer phase filter, audiable effects should be miniscule. So, in integer ratio conversion (like 88->44), quality
depends just on filter performance. In theory, conversion 96->48 could be slightly more "transparent" than 88->44 because of wider transition
to stop band, but this is hardly an issue.
OTOH non-integer ratio conversion is whole different can of worms. ASRC chips usually do drastic upsampling  and than downsampling, while ITB
you could use a mix of upsampling and interpolation. Just remember that you must do antialias filtering every time you downsample (and even if
you upsample, if you are planning to do any nonlinear processing after upsampling). So, again it is about filter performance.
(here is one nice paper on interpolation for those interested: http://www.student.oulu.fi/~oniemita/dsp/deip.pdf).

Bottom of the story is, in theory integer conversion should be less comlpicated, and thus pottentially better, in practice it depends on
implementation.

cheerz
urosh
 
recnsci said:
Bottom of the story is, in theory integer conversion should be less comlpicated, and thus pottentially better, in practice it depends on
implementation.

cheerz
urosh

Hi,

Hmm, confused again, the Swiss interview ( http://www.gearslutz.com/board/music-computers/93177-daniel-weiss-interview-weiss-07-11-2006-a.html ) indeed states it's more work to go from 96 to 44.1 than 88.2 to 44.1, but that there need not be any sonic difference (apart from the slightly different source-recording made at 96 i.s.o. 88.2 that is):


What is the best sample rate for recording?

I guess you are alluding to the 48 kHz vs. 96 kHz discussion from earlier. Many would say that it is easier to scale down from 88.2 kHz to 44.1 kHz than from 96 kHz to 44.1 kHz. It does take more effort to scale down from 96 kHz to 44.1 kHz; as bigger filters are needed.
*A sample rate converter is basically a low-pass filter, with the respective management of the coefficient of the filter(digital filters).*
When converting from 88.2 kHz to 44.1 kHz, one must use a low-pass filter that separates at 22.05 kHz. When converting from 96 kHz to 44.1 kHz, there are more filter coefficients, and, as a result, more memory is required. That is a disadvantage, but not a huge one in and of itself. The conversion from 96 kHz to 44.1 kHz can absolutely be as good as one that converts from 88.2 to 44.1 kHz. A bit more resources are required, but in these days, that is no problem.

There can be sound differences among various converters, depending on the sampling rate used, because, as stated before, the sample rate is converted at the input, by the implementation of filters. These filters vary from brand to brand. With the various filters come artifacts, and different sound results.

Bye,

  Peter

 
 
clintrubber said:
recnsci said:
Bottom of the story is, in theory integer conversion should be less comlpicated, and thus pottentially better, in practice it depends on
implementation.


cheerz
urosh

Hi,

Hmm, confused again, the Swiss interview ( http://www.gearslutz.com/board/music-computers/93177-daniel-weiss-interview-weiss-07-11-2006-a.html ) indeed states it's more work to go from 96 to 44.1 than 88.2 to 44.1, but that there need not be any sonic difference (apart from the slightly different source-recording made at 96 i.s.o. 88.2 that is):

That is, if you use SRC designed by Mr. Weiss, it doesn't matter if you go for integer or noninteger conversion. OTOH, I could code transparent integer SRC in few hours, but I doubt I could come up
with eqaly good arbitrary rate converter.

cheerz
urosh
 
I started my recording project yesterday and I decided to go with 88.2k.  I was actually surprised that my tired old ears could hear that much of a difference! I can only say that things sounded smoother, warmer and more real to me. Now we'll see what the conversion will do and if it's worth the extra computer crunching.
Thanks everyone for your input.
By the way, I used my brand new 8 channel NYDave one bottle mic pre for the session. I finished troubleshooting it about 1/2 hour before the session started. This thing sounds incredible :eek:
And soooo quiet...the noise floor is so low....nothin' but inky black silence. Thanks for the beautiful design Dave!
Freddy
 
I guess it really does not matter if you record at 88.2 or 96 if you mix out through a analog console or a sumthing type unit. Better would be 96 over the two.
 
Freddy,

I don't get around here much anymore, but I happened to see this post. You built 8 channels of that thing? Wow. Did you ever post about it? If you did, please provide a link. I'm really under the weather and could use a little cheering up. 

Sorry for the side-track. Carry on.
 
44.1kHz. Since I'm mostly doing 2-track stuff these days (or mono), I'm not mixing outside the box, and I hear the sample rate conversion more than I'd like. (Yes, the one in the Benchmark DAC sounds bloody awesome, but I don't have that to do my conversion.)

For multitrack...well, that's another matter. I like tape best.

Peace,
Paul
 
Dave,
You got it my man!
About the 8 channel box...It's in a 2 rack space par-metal chassis. The power supply is in a seperate rack mount chassis and they are connected with a 6 pin amphenol connector. I've got 296 volts for B+, 6.2 volts DC on the heaters, 48 volts for phantom and 5 volts for my nifty little illuminated pushbuttons for polarity, pad and P48.
For the amplifier I had 2 double sided PC boards manufactured (it was cheaper than 1 large board). Each are 6" by 8" wide and fit side by side very snuggly in the case. Each board does 4 channels. I laid out the boards with ground planes on the top and bottom and I made a dual footprint for cinemag and also lundahl input transformers. So just for variety I installed 4 cinemag 10bpc in the first 4 channels and 4 Lundahl 1636 in the other board. The cinemags sound very beefy and a bit dark...that "iron" thing. And the lundahls seem to have more high end sparkle and detail in the mids.
I used no zobels on either. Tubes are NOS GE JAN dated 1977. All resistors are PRP metal films except for R9 which is a Kiwame.I remembered that somewhere in that monster one-bottle thread you said that your original value of 10k for R12 was too low and 22k was better so I went with that.  You also said the we could then lower the value of C5 which I did. I didn't know how low though so I just took a guess and used 10uf (waddya think? too low? go lower?)I also bypassed C5 with a 0.01 uf cap.  All other component values are as per your scheme except for C4 (as you noted) I used a 4.7 uf film cap. R10 was eliminated....transformer out only. Output transformers are Edcor WSM600/15k. I will have a front panel made by Front Panel Express soon but I just needed to get this baby up and rockin for this session.That's it!
I used it on drums for it's inaugural recording...
Channels 1-4 (cinemag)were: kick (Sennheiser421), snare (sm57), rack tom(421), and floor(421)
Channels 5-8 (lundahl) were: overheads left and right (Josephson small diaphram condensors), drum room (Crowley and Tripp Naked Eye ribbon mic) and channel 8 was electric guitar amp mic'ed with a Royer R121. Electric bass went direct into my NYDave two-bottle ;D
I wrapped up bed tracks last night and I gotta say....this is probably the best recording I've made.
The drum sound is absolutely enormous! The musicians were just freaked out when I played it back on the monitors. No eq or processing of any kind except for the room mic which i crushed with my DIY La2a.
I'll post some pictures of the guts and all but I will be tracking overdubs for maybe another week or so before I can pull it out of the rack OK?
Best Regards,
Freddy
 
NewYorkDave said:
Freddy,

I don't get around here much anymore, but I happened to see this post. You built 8 channels of that thing? Wow. Did you ever post about it? If you did, please provide a link. I'm really under the weather and could use a little cheering up. 

Sorry for the side-track. Carry on.

Hey Dave, speaking of the sumthing, did you have some sort of project like that that the sumthing was based off? I read a post where someone had a link to it or what ever it was in the group photo site but the links are dead.
 
Well, I work analog too, and record at 88.2, and it is clearly audible when you switch the sample rate down on like, choirs and orchestras. The sound becomes less silky and more jaggy, I'd say, more coarse. But what I can't really make out is the difference between recorded and produced at 88.2 and then downsampled or downsampled and produced, but the good sound in general makes me wonder: if you produce almost-all analog anyway, why not just haul off and record at 44.1 and call it a day? Noone is gonna ask you for an SACD, you know?
 

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