hey Sork, the converter in WaveLab is not very good, i'll tell you that for a fact. I think its best to record and mix in 44.1, 24 or 32 bit, simply because if you only have WaveLab, and are doing your own mastering, you'll get much better quality without WaveLab's converter. 8-)

also, if your recording program allows you to export at 44.1 from a recorded 48, i would go ahead and do that, (like you record at 48 and you bounce at 44.1), otherwise stuck to recordnig at 44.1 with a high bitrate.

Ok, kind of the answer I expected! :wink:
What if then, I did my recordings at 48k, and recorded into wavelab with a digital cable (real time), with wavelab set at 44.1? Will this sound just as bad as doing the conversion with wavelab? And will setting wabelab to record at 32bit floating make any difference from recording at 24bits? :oops:

why would you record into wavelab? another cable? bounce it out of your seq. and then import it into wavelab.

Because I've heard that the mixdown engine in my DAW (Standalone Roland Vs-2480) isn't very good, so I just skip mixing down on the Roland, and just record my mix to a stereofile in Wavelab! And again, will it sound good if I have a project thats at 48k recorded realtime into wavelab at 44.1?

Sork,

Your problem seems like a fairly simple one. First though, a little bit of terminology. When changing sample rates, there is no dithering that goes on. That is when you change bit depths. However, what is done is, the program removes, based on a predetermined algorithm, approximately 4000 samples. Very few programs do this well, and for that matter, very few outboard sample rate converters do this well.

Your best bet may be one of the two following options:
1. Record in 44.1 to begin with. There will be very little (if any perceptable) difference between 48 and 44.1. Any such difference will be destroyed by the downsampling.

2. Come out of your DAW in analog, into a mixer with decent summing (such as an inexpensive Soundcraft/A&H) and feed the stereo analog signal into Wavelab. In this method, you can insert some outboard gear inbetween the two DAWs and forget messing with as many plug-ins.

If you need a couple channels of inexpensive conversion, check out ART's DI/O box - it does a decent job for $150, though it is unbalanced only. I'm guessing by your set-up that you don't want to blow too much more on conversion.

My $.02

J...

Ok then, I'll continue recording at 44.1! Thanks for the helpful answers!

Cucco wrote: Sork,

When changing sample rates, there is no dithering that goes on. That is when you change bit depths.
J...

Wrong.... when ever there's a mathmatical calculation the word leng increases & you must use dither. src is not basic math.

What i would do if i were you, is record at 24/48 burn your mixes to disc & send them off to a mastering house...or use a good converter at 44.1 going into your sound card & capture in wavelab at 32/44.1
I do this every day & it sounds great.
Ed

Ed,

I'm glad you do this all the time, but I'm afraid there technically is no dithering in Sample Rate Conversion. The act of ditheriing, by it's very definition is to add information in small particles to an object so that when it is reduced, the reduction does not appear as nothingness. This simply is not what Sample Rate Conversion is doing. Sample Rate Conversion is the state of taking 48,000 samples per second and transforming it to 44,100 samples per second. The only way to do this is to selectively remove some of the samples (nearly 10%). However, this cannot be done simply by randomly removing samples (despite the fact that this is what some Sample Rate Conversion programs do.) One has to be very selective at removing samples.

For example, you can remove more samples at lower frequencies than you can at higher frequencies. However, despite this, you simply cannot remove samples in an exponentially repeating pattern; the human ear recognizes this pattern. Therefore, you must do kind of a combination of the both (with a little bit of planning behind it.)

I'm pretty sure I'm correct on this, since, for my day job when I'm not recording orchestras, I write this kind of software for the military intelligence field - and yes, it's entirely based on algorithmic computations.

Also, while I whole heartedly agree that the best solution would be to record at 32 bit floating point/44,100 samples per second using a high quality converter, I'm afraid you missed the essence of Sork's message. No offence intended towards Sork, but he's making do with the best equipment he can for the money - note the Roland VS-2480. How is it helpful to people on this forum to suggest a $3000 solution to a $5 problem? Yes, the outboard converter would be the best option - but he's asking for a solution that he may already possess.

I will agree with you again that outsourcing the mastering will yield him the best results, but why does everyone assume that EVERY project must be sent to a mastering engineer? Maybe the guy wants to learn a little of it himself...maybe he's having a little fun...maybe he doesn't have the funds.

Please, I urge you, check your facts before you post information, and answer the question at hand, not what you interpret to be the question. You'll find it's a lot more helpful. :)

Thanks,

J...

Correct, there is no dithering in src but dithering should be applied afterwords.

I meant to say when reducing wordlegth you must use dither. when any digital process is applied it expands the word length including src.If you need to reduce the word length from that point you need to dither.

in your quote it looks like your impling that dithering only happens when changing bit deapths. For those that don't know, you have to apply dithering as a last step, it does not happen automatically during that procces. I also urge you to chose your words more clearly.
Maybe you meant truncation.

Thats why all the good src hardware boxes have dithering.

http://www.weiss.ch/sfc2/sfc2.html
Altronics Amp

Lastly i didn't miss his point. you can get a good adc for pretty cheep now a days.
Ed

All of the top hardware SRC's do have a dither option but that's because of their ability to reduce the bit depth. I don't think that they apply dither as the result of SRC'ing. I could be wrong though. I checked the flow chart on my new apogee rosetta 200 which has a/d/a, src, and bit reduction. In the flow chart, dither is applied before the SRC. Now i'm not sure why the SRC follows the bit reduction, but it does. Logically, I would like to see dither as the very last step in this chain but for some reason it doesn't. Maybe it's the design of this unit only for reasons I don't know yet.

hmm, dither before src.
i think most of those units have a higher internal sample rates. the weiss proccesses 40bit float/32 fixed, so i don't know first hand but based on that I would guess dither is offered after src.

I remember on glenns board a conversation about this, & quite a few good engineers missed this point until dave collins & goran finnberg set every body strait . at least the fact that src needs dither afterword.
Ed

sork,
if you use wavelab to src, don't use the process src, use the resample 192 plugin in the master section. even pg the designer of the program recomends that plug over the internal one.
Ed

ok, but back to the original question....

What if he comes out of the 2480, into a good summing mixer as mentioned, then out of that into a dat recorder, say a panasonic 3800, to utilize the converters in there?

Digital out of the 3800 into the other comp.

I noticed that the panny's were the ones you saw in ALL the studios about 5-10 yrs ago, so I'm guessing the converters are decent?

Ed,

I think it's important that you understand that dithering and SRC (Sample Rate Conversion) are completely independent of each other. You can put the SRC anywhere in the chain that you would like, as there is no dithering that occurs at this point.

Also, I said nothing about bit depth reduction or truncation, which is two different processes, so what you are referring to by - "in your quote it looks like your impling that dithering only happens when changing bit deapths" does not seem to make any sense. However, dithering DOES only occur with bit depth reduction.

When you simply truncate a word from 24 bits to 16 bits, you are, in essence removing not just the extended dynamic range of the recorded work, but you are also changing how the digital hardware deals with the analog voltage. In a 24 bit word, the incoming analog voltage (given in whole integers with an infinite number of decimal places following) are represented a binary number representing that voltage, up to about 4 decimal places. In a 16 bit word, you get the same principle, but a representation of up to only about 2 decimals. Therefore, the transition between sounds of different amplitude is much less smooth - almost represented as a stairstep effect if graphed.

While you can simply truncate the bit depth (easily done by simply patching a 24 bit signal (source) into a 16 bit medium (destination) with no intermediary hardware to apply dithering) it will produce detrimental sounds easily audible to the average listener. However, by applying a shaped noise, similar to what is going on within the program material (and often derived from it), you fool the human ear into thinking it hears smooth transitions. This is dithering and has nothing whatsoever to do with changing the sampling rate.

The reason that the two pieces of hardware that you mentioned include dithering is because they will perform both sample rate conversion (important!) and bit depth reduction (NOT TRUNCATION).

Many will argue which of these items needs to come first - SRC or bit depth reduction (often referred to somewhat incorrectly as word length reduction). By placing the SRC first, you remove much of the audio material required by the dither device to create the artificial noise. However, if you place the SRC second, you may actually be removing samples of the carefully created dithering noise as well as program material. Neither of these is the best solution. That is why several people prefer to mix and record in the same frequency that will ultimately be used in playback.

Of course, I can point out one other obvious point. When recording at a sample rate that is a multiple of your final destination frequency, the Sample Rate Conversion that takes place is simply a removal of evenly spaced samples. For example, when recording at 88.2Khz and down-sampling to 44.1 Khz, you simply need to remove every other sample. You will still get a very accurate picture of the original wave form, with only minor discrepencies at the highest of frequencies (As indicated in the Nyquist Theorum.)

I think if Dan (Weiss) were to chime in (I understand he lurks around here sometimes) he would agree that even his sample rate conversion follows these basic rules.

Thanks,

Jeremy :)

wiz1der wrote: ok, but back to the original question....

What if he comes out of the 2480, into a good summing mixer as mentioned, then out of that into a dat recorder, say a panasonic 3800, to utilize the converters in there?

Digital out of the 3800 into the other comp.

I noticed that the panny's were the ones you saw in ALL the studios about 5-10 yrs ago, so I'm guessing the converters are decent?

Wiz1der:

You're thinking creatively, and that's good. However, I don't think that this is the best solution.

How about this:
Come out of the 2480 in analog
Go into a good sound card via analog (RME or Lynx - the Lynx is big $$$, but worth it. The RME can be had for little $$$ and is a good contender. Hell, you can even run into an Echo Mia)

-OR-

Come out of the 2480 in analog into a decent converter
Decent converter sent back into pc in digital in 44.1/16.
This is essentially what you said, but I don't understand the need to use the converters on the DAT. They're not THAT good.

-OR-

Come out of the 2480 in analog
Go into a decent outboard effects box (Lexicon MPX550, TC digital compressor) which will send the signal to the PC in digital.

You just have to be careful not to go crazy with the effects, cuz you can't undo them.

These are just a couple of options based loosely on your suggestion. Of course, all of these are IMHO.

Thanks,

Jeremy :)

Ed Littman wrote: [quote=Cucco]Sork,

When changing sample rates, there is no dithering that goes on. That is when you change bit depths.
J...

Wrong.... when ever there's a mathmatical calculation the word leng increases & you must use dither. src is not basic math.

What i would do if i were you, is record at 24/48 burn your mixes to disc & send them off to a mastering house...or use a good converter at 44.1 going into your sound card & capture in wavelab at 32/44.1
I do this every day & it sounds great.
Ed

Wrong Ed!

I would be carefull about sending a mastering engineer 48k when I've not spoken with him about it.

Best Regards

Cucco wrote: Ed,

I think it's important that you understand that dithering and SRC (Sample Rate Conversion) are completely independent of each other. You can put the SRC anywhere in the chain that you would like, as there is no dithering that occurs at this point.

Of course, I can point out one other obvious point. When recording at a sample rate that is a multiple of your final destination frequency, the Sample Rate Conversion that takes place is simply a removal of evenly spaced samples. For example, when recording at 88.2Khz and down-sampling to 44.1 Khz, you simply need to remove every other sample. You will still get a very accurate picture of the original wave form, with only minor discrepencies at the highest of frequencies (As indicated in the Nyquist Theorum.)

Thanks ,I understand that src & dithering are different procceses, but that doesn't mean you should not dither after.

src will expand the word legth(32float etc.).
when ever you reduce word length you dither(back down to 24 or 16bits)
your qoute was not clear of that fact & so i'm trying to clarify it.

also, now a days all sample rate converters up sample to a very high frequency around 15mhz thats an even multiple of all of the other sample rates involved. Then is down sampled to the destination sample rate using even multiples reducing any funky math. So it does not matter if you src from 96 down to 44.1.

upsampling> reconstruction filter>high sample rate>anti-aliasing filter> down sampling

Ed

[quote="Ammitsboel
Wrong Ed!

I would be carefull about sending a mastering engineer 48k when I've not spoken with him about it.

Best Regards

What do you mean by that. I did'nt say he should not speek to the mastering engineer first.
Did I miss your meaning here?

Ed Littman wrote: [quote="Ammitsboel]
Wrong Ed!

I would be carefull about sending a mastering engineer 48k when I've not spoken with him about it.

Best Regards

What do you mean by that. I did'nt say he should not speek to the mastering engineer first.
Did I miss your meaning here?

It just didn't came out clear in your message, that's all.
By looking at your message he could think that it was ok to deside for himself rather than deside together with the ME.

Best Regards

Ed Littman wrote:
Thanks ,I understand that src & dithering are different procceses, but that doesn't mean you should not dither after.

Actually, it does. You should not dither AFTER. You should dither at the same time as you reduce word length.

Ed Littman wrote:
src will expand the word legth(32float etc.).
when ever you reduce word length you dither(back down to 24 or 16bits)
your qoute was not clear of that fact & so i'm trying to clarify it.

WHAT?????????????
Sample Rate Conversion does no such thing!!!!!!!!
At no point, when you change the sample rate of a digital recording, do you change the bit depth. If AND I MEAN IIIIIFFFFF you do, it is purely your choice - and not a good one at that. Sample Rate Conversion and Bit Depth have absolutely NOTHING to do with eachother and in every situation they are done seperately. Whether or not you are aware of that is a different situation as most programs/outboard gear are capable of doing these things at the same time.

However, I must stress again. NO, when you downsample, you DO NOT automatically go to 32 bit floating point.

Ed Littman wrote:
also, now a days all sample rate converters up sample to a very high frequency around 15mhz thats an even multiple of all of the other sample rates involved. Then is down sampled to the destination sample rate using even multiples reducing any funky math. So it does not matter if you src from 96 down to 44.1.

upsampling> reconstruction filter>high sample rate>anti-aliasing filter> down sampling

Ed

No, not all sample rate converters do this action. You are referring to a converter that will multiply the sample rate so that whatever the source sample rate and the destination sample rate are, they are common denominators of some insanely large number and then they remove the remaining samples. If you do the math on this, you will find that this is problematic at best ( (44100*48000)/100=21,168,000 samples - this is the number at which these two sampling frequencies are common denominators. That would mean that your microprocessor, if doing nothing else will have to perform this many actions per second multiplied exponentially by the bit depth, with near zero latency. I don't know of a processor on the planet capable of doing that with any degree of stability!)

Ed, I try to respect everyone's opinion on this board, but you are making some very incorrect statements that you should research before you provide conjecture. There are some of us on this board that have studied the hell out of this stuff in college and beyond and unless you're sure of yourself, you could come out looking pretty foolish.

Sorry if I'm a bit harsh, but it aggrevates me when people stick to their guns and they really only are providing what they believe to be true. I will give you credit though, you are being curteous in your discussions. I thank you for that. Let's face it, we're all friends in this business - until you move to my market, then of course you're my competitor! (LOL)

J...

p.s. please don't take my capital letters above as YELLING. I merely use them for emphasis of my point.

If the bit depth is increased to 32 bit float, there is still a 24 bit mantissa. Is dither even needed then? The only truncation would be of the 8 bit exponent and not the 24 bit word.

I could certainly understand the need for dither if it were upsampled to something other than 32 bit float or if there is a need to drop down to 16 bit.

Thanks,
Erik

soundfreely wrote: If the bit depth is increased to 32 bit float, there is still a 24 bit mantissa. Is dither even needed then? The only truncation would be of the 8 bit exponent and not the 24 bit word.

I could certainly understand the need for dither if it were upsampled to something other than 32 bit float or if there is a need to drop down to 16 bit.

Thanks,
Erik

Erik,

I see the logic in your question - and yes, it is possible simply to truncate the additional bits from 32 bit float to 24. However, in true 32 bit float, you really are adding bits, but only when necessary. Most dynamic audio sources will need quite a bit of bit addition.

The really cool thing about 32 bit floating point is that it is far more efficient than straight 32 bit and it yields a dynamic range of nearly 200dB! But, because bits are added (taking the voltage representation as far out as 7 or more decimal places), dithering is still required to get back to 24 bits without a (arguably) noticeable difference.

I guess the bottom line is, if you change bit depths at all, you would be better off performing some type of dithering.

J...

I really suck at this bit depth stuff but I just want to know what's the best bit depth and sample rate you can burn to a CD. 24 bit 44.1K?

radioliver wrote: I really suck at this bit depth stuff but I just want to know what's the best bit depth and sample rate you can burn to a CD. 24 bit 44.1K?

Nope, sorry. 16bit 44.1K. That's it. Of course, this is still really good.

I've done listening studies with a broad based listener group including several very talented and accomplished classical musicians. Out of the 35 people surveyed, only 3 could identify an original, unchanged DSD master compared to the downsampled CD (PCM) master. The equipment is good(listed below), they just couldn't tell.

Equipment used in test:
Pyramix workstation (DSD Master)
Genex DA converter with DSD to Analog output option
Rotel Preamp
Rotel 5 channel amplifier (times 2 - all speakers bi amplified)
MIT Shotgun interconnects and speaker cables
NHT 2.5i speakers (4, plus matching center)
REL Storm Subwoofer

Makes me wonder if it's all worth it - then I listen to the DSD and I know it is...

J...

Cucco wrote: [quote=radioliver]I really suck at this bit depth stuff but I just want to know what's the best bit depth and sample rate you can burn to a CD. 24 bit 44.1K?

Nope, sorry. 16bit 44.1K. That's it. Of course, this is still really good.

I've done listening studies with a broad based listener group including several very talented and accomplished classical musicians. Out of the 35 people surveyed, only 3 could identify an original, unchanged DSD master compared to the downsampled CD (PCM) master. The equipment is good(listed below), they just couldn't tell.

Equipment used in test:
Pyramix workstation (DSD Master)
Genex DA converter with DSD to Analog output option
Rotel Preamp
Rotel 5 channel amplifier (times 2 - all speakers bi amplified)
MIT Shotgun interconnects and speaker cables
NHT 2.5i speakers (4, plus matching center)
REL Storm Subwoofer

Makes me wonder if it's all worth it - then I listen to the DSD and I know it is...

J...

He he... but I'm sure they all could tell a good from a less good converter...?
There are many things involved here.... so many that it makes my head spin around just thinking about it :shock:

Cucco wrote: [Ed, I try to respect everyone's opinion on this board, but you are making some very incorrect statements that you should research before you provide conjecture. There are some of us on this board that have studied the hell out of this stuff in college and beyond and unless you're sure of yourself, you could come out looking pretty foolish.

Sorry if I'm a bit harsh, but it aggrevates me when people stick to their guns and they really only are providing what they believe to be true. I will give you credit though, you are being curteous in your discussions. I thank you for that. Let's face it, we're all friends in this business - until you move to my market, then of course you're my competitor! (LOL)

.

I'm not going to get into a math argument with you,& frankly i'm not that interested in it.
But you may want to check out the glenn meadows mastering board
http://webbd.nls.net:8080/~mastering/login
There are some well known equipment designers & legendary mastering engineers that post on a reguler bassis(even our humble host MF has held the discussions there in high regard).
Most all of the members would disgree with you flat out & give you a run for your $$ on this topic.

Maybe Dave Collins, Michal Jurewicz or bob katz & the list goes on are incorrect fools. you should go find out.
Ed

Ammitsboel wrote: [
It just didn't came out clear in your message, that's all.
By looking at your message he could think that it was ok to deside for himself rather than deside together with the ME.

Best Regards

Most mix engineers make that decision before it gets to me.
I always ask for the highest resolution files that are not digitally limited & not dithered to 16bit if they statrted at a higher sample rate. But if it was recorded at the same resolution that there sending me what do I care what bit/sample rate it is as long as i can open the file up correctly?
How bout you?
Ed

Ed Littman wrote:
I'm not going to get into a math argument with you,& frankly i'm not that interested in it.
But you may want to check out the glenn meadows mastering board
http://webbd.nls.net:8080/~mastering/login
There are some well known equipment designers & legendary mastering engineers that post on a reguler bassis(even our humble host MF has held the discussions there in high regard).
Most all of the members would disgree with you flat out & give you a run for your $$ on this topic.

Maybe Dave Collins, Michal Jurewicz or bob katz & the list goes on are incorrect fools. you should go find out.
Ed

Hmm... your not interested in the math. I think that says it all right there.

What, praytell, would they disagree with me about? I think I've made my case quite scientifically here.

If you are telling me that they would disagree that SRC requires dithering, then you are right, they are incorrect fools. (And by no means is Bob Katz a fool - he knows exactly what he's talking about, and he would not disagree with me.)

If you are saying that they would disagree with me that SRC causes a signal to upsampled to 32bit floating, then again, they are idiots and fools.

Look, I spend all day doing this sh*t with people that have far more education than you or I could ever obtain, yet still I have retained my job. So that either means that I am always quiet (and I think no one here would think that) or I am right and I know what I'm talking about.

Please learn before you speak.

Notice, Michael F. and Tom B. aren't chiming in saying how wrong I am, and they're probably the two most talented and knowledgeable mastering engineers on this forum.

Until you can *PROVE* these strange conjectures that you keep coming up with (I really like the one about bit addition being a part of Sample Rate Conversion), don't tell others they are wrong. As I stated above, I've actually given proof for my side - where's your proof?

J...

P.s. I want to make it very clear, I didn't say that no SR Converter does the least common denominator algorithm, I said that most don't.

Ed Littman wrote:
I always ask for the highest resolution files that are not digitally limited & not dithered to 16bit if they statrted at a higher sample rate. Ed

You're still killing me here... What do higher sample rates have to do with a 16 bit or 24 bit recording? They are *independent*. They have profound influences on completely different aspects of the digital "waveform."

So, by your statement above, if I gave you a 44.1khz sampled track, you would be fine if I gave it to you at 16bit post-dither. But, if it were 96khz, you would ask for the 24 bit copy? Why? Specifically, why do you not care if the lower sample rate version has been dithered, not why do you not want the higher sample rate dithered.

J...

well, I am more intersted in what I hear...thats my job
& yes, in my qoute I meant bit rate not sample rate.

I was referring to that many well respected engineers That I have learned from on glenns board have said that any calculation will expand the word length including src.& when reducing wordlength you dither. Maybe the designer will implement it and not always
give option to the user Like a simple fader movement is going
to increase the utilization of bits and return after the maneuver, yet their are no user dithering options. Also, many have told me that current sample rate converters upsamples first so from going 96 down to 44.1 is no different than 88.2 down to 44.1

i guess it comes down to... you say
"At no point, when you change the sample rate of a digital recording, do you change the bit depth"
& then you say..

"If you are telling me that they would disagree that SRC requires dithering, then you are right, they are incorrect fools. (And by no means is Bob Katz a fool - he knows exactly what he's talking about, and he would not disagree with me.) "

does this not mean you agree?

are we not saying the same thing here?

I dont claim I know more than you or that i'm better at my job than you(i still have mine too), but i saw your quote & it was contrary to what I have learned from other respected sources.

I'm hoping at least to anybody that reads this thread there is somthing to learn from.
Ed

Yes, we all are here to learn. Let's keep it clean, clear and informative. I know how stuff sounds and what I like, but do I know every detail involved? no. I try to keep informed but there are a lot of things that go on in these boxes that aren't very apparent and can get you pinned into a corner without you knowing. Sooooo, instead of slinging mud, lets keep it a little cleaner so we all can sift through the crap and make good music. When I get back from san fran, I'm going to do some tests with my hardware src and see if I can come up with anything informative on at least this unit. It does sound good so at least it has that going for it. whether or not it's doing it correctly, I would assume that someone did their homework.

Ed Littman wrote: i guess it comes down to... you say
"At no point, when you change the sample rate of a digital recording, do you change the bit depth"
& then you say..

"If you are telling me that they would disagree that SRC requires dithering, then you are right, they are incorrect fools. (And by no means is Bob Katz a fool - he knows exactly what he's talking about, and he would not disagree with me.) "

does this not mean you agree?

are we not saying the same thing here?
Ed

Ed,

I think my inflection did not come across right in this quote. I mean to say "If you are telling me that they would disagree AND THEY BELIEVE that SRC requires dithering..." So in that case, I am consistent. However, I can see by my poor choice of words that it came across wrong.

Michael's right, we should all try to learn and not mud sling. I'm afraid I get very passionate about this stuff. I know that some people (who know an awful lot) are good engineers, but may not know all of the intracacies of how each device works. That doesn't mean they can't do their jobs well. I also know that there are some people who are obsessive and must know exactly how something works. Unfortunately, I'm one of those geeks. So, for those who are like me, I try to explain technically why things are the way they are AND make sure that when someone posts something contrary to that, I'm quick to jump in.

By no means do I ever intend to personally attack anyone (which I think, overall, you and I have avoided personal attacks). However, these "heated" debates are sometimes where some of the best information comes from.

Cheers friend,

J... :D

Cucco, I don't think anyone minds heated debates. But it's not life and death, just dither. I try and learn as much as I can grasp but it take a lot of time to know everything involved in making music.

Ed Littman wrote: [quote=Ammitsboel][
It just didn't came out clear in your message, that's all.
By looking at your message he could think that it was ok to deside for himself rather than deside together with the ME.

Best Regards

Most mix engineers make that decision before it gets to me.
I always ask for the highest resolution files that are not digitally limited & not dithered to 16bit if they statrted at a higher sample rate. But if it was recorded at the same resolution that there sending me what do I care what bit/sample rate it is as long as i can open the file up correctly?
How bout you?
Ed

I don't like getting other sample rates than 44.1 because of sound quality issues and that it will take about 1 hour for a "normal" album to be converted in my daw.
I also don't like SRC'ing if i don't need it, so i advice mix engineers to stick with 44.1 unless they are dealing with sources from another rate, then it can be discussed if their SRC or mine SRC is best.

Best Regards,

I used to think the same way until I got my apogee SRC. It sounds great and I like to be able to do all of my processing at 96khz and then SRC down to 44.1. To me, it sounds great and the benifit of processing at 96khz outweighs any reduction in quality from SRC'ing. I have also been using it to reclock signals, which I find improves the sound on poorly clocked mixes. I can also do all of this in realtime so it takes the wait out of the equation.

Well Michael, put it back in the shipping box and sent it to me!!

Michael Fossenkemper wrote: I used to think the same way until I got my apogee SRC. It sounds great and I like to be able to do all of my processing at 96khz and then SRC down to 44.1. To me, it sounds great and the benifit of processing at 96khz outweighs any reduction in quality from SRC'ing. I have also been using it to reclock signals, which I find improves the sound on poorly clocked mixes. I can also do all of this in realtime so it takes the wait out of the equation.

Ahh. Now Michael's touching on an excellent point. Not all is "lost" when you sample rate convert. There are some serious benefits to recording at the higher sample rates IF and only IF you have a good means of getting that back down to 44.1 without detriment.

Unfortunately, reverting to the original question at hand, the SRConverter built into WaveLab is really not all that good.
J...

I find that all software based SRC is not my cup of tea. The main reason is I can't hear it until after it's done. the next reason is time. the 3rd reason is most suck. I love to hear everything from processing, src, dither all in realtime. No supprises.

[quote="CuccoAhh. Now Michael's touching on an excellent point. Not all is lost when you sample rate convert. There are some serious benefits to recording at the higher sample rates IF and only IF you have a good means of getting that back down to 44.1 without detriment.

Unfortunately, reverting to the original question at hand, the SRConverter built into WaveLab is really not all that good.
J...

One of the reasons I don't mind if i getting high sample rates from clients & don't wory about src. is that i procces through an analog chain & my adc is set to 44.1 upon capture.

also, as stated somewere above, that the resampler 192 plug in the master section of wavelab is recomended by the designer & better than the internal version.
ed

Ed Littman wrote:
One of the reasons I don't mind if i getting high sample rates from clients & don't wory about src. is that i procces through an analog chain & my adc is set to 44.1 upon capture.

also, as stated somewere above, that the resampler 192 plug in the master section of wavelab is recomended by the designer & better than the internal version.
ed

Well, you'll get no arguments from me there.

When I master classical, I take the digital signal from my multi-channel mix, allow all automation to take place in the box, while sending it out through my D/A converters in analog to my mastering EQ and then back into the box in 24/32Float, 44.1 and then apply necessary fades. Analog, to me, is the best way to Sample Rate Convert. (I know, it's not technically sample rate conversion...)

J...