48 or 41.1 khz?

[AlexLakis]

Have you tried the two out? Can you tell the difference between the two? Why not listen to what your ears have to tell you?

44.1 will allow you to have a slightly higher track count/processing headroom.

To me, bitrate is much more important than samplerate. I record in 44.1/24 most of the time. I can't tell the difference between 48 and 44.1 once the mix is all said and done.

[zetterstroem]

44.1 for cd..... 48 for dvd etc.

the quality lost in the sample rate conversion is greater than that gained by using 48...

[Bishbashbosh]

Definitely 24bit (there's a dramatic difference in quality from 16bit unless you are using monstrously expensive converters..... even then 24bit...!)

I'd suggest 44.1 UNLESS you are:

1. Doing music for film or TV, in which case 48kHz is the standard, so I usually record at the final delivery rate.
2. Sending to an ME who is going to use analogue processing in the mastering process.... in which case it doesn't really matter what sample rate you use as the ME will reduce back to 44.1 himself.
3. You are intending to mix OTB, and will be converting to your delivery rate from analogue anyway - in which case tracking rate doesn't really matter.

If you are using reasonably decent converters then 44.1 is absolutely fine, though you may find that in certain circumstances on certain DAW's that 48kHz yields slightly better results.
As you mentioned though, this improvement can be easily ruined by poor conversion back to 44.1

[Lagerfeldt]

Originally Posted by zetterstroem

44.1 for cd..... 48 for dvd etc.

the quality lost in the sample rate conversion is greater than that gained by using 48...

Exactly, usually at least.

[PlugHead]

Originally Posted by zetterstroem

44.1 for cd..... 48 for dvd etc.

the quality lost in the sample rate conversion is greater than that gained by using 48...

Ditto - I record at the intended (end) delivery rate to avoid the artifacts of bad SRC...

YMMV,

[BrianT]

There is one other factor. Because of converter design issues, some systems simply sound better at 48K. Really, some converters do a better job of A/D conversion at 48K. Put simply, some converters were optimized for 48K and 44.1K is a slight compromise in performance.

I'm not talking about 48K vs 44.1K as a pure math equation, as that difference is nominal. I'm talking about pragmatic aspects of the hardware design and the difference in how filters are optimized for one freq vs the other. I know several successful engineers who believe their systems sound enough better at 48K to offset any SRC artifacts in going to 44.1K after mix. BTW, SRC has gotten very good in the last couple of years.

Listen to your system recording the same material at both frequencies. Then decide.

[max cooper]

I did some acoustic stuff at 48kHz today, and did a quick mixdown from 48/24 to 44.1/16 using Pro Tools Tweak Head and I was expecting dreck. I had never tried it since so many have warned me to stay away from the uneven math involved.

I was totally surprised that it was not only not bad, but possibly I'm gonna do it this way for the near future.

I still have to do a more direct comparison, but I think it's better than 44.1/24 to 44.1/16.

Oh. Rosetta 800 A/D.

[JP06]

Originally Posted by max cooper

I did some acoustic stuff at 48kHz today, and did a quick mixdown from 48/24 to 44.1/16 using Pro Tools Tweak Head and I was expecting dreck.

Heh, one would have to have really supernatural ears to hear a difference in these speeds.

I still have to do a more direct comparison, but I think it's better than 44.1/24 to 44.1/16.

I don't think you'll hear a difference in your comparison. But technically the difference is there. Your signal is more natural (more in the way it was played) when it stays in 44.1kHz. Unlike when the signal is matematically compressed from 48kHz to 44.1kHz.

If you are going for cd, then use 44.1kHz or 88.1kHz.

[Bishbashbosh]

Originally Posted by JP06

Heh, one would have to have really supernatural ears to hear a difference in these speeds.



I don't think you'll hear a difference in your comparison. But technically the difference is there. Your signal is more natural (more in the way it was played) when it stays in 44.1kHz. Unlike when the signal is matematically compressed from 48kHz to 44.1kHz.

If you are going for cd, then use 44.1kHz or 88.1kHz.

Although I'm not usually a proponent of higher sample rates, as I mentioned in my earlier post, 48kHz CAN sound better than 44.1kHz given certain circumstances.
These circumstances usually involve anti-aliasing filters which when recording at 44.1kHz kick in at about 18kHz (which is just within the audible range).
Usually more modern, better quality converters sample at a higher rate and then convert back down to 44.1kHz, but older converters may use a simple filter which can be audible.
Using these same converters at 48kHz, the anti-aliasing filters are set at a much higher frequency (say 21kHz) which is outside the range of human hearing.
I would suggest that it's worth checking to see if 48kHz yields an improvement...... I certainly felt it was an audible improvement on my old 001.

Using a good quality sample rate converter (like Barbabatch) can also give improvements when converting down to 44.1kHz from 48k.

[BrianT]

Originally Posted by JP06

Heh, one would have to have really supernatural ears to hear a difference in these speeds.

But technically the difference is there. Your signal is more natural (more in the way it was played) when it stays in 44.1kHz. Unlike when the signal is matematically compressed from 48kHz to 44.1kHz.

I don't think you'll hear a difference in your comparison.

It's amazing how dubious information or incorrect conclusions somehow take on a life of their own on the internet. It's as though, by simple repetition, statements somehow become more and more true, regardless of their validity.

Has anybody in this thread had a recent listen to the various hardware and software Sample Rate Converters that have become available in the last couple of years? I have. It's almost a non issue at this point, sonically. The only real argument against good SRC is that it's one additional slightly inconvenient step to make a CD.

Listen to your own rig at both rates. Again, the difference is not just math, it's also in the design of your converters. What you're listening for is how your A/D sounds, and there may well be more difference than you expect, depending on your hardware. It's all about how they did the filters.

[markknopfler66]

Originally Posted by BrianT

It's amazing how dubious information or incorrect conclusions somehow take on a life of their own on the internet. It's as though, by simple repetition, statements somehow become more and more true, regardless of their validity.

Has anybody in this thread had a recent listen to the various hardware and software Sample Rate Converters that have become available in the last couple of years? I have. It's almost a non issue at this point, sonically. The only real argument against good SRC is that it's one additional slightly inconvenient step to make a CD.

Listen to your own rig at both rates. Again, the difference is not just math, it's also in the design of your converters. What you're listening for is how your A/D sounds, and there may well be more difference than you expect, depending on your hardware. It's all about how they did the filters.

Do you know about the converters on AKAI DR16 Pro? What would be your choice if you were using this machine?

Thanks

[AdamJay]

its been my experience that cheaper, less capable converters yield much better results at high sample rates than they do at low sample rates. (motu, m-audio, etc.)

where as better, more capable converters (lynx, apogee, etc.) don't exhibit nearly the drastic improvement when operating at higher sample rates.
My DAC-1 sounds about 99% as good @ 44.1khz as it does @ 96khz.

Clocking has alot to do with it as well.
Clock those cheap converters to a Big Ben and the quality gap between sample rates will shrink.

[pingu]

I asked a similar question at PSW

i will remain at 44.1 for the time being


For a bit of technical info.

http://recforums.prosoundweb.com/ind...7749bbfbf43759

[Lagerfeldt]

Originally Posted by AdamJay

its been my experience that cheaper, less capable converters yield much better results at high sample rates than they do at low sample rates. (motu, m-audio, etc.)

where as better, more capable converters (lynx, apogee, etc.) don't exhibit nearly the drastic improvement when operating at higher sample rates.
My DAC-1 sounds about 99% as good @ 44.1khz as it does @ 96khz.

Clocking has alot to do with it as well.
Clock those cheap converters to a Big Ben and the quality gap between sample rates will shrink.

Very true about 44/higher, it's got to do with the filter design.

However clocking to an external device like Big Ben is a much discussed issue. While it will fix problems with multiple units working together you can't improve on the internal clock, so it won't ever be better just by clocking. Physics simply don't allow it.

[Bob Olhsson]

Brian's point is critical to understand.

Sample rates and bits only tell us about POTENTIAL performance. Few manufacturers publish truly meaningful specs. so there is no substitute for listening.

Back around 1990 I was real excited about the performance of my new Apogee A to D converter. A friend of mine had tried it with his Panasonic 3700 and just didn't "get" why I was so impressed. Turned out that he had compared at 48k while I had compared at 44.1. He had heard the loss at 44.1 and assumed this was entirely because of the sample rate. I had not heard much of any loss at 44.1 with the Apogee and assumed the difference was mostly the converter. We were both right and wrong.

[AdamJay]

Originally Posted by Lagerfeldt

Very true about 44/higher, it's got to do with the filter design.

However clocking to an external device like Big Ben is a much discussed issue. While it will fix problems with multiple units working together you can't improve on the internal clock, so it won't ever be better just by clocking. Physics simply don't allow it.

Hmmm, a big ben made my old fireface800 more detailed and clearer. And it was alone, not connected to any other digital device. DAW-FF800-Ben.
That Son of a bitch, Ben, must have defied physics because my ears do not lie.

[theblue1]

Originally Posted by markknopfler66

Hi,
I have been recording my band's first album in our studio. Our HD recorders are capable of recording 48 or 41.1 khz. and 24 bit or 16 bit. I have read in many places that conversion of 48 khz to 41.1 khz in the latest process adds some noise or have other disadvantages when compared to directly recording in 41.1 khz. Is this true? Which one would you prefer if you were in the production of my album?

Many Thanks for your helps

Murat

Absolutely true.

And while various marketing departments make all kinds of wacky claims about SR conversion -- the bottom line is, if you convert an audio stream from one format to another that is not an even multiple of it (or vice versa) you will -- unavoidably -- end up with artifacts which will almost certainly be noticeable to careful listeners.

If you're outputting to 44.1 kHz -- work at 44.1 or at an even multiple of it, like 88.2 or 176.4 kHz -- NOT 48, 96 or 192 which are based on the old (but still current) video standard of 48 kHz.

(Now ALL of this insanity could have been avoided if the MORONS at Sony and Phillips hadn't compromised on SR to take the lesser SR of 44.1 kHz in order to deliver a one hour disc to the honchos... a one hour disc that has seldom been filled completely without resorting to the most egregiously superfluous filler... crappy alternate takes, even ads... For a quarter century we've all had to live with the deleterious effects of that incredibly venal, short-sighted decision. Just say "never again" to any leadership role from Sony or Phillips in this industry. Screw them.)


PS... and OF COURSE, you should use 24 bit. 24 bit is as near to win-win as we're gonna get. (That said, an input bit depth of 20 bits is probably sufficient to faithfully capture most any input signal -- but internal processing at at least 24 bits (if not 32 or higher) will help preserve the accuracy of your audio as you engage in more processing, mixing, etc.)

[Bishbashbosh]

Originally Posted by AdamJay

Hmmm, a big ben made my old fireface800 more detailed and clearer. And it was alone, not connected to any other digital device. DAW-FF800-Ben.
That Son of a bitch, Ben, must have defied physics because my ears do not lie.

The quality of the DAC can be dramatically improved by an external clock, if the original internal clock isn't too great.
This doesn't mean that the mix in the box is any different, it simply means that you're hearing it more clearly on YOUR system.

However I'd be willing to bet that ADC can be improved also, which WILL make a real difference to the actual sound recorded, rather than just an improvement to monitoring.

[AdamJay]

Originally Posted by Bishbashbosh

The quality of the DAC can be dramatically improved by an external clock, if the original internal clock isn't too great.
This doesn't mean that the mix in the box is any different, it simply means that you're hearing it more clearly on YOUR system.

However I'd be willing to bet that ADC can be improved also, which WILL make a real difference to the actual sound recorded, rather than just an improvement to monitoring.

sorry, i should have clarified.
i was referring to the ADC on the Fireface w/ Big Ben being improved.
which apparantly defies physics, according to Lagerfeldt

[Lagerfeldt]

Originally Posted by AdamJay

Hmmm, a big ben made my old fireface800 more detailed and clearer. And it was alone, not connected to any other digital device. DAW-FF800-Ben.
That Son of a bitch, Ben, must have defied physics because my ears do not lie.

You have to differenciate between the ADC and DAC. While your particular listening experience is influence by the DAC, slaving the ADC via word clock is not going to improve matters in a 2 unit setup.

There's plenty of detailed litterature on the internet about how converters work. You can also read books such as "Mastering Audio - The Art and the Science" by Bob Katz. Besides providing a good insight into mastering there are lots of well explained chapters on digital audio and converters. And one chapter in particular about this issue, chapter 19 called "Jitter - Separating the Myths from the Mysteries".

For copyright reasons I can't print the chapter here, but feel free to read the book yourself. It'll do a far better job than me explaining things in English.

I've taking the liberty of quoting Dan Lavry (maker of Lavry converters, I guess you know them...) on the subject:

"You are correct to say "Dan Lavry believes that external clocks can RARELY (if ever) make any system sound better than using a device's internal clock".

I would go much further then that: it is not about what I believe, it is about engineering.

I believe that people hear a difference between external clock boxes, and between internal and external clocking. Any and everyone is entitled to their subjective likes and dislikes.

It is a fact that jitter creates rather unpleasant distortions (non harmonic) and in most cases it increases the noise floor.
Yet, one can decide they like more noise, harsh distortions... all that is rather surprising, but not arguable because it is subjective.

But those that push the notion that an external clock will improve the sound, are not just saying that the sound will be altered, that some people will like it, others will not. The claim here is that there will be an improvement. That claim really translates to "more jitter is good". Of course no one is ready to state publicly that "more jitter is good", and most often the proponents of external clocking will go out of their way to convince you that they are providing with less jitter.

Less jitter the internal clock? A good internal clock provides a lot less jitter than the best of all external clocks.

In the case of internal clocking, one relays on a local fixed crystal aimed at keeping the frequency constant. That crystal circuit would be positioned near the AD, to minimize interference.

For external clocking, one may Begin with an equal quality crystal circuit inside the clock box, but the signal has to go through a driver, cable, receiver and yet another crystal circuit, all with some frequency locking circuitry...All of that is a lot more jitter.

So in order to overcome the above stated reality, there are stories floating around, that a clock box sends signals that improve the AD clocking. This is of course impossible, because one can not improve without knowing what needs to be improved. The clock box gets to send the signal to an AD, but it is a one way communication, and the AD does not tell the clock box what it is doing. In other words, a doctor can not heal a patient without seeing talking or having any communication with the patient.

The clock box can not send a better word clock then a simple square wave with the lowest possible jitter. Putting the same clock inside, eliminates many of the elements piled up in series between the clock source and the AD.