Up Sample Conversion Software

[jtwrace]

What are you using? How hard is it?

 

[Lars]

Wave Editor or iZotope RX Advanced. Both are very easy to use. Wave Editor is the way to go if this is all you want to do.

 

[jtwrace]

Maybe someone could do some screen shots or Chris could do a How to??? **hint** **hint**

 

[ted_b]

And when? What are the benefits?

 

[BobH]

in the past. It was very easy to use.

 

How much difference did it make? It made the file sizes a lot bigger!

 

Honestly, I really could not detect a difference either way except, if really pushed to make comment then I would say it made things slightly indistinct and harder to hear - but I'd be struggling to justify those comments! I tried upsampling to 24/44.1, 24/88.2 and 24/96 and always got the same result - if anything upsampling to 24/88.2 had less impact than any of the others but, like I said, any differences were far too subtle for my system and/or ears!

 

YMMV of course but I count in the 'not worth the effort' group!

 

[ted_b]

So i did a quick search and came up with the facts that Barry Diament (renowned engineer, HiRez DAC enthusiast and CA contributor) says that maybe some DACs will perform better at higher sample rates (i.e play to their sweetspots)....maybe. Steve Nugent (Empirical Audio) likes the results of Wave Editor (same Isotope 64 bit SRC) almost universally...for his good redbook. Others are less definitive, but few if any describe sonically what the reward is of this massive gain in file size!

 

Ted

 

[jtwrace]

For me, I would like to try it. The WE is only $79 so not much lost there considering...

 

I would only try my files that I listen to more and that I really know well. I just thought I would ask around before I spend time doing this. I upload all my Cd's in the AIFF Encoder format. Hopefully more will coment on this. Weiss? Steve N?

 

[vortecjr]

and with my wife selecting between 16/44.1 and an upsampled version of the same title done on sony's soundforge....I could pick them out ever time! Now I was sitting right in front of the speaker and really listening out for the difference. I dont think I could tell them apart from far away during casual listening.

 

I have also stated in another post that, "We also did some software upsampling with dbpoweramp during a rip for a friend who insisted on hearing Gilmore at 24/96 and we were quite impressed. We heard a much larger soundstage (that sold him....so add one one more computer audiophile to the gang) even if my hard core audio buddy insists its not real music if we upsample."

 

Hope this helps!

 

ps you can get sony's soundforge on a 30 free trial from the sony store....

 

[cfmsp]

Jtwrace,

 

I tried it, but couldn't tell much of a difference, if any, although I'm no where near as able to hear things as others, at least I rarely hear 'night and day' differences when other people claim to.

 

I did check around before trying it, and Wave Editor is absolutely the cheapest way to get the industry leading upsampling software from iZotope. The only real competitor is the much more expensive Weiss product - Saracon?

 

Another plus is that the Wave Editor guys (Audiofile Engineering) may be working on an Amarra-like product. They're great guys. They also have a trail period I think.

 

Snnic has licensed iZotope for inclusion in Soundblade, and apparently it will be included in a future release of Amarra.

 

enjoy, and let us know what you find,

clay

 

 

 

[PeterSt]

Maybe, before you dive into upsampling, it is good to wonder what actually happens anyway inside of your DACs.

 

:-)

 

[vortecjr]

I understand your comment and actually it's a curiosity of mine. I don't have a complete opinion on this yet and I was wondering if you could comment further. Given the choice of software up sampling or hardware up sampling is one better than the other as a rule or does it have to do with the quality of the algorithm that each uses. If you choose to software up sample to say 24/192 and play it through a 24/192 dac does the dac just convert the file to analog and by-pass the algorithm all together. Finally, does a dac that receives a 24/48 singnal upsample or not or does it depend on the dac?

 

thank you and sorry for all the questions

 

Jesus

www.sonore.us

 

[PeterSt]

Hello Jesus,

 

If you scroll through this article (picked it up from another thread at CA) : http://www.acourate.com/freedownload/SRCFilterInvestigations.pdf

you get an idea of the net effects of filtering. Besides a filter does good things, it also does bad things, and it is not difficult to get that from the article. For now (not that I'll come back on that in here) let's assume the net result is good.

 

This type of filtering (which is about reconstructing (recovering) the waves as good as possible from a too low sample rate (red book) into something which better represents the wave *after* upsampling) is always necessary, or otherwise the "better represents" is not there. Think about the interpolation (read : inventing of new sample points), which you -simplified- best can see as creating more dots on a virtual sine, assumed it is about a sine. Thus, when a 11025 Hz sine -sampled at 44100- consists of 4 dots, where are the new 4 dots to go when upsampled to 88200 to mimic the sine (etc.) knowing that the original 4 dots are not at nice places to begin with (this is hard to understand, but also less important for the story). What is important though :

 

After a stage of upsampling, and caused by the reconstruction filter, there is this ringing. Ringing is no virtue by itself. Thus, as said, the net result we assume ok, but the ringing "half" of that result sure is not.

Now, what happens if you upsample in front of the DAC ? You will feed it with ringing material. What will happen next ? your oversampling DAC will again add ringing, because that too "upsamples" (though many more times when it is sigma-delta), and of course nothing knows about your added ringing, and it is treated as normal material (music data), and I don't think this is a good idea.

 

Thus, additional ringing, and it won't / can't be removed.

 

What I said elsewhere, and this may even be more important (may), is that for 100% sure you will be applying two different methods of filtering; one from your own (whatever it is today, and whatever it may be tomorrow), and one from the SRC in the DAC. Don't let this be confusing : the filtering I talk about is the digital filter, going along with your upsampling and going along with the oversampling in the DAC (the SRC in there).

These two different methods may cause additional anomalies. Thus, where one method creates a consistent type of ringing (which will be perceived at listening somehow, although possibly not directly recognizeable) is now crossed with another type. Also, both will not ring at the same places, which by itself may cause additional anomalies. May, because I just expect that (hence, not sure).

 

Lastly, at using sigma-delta (which will be, say, 99% of cases) that DAC may upsample 64 to 512 times, while you too add 2 or 4 times. Yes, it is 2 or 4 times more, but what is the use ?

4 times (and even 2 times) is the most useful because it shifts away the Nyquist frequency to the inaudible area (aliasing), but 64 or 512 times is totally useless, let alone 2 or 4 times that (making it 128/256 to 512/1024 times). Sadly the 64-512 times the sigma-delta applies is just a necessity to let the (1 bit) DAC operate, and it can't be shut off, nor can its necessary filter.

 

Only when you are using a DAC that can operate at the input rate (we call that NOS - oops), upsampling yourself it not only very useful (shifting away Nyquist), but according the books even a necessity (moving out aliasing to the inaudible area).

 

All 'n all, upsampling is useful, filtering belongs to it (taking away high frequency burden of your amps), but one instance doing it, really is enough. Two instances will interfere with unpredictable results.

From all follows that those using sigma-delta should never upsample themselves.

 

I hope it is clear a byte ?

Peter

 

(PS: don't let follow from this that I like upsampling w/ filtering, which just is a subject by itself; well, you know).

 

[PeterSt]

If you choose to software up sample to say 24/192 and play it through a 24/192 dac does the dac just convert the file to analog and by-pass the algorithm all together. Finally, does a dac that receives a 24/48 singnal upsample or not or does it depend on the dac?

 

At using sigma-delta, this always operates at a higher sample rate; thus, when the sample rate 2882.4 (64 times), it will oversample whatever it is fed with to 2882.4.

 

At using multi bit DACs it depends how the architecture has been setup;

Theoretically it can work like what comes in goes out, but this is very uncommon, because it would imply NOS, and afaik the few known 24 bit multibit DACs are not setup NOS. This means they upsample, and again it is a choice of architecture, to what. It can be to 192 but also to 384. But ...

 

This too is not much common, and you can also encounter something like 210.9.

Now it is getting confusing, because this is related to the SRC which may be in there (these days you could say "will be in there", and that runs on its own speed, but runs on a speed that comes handy for clock christals.

 

We now slowy dive into the matters of why some DACs are cheap and others are expensive, because, for example, a DAC which runs good at 48KHz will not run good at 44.1. It needs another clock for that (but one clock can serve 48, 96 and 192, and another does 44.1, 88.2, 176.4).

 

More complex it gets when no "common denominating" SRC is used, and the output is equal to the input (NOS). Now there is no anti-jitter device (the SRC), so all kind of provisions must be taken to achieve good jitter specs (ehh, which 4 years ago were unthinkable).

 

To answer your question : it depends !

If you have a sigma-delta, it is clear. No choices there. Output is always at the rate the DAC runs at.

If you have a multi bit you can expect anything. BUT, my last mentioned case will not be found I think (ok, not yet, haha). It is too much against good theories, too difficult (with todays techniques like the SRC), and too costly. I think the only one which *had* that was MSB (but up to 24/96 only). Today they too upsample.

 

Additionally, notice that there's also something like an anolgue filter, which filters the high frequency remainders of or the native digital data, or the sh*t happening within sigma-delta. This *is* important, because this filter should adapt to what happened in front of it. Thus (for best sonic results), it should filter differently when the DAC is fed with 96KHz vs. 44.1. Never mind this by itself, but this again is a reason why DAC manufacturers will like to output at one rate only. And for this case try to see that the input to the DAC chips is the output of the SRC, if there, and it is that output of the SRC which is always the same, making the input to the DAC always the same, making the analogue output always the same, making the analogue filter allowed to be always the same.

If the ouitput is not always the same, the filter still can be the same, but this is not optimal.

 

Peter

 

[cfmsp]

 

"I hope it is clear a byte ?"

 

Yes, Peter, thank you.

 

clay

 

[vortecjr]

Thank you very much for your time and effort on this! I read it through very quickly this morning. I will sit down and read it again after the boys are sleeping and its quiet!

 

Best regards

 

Jesus R

www.sonore.us

 

[vortecjr]

just got around to some quiet time and read your post again. I'm going to forward it via e-mail to some friends interested in upsampling and dacs in general. Speaking of e-mails can you contact me at [email protected] I want to chat with you about some things off line. Thanks for everthing!

 

[DSD_mastering]

We've tried dozens and the best we have heard and now use is Weiss Saracon. With lots of user adjustablilty, you can't beat it.

 

 

Regards,

 

 

[Thomas J]

I use Weiss Saracon likewise. Excellent product.

 

[Audio_ELF]

Bruce ...

Given that Saracon is £1000 or so (£500 without DSD capabilities) ... can you give any indication of any software that gets close for less ... i.e. WavEditor (Audiofile Engineering) comes with iZotope for $80. Does this get close at all? Can you maybe indicate some of the other re-sampling options you tried and found close to Saracon?

 

Eloise

 

[DSD_mastering]

For PCM work, I've also liked R8Brain Pro. There is also a free app.

 

http://www.voxengo.com/product/r8brainpro/

 

For DSD work, there is a free app that comes with the Korg MR units.

 

 

Regards,

 

 

[cfmsp]

 

 

Eloise,

 

I think you may know this, but thought I would reiterate for those who are following this thread.

 

Not only does Wave Editor provide the highly-regarded iZotope for $79, Sonic Studio also licensed iZotope for inclusion in Soundblade, and appear to be on the verge of supplying it with Amarra in a future release, which would seem also to be a vote of confidence re its quality.

 

clay

 

 

 

 

 

 

 

[DSD_mastering]

As does Sony Sound Forge

 

 

Regards,

 

 

[cfmsp]

 

thanks Bruce

 

Great, so Sound Forge is a solution for our PC brethren, as Wave Editor only runs on a Mac.

 

clay