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Author Topic: 24/96 to 16/44.1 - Best Method?  (Read 5718 times)

mark4man

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24/96 to 16/44.1 - Best Method?
« on: June 02, 2004, 11:41:27 pm »

Crew,

All set to perform bit format & sample rate conversions (in WaveLab) for CD pre-mastering...

& I ran across an old e-mail response from one of the tech brains at Cakewalk (I was about to upgrade to SONAR; & asked them some basic questions regarding 24/96 to 16/44.1 conversions.)

Up until I read this e-mail this evening, I was convinced I would perform the conversions in the digital domain...

but the Support Tech from Cakewalk suggested that...because of the interpolation artifacts that the (digital) process introduces (no matter whose sample rate converters are used)...it's actually better to output the 24/96 file as analog; & recapture at 16/44.1...that the results have less inherent noise.

So, I wanted to ask...

What's the consensus in the forum?

Who converts in the digital domain...who recaptures the analog file...

...what kinds of results do you get; & why.

Thanks very much,

Sincerely,

mark4man
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jfrigo

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #1 on: June 03, 2004, 12:47:10 am »

mark4man wrote on Wed, 02 June 2004 20:41


Who converts in the digital domain...who recaptures the analog file...


Digital. A good sample rate converter like the Lavry 3000 or Sonic Solutions HD sounds great. Back in the earlier days of SRC, there were many nasty sounding ones, and a trip through a good set of converters may have been a potential improvement, but today, keeping it digital offers the best performance in my experience. However, we did rent a Lavry DAC924 for the last SACD session (to feed the Meitner AD), but 96k PCM to DSD conversion isn't the same thing as going from 96 PCM to 44.1 PCM. It's still early in the life of SACD.

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bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #2 on: June 03, 2004, 07:59:07 am »

At this point, I avoid SRC like the plague as I have found nothing in software that is invisible and don't have a hardware SRC. Hopefully I'll have something soon to alleviate. More later.

jfrigo wrote on Wed, 02 June 2004 23:47

However, we did rent a Lavry DAC924 for the last SACD session (to feed the Meitner AD), but 96k PCM to DSD conversion isn't the same thing as going from 96 PCM to 44.1 PCM. It's still early in the life of SACD.

Cool, I didn't know you had done any SACD. What was the session you mastered for SACD, Jay?
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Brad Blackwood
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jfrigo

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #3 on: June 03, 2004, 03:53:33 pm »

I just posted a long response to Brad's post, but I got an error when I tried to post and then couldn't "backstep" to the page with all the typing. Frustrating. Let me try again....
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jfrigo

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #4 on: June 03, 2004, 04:13:13 pm »

bblackwood wrote on Thu, 03 June 2004 04:59

At this point, I avoid SRC like the plague as I have found nothing in software that is invisible and don't have a hardware SRC. Hopefully I'll have something soon to alleviate. More later.


As I was saying...
Brad,
I think you would be happy with the Lavry 3000, and the Weiss also seems like it would be worth looking into, but I haven't actually heard that one yet, so I can't say for sure. The Sonic HD background SRC sounds great and has some filter options which is nice, but it's not in-line/real-time like the hardward boxes, so it's not as convenient for many things.

When you get 96k sources, I guess you just D/A at 96 and go through the analog chain and A/D at 44.1? Once you get SRC in there, you may like to try the A/D at 96k and run the limiter at 96k. Then again, you may not. It will interesting to hear your impressions at that time.
Quote:


jfrigo wrote on Wed, 02 June 2004 23:47

However, we did rent a Lavry DAC924 for the last SACD session (to feed the Meitner AD), but 96k PCM to DSD conversion isn't the same thing as going from 96 PCM to 44.1 PCM. It's still early in the life of SACD.

Cool, I didn't know you had done any SACD. What was the session you mastered for SACD, Jay?



Notice the "we" as I imagine you must have. It was one of the sessions DC ("Dave Collins" for the acronym impaired) and I worked on together a little while after he made Mastersuite's studio his home base, with Dave EQing and me editing. It was The Yellowjackets "Time Squared" for Heads Up/Telarc, and was nominated for a 2004 Grammy in the Best Contemporary Jazz category. Rich Breen (richbreen.com) did a nice mix and the project was recorded to and mixed from Pro Tools HD at 96k, so I know that system can sound good in the right hands.

After prepping, we took the output of the Sonic HD into the rented Lavry DAC924 out to the Meitner converters and Genex that Telarc had sent along. I thought the original 96k PCM sounded better, but that could be because I was listening to the PCM through the DAC924 and the DSD through the Meitners. It sounded a little murkier after going through the Meitners. I haven't actually heard the hybrid SACD/CD since, so I don't know how the SRC'd 44.1 PCM sound next to the DSD on a universal consumer player.

David Glasser from Airshow could probably tell us what the latest SOTA is in PCM to DSD transcoding and whether it is satisfying to his ears. With my limited DSD experience, I'll leave it to him to expand upon the issue if he would like to.
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OTR-jkl

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #5 on: June 03, 2004, 04:34:37 pm »

Quote:

Who converts in the digital domain...who recaptures the analog file...

...what kinds of results do you get

FWIW-
I had a 48/24 project a little while back and used the SRC within Samp7 to knock it down to 44/16. As Brad can attest, the results using that method lack something to be desired... I don't have a HW SRC either so I'm not sure what I will do with the next one that comes in. Fortunately, most of my stuff comes in at 44/24 so its rarely an issue - so far...
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mark4man

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #6 on: June 03, 2004, 06:50:53 pm »

Thanks all...

How about bouncing out to an Alesis Masterlink?

How are the converters in that device?

Thanks again,

mark4man
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bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #7 on: June 03, 2004, 07:21:31 pm »

Not so great in my limited experince, Mark.

I have chosen to simply load twice when I need 48k stuff for DVD release as well as the normal 44.1k stuff for CD. I'd really like to have an async hardware box and am looking into an interesting option right now...
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Brad Blackwood
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Re: 24/96 to 16/44.1 - Best Method?
« Reply #8 on: June 03, 2004, 07:23:00 pm »

jfrigo wrote on Thu, 03 June 2004 15:13

Notice the "we" as I imagine you must have. It was one of the sessions DC ("Dave Collins" for the acronym impaired) and I worked on together a little while after he made Mastersuite's studio his home base, with Dave EQing and me editing.

Nice. How much editing goes on during one of those YJ sessions? In my experince those jazz guys have their stuff down cold way before mastering...
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Brad Blackwood
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David Glasser

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #9 on: June 03, 2004, 08:31:47 pm »

Quote:

jfrigo wrote on Thu, 03 June 2004 14:13
David Glasser from Airshow could probably tell us what the latest SOTA is in PCM to DSD transcoding and whether it is satisfying to his ears. With my limited DSD experience, I'll leave it to him to expand upon the issue if he would like to.



For PCM to DSD & DSD to PCM, we've been using the Prism ADA-8, and I think it sounds fine. I also use the Sony SBM-Direct software which runs on the Sonoma to make Red Book masters from DSD - it's stereo only and only outputs 44k (16 or 24 bit). It sounds very good. I'm interested to see what the new Phillips Pro-Tools thing is all about.

Re: PCM src - for real time stereo, I use the Weiss SFC-2. Excellent. Or the Prism ADA-8. For out-of-real time, the Sonic HD src can't be beat by anyone.
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David Glasser
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jfrigo

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #10 on: June 03, 2004, 08:44:57 pm »

bblackwood wrote on Thu, 03 June 2004 16:23

jfrigo wrote on Thu, 03 June 2004 15:13

Notice the "we" as I imagine you must have. It was one of the sessions DC ("Dave Collins" for the acronym impaired) and I worked on together a little while after he made Mastersuite's studio his home base, with Dave EQing and me editing.

Nice. How much editing goes on during one of those YJ sessions? In my experince those jazz guys have their stuff down cold way before mastering...


It depends...sometimes it's quite a bit. Let's take a few edit projects I did with Dave as an example. The X-Men 2 soundtrack ended up actually having quite a bit of editing. I sat with Casey Stone, the scoring mixer, for at least a couple hours dealing with little fine-tuning, enhancements, volume rides etc. His quote to me at the end was, "You really earned your money today." Add to that the normal assembly, sequencing, and PQ coding and it's actually more work than just throwing some start IDs on there.

With Sandro Albert, another jazz project mixed by Rich Breen, the artist and producer stayed for quite a while trying different sequences, fades, spacings, crossfades, volume adjustments etc, so again it wasn't just a simple PQ job. I also had to come back for a resequence a couple weeks later. Sometimes the artists can't quite make up their minds, but the material was nice as was the artist, so no big deal.

For the War remasters for Rhino/Warner, the original producer was in attendance and he kept me quite late doing "new, never before released edits" of a couple songs (but they left the classic "Lowrider" alone), plus copying the edit from an earlier release of another song, plus the two CDs worth of assembly that there were changes on, and multiple refs of alternate versions of things. Again, a fair amount of work for an "edit" session.

As for the Yellowjackets, we did the normal amount of messing around with spacing and fades and maybe tried an alternative here or there, but in their case, I'd say they were pretty well prepared and it went pretty quickly. I did have to come back and babysit a second Genex transfer for some reason, but I can't remember if we had to change something or fix something or if they needed another copy or what - I just remember that I had to meet the mixer again the next day to do some more work.

So, sometimes edit session are quite involved, sometimes not so involved, but they seldom turn out to be a simple Sequencing and assembly anymore. But hey, billable hours are billable hours. Do you get lots of simple edits, or are people getting pretty creative? It sure seems that many if not most of my clients enjoy the positive difference we can make during the editing stage.
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bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #11 on: June 03, 2004, 08:47:55 pm »

jfrigo wrote on Thu, 03 June 2004 19:44

As for the Yellowjackets, we did the normal amount of messing around with spacing and fades and maybe tried an alternative here or there, but in their case, I'd say they were pretty well prepared and it went pretty quickly.

Kinda what I figured...

Quote:

Do you get lots of simple edits, or are people getting pretty creative? It sure seems that many if not most of my clients enjoy the positive difference we can make during the editing stage.

Hardly any editing nowadays (thank goodness)...
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Brad Blackwood
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jfrigo

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #12 on: June 03, 2004, 08:50:39 pm »

bblackwood wrote on Thu, 03 June 2004 17:47


Quote:

Do you get lots of simple edits, or are people getting pretty creative? It sure seems that many if not most of my clients enjoy the positive difference we can make during the editing stage.

Hardly any editing nowadays (thank goodness)...


Did you hire somebody to do your editing or is it just that there aren't client attended sessions with people wanting to try things?
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jfrigo

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #13 on: June 03, 2004, 08:56:05 pm »

bblackwood wrote on Thu, 03 June 2004 17:47

jfrigo wrote on Thu, 03 June 2004 19:44

As for the Yellowjackets, we did the normal amount of messing around with spacing and fades and maybe tried an alternative here or there, but in their case, I'd say they were pretty well prepared and it went pretty quickly.

Kinda what I figured...



They've made a record or two so I'd hope they've got it down by now. Plus, it's not like some of the classical guys who want to do hundreds of edits or the rap guys who crossfade in and out of skits every other song. Most jazz sessions are pretty smooth.
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bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #14 on: June 03, 2004, 09:12:24 pm »

jfrigo wrote on Thu, 03 June 2004 19:50

Did you hire somebody to do your editing or is it just that there aren't client attended sessions with people wanting to try things?

Haven't hired anyone yet, will most likely do so with the new room. Most clients do not attend, but even those that do don't usually want much editing - they usually take care of that at  the studio before-hand...
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Brad Blackwood
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bbalin

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #15 on: June 06, 2004, 02:45:45 pm »

Sorry for jumping in here - but this topic really makes me curious :
@Brad :
Quote:

At this point, I avoid SRC like the plague as I have found nothing in software that is invisible

Brad, what are the problems you're seeing with software SRCs ?

I remember our (small) discussion on the samplitude forum but unfortunately it did not go on.

-Boris
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bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #16 on: June 06, 2004, 04:06:49 pm »

bbalin wrote on Sun, 06 June 2004 13:45

Brad, what are the problems you're seeing with software SRCs ?

I remember our (small) discussion on the samplitude forum but unfortunately it did not go on.

Hi Boris, welcome to the board...

When using any software SRC I've used, I notice a slight collapse of the soundfield along with an overall darkening of the track. The very highest setting in Samp7, for example, yields a narrower soundfield an sounds like you reduced 10kHz by a good .5dB. Not acceptable to me.

The best hardware SRCs do not do this, but even the finest ones (Weiss and Lavry) use chips that have been surpassed.

Once I figure out whether the new box works or not, I'll let you know...
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Brad Blackwood
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OatBran

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #17 on: June 06, 2004, 11:25:22 pm »

  I have a quick question that applies indirectly to this topic.  In my experience I have always found the conversions from 88.2 to 44.1 more seemless then the 96 to 44.1 conversion.  Not much more, but apparent none the less.  I haven't used any hardware src to compare with, but all of my listening with the software side on various projects has shown the few 88.2's I have gotten to be a little cleaner transition then I was able to get with the 96's.  Most of these projects with higher sampling rates have been orchestral in nature... maybe that has something to do with it.  I am guessing that the reasons for this might have something to do with a more linear conversion (x2).

 Is this just in my head/ears or has anybody else found this?  And if so, why did we as an industry go 96 instead of 88.2 if our market is still 44.1?
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TotalSonic

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #18 on: June 07, 2004, 12:21:08 am »

OatBran wrote on Mon, 07 June 2004 04:25

 
 Is this just in my head/ears or has anybody else found this?  And if so, why did we as an industry go 96 instead of 88.2 if our market is still 44.1?


I think the 96kHz decision might reflect more the fact that digital film/video production has been using 48kHz as its standard for quite some time now.  So I guess the market envisioned is more towards video standards than audio ones.  Kind of reflects the trend that DVD-V sales are going up while CD sales are pretty flat. Cue The Buggles "Video Killed The Radio Star"

Best regards,
Steve Berson

OatBran

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #19 on: June 07, 2004, 05:29:47 am »

  Ah, indeed!  A very good reason and I now remember having that presented to me as an answer before...

Thanks Very Happy
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bbalin

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #20 on: June 07, 2004, 04:53:06 pm »

Quote:

When using any software SRC I've used, I notice a slight collapse of the soundfield along with an overall darkening of the track. The very highest setting in Samp7, for example, yields a narrower soundfield an sounds like you reduced 10kHz by a good .5dB. Not acceptable to me.


I have yet to do these tests myself.
As I do not own 88/96k capable equipment up to date I was limited to bench testing using upsampled or synthetic signals.
What I found out :
When upsampling : Both Wavelab and Samp do create alias tones and do not have a proper passband filter, Wavelab being worse than Samp (in highest quality setting).
On the contrary Cool Edit Pro (now Adobe Audition) performs nearly ideal in this respect.
Incomplete filtering also is a problem when downsampling in Wavelab and Samp, again Cool Edit Pro is yielding smooth results.

As for clipping/overload after SRCing I'm pretty sure that you simply cannot avoid it (depending on the type of source material of course) because the SRC has to deal with the intersample peaks and the shifted sampling points will hit the waveform at some arbitrary point.
As you know using 32bit float format will overcome at least the clipping.
In this respect Cool Edit Pro offers a surprise again : it has a waveform display actually showing the intersample peaks so you can easily watch the potential problems before conversion.
And you could modify the settings of your limiter/maximizer Wink

Boris
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mark4man

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #21 on: June 07, 2004, 08:09:13 pm »

Quote:

In my experience I have always found the conversions from 88.2 to 44.1 more seemless then the 96 to 44.1 conversion. Not much more, but apparent none the less. Is this just in my head/ears or has anybody else found this?

Yes.  88.2 to 44.1 is a direct (even) divisible (not the correct mathematical term...but all I can think of, right now.)  The SRC is optimized for integer scaling with direct multiples.  Far less rounding errors.

Oh..it's Integer Factor.  88.2 to 44.1 is represented by an integer factor of 2:1.

mark4man

BTW - I know my post was relative to 96 to 44.1...but that's because everything our band members give me to produce is 24.96.  They do not have 88.2 capability.
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OatBran

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #22 on: June 07, 2004, 09:06:17 pm »

Thanks mark4man.  That was my understanding as well.  So why is it then that the music industry is still using 96k?   I understand the video aspect and its compatibility, but rarely is that an issue with most/any of the projects I deal with.  Not sure about you guys.  But if 88.2 sounds just as good as good as 96 (or damn close), then why would we use 96k and allow this src to be so nasty?  Should we be using 88.2 unless we know that video might be involved?  Would more mastering engineers prefer that?  Do 9 out of 10 dentists agree?

Thanks.
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bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #23 on: June 08, 2004, 02:22:28 am »

Well, iirc, most modern async SRCs upsample to a very high rate then convert to the lower rate. The benefit of 2:1 ratios only exists in sync converters.

Again, feel free to correct, but I'm pretty the good asyncs can do it equally well...
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Brad Blackwood
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Re: 24/96 to 16/44.1 - Best Method?
« Reply #24 on: June 10, 2004, 10:01:53 pm »

OatBran wrote on Sun, 06 June 2004 22:25

  I have a quick question that applies indirectly to this topic.  In my experience I have always found the conversions from 88.2 to 44.1 more seemless then the 96 to 44.1 conversion.


I think that this points to the algorythms at work and the precision with which they are carried out.  With the Weiss SFC2 I can detect no difference going between any of the 4 sample rates offered, up or down.
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chrisj

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #25 on: June 11, 2004, 03:18:38 pm »

I'm honestly surprised by this thread as well- Brad's way ahead of me in so many things that I'm puzzled. Maybe I can help. Try this:
http://www.mega-nerd.com/SRC/
This algorithm is present in the open source tool 'Audacity'. It's based on an algorithm by Julius O. Smith using (insert geek block HERE) windowed sinc interpolation at an arbitrary quality level- you trade off between speed and quality.
What this means is, I've fooled with a lot of poorer SRC and done things like tested them with a sine sweep from 10K to 26K (at 48K sampling rate) and converted it to 44.1K. You get really LOUD birdies and artifacts if you do that. Normal software SRC is appalling under those circumstances, or if you have particularly nasty abrasive program material as I've sometimes had. The windowed sinc SRC I mentioned above successfully converts the sweep-to-above-20K so well that you still (I still...) can't hear the sine as it goes above about 15K. Any other software SRC (maybe not Barbabatch) and you totally hear lots of loud artifacts the whole time.
The quality settings work so that at low quality you still don't get artifacts, but it starts rolling off at 14-16K. At the very highest quality you can get arbitrarily close to perfect passband up to the Nyquist frequency, and still perfect rejection after that- absolutely no audible artifacts.
The concept is called an emulated conversion of the original data to analogue and then resampling at the new sample rate- which is of course what you're doing now. If this sound library is helpful to you, please let me know! I'd like to think I was able to come up with something to help you out, and I know this algorithm performs fantastically. I'd love to see other software incorporating it (it would have to be open source to incorporate this 'Secret Rabbit Code' library, but again there is already usable OSS software that does, like Audacity)
Good luck! I suspect it should be possible to do this operation in the digital domain as well or better than in analog, because you're not supposed to be adding any color- it's all about rejecting the artifacts, and it's all about finding ways to throw more computational power at the problem. Windowed sinc has that capability, and should solve your problem.

chrisj

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #26 on: June 11, 2004, 03:30:10 pm »

Oh- an interesting side effect of the windowed sinc is that you can generate a sweep at 48K and have it stay fairly audible throughout, due to artifacts that apparently interact with the D/A in some way. Generators aren't always perfect. You see the meters on the power amp light up (if you got 'em) and hear a high pitched sweep that's actually supported by artifacts. Convert with windowed sinc to 44.1K, and the sweep becomes inaudible, meters still light, and if you look at the waveform it's plainly still a sweep, but you've cleaned up artifacts present in the original file. I'm not entirely sure how that works, and doubt it would happen if you weren't downsampling, but it does...

bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #27 on: June 11, 2004, 03:42:24 pm »

chrisj wrote on Fri, 11 June 2004 14:18

I'm honestly surprised by this thread as well- Brad's way ahead of me in so many things that I'm puzzled. Maybe I can help.

Umm, I don't get it - what does this mean?

FWIW, I'm quite sure I'll end up with a hardware async SRC when all is said and done...
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Brad Blackwood
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chrisj

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #28 on: June 11, 2004, 04:10:09 pm »

bblackwood wrote on Fri, 11 June 2004 15:42

chrisj wrote on Fri, 11 June 2004 14:18

I'm honestly surprised by this thread as well- Brad's way ahead of me in so many things that I'm puzzled. Maybe I can help.

Umm, I don't get it - what does this mean?

FWIW, I'm quite sure I'll end up with a hardware async SRC when all is said and done...



Easy- it means 'hey, that's weird- I'm using this SRC that's essentially perfect and Brad doesn't seem to have heard of it'. Maybe it's because I'm hanging around with the OSS programmers where the idea germinated? Some of you guys are treated with such, um, reverence (excepting Mixerman of course) that it's surprising to see you hitting a wall and not having solutions to a thing. At the same time, I totally understand your finding software SRC inadequate, because I've done the tests and even experimented with SRCing through extraordinarily high intermediate sample rates, manually. It doesn't really help, and the windowed sinc does.

If you can get a hardware async SRC, it would be because that is using some type of similar algorithm (unless it's literally got an analog stage, which would be an interesting approach and I'd be fascinated to hear about anybody doing that- an analog stage that was about a half inch of circuit board traces heavily shielded, that would be a heck of a good way to bounce off the analog domain).

Since libsamplerate ('Secret Rabbit Code', as used in Audacity) is _software_ async SRC using essentially a virtual analog intermediate stage and NOT just a very-high-rate intermediate stage, I still think you should check it out. Honestly, it performs very well. Very very well. I am dubious a hardware box could be better sonically, though I'm sure one could be as good.

bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #29 on: June 11, 2004, 04:31:40 pm »

chrisj wrote on Fri, 11 June 2004 15:10

Since libsamplerate ('Secret Rabbit Code', as used in Audacity) is _software_ async SRC using essentially a virtual analog intermediate stage and NOT just a very-high-rate intermediate stage, I still think you should check it out.

What does that mean? The best SRCs I've seen do simply upsample to a very hi fs before downsampling. I might have missed it but what on earth is 'a virtual analog intermediate stage'?
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Brad Blackwood
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chrisj

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #30 on: June 11, 2004, 05:33:44 pm »

bblackwood wrote on Fri, 11 June 2004 16:31

chrisj wrote on Fri, 11 June 2004 15:10

Since libsamplerate ('Secret Rabbit Code', as used in Audacity) is _software_ async SRC using essentially a virtual analog intermediate stage and NOT just a very-high-rate intermediate stage, I still think you should check it out.

What does that mean? The best SRCs I've seen do simply upsample to a very hi fs before downsampling. I might have missed it but what on earth is 'a virtual analog intermediate stage'?


Maybe Zoesch can help with this one if you're serious about wanting to know. There's a reason why libsamplerate gets qualitatively different results than hi-fs converters- if you use a really low quality level what happens is you start losing the sharpness of the brickwall filter, rather than you start getting artifacts. I'll try to explain though... I got it a bit wrong, we're talking about 'periodic sinc interpolation', not windowed sinc interpolation. However it works out to be a window anyway- or sort of like a wavelet.

http://www-ccrma.stanford.edu/~jos/resample/Theory_Ideal_Ban dlimited_Interpolation.html

I don't understand a lot of that math either, but look at the pictures! Very Happy

Essentially, in order to do this, you set up a sinc function that looks like a tiny sine wave with a lot of pre-ring and post-ring. It forms a sort of filter. For every sample, you juxtapose your original waveform's samples against this sinc waveform, and you take the points where it coincides with the source samples and add them all together to produce a composite result that's the convolution of the original file with this funny sinc wave thing.

That works as a filter. Specifically, it works as a really good brickwall filter at the resulting sample rate you're converting to. It also means that the more surrounding samples you use in the convolution, the better accuracy you can get. So to do a super-high-quality version you might be using all the samples from a tenth of a second around the immediate sample- so you'd be doing math on about four thousand samples for every output sample. It gets slow, but that's how it works.

And because you're convolving the input source data against a mathematical construct, the output accuracy isn't limited by an intermediate sample rate at all. There is no intermediate sample rate, just the mathematical shape of the desired waveform at whatever accuracy your computer will support. It becomes strictly a matter of how many adjacent samples you're willing to consider- in other words, CPU load.

This shows some pictures of how it works when your output sample time position is nowhere near the input sample position- http://www-ccrma.stanford.edu/~jos/resample/Implementation.h tml

The mathematical formula for that wave is what moves along with the output sample position, and it sums the intersections of the input samples as they go across the wave. Also note that some of the samples- like the ones right near the point you're working on- are gonna be subtracted, not added! If it's under the horizontal line representing zero for the sinc function, you'll be taking an attenuated version of that sample and subtracting, not adding it. How attenuated? That depends on how far away from the zero line the sample's time position falls- and again, that's a mathematical calculation with NO intermediate sample rate to be concerned with. If you were using a high-resolution math library (like the ones that can give you 100 pages of the value of pi) you could reasonably expect to get billions of times the 'intermediate resolution' of the high-fs converters. In practice, you just get way more 'intermediate resolution'. It's like the difference between 16 bit fixed and 32 bit float- hard to specify exactly how much better the resolution is when you're using a floating-point variable mantissa, but in practice it's 'way better'. Less crunchy.

*chirp* *chirp*

Sorry. Zoesch? Bueller?

Anyway- it's good, mmkay? Very Happy

bblackwood

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Re: 24/96 to 16/44.1 - Best Method?
« Reply #31 on: June 11, 2004, 05:39:59 pm »

Wow, that looks fascinating. Gonna have to read more about this...
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Brad Blackwood
euphonic masters
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