Weiss Saracon

fas42

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6. I know some SACD's are recorded pretty hot and any conversion to PCM will be clipped. I'd suggest keep it at 0 and if you run into any clipped files, reduce 3dB.
I don't follow this, Bruce, why would they clip? Of interest, I'm currently heavily investigating the possibilities of digital rescue, and I'm finding some of the gross "distortion" mastered into releases over the last few years quite amazing: the industry has dug itself into quite a mess ...

Frank
 

billh

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1. No dithering
2. No Quantization
3. I'm not familiar with "Hearing curve", but try each to see which one is better for you.
4. I'd use 176.4
5. 24-bit
6. I know some SACD's are recorded pretty hot and any conversion to PCM will be clipped. I'd suggest keep it at 0 and if you run into any clipped files, reduce 3dB.

Don't worry about blowing out tweeters with UHF noise. Scarlett book specs limits the UHF noise of 20-50kHz and 50-100kHz when they're put out for production.

Great - Thank you very much!:)

I saw somewhere else that I should convert them at -6dB so I'll play around with that one - I guess I can play them back on foobar and use the meter to see if they are peaking.

Again - Thanks!
 

fas42

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Full modulation of a DSD signal is approx +6dB.
Okay, got a handle on it, it's a definition thing: 0dB on PCM is full modulation, 100% voltage swing; 0dB on DSD has been specified to match a 50% voltage swing, probably as a convenience to sound engineers. I note that converters of DSD to PCM have simple switches, so that the 0dB levels of the two media can be set to match ...

Frank
 

billh

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Is there any way to compare a file converted from DSD to PCM (wav, aif, flac) using Audiogate, Saracon or Pyramix to the original DSD source (other than by ear)? Just wondering how one can be sure the resultant converted files are exact duplications.
 

Bruce B

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Is there any way to compare a file converted from DSD to PCM (wav, aif, flac) using Audiogate, Saracon or Pyramix to the original DSD source (other than by ear)? Just wondering how one can be sure the resultant converted files are exact duplications.

Of course... you can either do a null test or just line up the PCM files down to the sample and check several places in the file if the samples are still aligned.
 

billh

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Of course... you can either do a null test or just line up the PCM files down to the sample and check several places in the file if the samples are still aligned.

Could I also use a program like Audacity to accurately compare 2 music files?
 

fas42

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There's a simple technique that Audacity and such programs can do: first make sure that they are time aligned, correct to a sample, or digital point, then invert one file and mix the two together to create a third track or file. With this file, no matter how much you amplify it, it should always be pure silence ...

Frank
 

billh

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OK - I downloaded both Audacity and the trial version of Soundforge Pro and it looks like neither one can handle anything over 96000 hz so I guess I am going to have to make my experiments with 96000 hz sample files and just assume if they are OK (at 96000 hz) then ones created and higher sample rates (176, 192) will be OK too?

EDIT: I just realized Soundforge can handle the higher sample rates but Audacity doesn't look like it can as it is having trouble recognizing the file type for anything higher than 96000. So I am good to go.
 
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billh

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Sorry to keep flogging this thing but I see how to get a graph of the FLAC, PCM or WAV file but how to I get the snapshot of the original DSD file to compare against?

Sorry to be so dense - I'm sure you can tell I am a noob at this.
 

Bruce B

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Sorry to keep flogging this thing but I see how to get a graph of the FLAC, PCM or WAV file but how to I get the snapshot of the original DSD file to compare against?

Sorry to be so dense - I'm sure you can tell I am a noob at this.

You'll need a DSD editor like Sonoma or Pyramix.
 

fas42

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OK - I downloaded both Audacity and the trial version of Soundforge Pro and it looks like neither one can handle anything over 96000 hz so I guess I am going to have to make my experiments with 96000 hz sample files and just assume if they are OK (at 96000 hz) then ones created and higher sample rates (176, 192) will be OK too?

EDIT: I just realized Soundforge can handle the higher sample rates but Audacity doesn't look like it can as it is having trouble recognizing the file type for anything higher than 96000. So I am good to go.
Not sure what your trouble with Audacity is, I have no trouble looking at higher sampling rate files ...

Frank
 

fas42

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Sorry to keep flogging this thing but I see how to get a graph of the FLAC, PCM or WAV file but how to I get the snapshot of the original DSD file to compare against?

Sorry to be so dense - I'm sure you can tell I am a noob at this.
My understanding is that to actually look at DSD in a meaningful way, it effectively has to be converted back to a PCM format, which I imagine these high end editors do. DSD in itself is pure two state binary, a massively long sequence of 0's and 1's, which is just gibberish to look at. It is somewhat illuminating to think that DSD is ultimately nothing more than a simple on/off switch being flicked back and forth at a ferocious rate ...

Frank
 

billh

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My understanding is that to actually look at DSD in a meaningful way, it effectively has to be converted back to a PCM format, which I imagine these high end editors do. DSD in itself is pure two state binary, a massively long sequence of 0's and 1's, which is just gibberish to look at. It is somewhat illuminating to think that DSD is ultimately nothing more than a simple on/off switch being flicked back and forth at a ferocious rate ...

Frank

I guess what I am looking for is to just make sure I get the best DSD to PCM transfer possible. It sounds like any of the programs mentioned already should be capable of doing this without any loss of fidelity (Philips Audio Format Converter, Audiogate, Saracon, Pyramix and maybe even Foobar).

I gather the conversion is simple bits and math so I will do some experimenting to see which one sounds best. Saracon and Pyramix are for pros and I really can't spend that kind of money on something I might not use much especially if the Audiogate and Foobar converters do as good a job for a basic DSD to PCM conversion.
 

Bruce B

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I gather the conversion is simple bits and math so I will do some experimenting to see which one sounds best. Saracon and Pyramix are for pros and I really can't spend that kind of money on something I might not use much especially if the Audiogate and Foobar converters do as good a job for a basic DSD to PCM conversion.

The conversion is simple math but what the converters do with those bits are the problem. Some pass the information along unharmed and some just don't sound good to me at all. What I don't like about Saracon is the steep filter. At least you have several filter setting in AudioGate and Pyramix. Another DSD SRC I forgot about is DiscWelder.
 

billh

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So, it sounds like you are OK with Audiogate - what do you suggest for multichannel? I have an older copy of the Philips AFC software I think may work with multichannel but haven't played around with it yet.

All I'm going to be doing is converting my SACD collection to FLAC so I can stream in my home via an NAS hard drive and Twonky (all my CD's and DVD-A's are already done). Working on my SACD's now and then will move to my vinyl favorites.

Thanks again for all the help.
 

billh

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1. No dithering
2. No Quantization
3. I'm not familiar with "Hearing curve", but try each to see which one is better for you.
4. I'd use 176.4
5. 24-bit
6. I know some SACD's are recorded pretty hot and any conversion to PCM will be clipped. I'd suggest keep it at 0 and if you run into any clipped files, reduce 3dB.

Don't worry about blowing out tweeters with UHF noise. Scarlett book specs limits the UHF noise of 20-50kHz and 50-100kHz when they're put out for production.

Bruce,

Getting back to the Quantization - my choices are TRUNCATATION or ROUNDING - rounding sounded better to me so that is what I used but what is the difference between the 2? Am I right to use rounding?

Thanks.

Bill
 

Bruce B

WBF Founding Member, Pro Audio Production Member
Apr 25, 2010
7,006
512
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Snohomish, WA
www.pugetsoundstudios.com
Bruce,

Getting back to the Quantization - my choices are TRUNCATATION or ROUNDING - rounding sounded better to me so that is what I used but what is the difference between the 2? Am I right to use rounding?
Thanks.
Bill

Well if it sounds better to you, then by all means use it. Truncation is just cutting off or dropping. Rounding is using approximation to fill in.
 

billh

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Well if it sounds better to you, then by all means use it. Truncation is just cutting off or dropping. Rounding is using approximation to fill in.

Hehe.........I haven't actually listened to it. When I said it sounded better I meant the "term" ROUNDING sounded better than the term TRUNCATION. Which would you recommend? I am just getting started - looks like a long weekend. This is a slow process, especially the multichannel discs.
 

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