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Old 13th May 2012, 10:52 PM   #91
Bunpei is offline Bunpei  Japan
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Playing DSD DFF files of 48 kHz series is implemented on Chiaki's SDTrans384.
I felt the 48 kHz series DSD is more natural when original sources are of 48 kHz series and converted into DSD.
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Old 4th August 2012, 01:22 PM   #92
Bunpei is offline Bunpei  Japan
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Default A program for direct conversion from PCM 44.1kHz/16bit WAV to DSD256 DFF

Based on the source codes released by Koon, Sunacchi, a Japanese user of SDTrans added his coding that makes direct conversion from PCM 44.1kHz/16bit WAV to DSD256 DFF.
His program can be downloaded from here http://ama-audio.up.seesaa.net/image/wav2dff-5c465.zip.
No major modifications are made to Koon's original algorithm.

----------------------------------------------
Wave to Dsdiff Converter - Version1.3
----------------------------------------------
1. Command line example;
> wav2dff "c:\data\aaa.wav" /DSD256 /t:32768

2. Usage;
wav2dff.exe infile <outfile> </DSDrate> </t:nnnnn> </k:mm> </p:5>

Option parameters are,
Rate of DSD: /DSD64(default) or /DSD128 or /DSD256
Number of taps in FIR: /t:2048(default) or /t:4096 or /t:8192 or /t:16384 or /t:32768
Kaiser Window Alpha Value: /k:4 .. /k:16 /k:10(default)
Noise Shaper Zero parameter set pattern number: /p:1 .. /p:5 /p:2(default)
/p:1 NoiseShaper param Zero1=0.94, Zero2=0.73, Zero3=0.41
/p:2 NoiseShaper param Zero1=0.95, Zero2=0.74, Zero3=0.42
/p:3 NoiseShaper param Zero1=0.96, Zero2=0.76, Zero3=0.43
/p:4 NoiseShaper param Zero1=0.97, Zero2=0.77, Zero3=0.43
/p:5 NoiseShaper param Zero1=0.98, Zero2=0.78, Zero3=0.44

Sunacchi and Bunpei would like to express our sincere gratitude to Koon because he generously releases his valuable source code to public. Especially his source code helped us to have a clear understanding how a DSD signal is practically generated through a Delta-Sigma modulation technology as well as an upsampling pre-processing using a FIR.
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Old 4th August 2012, 02:12 PM   #93
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Thanks Bunpei, Sunachi & Koon, can't execute the programm, how have to do?
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Old 4th August 2012, 02:14 PM   #94
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Hi Bunpei, thanks to read and upgrade my terrible source code
In my PCM2DSD source code, FIR is processed by simple C routine, but I also have CUDA and AVX or SSE FIR converter.
Please see https://sites.google.com/site/koonau...fir-vst-plugin for SSE and AVX
https://sites.google.com/site/koonau...hannel-divider for CUDA
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Old 4th August 2012, 03:05 PM   #95
Bunpei is offline Bunpei  Japan
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Quote:
Originally Posted by merlin el mago View Post
... can't execute the programm, how have to do?
Oh, you are right.
I ran the program on my Windows 7 64bit Edition PC. The program terminated without any error message during its execution.
I'm now checking it on my Windows XP 32bit slow notebook PC. It will take more time to get a result.
Would you tell me your execution environment?

I will ask Sunacchi to find a work around.
I appreciate your patience.

Bunpei
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Old 4th August 2012, 05:43 PM   #96
Bunpei is offline Bunpei  Japan
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I found a possible cause of the error.

Please add "/r" option in your command line, for example,
> wav2dff "c:\data\aaa.wav" /r /DSD256 /t:32768
and please execute it again.

I omitted options "/r" and "/n:xx" in my explanation of the previous post because I think they were so complicated. But now, I need to explain these.

When the program encounters a "clipping" event, it recognizes the audio level value as erroneous and terminates the processing. That might be why you couldn't get any resulting file.

The program has an internal value named as "Normalization factor" which initial value is proportional to a given DSD sampling rate. The factor is used to re-adjusting the audio level values.

If you specify the "/r" option, the program automatically try to re-generates a DSD output file by decreasing the Normalization factor value by 1 at the occurrence of the first clipping. In the case that the program finds a new clipping again, it will try processing again by decreasing the Normalization factor by 1. The iteration limit is 10.
If you can not get any result even within 10 decrements, you need to specify a larger decrement value by using the option "/n:-xx", for example, "/n:-16". (The value "-16" is named as "Normalization Offset" by Sunnachi.)

Bunpei

Last edited by Bunpei; 4th August 2012 at 05:48 PM.
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Old 5th August 2012, 05:46 PM   #97
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Quote:
Originally Posted by Bunpei View Post
Oh, you are right.
I ran the program on my Windows 7 64bit Edition PC. The program terminated without any error message during its execution.
I'm now checking it on my Windows XP 32bit slow notebook PC. It will take more time to get a result.
Would you tell me your execution environment?

I will ask Sunacchi to find a work around.
I appreciate your patience.

Bunpei
Thanks Bunpei, I run Windows XP & Windows Vista

Felipe
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Old 5th August 2012, 11:24 PM   #98
Bunpei is offline Bunpei  Japan
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Hi, Felipe,

Do you still have the problem?
When you have any, would you tell me your command line and screen dump images?
What kind input WAV files did you supply to the program?

Bunpei
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Old 6th August 2012, 11:59 AM   #99
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Hi Bunpei

I still can't run the programm. I'm here:

C:\Documents & Settings\usuari>

How can do to execute the programm for windows XP?

Cheers

Felipe
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Old 6th August 2012, 04:19 PM   #100
Bunpei is offline Bunpei  Japan
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Quote:
Originally Posted by merlin el mago View Post
I'm here:

C:\Documents & Settings\usuari>

How can do to execute the programm for windows XP?
Hi, Felipe,

You seems to be unfamiliar with running programs on PC.

First, this is the easiest way for checking.
Please put wav2dff.exe program in the zip file in your appropriate folder.
(Please tell the folder name to me later.)
You will be able to find its icon in a Windows Explorer window.
Please drag your wav file and drop it on the icon.
A command prompt screen will appear automatically and you will be able to read some information on the screen.
The converted output file will be created in the same folder of your wav file.

Bunpei
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