|Vendor's Bazaar Commercial Vendors large & small hawking their wares|
Please consider donating to help us continue to serve you.
Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving
||Thread Tools||Search this Thread|
|17th March 2013, 11:57 AM||#1351|
Join Date: Feb 2006
Thanks for your explanations Bunpei. They are very well put, articulate, clear and easy to understand. Very impressive for a non-native english speaker.
|17th March 2013, 12:06 PM||#1352|
Join Date: Jun 2004
Location: Bath, UK
It seems, with some work, I may therefore be able to make use of the Amanero I2C data after all. If I can get the PIC to receive the data and then decide how best to send it on to the 9018 this could open lots of very useful possibilities, like volume control from the PC, etc.
Seems my project has just got bigger again
|17th March 2013, 12:54 PM||#1353|
Join Date: Aug 2008
Hi, merlin el mago, dickiegeorge and to those who are interested in creating DSD256, DSD512 source files,
I'd like to explain briefly a method to get DSD256, DSD512 files from your own existing sources.
Key messages are;
1. Sound quaility of resultant DSD sources depend on the software you use
Some Japanese audiophiles favor the most complicated and time consuming method.
2. For creating a DSD256 of DSDIFF format by converting an original 44.1 kHz/16 bit PCM WAV file, the most easy way is to use sunacchi's direct conversion program "PCM - DSD Converter V1.5 "
with its GUI front-end program "Wave to Dsdiff Converter - Version1.5 - GUI" by YUKI-san.
The sunacchi's program is based on Koon's pioneering original source,"Wav2DSFconverter01_20111129.cpp" shown here.
3. Essential steps to get DSD256, DSD512 files
A. Upsampling original 44.1 kHz/ 16 bit PCM WAV source to 176.4 kHz or 352.8 kHz/ 24 bit PCM WAV
(If you use original WAV files of 176.4 kHz or 352.8 kHz, you can omit this step.)
B. Re-write a sampling frequency dependent field values in a WAV header to those of 44.1 kHz
C. Apply delta-sigma modulation to the 44.1 kHz file by using Korg AudioGate, Sony DSD Direct, Weiss SARACON etc. [WAV-> DSDIFF or DSF file conversion]
D. Re-write a sampling frequency field value of DSDIFF header to that of DSD256 or DSD512
For assisting these steps, sunacchi provides a set of useful tools.
chfsWav441.exe --- drag & drop capable program for B
chfsDff256.exe --- drag & drop capable program for D (DSD256 only)
Bunpei uses FUSE program for initial upsampling, SONY DSD Direct for delta-sigma modulation and Korg Audio Gate for DSF -> DSDIFF format conversion. When DSD64 or DSD128 sources are to be upsampled, Bunpei uses Korg AudioGate for initial DSD -> PCM conversion.
The number of taps in FIR employed for initial upsampling has a significant effect on sound quality of final DSD sources.
For 48kHz series data, a direct application of the tools explained above is not effective.
sunacchi's "wav2dff" program terminates at the middle of processing when a upsampled data clips. In the case, please adjust the offset parameter to remove the clipping.
The options in the sunacchi's "wav2dff" program are associated with internal parameters.
PCM to DSD conversion consists from two major steps.
i. Upsampling from 44.1kHz to 11.2896MHz (x 256)
Zero insertion and low pass filtering by a FIR digital filter
ii. Delta-Sigma modulation
A. Tap counts
The number of taps used in the FIR digital filter
In general, the bigger number brings the better sounds. However, the longer processing time.
B. Kaiser window alpha value
The digital LPF uses a "window" function (the same term that appears in DFT(Discrete Fourier Transform))
Kaiser type window has a characteristic parameter "alpha" which determines a shape of window.
C. Normalization offset
In the FIR calculation for x 256 upsampling, a default normalization factor is 256. However, when an input sound data has a "clipping", it causes abnormality in the calculation. In order to avoid the error, an "offset" value for adjustment was introduced by Sunacchi as a work-around.
If your source contains clippings, you need to set a certain negative value, for example, -5 for this option. In this example, the effective normalization factor is 256+(-5)=251.
D. Pattern number for control parameter set
A noise shaper is a digital control circuit with a feedback. A z-transform notation is used for defining its transfer function. The noise shaper employed in the program is of 7th order. In general, the stability of the function can be estimated by the positions of "poles" and "zeros" in a Z-plane.
In this program, values of parameter zero are predefined in several sets. The pattern number is used to select one of the set.
Bunpei, sunacchi and YUKI-SAN are not responsible for the resultant damaged or lost data you may suffer in your processing. In order to avoid those unexpected damages, please keep backup of your original data or please apply programs introduced above to your copy data.
Sound quality of resultant sources heavily depend on your PC and DAC environment. Bunpei gives no guarantee for your satisfaction.
Last edited by Bunpei; 17th March 2013 at 01:19 PM.
|17th March 2013, 11:24 PM||#1357|
Join Date: Aug 2008
|17th March 2013, 11:57 PM||#1358|
Join Date: Jan 2011
Location: Madrid - Spain
|Thread Tools||Search this Thread|
|Thread||Thread Starter||Forum||Replies||Last Post|
|HLLY USB SPDIF/ I2S CONVERTER||kp93300||Digital Source||12||29th January 2012 02:55 PM|
|exaU2I - 32bit/384kHz Multi-Channel Asynchronous USB to I2S Interface||exa065||exaDevices||0||11th November 2011 11:41 AM|
|$40 USB to I2S converter||sharpi31||Digital Line Level||7||12th July 2009 04:09 PM|
|New To Site?||Need Help?|