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DSD Playback system, DSF Player + USB DDC + DSD Amplifier
DSD Playback system, DSF Player + USB DDC + DSD Amplifier
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Old 20th November 2011, 01:24 PM   #11
SoNic_real_one is offline SoNic_real_one  United States
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Nobody "commends" him what to do.
This is a forum and the purpose of it is the discussion, sharing ideeas and experiences. Blogs are more appropiate for talking to yourself.

Back to forum mode:
I think that if you don't have a feedback loop, and stay purelly digital, you need voltage switching. Curent sources are analogic only in my experience.
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Old 20th November 2011, 07:54 PM   #12
SOUND_Rays is offline SOUND_Rays  Slovenia
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Quote:
Originally Posted by SoNic_real_one View Post
Nobody "commends" him what to do.
@ SoNic_real_one

Couldn't more than 100% agree with You

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Old 20th November 2011, 09:30 PM   #13
KOON3876 is offline KOON3876
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Sorry I'm just scribbling what I'm interested / I made

Dead_Time control, should be something like attached?
drive signal = function(input, input_delay1, input_delay2)
(1) High drive signal should have same length, as pulse width
(2) Low drive signal has narrow on time than High
(3) each signal should pass through same number of logic gate

Voltage_switching: now looking signals, with modified R-C and Schottky diode between D-S. I hate switching noise
1 bit Current Push-pull, if I try: switching was very smooth, I remember. with my understanding, this is not analogic because.. each FET does not know "analog signal voltage" itself. FET/driver only knows current bit 1/0.
NFB: there are no feedback for signal, (to add, I need 1 bit A/D) so Power supply must have feedback.
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File Type: png Dead-Time-control.png (31.6 KB, 486 views)
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Old 21st November 2011, 12:15 AM   #14
KOON3876 is offline KOON3876
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Default PCM to DSD conversion?

Now I want to understand DSD,
I made x64 upsampler, so now I get 2.822MHz float PCM.
(Image, Right, shows 1kHz sign wave - 64 upsampled - 22.5k FIR LPFed)

next I need to apply 7th order noise shaper.
I could design H(z) = A(z)/B(z) as attached image (left).
here is Scilab command.

z=%z;
H=(1-6.996482471749818*z^(-1)+20.982415870011799*z^(-2)-34.964835250398139*z^(-3)+34.964835250398139*z^(-4)-20.982415870011799*z^(-5)+6.996482471749818*z^(-6)-1.000000000000000*z^(-7));
D=(1-5.250000000000000*z^(-1)+11.812500000000000*z^(-2)-14.765625000000000*z^(-3)+11.074218750000000*z^(-4)-4.983398437500000*z^(-5)+1.245849609375000*z^(-6)-0.133483886718750*z^(-7));
sys=syslin("d",H/D);
clf();
f1=scf(0);
plzr(sys)
f1=scf(1);
gainplot(sys,1e-3,1e-1,0.001)

Now I'm wondering how to program this H as IIR filter.
anyone know how to program IIR in DSP?
Attached Images
File Type: png Scilab_simulation.png (35.3 KB, 482 views)
File Type: png WAV2DSF_Oversampling.png (20.3 KB, 464 views)
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Old 21st November 2011, 02:18 AM   #15
KOON3876 is offline KOON3876
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Attachment, white noise maker
sorry, I'm trying to convert PCM to DSD.

(1) read wave file to memory
(2) prepare DSD header
(3) prepare FIR filter
(4) prepare Shaper parameter NG?
Loop
(5) oversample x64
(6) apply FIR LPF
(7) apply 7th noise shaper IIR NG?
(8) pack to DSD, write to file
Loopend

If this program run, I can convert PCM to DSD anytime.
(realtime, in the ASIO driver, or convert wav to DSD through x64 float upsampling)
Attached Files
File Type: txt Wav2DSFconverter01(2).cpp.txt (19.1 KB, 51 views)

Last edited by KOON3876; 21st November 2011 at 02:26 AM.
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Old 21st November 2011, 03:10 AM   #16
SoNic_real_one is offline SoNic_real_one  United States
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For DSD programming - maybe you can talk to kode54?
For dead time - now you are showing a bridge...
Anyway, the timings for B side can be the same like the ones on the A side, no need for extra delay there (you have the load between A and B, it won't "short" the supply).
Analog timings are instable, you would have to give a big safety margin. I would delay them in software.

Last edited by SoNic_real_one; 21st November 2011 at 03:17 AM.
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Old 21st November 2011, 04:12 AM   #17
KOON3876 is offline KOON3876
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Hi Sonic
Yes that's right, I visited HidrogenAudio now. I hope Kobe54 knows how to for PCM to DSD.

timing control by digital clock.. maybe 10ns + / - shift, by another 100MHz, inside FPGA.
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Old 23rd November 2011, 03:45 PM   #18
KOON3876 is offline KOON3876
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I was told some parameter and how to implement 7th order noise shaper on the book, and it is working. But still I can not understand how these parameters were calculated

Green dot: oversampled (0 inserted) 1kHz signal
Red line: 22k LPF passed, this is 2.8MHz sampling rate signal
Then (not on the graph, DSD signal = +1/-1)
Blue line: DSD signal passed LPF again. 1kHz signal reproduced. you can see FIR is starting around 1000 sample point.

10 seconds of WAV file is converted to DSD within around 5 seconds.

I will keep studying, but so far, I can produce DSD stream in realtime from PCM. it means I can modify my ASIO driver to produce DSD stream, to my 1bit amp. interesting I don't need any DAC chip, TI's amp, only need DSD byte stream and FET driver.
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File Type: png Input-OverSample-DSD-Reversed.png (23.9 KB, 439 views)
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Old 23rd November 2011, 07:41 PM   #19
SoNic_real_one is offline SoNic_real_one  United States
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Well... nice! That's how people get rich
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Old 23rd November 2011, 08:25 PM   #20
KOON3876 is offline KOON3876
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Now I'm studying more about noise shaper.

SoNic, "no where to tweak" means "that is the best" for me.. but it also means "nothing to DIY" omg.

I'm wondering Tube Vacuum amplifier which can accept Direct DSD signal.
Tube can work up to 10GHz, no problem at all for DSD 2.8MHz or 5.6MHz.
How do you think??
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