I have some DSD music file from SACD now. I want to listen to them as it is.
So I made prototype of full DSD playback system.
(1) DSD File player (source, DSF hex dump screenshot)
Read DSF format DSD file into memory, make byte(= 8ch) stream, send to USB DSD digital-digital converter.
Now 1 Byte data contains 2 bit (L,R) only, but for future I want to playback 5.1 DSD. so I use 1 byte for 2 bit sample.
(2) DSD DDC (Photo, Schematic, VHDL) Right side of the photo
This is FTDI UM232H + FIFO + FPGA, to make +- output to DSD amplifier.
Made for USB - I2S x 24ch (see "Digital Line" forum).
FPGA is too large for DSD.. only 27 FFs are used.
These (1) (2) are just making bit stream to the (3).
(3) DSD amplifier (Photo, Schematic, Signal Photo) Left side of the photo
MOSFET Driver + MOSFET, 1bit amplifier. just a High Speed H bridge and LPF.
Volume is controlled by Final stage voltage. (0.0 - 6V)
Note: I'm using LOW side driver so Final stage voltage is limited, driver - 5V.
If you use High/Low side driver, you can apply any power.
It sounds. 🙂
Byte stream, which contains DSD signal, and H bridge + LPF can playback the Audio (as DSD theory is)
I have no experience with SHARP 1 Bit amplifier. This one sounds like SHARP? I hope so.
So I made prototype of full DSD playback system.
(1) DSD File player (source, DSF hex dump screenshot)
Read DSF format DSD file into memory, make byte(= 8ch) stream, send to USB DSD digital-digital converter.
Now 1 Byte data contains 2 bit (L,R) only, but for future I want to playback 5.1 DSD. so I use 1 byte for 2 bit sample.
(2) DSD DDC (Photo, Schematic, VHDL) Right side of the photo
This is FTDI UM232H + FIFO + FPGA, to make +- output to DSD amplifier.
Made for USB - I2S x 24ch (see "Digital Line" forum).
FPGA is too large for DSD.. only 27 FFs are used.
These (1) (2) are just making bit stream to the (3).
(3) DSD amplifier (Photo, Schematic, Signal Photo) Left side of the photo
MOSFET Driver + MOSFET, 1bit amplifier. just a High Speed H bridge and LPF.
Volume is controlled by Final stage voltage. (0.0 - 6V)
Note: I'm using LOW side driver so Final stage voltage is limited, driver - 5V.
If you use High/Low side driver, you can apply any power.
It sounds. 🙂
Byte stream, which contains DSD signal, and H bridge + LPF can playback the Audio (as DSD theory is)
I have no experience with SHARP 1 Bit amplifier. This one sounds like SHARP? I hope so.
Attachments
Hi Sonic
You mean Dead-time control? Yes I selected the driver without timing control.
I loss some power (10ns/300ns or so) / I get correct pulse width.
dead time control reduces on-width of pulse = distortion.
same as analog amp, we need some power loss = idle current for quality.
This is prototype so I want to know which driver is the best, or thinking to design adequate driver circuit (for audio) by myself.
You mean Dead-time control? Yes I selected the driver without timing control.
I loss some power (10ns/300ns or so) / I get correct pulse width.
dead time control reduces on-width of pulse = distortion.
same as analog amp, we need some power loss = idle current for quality.
This is prototype so I want to know which driver is the best, or thinking to design adequate driver circuit (for audio) by myself.
IMO with MOS like IRF you get less distortion with a small dead-time. You won't get anyway a "clean" transition there, it's still at zero, being a short-circuit between rails. Reduced voltage dip on rails at zero crossing and lower temperature on finals will help lowering the distortions.
How small can that time be... is up to how fast are those finals.
Sorry, ideea is good, but I found out long time ago that the final stage needs to be analogic. At least this is the state of electronic devices in this decade. 0.2% is nothing to care. Add one more zero and we are talking replacing the analog final stage.
How small can that time be... is up to how fast are those finals.
Sorry, ideea is good, but I found out long time ago that the final stage needs to be analogic. At least this is the state of electronic devices in this decade. 0.2% is nothing to care. Add one more zero and we are talking replacing the analog final stage.
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Sonic,
0.2% is including cheap Pioneer 3 inch full range speaker and Samson C01U USB mic.
I crushed my USB audio interface🙁
How fast = how small FET, I think this is too large. I just need 4.5V for 6 feet listening.
maybe I need to try more smaller FET / my own fast driver.
0.2% is including cheap Pioneer 3 inch full range speaker and Samson C01U USB mic.
I crushed my USB audio interface🙁
How fast = how small FET, I think this is too large. I just need 4.5V for 6 feet listening.
maybe I need to try more smaller FET / my own fast driver.
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As I said, the ideea is good.
Dead-time will help IMO, for reasons exposed above.
But modern DAC's have an easyer job to convert DSD to analog (than PCM) because of the fact that they are Delta-Sigma in essence. So they are pretty good at it.
Dead-time will help IMO, for reasons exposed above.
But modern DAC's have an easyer job to convert DSD to analog (than PCM) because of the fact that they are Delta-Sigma in essence. So they are pretty good at it.
Sonic,
That is just a difference of our tastes, I want to "reduce".. the DAC chip, I/V, opamp, volume, JFET input, NFB, driver stage and Final stage and emitter register. and multiple toroidal and huge capacitors..
less analog is better for me.
for another idea,
https://sites.google.com/site/koonaudioprojects/power-dac
I once made Power DAC (PCM to current, to drive speaker) before.
It was PCM so bit accuracy was weak point, but by DSD I just need 1 current source / 1 current sink, big enough to drive speaker.
Which one do you think better for audio?
switching voltage by DSD, or current push / pull by DSD?
That is just a difference of our tastes, I want to "reduce".. the DAC chip, I/V, opamp, volume, JFET input, NFB, driver stage and Final stage and emitter register. and multiple toroidal and huge capacitors..
less analog is better for me.
for another idea,
https://sites.google.com/site/koonaudioprojects/power-dac
I once made Power DAC (PCM to current, to drive speaker) before.
It was PCM so bit accuracy was weak point, but by DSD I just need 1 current source / 1 current sink, big enough to drive speaker.
Which one do you think better for audio?
switching voltage by DSD, or current push / pull by DSD?
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It is an endless topic to discuss since taste of sound is different to human ears. For hobbyist, I recommend to go ahead your preference and regardless any diyers with opposite commend. For commercial interest, you build audio gears as per customer’s comment on taste of sound and independent of your taste of sound. For this reason, you shall feel happy if select your preference in hobbyist or make profit in commercial interest. It is the end of my philosophy for DIY sharing.
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.
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.
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.
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.
Attachments
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? 😕
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? 😕
Attachments
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)
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)
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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.
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.
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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.
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.
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.
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.
Attachments
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??
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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