I don't think power vacuum tubes would provide any advantage, sice you are using them as switches only. They are horrible when driven to saturation.
IMO, the DSD analog filter should be around 50kHz, slow slope... No need for 10GHz 🙂
IMO, the DSD analog filter should be around 50kHz, slow slope... No need for 10GHz 🙂
Hi Sonic, you noticed me that "Why it have to be Switch?". You are right.
I'm thinking dead_time control and I don't like dead_time control idea, it's not for the audio. 🙁
Maybe I can use good old analog circuit,
(a) logic out, volume(pulse height changed between 0V - 3.3V), DC cut C,
(b1) fixed gain 1Tr amp, diamond buffer Emitter Follower handles pulse, then + LPF.
(b2) Vacuum amp(no NFB) handles pulse, then it's output transformer is natural LPF.
By the way attached original 1kHz wav file "C:\TEMP\1KHz.wav"
and I converted to 1kHz.DSF by my program,
Then convert DSF back to WAV by KORG AudioGate. "C:\TEMP\KORG\1kHz.wav"
very little noise (below -110dB) was added, but should be OK.
and attached current source (dirty but anyway it works)
I'm thinking dead_time control and I don't like dead_time control idea, it's not for the audio. 🙁
Maybe I can use good old analog circuit,
(a) logic out, volume(pulse height changed between 0V - 3.3V), DC cut C,
(b1) fixed gain 1Tr amp, diamond buffer Emitter Follower handles pulse, then + LPF.
(b2) Vacuum amp(no NFB) handles pulse, then it's output transformer is natural LPF.
By the way attached original 1kHz wav file "C:\TEMP\1KHz.wav"
and I converted to 1kHz.DSF by my program,
Then convert DSF back to WAV by KORG AudioGate. "C:\TEMP\KORG\1kHz.wav"
very little noise (below -110dB) was added, but should be OK.
and attached current source (dirty but anyway it works)
Attachments
Analog power stage was my suggestion too...
But if you can switch MOS-FETs (and feed from power rails) at at least 2.8MHz, your aproach is valid.
But if you can switch MOS-FETs (and feed from power rails) at at least 2.8MHz, your aproach is valid.
Please forget Post #22. that is because of 16 bit limit. (final digit error)
I made 44.1/24bit 1kHz Wav file
converted it to DSF format by my program
converted back to 44.1/24 WAV by KORG AudioGate
and there are no noise over -140dB. 🙂
I wonder my mistake.. I checked final WAV, and there was KORG Tag. OK I'm not looking same file.
0079142c|4c 49 53 54 46 00 00 00 49 4e 46 4f 49 43 52 44 | LISTF...INFOICRD
0079143c|0c 00 00 00 32 30 31 31 2d 31 31 2d 32 33 00 00 | ....2011-11-23..
0079144c|49 53 46 54 26 00 00 00 4b 4f 52 47 20 41 75 64 | ISFT&...KORG Aud
0079145c|69 6f 47 61 74 65 20 76 65 72 2e 32 2e 32 2e 31 | ioGate ver.2.2.1
0079146c|20 28 57 69 6e 64 6f 77 73 20 37 29 00 00 | (Windows 7)..
now I have nice quality PCM to DSD converter, I can add this function to my another programs.
(VST for crossover, ASIO driver, etc)
I made 44.1/24bit 1kHz Wav file
converted it to DSF format by my program
converted back to 44.1/24 WAV by KORG AudioGate
and there are no noise over -140dB. 🙂
I wonder my mistake.. I checked final WAV, and there was KORG Tag. OK I'm not looking same file.
0079142c|4c 49 53 54 46 00 00 00 49 4e 46 4f 49 43 52 44 | LISTF...INFOICRD
0079143c|0c 00 00 00 32 30 31 31 2d 31 31 2d 32 33 00 00 | ....2011-11-23..
0079144c|49 53 46 54 26 00 00 00 4b 4f 52 47 20 41 75 64 | ISFT&...KORG Aud
0079145c|69 6f 47 61 74 65 20 76 65 72 2e 32 2e 32 2e 31 | ioGate ver.2.2.1
0079146c|20 28 57 69 6e 64 6f 77 73 20 37 29 00 00 | (Windows 7)..
now I have nice quality PCM to DSD converter, I can add this function to my another programs.
(VST for crossover, ASIO driver, etc)
Attachments
That's more like it! Try a wav file that is higher in frequency, since DSD it has less capability there. A 22kHz audio sampled with 48 or 96 would be sufficient.
PS: 144 dB is the theoretical limit of PCM 24 bit resolution. No audio DAC on market comes close of that.
PS: 144 dB is the theoretical limit of PCM 24 bit resolution. No audio DAC on market comes close of that.
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So, now we have over 140dB (DAC - LPF)? DSD only needs LPF to get back to analog.
you just need low noise LPF 🙂
you just need low noise LPF 🙂
OK 20kHz 24bit WAV -(mine)> DSD -(KORG)> 24bit WAV, still there are no noise > -140dB.
48/96 is not natural for DSD.. I can not convert them linearly.
Attachments
Yes, but you will see them in "nature" 😀
Can you extend the graph past 22kHz or is a brickwall there in the Korg?
Can you extend the graph past 22kHz or is a brickwall there in the Korg?
Hi, I can not see over 22050, by WaveSpectra.
if I convert 88200/24bit wav, I will see noise curve attached "DSD_noise_Shaper.png".
(my program only handles 44100 now)
I set Zero point at 0.95, 0.74, 0.42 = 20947Hz, 16317Hz, 9261Hz.
These points and noise_floor are trade-off. 😱
(noise lowered dB * lowered freq area) < (noise shaped dB * shaped freq area)
oh, KORG can handle 88/24. I tested 1kHz-88-24.wav to dsf to wav.
it shows -160dB range.
-101dB: 30000Hz
-88dB: 44000Hz
So, If I use LPF cutoff at 25000, 4th ButterWorth,
Then 44000Hz noise goes to -88dB - 10log(1+(44000/25000)^8) = -107dB.
maybe it's OK.
anyway I can not notice over 18500🙁
if I convert 88200/24bit wav, I will see noise curve attached "DSD_noise_Shaper.png".
(my program only handles 44100 now)
I set Zero point at 0.95, 0.74, 0.42 = 20947Hz, 16317Hz, 9261Hz.
These points and noise_floor are trade-off. 😱
(noise lowered dB * lowered freq area) < (noise shaped dB * shaped freq area)
oh, KORG can handle 88/24. I tested 1kHz-88-24.wav to dsf to wav.
it shows -160dB range.
-101dB: 30000Hz
-88dB: 44000Hz
So, If I use LPF cutoff at 25000, 4th ButterWorth,
Then 44000Hz noise goes to -88dB - 10log(1+(44000/25000)^8) = -107dB.
maybe it's OK.
anyway I can not notice over 18500🙁
Attachments
That should be OK, it's on par with what I know DSD can do...
I don't hear (now) over 18k, but I can tell if the 2nd and 3rd harmonics are "wrong" - to fundamantals in 10-14 kHz range. Tube amplifiers drive me on walls with that, even at medium volumes.
Young people might hear even better - I tested myself when i was 18 and I was hearing up to 21kHz. Very attentuated, I had to raise the volume some 20dB, but I was there.
I don't hear (now) over 18k, but I can tell if the 2nd and 3rd harmonics are "wrong" - to fundamantals in 10-14 kHz range. Tube amplifiers drive me on walls with that, even at medium volumes.
Young people might hear even better - I tested myself when i was 18 and I was hearing up to 21kHz. Very attentuated, I had to raise the volume some 20dB, but I was there.
I searched another PCM - DSD converter. any other?
(1) Weiss Saracon. $2500! -147dB S/N. mine looks like almost same performance.
(2) Korg AudioGate, Free but annoying auto-tweet. to unlock, KORG MR-2, $500
(3) Sony VAIO, some model has SonicStage Mastering Studio??
=======================
any math method to use FIR for delta-sigma modulation?
"delta" needs feedback, but maybe there should be some method to use FIR filter.
Then I can improve the high frequency performance, steep noise shaper.
(1) Weiss Saracon. $2500! -147dB S/N. mine looks like almost same performance.
(2) Korg AudioGate, Free but annoying auto-tweet. to unlock, KORG MR-2, $500
(3) Sony VAIO, some model has SonicStage Mastering Studio??
=======================
any math method to use FIR for delta-sigma modulation?
"delta" needs feedback, but maybe there should be some method to use FIR filter.
Then I can improve the high frequency performance, steep noise shaper.
If you are familiar with MATLAB, this can be regarded as a part of "PCM - DSD converter ".
Delta Sigma Toolbox - File Exchange - MATLAB Central
Hi Thanks Bunpei san
MATLAB, that is thousands $ software 🙁
If there are no free & easy WAV to DSF converter, only small number of people will play DSD.
maybe I should brush up DSF converter, and ask some frontend GUI programmer to support it??
for myself I want FLAC (LINN RECORDS etc) to DSF converter.
You Japanese have some DSD download store, I envy you!
MATLAB, that is thousands $ software 🙁
If there are no free & easy WAV to DSF converter, only small number of people will play DSD.
maybe I should brush up DSF converter, and ask some frontend GUI programmer to support it??
for myself I want FLAC (LINN RECORDS etc) to DSF converter.
You Japanese have some DSD download store, I envy you!
Yes 2L, but still "test bench" for DSD? Japanese already have Onkyo and Ototoy DSD store and selling from 2010.
and This machine looks great, DAC with DSD from SD playback.
HP-A8 to launch at CanJam @ RMAF - Fostex News
commercial version of Chiaki's SDTrans? 🙂
and This machine looks great, DAC with DSD from SD playback.
HP-A8 to launch at CanJam @ RMAF - Fostex News
commercial version of Chiaki's SDTrans? 🙂
https://sites.google.com/site/koona...ck-system/Wav2DSFconverter01_20111129.cpp.txt
I modified Wav to DSD converter, to handle 88.2kHz. maybe 176.4 also work.
for 96/192 flac, please use SOX to make 88.2 or 176.4kHz wav.
DSD requires 2.822MHz, that is x64 of 44100.
I modified Wav to DSD converter, to handle 88.2kHz. maybe 176.4 also work.
for 96/192 flac, please use SOX to make 88.2 or 176.4kHz wav.
DSD requires 2.822MHz, that is x64 of 44100.
DSD -> PCM -> DSD
- I do use an orig. ver. of THIS option . . . if need to convert from DSD to PCM . .
PM me if need any further info . .
Greetings,
- I do use an orig. ver. of THIS option . . . if need to convert from DSD to PCM . .
PM me if need any further info . .
Greetings,
He does the backwads conversion... from PCM to DSD.
Would be good if his software would do ant conversions "on the fly".
Would be good if his software would do ant conversions "on the fly".
Hi Sonic, yes,
DSD -> PCM : for signal processing.
PCM -> DSD : I think.. DSD makes DAC job almost done. so using DSD as final signal format can improve the sound.
now my PCM->DSD = oversampling to 2.8MHz + 2048 tap FIR LPF + 7th IIR.
I'm wondering about FIR LPF, because this LPF has no effect < 22050Hz. don't have to be linear.
So maybe I can use 16th IIR LPF instead of 2048 FIR LPF.
This is lightweight code, can be implemented in.. where? ASIO driver?
Or something interesting, "mathematical DAC"
I2S In -> MCU(>100MHz) -> DSD output -> passive LPF
DSD -> PCM : for signal processing.
PCM -> DSD : I think.. DSD makes DAC job almost done. so using DSD as final signal format can improve the sound.
now my PCM->DSD = oversampling to 2.8MHz + 2048 tap FIR LPF + 7th IIR.
I'm wondering about FIR LPF, because this LPF has no effect < 22050Hz. don't have to be linear.
So maybe I can use 16th IIR LPF instead of 2048 FIR LPF.
This is lightweight code, can be implemented in.. where? ASIO driver?
Or something interesting, "mathematical DAC"
I2S In -> MCU(>100MHz) -> DSD output -> passive LPF
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