The Best DAC is no DAC

R1 and R2 are resistors in grid for V1 and V2

Hello Jedy,

why do you not use/need a gridstopper resistor?

Ronny
Dear Mr. KOIFARM,
R1 and R2 are resistors in grid for V1 and V2 and low pass for DSD format.
In preamplifier Hi End Car, I dont use resistors of grid because not necessary.
If you desire to improve this preamplifier, you insert R3 and R4 before of R1 and R2.
If you have other questions, I answer to you with great pleasure.
Thanks and All the Best,
JEDY (mirco)
 
I'm SORRY for my distraction !!!

Dear Mr. KOIFARM,
R1 and R2 are resistors in grid for V1 and V2 and low pass for DSD format.
In preamplifier Hi End Car, I dont use resistors of grid because not necessary.
If you desire to improve this preamplifier, you insert R3 and R4 before of R1 and R2.
If you have other questions, I answer to you with great pleasure.
Thanks and All the Best,
JEDY (mirco)
Dear Mr. KOIFARM,
I'm SORRY, SORRY, SORRY for my distraction !!!
The Grid n°2 (pin 9) is connect directly to the Anode (pin 7).
 
Just a quick note to apologise to anyone waiting for an update on my filter board; stuff happens in life and I've had very little spare time to spend on this project and associated aspects and I have not resolved the mute relay issue.

Things have now moved on in this popular DSD subject area so I am not going to spend any more time on my project and instead will hook my wagon up to Acko's new project;

http://www.diyaudio.com/forums/group-buys/280763-direct-drive-dsd.html

Cheers

Ray

Never say never!

I had another look at this again this morning as the problem kept nagging at me. After some testing it was clear that there is a problem with the relay on the board I put together; although it clicks it seems to not changeover the connections most of the time. I tested another relay that hadn't been assembled to a board and it works perfectly.

Anyway, I assembled another board but this time, just in case it is a heat issue from soldering, I installed the relay into an IC socket. With this board I've just been listening to DSD256, courtesy of HQPlayer's upsampling, through my home cinema amp. I'm using the thing I've just got working as an HQPlayer NAA endpoint.

Oh, and with the relay installed I've not heard any clicks between tracks on an album. I don't know about when I change albums over as the HQPlayer engine PC is elsewhere in the house and I can't be in two places at once - will check when the lady of the house gets back later.

As that has gone well I'm going to take it a bit further and try it in my main system. I'm going to have to replace the 47K resistor with a 50K Alps Blue potentiometer so it has a volume control between the LP filter and the buffer I'm using first though.

A bit frustrating that I had the problem with the relay but I got there in the end.

Ray
 
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I don't understand in any way this schematic.
How the relay works?
Is it an high pass filter?

EDIT:
ok, from right to left, not vice-versa ;-)

When the TLP device goes on (LED emits because the port it connects to goes 'high') then a current will flow on the output. This will turn Q1 on. Then a current flows from pin 1 to pin 10 and activates the relay.
(The diode is to prevent bouncing/sparking of the relay coil when the relay goes off again.)
 
... I2SOVERUSB (bus or self powered; external clock connector) + HC74 + LL1527XL + XLR/RCA... first PBClayout placement... will change!

Jean-Paul
 

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I've been really enjoying the results of my project that came together in post #344.

I'm feeding it DSD256, upsampled with HQPlayer, which streams the native DSD to a Linux-based NAA and then to my JLSounds USB board. The Linux build on the NAA has a very recent kernel that enables DSD256 on the JLSounds board. The sound is amazingly good. With the mute feature now working and with everything being DSD256 I am not experiencing any pop or click issues.

On the basis of this result I have started work on a tube-based filter/buffer.

One thought, has anyone measured the actual analogue signal level derived from passing DSD through a low pass filter? The level on my build seems a little lower than a 'typical' source component.
 
I've been really enjoying the results of my project that came together in post #344.

I'm feeding it DSD256, upsampled with HQPlayer, which streams the native DSD to a Linux-based NAA and then to my JLSounds USB board. The Linux build on the NAA has a very recent kernel that enables DSD256 on the JLSounds board. The sound is amazingly good. With the mute feature now working and with everything being DSD256 I am not experiencing any pop or click issues.

On the basis of this result I have started work on a tube-based filter/buffer.

One thought, has anyone measured the actual analogue signal level derived from passing DSD through a low pass filter? The level on my build seems a little lower than a 'typical' source component.
Ray, I haven't measured output through the LPF but I agree that it's low. My 'balanced output' flip flop doubles the output voltage swing and output is still low compared to other source components. My set up is similar but not the same. I use HQPlayer in Linux (Unbuntu 14.04) and JLSounds USB board. Everything upsampled to DSD256. But no NAA.

What kernel are you using? I'm using Jussi's own kernel (3.13.0-55) available from the HQPlayer web site. Ubuntu keeps updating itself and it has installed 3.16.xx on my PC. When I boot I have to choose the correct kernel in GRUB. Standard Ubuntu kernel doesn't support native (non-DOP) DSD as far as i can tell - has it recently been updated?
 
What kernel are you using?

Hi Hazard.

I'm running HQPlayer on a Windows 10 workstation. I'm running Windows because it allows me to run music library applications that improve significantly on HQPlayers own library function. Roon is the one I'm mainly looking at but its interface to control HQPlayer isn't ready yet so I'm currently trying Muso, which is able to control HQPlayer and has a basic remote control function too.

The HQPlayer NAA enables the HQPlayer computer remote from the audio system by using a proprietary interface to stream the upsampled DSD256 over an Ethernet connection. Currently my NAA box uses a silent Atom-based computer. I'm running Debian Stretch, kernel version 4.2 which has native DSD capability out of the box. Installation was very simple and I only installed the core so no GUI interface. I just had to install a couple of libs required by the NAA package and then NAA itself and it all worked. The only minor glitch was that NAA is supposed to start as a daemon but didn't but following Jussi's advice I set a static IP address and all was well.

Here's a link to the Debian Stretch installer;

https://www.debian.org/devel/debian-installer/

and, in case its of interest, this is where you can find Muso;

http://klarita.net/muso.html

Ray
 
After getting my no-DAC DSD project up and running I have received a number of requests from people asking about the availability of the PCBs I had made up.

As I have now proved that the problem was a faulty relay and the boards work fine I'm happy to make the spare boards available to anyone interested in trying this DSD stuff out...

The boards are similar to the one pictured in this post (but are red instead of blue). They are designed to locate onto the header pins of the JLSounds I2SoverUSB board but could be used in other scenarios with a bit of creative hook up wiring.

http://www.diyaudio.com/forums/digital-line-level/273474-best-dac-no-dac-23.html#post4381033


The boards are not meant to be a definitive statement on the state of the art on DSD replay, just a cheap and simple prototype to find out what the fuss is about. There's no room for esoteric caps etc. so if you want to try that approach that's your problem to sort out locating them.

In my build, where I am only passing DSD256 to the USB board, I am not experiencing any pop or click issues but YMMV.

I have five boards available, first come, first served. PM me if you're interested and I'll give you a cost to send one to you, obviously dependent on where you live.

Ray
 
I believe I've replied to the PMs sent to me about the PCBs.

One thing prospective adopters might want to think about is the size of the DC blocking capacitor that was included in the original JLSounds schematic and which I carried forward into my initial project.

My understanding of these matters (and I profess no expertise) is that size of the DC blocking/coupling cap is defined by the load impedance and the required LF cutoff point.

The originally specified cap was 22uF and that will give a -3dB point of around 3Hz with a 20K load impedance. With a 100K load the -3dB point drops to 0.07Hz, alternatively, to retain the 3Hz -3dB cutoff with a 100K load you only need a cap of around 0.47uF.

I came across this when looking for some info on this topic;

Coupling Capacitor Calculator by V-Cap

Hopefully someone will pipe up and point out my errors if I have this wrong.

Ray
 
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@nautibuoy:

Hi - no I think your calculations are correct.

However, in my experience it's possible to make this without a coupling capacitor. By using a D flip-flop with a Q and Qnot (inverted) output the absolute DC level will be Vcc/2 (with a DSD signal), but since the output itself - the Q and Qnot pins - are self-referencing (differential output, i.e. not referenced to ground but to eachother) the resultant DC difference between the two pins in my experience is only a few mVs.

The clock for the D flip-flop is the DSDCLK.

In practice a simple setup can be the output pins Q and Qnot each followed by a suitable resistor and then there's a suitable capacitor connected between each of the resistor "ends" (the output side) to ground to remove HF noise from the DSD conversion process. Can be e.g. two 100 ohms resistors with a 15 nF capacitor to ground for an upper cut-off frequency of ~ 100 kHz.

A word of caution, though: This setup requires that the output be switched off (I short the output pins after the resistors with a relay) when there's no DSD signal. Otherwise the Q and Qnot pins will remain constant at either a high or a low level which quickly will give about Vcc on the output. With the Amanero board I have made this work, albeit not yet perfectly as I still get some rather loud clicks on the output. But maybe the JLSound's board is better here?

Cheers,

Jesper
 
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@nautibuoy:

Hi - no I think your calculations are correct.

However, in my experience it's possible to make this without a coupling capacitor. By using a D flip-flop with a Q and Qnot (inverted) output the absolute DC level will be Vcc/2 (with a DSD signal), but since the output itself - the Q and Qnot pins - are self-referencing (differential output, i.e. not referenced to ground but to eachother) the resultant DC difference between the two pins in my experience is only a few mVs.

The clock for the D flip-flop is the DSDCLK.

In practice a simple setup can be the output pins Q and Qnot each followed by a suitable resistor and then there's a suitable capacitor connected between each of the resistor "ends" (the output side) to ground to remove HF noise from the DSD conversion process. Can be e.g. two 100 ohms resistors with a 15 nF capacitor to ground for an upper cut-off frequency of ~ 100 kHz.

A word of caution, though: This setup requires that the output be switched off (I short the output pins after the resistors with a relay) when there's no DSD signal. Otherwise the Q and Qnot pins will remain constant at either a high or a low level which quickly will give about Vcc on the output. With the Amanero board I have made this work, albeit not yet perfectly as I still get some rather loud clicks on the output. But maybe the JLSound's board is better here?

Cheers,

Jesper

Hi Jesper. it sounds as though your approach is not dissimilar to Hazard's;

http://www.diyaudio.com/forums/digital-line-level/273474-best-dac-no-dac-9.html#post4323177

except he's using transformers as his LP filters... Do you have a schematic you could share?

I've been reading back through this thread over the last day or two and I'm drawn back to some of the earlier posts that were looking at using the flip-flop approach to getting a differential signal and then using a TVC as the LP filter - very neat.

Ray
 
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