The uRendu will function as a NAA. It will run the same Sonic Orbiter software.
Yup!
Maybe a stupid questions but would it be possible to use this "no dac" dac approach in an 8 channel manner to feed an active 4 way speaker system?
Are you kidding? Great question, man and that's something that's been on my mind for a good while. 🙂
Now that Jussi has added the matrix routing and processing in the DSD domain to HQ Player, this could be doable even for DSD.
Not tried yet though as I'm not willing to remove the internal XO of my speakers (they are a gift).
Sure would be nice if all 16/44 files could be upsampled to dsd and then split for active systems through this "no dac".....
The uRendu will function as a NAA. It will run the same Sonic Orbiter software.
fair enough...
Sure would be nice if all 16/44 files could be upsampled to dsd and then split for active systems through this "no dac".....
Absolutely!
Sure would be nice if all 16/44 files could be upsampled to dsd and then split for active systems through this "no dac".....
*IF* the source is ever in a PCM state, then many straightforward options exist to filter the data before up sampling to DSD. This can be done in linux/ALSA without destructive modifications to the PCM sampling rate. I'm even able to do it on a BBB (IIR filters), as discussed here.
I would still very much appreciate hearing any comparisons, conclusions, and/or 'lessons learned' from the principals in this thread!
I'm curious to see where this goes. Thank you for exploring such a simple and seemingly radical approach. Hopefully you keep posting your progress here.
I once was scanning a guide to working with DSD professionally with production. It recommended if you use PCM to convert to 88.2 24-bit. I am not sure if it doesn't allow anything that isn't an even resample of 44.1kHz in the standard but i am just going from memory.
If you are doing any re sampling, especially uneven resampling, i would recommend using SoX resampler plugin for foobar - it does much better on uneven resamples than most all.
If you are doing any re sampling, especially uneven resampling, i would recommend using SoX resampler plugin for foobar - it does much better on uneven resamples than most all.
Hi.
@nautibuoy:
very interested on your pcb/schematic.
Is it ok now?
I've got a jlsounds usb interface, double supply, etc.
Now it's time to go DSD.
In my system (only digital volume), popless playback is mandatory, so i'm very interested in a mute relay.
EDIT:
is this ok?
http://jlsounds.com/uploads/LME49710_schematic.pdf
@nautibuoy:
very interested on your pcb/schematic.
Is it ok now?
I've got a jlsounds usb interface, double supply, etc.
Now it's time to go DSD.
In my system (only digital volume), popless playback is mandatory, so i'm very interested in a mute relay.
EDIT:
is this ok?
http://jlsounds.com/uploads/LME49710_schematic.pdf
Last edited:
I have now received the correct PCBs for my simple direct DSD filter and mute module for my JL Sounds USB board. I have made a couple up, minus the mute relay, and installed one for testing.
I have played a native DSD256 album (dsf files) via HQPlayer - sounded v good. Obviously DSD128 (converted off line from FLAC CD rips) played very well too.
Not surprisingly, my Atom D610 equipped computer didn't have the grunt to do any on the fly upampling from 44.1KHz FLAC files to DSD.
I installed a mute relay and silence - the outputs are continuously grounded. There is something very strange going on with the relay that I don't understand so if anyone can shine a light; basically, I can measure the voltage change across the relay coil and can hear the click of the relay but the contacts don't seem to change. Here's the circuit;
An externally hosted image should be here but it was not working when we last tested it.
and I'm using one of these relays (4.5V coil, DPCO, non latching version);
http://www.farnell.com/datasheets/1683361.pdf
The pin annotations on the schematic relate to the relay pins.
With no music playing I measure 0VDC across the relay coil and 5VDC when music is playing, which is what I expect.
Ray
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 ;-)
Last edited:
I installed a mute relay and silence - the outputs are continuously grounded. There is something very strange going on with the relay that I don't understand so if anyone can shine a light; basically, I can measure the voltage change across the relay coil and can hear the click of the relay but the contacts don't seem to change.
Catching up on this thread... Is this still a problem? What happens if you route the signal through the relay? Ground on pins 2/9 (to ground when coil is de-energized) inputs on 3/8 and buffer outputs on 4/7? Best of luck!
Attachments
Last edited:
Has anyone found a unit that can output balanced DSD?
Hi Vladimir, HiFiMeDIY has a USB to I2S device with balanced DSD output
HiFime UH1 384kHz USB DAC, headphone amplifier and I2S/DSD interface
Attached is the pin out.
Alternatively you can use a flip flop (74HC74) to create your own balanced output. I've posted my circuit before but here it is again. I use this daily, it works a treat.
Attachments
That looks good. Is it sold just as a board that could be used with an isolator? Not sure if that is necessary, please advise
And is there a flip-flop board out there for sale?
And is there a flip-flop board out there for sale?
Last edited:
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
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
Hello everyone,
I have the JLsounds board and after reading the whole thread, I'm impatient to try this out.
I suspect the passive output filter must be very well built to filter out the leftover MHzs of DSD. A resistor/inductor with very low parasitic capacitance and a capacitor with very low parasitic inductance, maybe tantalum type. A transformer is not enough IMHO.
For now, I imagine a RLC LP filter with 16R , 30uH and 470nF. The R is there to damp the Q of the filter to ~0,5
I have the JLsounds board and after reading the whole thread, I'm impatient to try this out.
I suspect the passive output filter must be very well built to filter out the leftover MHzs of DSD. A resistor/inductor with very low parasitic capacitance and a capacitor with very low parasitic inductance, maybe tantalum type. A transformer is not enough IMHO.
For now, I imagine a RLC LP filter with 16R , 30uH and 470nF. The R is there to damp the Q of the filter to ~0,5
This is my Valve Preamplifier for DSD - DAC
Dears ALL,
this is my personal version of Valve Preamplifier for DSD - DAC.
In this special Preamplifier there only a capacitor of uncoupling among input and output.
Is very important that all capacitors not polarized are of high quality.
The polarized capacitors are long life and for use to 100Khz (for DCDC converter).
Thanks and All the Best,
JEDY
Thanks Ray.
Dears ALL,
this is my personal version of Valve Preamplifier for DSD - DAC.
In this special Preamplifier there only a capacitor of uncoupling among input and output.
Is very important that all capacitors not polarized are of high quality.
The polarized capacitors are long life and for use to 100Khz (for DCDC converter).
Thanks and All the Best,
JEDY
Attachments
- Home
- Source & Line
- Digital Line Level
- The Best DAC is no DAC