Hello,
how do you connect the coax out of the DAC to the S/PDIF input of the receiver module?
signal to input 1
ground to input 2 or 3? why are there 2 input there? Is that for balanced use?
how do you connect the coax out of the DAC to the S/PDIF input of the receiver module?
signal to input 1
ground to input 2 or 3? why are there 2 input there? Is that for balanced use?
promitheus said:
You would use Pin 1 and either Pin 2 or 3 as they will be joined by a jumper for SE use:
Pin 1 Signal
Pin 2 GND
Pin 3 GND
For balanced use you substitute R1(the jumper) for a 10K resistor or leave it open.
For balanced SPDIF you would use:
Pin 1 +
Pin 2 GND
Pin 3 -
All of this will be covered in the receiver manual which I am writing now.
Cheers!
Russ
What bitrates does the planned ethernet receiver support?
It's about time that they made a USB receiver which supports higher bitrates. I that case one could use upsamling in foobar to avoid the need for analog filtering. (I believe the dac already has an internal 200kHz analog filter.)
With a typical output of 2V on a 5V supply one could effectively drive a power amp directly from the dac output. Now that's an interresting thought for a minimalist. Wonder what the distortion figures would be like with such a load.
It's about time that they made a USB receiver which supports higher bitrates. I that case one could use upsamling in foobar to avoid the need for analog filtering. (I believe the dac already has an internal 200kHz analog filter.)
With a typical output of 2V on a 5V supply one could effectively drive a power amp directly from the dac output. Now that's an interresting thought for a minimalist. Wonder what the distortion figures would be like with such a load.
I am not sure yet, that project is still in the very very early stages (just planning right now), but with the devices I will be using It may indeed be possible to do 96/24. I am not just not sure yet.
My first attempt will simply be 44.1/16, and the clock I will use for that is not suitable for 48/24 or 96/24. So I think to support those rates too I would have to figure out a way to support multiple clocks. Not ruling it out, but the high bit rate version may be a second version after I get the 44.1/16 version working.
I will start a new thread about it when I have things better solidified.
Cheers!
Russ
My first attempt will simply be 44.1/16, and the clock I will use for that is not suitable for 48/24 or 96/24. So I think to support those rates too I would have to figure out a way to support multiple clocks. Not ruling it out, but the high bit rate version may be a second version after I get the 44.1/16 version working.
I will start a new thread about it when I have things better solidified.
Cheers!
Russ
I couldn't help but ask... pardon my ignorance, but looking at the schematics of both the DAC board and receiver board, I cannot seem to locate the occillator. Is there no need for an occillator?
Mazuki said:
There is no oscillator or clock on the DAC. The master clock is derived from the receiver module. 🙂 This will also be true for the USB module.
You could also if you like supply your own clock signal from a Tent clock or whatever you like.
Cheers!
Russ
Mazuki said:
Hi, if I'm not false, the standard config for the cs8416 is the PLL clock -> The clock is regenerated from the SPDIF clock.
I can add a question : is it possible to plug a hight quality input clock like XO clock or trendlab ?
EDIT : how can I manage to put this clock ?
Others questions :
- does anyone made a tube stage for this DAC ?
- how does this DAC sounds compare to NOS DAC ?
Thanks
The master clock coming from the CS8416 is actually quite low jitter, and works extremely well when you operate the receiver PLL at high update rate which is what I show as the default configuration in the documentation.
Connecting any other clock you like (as long as you calculate the correct frequency) is as simple as connecting it to the master clock terminal on the DAC. I have tested this with a simple clock of my own making.
This DAC has significantly better SNR and lower distortion than any NOS DAC you will ever hear. The big differentiator though is balanced output and incredible detail. No IV stage at all helps I think, there is less noise and distortion because there is the absence of that stage.
I don't know of anyone who has yet, but there is nothing to prevent you from doing a tube buffer after the DAC, just remember the output of the DAC is voltage and not current.
Cheers!
Russ
Connecting any other clock you like (as long as you calculate the correct frequency) is as simple as connecting it to the master clock terminal on the DAC. I have tested this with a simple clock of my own making.
This DAC has significantly better SNR and lower distortion than any NOS DAC you will ever hear. The big differentiator though is balanced output and incredible detail. No IV stage at all helps I think, there is less noise and distortion because there is the absence of that stage.
I don't know of anyone who has yet, but there is nothing to prevent you from doing a tube buffer after the DAC, just remember the output of the DAC is voltage and not current.
Cheers!
Russ
Thanks, very fast reply.
I've open a specific thread on a french forum. Perhaps we will launch a "group by" order.
The buffer stage is to convert balanced to unbalanced in a mono configuration. No amplification required.
But we will try it first without this stage !
Very nice project. The DAC is affordable and very modular !
I've open a specific thread on a french forum. Perhaps we will launch a "group by" order.
The buffer stage is to convert balanced to unbalanced in a mono configuration. No amplification required.
But we will try it first without this stage !
Very nice project. The DAC is affordable and very modular !
ezkcdude said:
RMAA? It is very difficult to measure balanced signals for THD etc with most computer gear. I fact I know of none that does it well. But I have asked my friends at the university to give Opus a look here very soon, in fact the next chance I have to run down there. So no I have not posted measurements yet, my facilities are not good enough yet to provided decent data. 🙂 Soon, very soon. I am confident my current sound card does not have low enough noise floor for good data even single ended.
Still lets start with the basis. When I spoke of noise and THD I was talking about the data sheets. I think anyone who has built and Opus will agree the performance backs up the DS specs.
The specs for the TDA15xx chips used in NOS DACs just simply are nowhere near as good(not even close). Also anytime you add things to the analog side (like IV stages) you introduce the noise and distortion of the active and passive devices added. This can also be true on the digital side.
The true test is in the hearing, and once you hear one it would be difficult to go back. 😉 Go ahead give it a build! 🙂
Also you cannot underestimate the value of the balanced signal into balanced gear. It makes a world of difference in terms of negating noise that would otherwise be present on a single ended source/cable to the next stage. That is a major point here, and the main reason there is an "Opus".
Cheers!
Russ
Russ White said:
RMAA= Right Mark Audio Analyzer
Gear needed? E-MU 1212M would be fine for balanced and would give you the measurements you need for ~$200. Now you know...
Did you think I was looking for a new sound card? 🙂
That card is quite nice, but not any better than what I have currently.
What I am trying to say is that there are inherent problems with trying to measure anything with > 100db resolution with most PCs (nice sound card or not), not that it can't be done. In fact with the new rig I am I am building I hope to get a lot closer. 🙂
The lab is much better, and I have it to use, so why not?
Please feel free to build one and measure the balanced output for yourself. 🙂
Cheers!
Russ
That card is quite nice, but not any better than what I have currently.
What I am trying to say is that there are inherent problems with trying to measure anything with > 100db resolution with most PCs (nice sound card or not), not that it can't be done. In fact with the new rig I am I am building I hope to get a lot closer. 🙂
The lab is much better, and I have it to use, so why not?
Please feel free to build one and measure the balanced output for yourself. 🙂
Cheers!
Russ
hi
j have just received and mounted my opus dac.
But no sound ....
j do not know how to connect it to my CDPro2 LF with I2S mode
On the cd pro j have
Pin 1 MISC
Pin 2 SCLK
Pin 3 WCLK
Pin 4 DATA
Pin 5 GND
Pin 6 V4 (serial cdg outpout)
on the opus dac j have :
BCK
SCK
LRCK
Din
G
J have put the the 6 connectors as it is explained on the manuel on the site of twister.
Thanks a lot for your response. For the beginning j am testing with a battery car about 12V
Philippe
j have just received and mounted my opus dac.
But no sound ....
j do not know how to connect it to my CDPro2 LF with I2S mode
On the cd pro j have
Pin 1 MISC
Pin 2 SCLK
Pin 3 WCLK
Pin 4 DATA
Pin 5 GND
Pin 6 V4 (serial cdg outpout)
on the opus dac j have :
BCK
SCK
LRCK
Din
G
J have put the the 6 connectors as it is explained on the manuel on the site of twister.
Thanks a lot for your response. For the beginning j am testing with a battery car about 12V
Philippe
The only thing I can note off the bat is that you seem to be one clock short...
I don't have any data on you transport,
guessing here:
SCLK = either Serial clock(bit clock) or it is system clock(master clock).
WCLK = Word clock which corresponds with LRCK on the Opus.
DATA = Serial data which corresponds with DIN on the Opus
That is only three digital outputs, you need four.
So I would need to know what SCLK is from your transport. From that you will likely have to generate another clock. Or use a separate clock.
I will google and see if I can be more help.
I don't have any data on you transport,
guessing here:
SCLK = either Serial clock(bit clock) or it is system clock(master clock).
WCLK = Word clock which corresponds with LRCK on the Opus.
DATA = Serial data which corresponds with DIN on the Opus
That is only three digital outputs, you need four.
So I would need to know what SCLK is from your transport. From that you will likely have to generate another clock. Or use a separate clock.
I will google and see if I can be more help.
Ok looking at the manual
Pin 1 MISC = has a clock 2 X the SCLK in audio mode. This can probably be used as your master clock (SCK on opus).
Pin 2 SCLK = is your bit clock (BCK on opus)
Pin 3 WCLK = is LRCK on opus
Pin 4 DATA = Din
Pin 5 GND
Pin 6 V4 (serial cdg outpout)
Please note the manual says something about the modules having to be in audio mode for this to work. Page 10 is where I was looking.
If using the MISC pin does not work then you will need to generate the clock using a PLL or Use another clock circuit. You could use a 2X or 4X PLL on the bit clock (SCLK).
I hope that helps. 🙂
Cheers!
Russ
Pin 1 MISC = has a clock 2 X the SCLK in audio mode. This can probably be used as your master clock (SCK on opus).
Pin 2 SCLK = is your bit clock (BCK on opus)
Pin 3 WCLK = is LRCK on opus
Pin 4 DATA = Din
Pin 5 GND
Pin 6 V4 (serial cdg outpout)
Please note the manual says something about the modules having to be in audio mode for this to work. Page 10 is where I was looking.
If using the MISC pin does not work then you will need to generate the clock using a PLL or Use another clock circuit. You could use a 2X or 4X PLL on the bit clock (SCLK).
I hope that helps. 🙂
Cheers!
Russ