DSP Xover project (part 2)

Ok, Nick don´t laugh now please. :) Ok, you can laugh, as long as I get some help in the end.

My Najda is running fine, and the only thing missing is the optocoupler trigger, so I can tell my amp to to go standby automatically.

The amp is DIY so I can put circuits in there, and it has an Arduino for other things.

So, here´s what I have in mind:

From the amp, send 5v the optocoupler, and have a return wire back which goes to an input pin on the Arduino (it is 5v compliant). This input pin gets an external pulldown resistor to ground. So when the Najda is on, the input pin will be pulled high, and the the Najda is off the input pin will be pulled to ground.

I can then use the Arduino to make the nice leds go on, turn on the relays for the PSUs, bring the PSU´s out of standby etc in a nice sequence.

Does this make sense or do I still misunderstand how an optocoupler works..?
 
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Hi! Long time no see ;)

Nick,
are we able to use FIR and IIR filter combinations in the same preset (ver1.1.2)? If so how?
thanks


No this is not possible yet. Some of you have been waiting for this for a long time, but it's eventually going to come.


Ok, Nick don´t laugh now please. :) Ok, you can laugh, as long as I get some help in the end.

My Najda is running fine, and the only thing missing is the optocoupler trigger, so I can tell my amp to to go standby automatically.

The amp is DIY so I can put circuits in there, and it has an Arduino for other things.

So, here´s what I have in mind:

From the amp, send 5v the optocoupler, and have a return wire back which goes to an input pin on the Arduino (it is 5v compliant). This input pin gets an external pulldown resistor to ground. So when the Najda is on, the input pin will be pulled high, and the the Najda is off the input pin will be pulled to ground.

I can then use the Arduino to make the nice leds go on, turn on the relays for the PSUs, bring the PSU´s out of standby etc in a nice sequence.

Does this make sense or do I still misunderstand how an optocoupler works..?

Oh - and are there any issues with a cable "leaving my amp" and entering the Najda box, in terms of ground loops etc, or is this a non-issue when using an optocoupler?

Hm can't see what's funny? :goodbad:
I'm not sure I quite understand what you intend to do with the remote output.
The best thing you can do is checking the optocoupler datasheet (FOD852S). There's a test circuit diagram in it that could get you started. The optocoupler is an isolated switch: it's either open or closed, and it's safe in terms of loops and injected noise.

Cheers

Nick
 
Hello Guys,

i found out, right the way, it is not possible to play hires Files with Najda (96kHz/ 192kHz on Digital IN)

all outs Stoppband frequencies was at 24kHz CutOff even on digital outs with V90DAC)

FS:96kHz or 192kHz have nothing to do with the samplerate converter on digtal in!!!
all signals comming from DIGTAL INS will downsamplet to 48kHz!!!

If you will use the Analogue INS with 30kHz cutoff you will get 30kHz cutoff on analogue and digita outs ! But this is the only way for Hires Frequencies !

Im not sure if anyone alredy wrote this!

So for me i will go back to my NANODIGI 2x8 with 3 V90DACS thats work fine with Hires files on DIGITAL INS the only thig is You need an 8Chanel volumecontrol....

damm probably i will najda use for analogue Volumecontrol so it isnt worthless at the end!!!

Sorry for my ugly English
 
Hello Guys,

i found out, right the way, it is not possible to play hires Files with Najda (96kHz/ 192kHz on Digital IN)

all outs Stoppband frequencies was at 24kHz CutOff even on digital outs with V90DAC)

FS:96kHz or 192kHz have nothing to do with the samplerate converter on digtal in!!!
all signals comming from DIGTAL INS will downsamplet to 48kHz!!!

If you will use the Analogue INS with 30kHz cutoff you will get 30kHz cutoff on analogue and digita outs ! But this is the only way for Hires Frequencies !

Im not sure if anyone alredy wrote this!

So for me i will go back to my NANODIGI 2x8 with 3 V90DACS thats work fine with Hires files on DIGITAL INS the only thig is You need an 8Chanel volumecontrol....

damm probably i will najda use for analogue Volumecontrol so it isnt worthless at the end!!!

Sorry for my ugly English

When you start a project you have to choose a sampling frequency. Later the
samplerate converter will up or down sample the digital input material to the
frequency of your choise. A four-way filter may be difficoult to set to 192khz
if you need to do a lot of settings. Not all but most settings require a few
percent of the total dsp recources. Higher samplerate equals less number of settings.

It's described at page six in the user manual, options are 48 / 96 or 192khz. http://www.waf-audio.com/doc/Najda/Najda_Under_Control_Manual.pdf
 
Has you ever measured the Frequencieresponse at Analogue out if you choose the Digital in with a 96kHz capable source)?

i did it and the Stopband frequencie was at 24kHz!!!

My setup was FS: at 96kHz and my source was my soundcart with ASIO 96kHz setings and REW at 96kHz and i dont got 48kHz Stopband frequencies like sugested on Analogue Qouts and only 24KHz Stopband at Digital Out with V90DAC conected!!! (Coaxial SPDIF IN and OUTs was USED)

If i remove the NAJDA Board between Soundcard DIG OUT and V90DAC DIG IN and i connect the V90DAC to my Soundcard without any changes of Sounddriver and REW i get the suggested 48kHz Stoppband (96kHz Samplingfrequencies) but not with NAJDA!

I Atached the REW MEasurement Curves with all done Meaturements. Some Measurements i did with different DC Capacitors. 10uF seem to be a goot choice (WIMA 10uF)
 

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BTW:

This information i got from WAF-AUDIO!!!

"The band-limiting you have observed comes actually from the digital inputs. Indeed, these signals go through the sample rate converter, and that's the element in the chain which is is band-limiting the signal to roughly 24 kHz. So for example if you inject a 30 kHz signal through the analogue inputs, then you should find a 30 kHz at the outputs as well (converters are band-limiting the signal to roughly 40 kHz when sample rate is 96 kHz or greater).
In order to have the full band of the signal, we should skip the sample rate converter. This is possible technically, but that would require that you precisely keep control over the sampling frequency of your signal. In other words, a signal with wrong sampling frequency could potentially damage your speakers - because the filtering coefficients computed for one sampling frequency are not correct if you use another sampling frequency. That's the reason why we force for now the routing of digital inputs though the sample rate converter.

Hope this clarifies the situation. Should you have another question, please don't hesitate to get back in touch."
 
BTW:

This information i got from WAF-AUDIO!!!

"The band-limiting you have observed comes actually from the digital inputs. Indeed, these signals go through the sample rate converter, and that's the element in the chain which is is band-limiting the signal to roughly 24 kHz. So for example if you inject a 30 kHz signal through the analogue inputs, then you should find a 30 kHz at the outputs as well (converters are band-limiting the signal to roughly 40 kHz when sample rate is 96 kHz or greater).
In order to have the full band of the signal, we should skip the sample rate converter. This is possible technically, but that would require that you precisely keep control over the sampling frequency of your signal. In other words, a signal with wrong sampling frequency could potentially damage your
speakers - because the filtering coefficients computed for one sampling frequency are not correct if you use another sampling frequency. That's the reason why we force for now the routing of digital inputs though the sample rate converter.

Hope this clarifies the situation. Should you have another question, please don't hesitate to get back in touch."

I had no idea that this was happening and for what its worth its not a problem
for me. I guess this is something that's relevant for both i2s and spdif digital inputs ?
Is there a way to lift this limit to what you would expect with another firmware ?
 
I had no idea that this was happening and for what its worth its not a problem
for me. I guess this is something that's relevant for both i2s and spdif digital inputs ?
Is there a way to lift this limit to what you would expect with another firmware ?

Yes it is relevant for all conection thrue the ASRC i guess.
Im not sure if its possible to change the firmware for this problem. Depends on how the ASRC is used (Hardwaremode or softwaremode)?

If it in Hardwaremode may be i could change the default samplefrequenz to the target FS of 96kHz with solder some pins to Vcc or ground need to read the dokumention of ASRC

But my bigest question is...why the ASRC isnt switch the default Samplingfrequenz of selectet FS ...only to 48Hz by default ...WHY:eek:

The only reasen for me is they use this thing in Hardwaremode and routed his pins to default 48kHz the easiest way may be:(


but how this could work ...the dsp is ste to fs:96kHz----and the inputfrequenz to DSP from ASRC is 48kHz ...where is the logic by this???? :scratch:

but im not the engineer who build this.....
 
I had no idea that this was happening and for what its worth its not a problem
for me. I guess this is something that's relevant for both i2s and spdif digital inputs ?
Is there a way to lift this limit to what you would expect with another firmware ?

Hi,
did I undestood it correctly? If you want to play a HiRes file, the only way would be through the analog inputs and selection of the Najda sample rate equal to the original sample rate of the HiRes file?
Cheers
Sigi
 
Hi,
did I undestood it correctly? If you want to play a HiRes file, the only way would be through the analog inputs and selection of the Najda sample rate equal to the original sample rate of the HiRes file?
Cheers
Sigi

Right, this is the Bad of NAJDA for me this Board is only to use it as an analogue Volumecontrol fed in the analogue outs of my V90Dacs thru the 8ch Volumechip!

If i had to know it befor i didnt bought this Board!

I was planing to use 2 or 3 of them for Surround configuration but now i need to find an other option....:mad:
 
Indeed. I don't think anyone here can hear frequencies above 24 kHz. My pet bat is another issue altogether, but we can't agree about taste of music anyway...

That is not the Point for me to discus.

The Point is in the Specification of the NAJDA Board i can Read this:

Digital Section:
Input sampling frequency: from 32 to 192 kHz
Computing Unit:
Processing sampling frequency: 48, 96 or 192 kHz selectable

Ther isn't any information abou digital in degrade all inputfrequencies to 48kHz!!!

I will never use an Analogue in (dubbleconversion) for Digital Files from NAS (WAV only!)

Would you ever buing a CAR With specification V8 600HP but no wort about degrade with electronik to 90mph Vmax???? You can still drive to youre Work with it, right?
 
That is not the Point for me to discus.

The Point is in the Specification of the NAJDA Board i can Read this:

Digital Section:
Input sampling frequency: from 32 to 192 kHz
Computing Unit:
Processing sampling frequency: 48, 96 or 192 kHz selectable

Ther isn't any information abou digital in degrade all inputfrequencies to 48kHz!!!

I will never use an Analogue in (dubbleconversion) for Digital Files from NAS (WAV only!)

Would you ever buing a CAR With specification V8 600HP but no wort about degrade with electronik to 90mph Vmax???? You can still drive to youre Work with it, right?
Hi,
read the post 1501 from Nick! It seems,that it is possible to avoid ASRC! Nick wrote:
"The current clocking strategy is master clock generated onboard and all external signals sample-rate-converted.
A next release will include the option of using an external clock instead, and bypass the ASRC.
Whether this is going to be beneficial for SQ is another topic"
Cheers
Sigi