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Okto 8 Owner’s Thread

Zooqu1ko

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The Okto DAC8 PRO seems to include some kind of filters. Can they be used to create high/low pass filters for a 4-way active speaker? What kind of limitations do they have, for example the amount?
Those filters are merely low pass filters at around 1/2 the sample rate of the current input signal. There are a few nice graphs that illustrate the trade offs of the various filters at https://www.stereophile.com/content/okto-dac8-stereo-da-processor-measurements

There's an alternative firmware that does what you're looking for:
https://www.audiosciencereview.com/...kto-dac8-8ch-dac-amp.7064/page-89#post-536652
 

Machismo

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Those filters are merely low pass filters at around 1/2 the sample rate of the current input signal. There are a few nice graphs that illustrate the trade offs of the various filters at https://www.stereophile.com/content/okto-dac8-stereo-da-processor-measurements

There's an alternative firmware that does what you're looking for:
https://www.audiosciencereview.com/...kto-dac8-8ch-dac-amp.7064/page-89#post-536652

Ok thanks. I already happened to modify my original post since I did some reading too and yes those filters are only for that one narrow purpose.

Interesting that alternative firmware!

I also red that originally there were plans for a official DSP version of Okto 8 but I guess there is not enough time to do that in the near future.

From the link I also found the information that any cable works as long as it's short:

"AES inputs in the DAC8 PRO are terminated with 110 Ohms as required by the standard, while the SPDIF expects 75Ohms. So a 220-240Ohm additional termination resistor will do the justice, since the digital receivers in the DAC8 PRO handle both SPDIF and AES voltage levels. But in fact, no resistor is needed at all if the cables are not very long. So a simple XLR M/RCA F adapter might work just fine in that case.
Pavel, Okto Research "
 
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Zooqu1ko

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Ok thanks. I already happened to modify my original post since I did some reading too and yes those filters are only for that one narrow purpose.

Interesting that alternative firmware!

I also red that originally there were plans for a official DSP version of Okto 8 but I guess there is not enough time to do that in the near future.
The DSP version isn't happening, because the DSP board in question was discontinued.

I haven't tried the specific combination with a nanodigi 2x8. A way more expensive solution would be a miniDSP DDRC 88d, that one already speaks AES/EBU. However, for short distances, you can get away with 75 Ohm coax and just an XLR plug at the end. A better solution would be a bundle of Canare BCJ-XP-TR[ABC]; I believe the equivalent devices from Neutrik may be too wide to fit side by side in the Okto. You'd still need a RCA<>BNC cables, those are easier to find, e.g. here: https://www.bluejeanscable.com/store/digital-audio/index.htm
 

Machismo

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The DSP version isn't happening, because the DSP board in question was discontinued.

I haven't tried the specific combination with a nanodigi 2x8. A way more expensive solution would be a miniDSP DDRC 88d, that one already speaks AES/EBU. However, for short distances, you can get away with 75 Ohm coax and just an XLR plug at the end. A better solution would be a bundle of Canare BCJ-XP-TR[ABC]; I believe the equivalent devices from Neutrik may be too wide to fit side by side in the Okto. You'd still need a RCA<>BNC cables, those are easier to find, e.g. here: https://www.bluejeanscable.com/store/digital-audio/index.htm

Ok! I thought about DDRC 88d too, but noticed that it processes internally at only 48 khz, so I didn't want it anymore. Instead I bought the DDRC-22D (internal operating sample rate: 96 kHz) to use with Dirac and from that I plan to route the signal to the nanoDIGI 2x8 B.

Not a very elegant solution, but quality wise difficult to beat, huh? I am still open to ideas however.

I now also just put my order in for the dac8 PRO, I hope it will come in a few months..
 

mdsimon2

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Ok! I thought about DDRC 88d too, but noticed that it processes internally at only 48 khz, so I didn't want it anymore. Instead I bought the DDRC-22D (internal operating sample rate: 96 kHz) to use with Dirac and from that I plan to route the signal to the nanoDIGI 2x8 B.

Not a very elegant solution, but quality wise difficult to beat, huh? I am still open to ideas however.

I now also just put my order in for the dac8 PRO, I hope it will come in a few months..

What is your use case? Implementing a 4 way active crossover in the nanodigi with stereo Dirac correction in the DDRC-22D?

Any reason you are particularly worried about the 48 kHz sample rate in the DDRC-88D?

Seems to me if you are worrying about details like 48 kHz vs 96 kHz sample rate you should also worry about the double ASRC you will have with the DDRC-22D and the nanoDIGI :).

Michael
 

Machismo

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What is your use case? Implementing a 4 way active crossover in the nanodigi with stereo Dirac correction in the DDRC-22D?

Any reason you are particularly worried about the 48 kHz sample rate in the DDRC-88D?

Seems to me if you are worrying about details like 48 kHz vs 96 kHz sample rate you should also worry about the double ASRC you will have with the DDRC-22D and the nanoDIGI :).

Michael

Yes that's the use case.

Well, I was thinking that if the source is 96khz and that goes to DRC-88D, it's converted to 48khz for internal processing and then perhaps back to 96 on the output, it's better to keep it at 96 all along? But since I haven't designed these devices I don't of course know exactly know what is the optimum. So feel free to explain why the 48 conversion is better?

And also usually processing data at lower resolution is worse, since it's not just about simple sample rate conversion then. It feels unlikely that the Dirac processing is identical when done in much lower 48khz resolution. As a matter of fact I should ask this straight from Dirac to find out.
 

mdsimon2

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Yes that's the use case.

Well, I was thinking that if the source is 96khz and that goes to DRC-88D, it's converted to 48khz for internal processing and then perhaps back to 96 on the output, it's better to keep it at 96 all along? But since I haven't designed these devices I don't of course know exactly know what is the optimum. So feel free to explain why the 48 conversion is better?

And also usually processing data at lower resolution is worse, since it's not just about simple sample rate conversion then. It feels unlikely that the Dirac processing is identical when done in much lower 48khz resolution. As a matter of fact I should ask this straight from Dirac to find out.

I was giving you a bit of hard time but I did want to point out that each miniDSP will asynchronously convert the incoming signal. So if you feed the OpenDRC-22D a 96 kHz signal you will get a 96 kHz signal out but that 96 kHz output signal will have been converted based on the clock in the OpenDRC-22D. Same thing will happen in the nanoDIGI. Some people claim that these conversions will degrade the sound but I have not personally found that to be the case.

Similarly I think the difference between processing at 48 kHz and 96 kHz is not a big deal. If anything I prefer 48 kHz as Dirac will have a lot more FIR processing power especially at low frequencies. You will get more high frequency warping at the lower sample rate but to me the benefit of Dirac is much more at lower frequencies.

Of course the 88D and the 22D are quite a bit different in that the 88D can do multichannel correction so the 22D may actually have more power devoted to 2 channels than the 88D.

Either way I think the DDRC-88D or a DDRC-22 (or SHD Studio) + nanoDIGI are great options.

Michael
 

Machismo

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I was giving you a bit of hard time but I did want to point out that each miniDSP will asynchronously convert the incoming signal. ...

No problem, it was good to me understand. I red about it and found out that it's usually done ASRC way mostly to avoid jitter, which can create some quality problems too.
So now I figured that the higher and more constant I can keep the khz rate through the chain of all devices, less the original music signal changes in percentage due to the small random noise that the ASRC process always creates by not being bit perfect.

Also since converting 96 => 48 => 96 in any case also includes the ASRC, to me it's logical that then I should at least avoid the frequency conversion part.

Btw, when my Umik-2 will arrive, I'll have a handy way to check from the speaker if the response goes to, for example 25 khz or over. If it does, then the whole chain must be operating above 48 khz and everything is in order. I already have the Umik-1, but it cannot not measure so high.
We all know how easily one setting can be accidentally wrong, so I am really looking forward to this.

I will of course also try to reduce the number of devices in the future as new hardware come out, but now I will have to go with the SHD Studio + nanodigi + Okto combination. I cancelled the order of 22D, since Studio has the same internal 96 khz resolution and added features which might come handy later. Also with SHD studio I don't have to use the computer soundcard since the Studio seems to have usb streaming. I am guessing that leaves out one ASRC compared to the other digital inputs. So right away this conversation has started to bear fruit regarding optimizing the signal path...

I've used Dirac for years in software form on Windows and it has a big impact also on the sound on the mid and high frequencies.
I certainly would not approve any kind of audible warping if there is such a thing. In software version I never noticed anything like that and I always ran it full range. Only thing that was a bit weird, was how the UMIK-1 compared to my traditional analog high quality measurement mic. I felt Umik was not so good in results. And now, behold, the UMIK-2 came out. Very interesting to see if that works better, or was it just my imagination.

I also sent the question to Dirac in Sweden about the 48 vs 96 khz internal processing, they now answered that they have got two kinds of opinions, some people notice and some don't notice any difference. Well that of course is true on almost any audio device out there... I was hoping to hear more kind of an engineering aspect with measurement etc. results.
However, if one device has to process 8 channels and the other only 2, there might be a difference in delay or some other parameter, since there is more work to do. Hard to say, but probably certainly there is no beneficial effect, so I will be on the safe side with just 2 channels.

FYI, dac8 PRO order 208 shipped today.

Allright, then there is over hundred orders waiting.
 
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mdsimon2

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No problem, it was good to me understand. I red about it and found out that it's usually done ASRC way mostly to avoid jitter, which can create some quality problems too.
So now I figured that the higher and more constant I can keep the khz rate through the chain of all devices, less the original music signal changes in percentage due to the small random noise that the ASRC process always creates by not being bit perfect.

Also since converting 96 => 48 => 96 in any case also includes the ASRC, to me it's logical that then I should at least avoid the frequency conversion part.

Btw, when my Umik-2 will arrive, I'll have a handy way to check from the speaker if the response goes to, for example 25 khz or over. If it does, then the whole chain must be operating above 48 khz and everything is in order. I already have the Umik-1, but it cannot not measure so high.
We all know how easily one setting can be accidentally wrong, so I am really looking forward to this.

I will of course also try to reduce the number of devices in the future as new hardware come out, but now I will have to go with the SHD Studio + nanodigi + Okto combination. I cancelled the order of 22D, since Studio has the same internal 96 khz resolution and added features which might come handy later. Also with SHD studio I don't have to use the computer soundcard since the Studio seems to have usb streaming. I am guessing that leaves out one ASRC compared to the other digital inputs. So right away this conversation has started to bear fruit regarding optimizing the signal path...

I've used Dirac for years in software form on Windows and it has a big impact also on the sound on the mid and high frequencies.
I certainly would not approve any kind of audible warping if there is such a thing. In software version I never noticed anything like that and I always ran it full range. Only thing that was a bit weird, was how the UMIK-1 compared to my traditional analog high quality measurement mic. I felt Umik was not so good in results. And now, behold, the UMIK-2 came out. Very interesting to see if that works better, or was it just my imagination.

I also sent the question to Dirac in Sweden about the 48 vs 96 khz internal processing, they now answered that they have got two kinds of opinions, some people notice and some don't notice any difference. Well that of course is true on almost any audio device out there... I was hoping to hear more kind of an engineering aspect with measurement etc. results.
However, if one device has to process 8 channels and the other only 2, there might be a difference in delay or some other parameter, since there is more work to do. Hard to say, but probably certainly there is no beneficial effect, so I will be on the safe side with just 2 channels.



Allright, then there is over hundred orders waiting.

USB input will still convert everything to 96 kHz (DSP runs at 96 kHz) so you will still have an ASRC. That being said the SHD Studio is great (I have one myself) and I think it makes a lot of sense to choose it over the DDRC-22D. You get a newer generation SHARC DSP running at a high clock rate, more inputs/outputs and a display for not much of an increase in price.

As you mentioned the SHD Studio is essentially treated as a sound card for your computer. One of the under appreciated implications of this is that it can be used to record the frequency response after Dirac is applied which allows you to know how much to cut to avoid digital clipping. This is very important if you are using downstream volume control in the Okto.

Michael
 

Machismo

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USB input will still convert everything to 96 kHz (DSP runs at 96 kHz) so you will still have an ASRC.

Ok, I am not familiar how usb soundcards function. In any case, I already have an usb soundcard with digital out, so at least I can leave that one out.

But I hope there is an 96 khz output format selectable in windows sound settings for the SHD Studio when connected with usb?
 

mdsimon2

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Ok, I am not familiar how usb soundcards function. In any case, I already have an usb soundcard with digital out, so at least I can leave that one out.

But I hope there is an 96 khz output format selectable in windows sound settings for the SHD Studio when connected with usb?

I am Mac user so cannot help with anything Windows related. In Mac you can choose a variety of sample rates from 44.1 kHz to 192 kHz (see attached), I assume Windows has something similar. That being said whatever you choose the SHD is just going to convert to it's own 96 kHz internally. I leave it at 96 kHz / 24 bit and don't worry about it.

Michael
 

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DWPress

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FYI, dac8 PRO order 208 shipped today.

And just 2 weeks ago Pavel told me my order #222 was shipping out in 2 weeks. :rolleyes:

I'm getting the Okto specifically to get far away from Minidsp. I've had the 4x10 for almost 10 years doing XO duties feeding it a 96kHz digital signal to bypass the ADC but then it performs another digital conversion anyway. I ran it with analog signals in too and honestly couldn't complain about sound quality either way.

However, the 4x10 died earlier this summer which was rather traumatic since I need 8 channels and XO somehow to have music in my studio building. So while I was waiting for the replacement board to come from Hong Kong I cobbled together a little Windows 10 box and executed the XO in JRiver via the procedural EQ in addition to the 2 channel convolution filters I usually use for DRC all output to a crappy ASUS Xonar U7 MKII.

It sounded great.

The 4x10 is only capable of IIR filters for its XO and paraEQ duties, comparable to what I did in JRiver procedural EQ I'd think. The huge difference I think is the amount of headroom the MiniDSP sucks up merely by turning it on. Add on top of it a level of DRC in FIR filters which will naturally take their share of headroom as well and pretty soon I'm turning the amps up more which is just making a lot of amps run harder.

It all sounds good and measures well enough but I'm looking forward to the Okto and doing all the XO and DRC purely via software on the computer and be free to use whatever sample rate I want with unlimited taps for filters + the Okto has a big honkin volume knob with a display and I can use it so many different ways!

I'm sure the SHD and DDRC line are a step up from the 4x10 but there are better ways to add Dirac or any other DSP to a signal if you're willing to use a computer in the chain and it can be implemented systemwide either in Windows, Linux or Mac.
 

phoenixdogfan

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"I cobbled together a little Windows 10 box and executed the XO in JRiver via the procedural EQ".

Not sure what you mean by "the procedural EQ" I know JRiver has a parametric EQ function in its DSP stack, did you mean that? It also offers a crossover function in the room correction part of that stack. Did you build the xover from the parametric EQ and why not use the crossover in the room correction? Are there any limitations to the room correction crossover function? And do you think JRiver can do that function as well as you were able to do it with the miniDSP interface?

I'm also have an Octo 8 Pro on order and am looking to maybe ditch my miniDSP 2 x 4 HD.

I am wondering if the simplest and cheapest way for me to do it is to replace miniDSP unit with JRiver, provided, of course, JRiver is up to the job.
 
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DWPress

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My mistake, I'm using Roon now and didn't fire up JRiver to confirm my process. Yes, XO and EQ with the regular parametric EQ.

The Room Correction part of the DSP stack is limited in XO slopes and frequency selection. I believe it is more targeted for multichannel audio or video separation of LF content from other speakers. There are quite a few examples of using the paraEQ for XO if you search around a bit. If the options available in Room Correction are adequate for your needs then that might suffice but I needed to direct frequencies to 9 speakers and 12 channels.

JRiver can certainly do the job as good or better than the 2x4HD. So can Roon. Here's a pic for a taste of the type of routing you'd need to configure:
 

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phoenixdogfan

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My mistake, I'm using Roon now and didn't fire up JRiver to confirm my process. Yes, XO and EQ with the regular parametric EQ.

The Room Correction part of the DSP stack is limited in XO slopes and frequency selection. I believe it is more targeted for multichannel audio or video separation of LF content from other speakers. There are quite a few examples of using the paraEQ for XO if you search around a bit. If the options available in Room Correction are adequate for your needs then that might suffice but I needed to direct frequencies to 9 speakers and 12 channels.

JRiver can certainly do the job as good or better than the 2x4HD. So can Roon. Here's a pic for a taste of the type of routing you'd need to configure:
Thanks. My xovers will be limited to a LW at 100 hz to hi-pass my LS 50's, and a low-pass also at 100 hz for my sub, so very rudimentary. Also an 8 band parametric EQ to correct the listening window of the LS 50's.
 

dualazmak

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Hello DWPress and phoenixdogfan,

As you may know,,, in Windows 10 Pro 64 bit PC,

Using ASIO4ALL, VB Audio Hi-Fi Virtual Cable+ASIO Bridge and software crossover EKIO (very nice and simple GUI operation), we can achieve very much flexible XO-EQ-DSP with umlimited numbers of channels into DAC8PRO's 8 channels through DIYINHK ASIO USB driver using one single USB cable...

In this I/O configuration, the audio (audio visual) player can be JRiver, Roon, other playes, web browsers, etc., as far as they can output audio signal into ASIO4ALL or VB Audio Virtual Cable (as virtual sudio device) into EKIO; all the processing can be done in 192 kHz 24 bit.

I recently summarized again such full ASIO I/O configurations at here and here.

If you would like to use another single DAC, while still using EKIO's crossover, EKIO can easily direct all the output channels into one single DAC using the specific ASIO driver, just like I frequently do so when I would like to use OPPO SONICA DAC or KORG DS-DAC-10 in my living room and/or my office upstairs on other PCs + audio system.
 
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phoenixdogfan

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Hello DWPress and phoenixdogfan,

As you may know,,, in Windows 10 Pro 64 bit PC,

Using ASIO4ALL, VB Audio Hi-Fi Virtual Cable+ASIO Bridge and software crossover EKIO (very nice and simple GUI operation), we can achieve very much flexible EQ-DSP with umlimited numbers of channels into DAC8PRO's 8 channels through DIYINHK ASIO USB driver using one single USB cable...

In this I/O configuration, the audio (audio visual) player can be JRiver, Roon, other playes, web browsers, etc., as far as they can output audio signal into ASIO4ALL or VB Audio Virtual Cable (as virtual sudio device) into EKIO; all the processing can be done in 192 kHz 24 bit.

If you would like to use another single DAC, while still using EKIO's crossover, EKIO can easily direct all the output channels into one single DAC using the specific ASIO driver, just like I frequently do so when I would like to use OPPO SONICA DAC or KORG DS-DAC-10 in my living room and/or my office upstairs on other PCs + audio system.
I'm exploring that option as well. I'm a little bit confused on whether it calls for ASIO4ALL AND VB Audio Hi_Fi cable or whether it calls for ASIO4ALL OR VB Audio HiFI cable. Is it one of the other or both softwares and if both why?
 
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