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Okto 8 Owner’s Thread

Dueprocess

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This is such a waste of money for a 2 channel system.

Michael
Personally, I agree. I don't use any MiniDSP products (and prefer USB, especially considering measurements show it is cleaner). That said, it's a far more simplistic option than the OP's suggestion (SHD Studio x3) and leaves them tied into the MiniDSP ecosystem, which they seem to have an attachment to.
 

Machismo

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Multiple SHD Studios will not work with the Okto. The Okto requires that all AES inputs be on the same clock domain and each SHD Studio will ASRC using a local clock.

For a stereo active system the best way to go is to do DSP in software (see link in my signature for how to do this with a RPi4). The Okto is literally built for this with the AES / USB mode. In such a setup using a single SHD Studio upstream of the Okto does solve a lot of issues, namely it gives you source selection, Dirac and ASRCs all inputs to 96 kHz so you can run your DSP software at a constant sample rate.

Michael

Good point, I didn't think about the clock issue.

DDRC88D might work yes. For me it would not be a waste of money, if it would work perfectly and anyway I think money is just a tool which has no value on it's own and therefore cannot be 'wasted'. On the other hand, bombing a city to ashes or burning high end speakers is quite a wasteful, in my opinion.

But since I use my system also with deejays playing live at my home with vinyl records, adding more milliseconds with DDRC88D would not be good. For example Minidsp 4x10 alone is 1.7ms which is still fine, but when I had Shd Studio with dirac on + 4x10 the combined delay of those two is 13ms which dj right away noticed. I don't use separate monitor speakers for deejays as they are in the front of the main system. That way the sound quality is also higher to the guests, since the sound don't come also from the monitors at the same time.

That's why I usually bypass the Shd in those situations. However I would not be able to bypass the DDRC88D, otherwise there would be no sound at all. Only by buying the DDRC88D I would know how much it adds delay compared to 4x10. Okto I have not yet measured either.

You mentioned DSP in software, what kind of delays does that add, any rough estimate?
 

mdsimon2

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Good point, I didn't think about the clock issue.

DDRC88D might work yes. For me it would not be a waste of money, if it would work perfectly and anyway I think money is just a tool which has no value on it's own and therefore cannot be 'wasted'. On the other hand, bombing a city to ashes or burning high end speakers is quite a wasteful, in my opinion.

But since I use my system also with deejays playing live at my home with vinyl records, adding more milliseconds with DDRC88D would not be good. For example Minidsp 4x10 alone is 1.7ms which is still fine, but when I had Shd Studio with dirac on + 4x10 the combined delay of those two is 13ms which dj right away noticed. I don't use separate monitor speakers for deejays as they are in the front of the main system. That way the sound quality is also higher to the guests, since the sound don't come also from the monitors at the same time.

That's why I usually bypass the Shd in those situations. However I would not be able to bypass the DDRC88D, otherwise there would be no sound at all. Only by buying the DDRC88D I would know how much it adds delay compared to 4x10. Okto I have not yet measured either.

You mentioned DSP in software, what kind of delays does that add, any rough estimate?

Latency depends on what chunk size you use and whether you use resampling and whether or not you are doing anything that adds additional latency like FIR filters. One of the nice things about the Okto is that in AES / USB mode it has an internal rate adjust functionality to bridge the clock domain from the AES input and the clock domain of the internal oscillators at the DAC so no resampling is required in the DSP software.

Without resampling, 96 kHz sample rate, 2048 chunk size and a RPi4 latency is 20-30 ms IIRC. Can get lower with a lower chunk size but like the Dirac enabled miniDSPs such as the DDRC-88D and SHD it will never be low enough for live use. For reference latency on the DDRC-88D is also 13 ms like the SHD -> https://www.minidsp.com/support/for...-couple-of-questions-before-buying-a-ddrc-88d.

The DDRC-88D is great for the application it was built for, multichannel Dirac. But IMO using it in a stereo active system and not using Dirac doesn't make much sense when you can get much more processing power at higher sample rates with a sub-$100 RPi running CamillaDSP.

Honestly sounds like the nanoDIGI is a better fit for you. Will have much lower latency (should be similar to the 4X10HD) and it runs at 96 kHz compared to the DDRC-88D 48 kHz which is beneficial when using IIR filters. For 6 channels it works great and on previous Okto firmwares 8 channels worked well. I never reached out to Pavel directly via e-mail (only tagged him on here without response) so it is possible he can set you up with a firmware that works with 8 channels, although to be honest I wouldn't count on it as their support is a bit lacking.

Michael
 

dualazmak

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Latency, delay, synchronization, time alignment are always critically important in multichannel DAC system with DSP at upstream digital domain within PC.

Please note that my discussion here is limited to pure audio-only system, and excluding audio-visual system where you need "time alignment adjustment" not only for the SPs but also for visual images/movies.

In the audio digital signal processing, we have so many buffers or latencies; JRiver output buffer, ASIO4ALL's I/O buffers, DSP's (in my case EKIO's) processing buffer, DIYINHK USB ASIO driver's buffer, and so on. It is not straightforward, therefore, to exactly measure the "absolute delay" between the JRiver's "shout" and the final air sound kick-up by SP.

As for the synchronization between the channels...
In my system using OKTO DAC8PRO (firmware 1.32) and DSP software EKIO feeding the cross-overed (EQ-ed) 8 channels into DAC8PRO through DIYIHNK USB ASIO multichannel driver, I have no synchronization issue; all the 8 channels are in complete sync.

I usually set all the buffers in the digital domain in rather large size, so that I should not have any latency or delay problems; in our audio setup, we have no problem at all if all the bunch of the digital and analog signal (15Hz - 30 kHz) have identical amount of delay time from the signal origin at music player such as JRiver, and this is always the case in our digital (PC based) audio system.

The relative delay between the sound of SP units, or "time alignment" in multiple SPs, however, is always one of the critical issues in audio system, especially in the multichannel multi-driver multi-amplifier system, as you may agree.

Even though some of the advanced audio measurement software, like REW and Equalizer APO, feature some kind of "delay and/or time alignment" measurement(s) (using microphone) between the SP units, the internal procedure for the measurement would be somewhat "black-box", and if it includes "absolute delay measurement" type approach, the given "delay and/or time alignment" results are not always accurate and stable; this issue have been well pointed by @zerxia in his post here and here. He wrote, "Through this test, I found that the channel delay displayed by the Equalizer APO is not accurate."

Consequently, we need to have reliable reproducible high-precision (0.1 msec to 1 msec precision) measurement method(s) for the relative delay (time alignment) by using only the air recorded room sound with measurement microphone together with reliable audio interface (ADC) of fixed latency.

Having the above mentioned issues in my mind, I recently developed rather primitive but really accurate reproducible measurement method for time alignment between the SP units by using only the air recorded sound analyzed with Adobe Audition 3.01 (or Audacity, if you like).
If you would be interested, please visit these posts on my project thread;
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507

Please simply PM me, if you would be seriously interested in using the test tone signals I prepared and used in my above time alignment measurements.
 
Last edited:

dartinbout

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Hey all you "Ocktopi"! Here's a cautionary tale that might allow you to go back to listening to music on this genius of the deep. I change the connection on my Nvidia card to HDMI (for 60hz refresh). The Okti went silent and would not acknowledge any signal. I tried a mess of things but I had to uninstall the aforementioned DIY ASIO Driver, boot, then re-install. You would think I would scream "Viola" but no0000...the routing disappeared and would not allow me to manually change it (center going to rt channel, sub to rt rear, etc). I had to reset the unit to default for the routing to return. I had recorded to the volume settings for each channel before (pessimistic IT person that I am). No back to your regularly sdheduled programming.
 

dartinbout

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Latency, delay, synchronization, time alignment are always critically important in multichannel DAC system with DSP at upstream digital domain within PC.

Please note that my discussion here is limited to pure audio-only system, and excluding audio-visual system where you need "time alignment adjustment" not only for the SPs but also for visual images/movies.

In the audio digital signal processing, we have so many buffers or latencies; JRiver output buffer, ASIO4ALL's I/O buffers, DSP's (in my case EKIO's) processing buffer, DIYINHK USB ASIO driver's buffer, and so on. It is not straightforward, therefore, to exactly measure the "absolute delay" between the JRiver's "shout" and the final air sound kick-up by SP.

As for the synchronization between the channels...
In my system using OKTO DAC8PRO (firmware 1.32) and DSP software EKIO feeding the cross-overed (EQ-ed) 8 channels into DAC8PRO through DIYIHNK USB ASIO multichannel driver, I have no synchronization issue; all the 8 channels are in complete sync.

I usually set all the buffers in the digital domain in rather large size, so that I should not have any latency or delay problems; in our audio setup, we have no problem at all if all the bunch of the digital and analog signal (15Hz - 30 kHz) have identical amount of delay time from the signal origin at music player such as JRiver, and this is always the case in our digital (PC based) audio system.

The relative delay between the sound of SP units, or "time alignment" in multiple SPs, however, is always one of the critical issues in audio system, especially in the multichannel multi-driver multi-amplifier system, as you may agree.

Even though some of the advanced audio measurement software, like REW and Equalizer APO, feature some kind of "delay and/or time alignment" measurement(s) (using microphone) between the SP units, the internal procedure for the measurement would be somewhat "black-box", and if it includes "absolute delay measurement" type approach, the given "delay and/or time alignment" results are not always accurate and stable; this issue have been well pointed by @zerxia in his post here and here. He wrote, "Through this test, I found that the channel delay displayed by the Equalizer APO is not accurate."

Consequently, we need to have reliable reproducible high-precision (0.1 msec to 1 msec precision) measurement method(s) for the relative delay (time alignment) by using only the air recorded room sound with measurement microphone together with reliable audio interface (ADC) of fixed latency.

Having the above mentioned issues in my mind, I recently developed rather primitive but really accurate reproducible measurement method for time alignment between the SP units by using only the air recorded sound analyzed with Adobe Audition 3.01 (or Audacity, if you like).
If you would be interested, please visit these posts on my project thread;
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507

Please simply PM me, if you would be seriously interested in using the test tone signals I prepared and used in my above time alignment measurements.
Thanks "Dual" for your efforts to improve our use of this beast. I do have a question in regards to everybody's experience of multichannel music. I have over 7k of DSD128 and under multichannel releases. I have been playing in this world for a while with a variety of system setups. Am I the only who noticed that the mixing engineers don't seem to give a dookie for our adjustments and set channel levels any darn way they please? I find myself needing to make changes per album and what ever drugs the engineer was taking that day. Anybody else have this issue or do I just need to take the same drugs they were taking in the booth that day.

 

Kal Rubinson

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Am I the only who noticed that the mixing engineers don't seem to give a dookie for our adjustments and set channel levels any darn way they please?
Not that I've found but it may have to do with your choice of repertoire and with personal preference. I have stumbled on a few that benefitted from readjustment but that's uncommon.

P.S.: You say you have 7k of DSD128. 7k what? Tracks, albums, TB?
 

dualazmak

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Thanks "Dual" for your efforts to improve our use of this beast. I do have a question in regards to everybody's experience of multichannel music. I have over 7k of DSD128 and under multichannel releases. I have been playing in this world for a while with a variety of system setups. Am I the only who noticed that the mixing engineers don't seem to give a dookie for our adjustments and set channel levels any darn way they please? I find myself needing to make changes per album and what ever drugs the engineer was taking that day. Anybody else have this issue or do I just need to take the same drugs they were taking in the booth that day.

First of all, let me inform you that now I usually play all the music tracks in 88.2 kHz (or 96 kHz) by on-the-fly conversion by JRiver (or Roon) into software XO/EQ EKIO (capable of up to 192 kHz) for crossover/EQ processing, even though my digital music library of about 25,000 files consists of mixture of various formats;
  • 16-bit/44.1kHz CD ripped non-compressed aif (majority!),
  • 24-bit/192kHz down-sampled or up-sampled aif,
  • 24-bit/96kHz flac,
  • 24-bit/192kHz flac,
  • 1-bit/DSD64(1x) 2.8MHz dsf,
  • 1-bit/DSD128(2x) 5.6 MHz dsf,
  • 1-bit/DSD256(4x) 11.2 MHz dsf,
You would please refer to my specific post here for the detailed reasons and background for "now play all in 88.2 kHz or 96 kHz".

OK, regarding your inquiry on "relative gain adjustments" for the crossover-ed channels in various music tracks (if I understand correctly of your point), I essentially agree with above Kal's comment, but I sometimes flexibly re-adjust the relative gains depending on the genre repertoire my-personal-preference and/or characteristics of the recording which you mean the preference of the recording engineers.

Here "the flexible relative gain control" in my setup is for five Fq zones/channels covered by sub-woofers (15 Hz - 50 Hz), woofers (45 Hz - 600 Hz), Be-squawkers (600 Hz - 6 kHz), Be-tweeters (6 kHz - 25 kHz) , and metal-horn-super-tweeters (8.8 kHz - 25 kHz), each of them is directly driven by dedicated amplifier.

In my setup, to enable "safe and flexible relative gain control" on-the-fly is one of the main reasons for my utilization of four HiFi "integrated amplifiers" and active sub-woofers with gain/volume control. (Please refer to my post here for the general aspects of my amplifier selections.)

As you may agree with me that I do not like to flexibly control/adjust software XO/EQ EKIO's relative gain controllers on-the-fly (even though EKIO is fully capable of on-the-fly relative gain adjustment) avoiding possible mis-adjustment for damages to SP drivers.

Consequently, I usually "keep" the relative gains in EKIO so that giving my "best tuned" total Fq response like;
WS002906 (1).JPG


and the further on-the-fly flexible fine tuning of the relative gain controls, only if needed, can be given by the four integrated amplifiers and active sub-woofers each of them has IR remote controller for flexible on-the-fly volume/gain control, as I shared one typical example situation;
- A serious jazz fanatic friend came to my home for audio sessions using my multichannel multi-driver multi-way multi-amplifier stereo system: #438

This diagram also shared there will give you proper understandings on the flexible fine tuning of the relative gains on-the-fly in my setup;
WS002476 (1).JPG




After my recent DIY installation of 12-VU-Meter Array, now I can monitor the actual room sound gains of all the SP drivers by the nice VU meters;
WS003854.JPG


Just like Kal's stance, you would please understand that I (we) seldom re-adjust the relative gains out of my preferable "best tuned" configuration shown in the above diagrams; even in the rare cases of re-adjustments, the fine tuning would be usually within plus/minus 6 dB for SP drivers.

Also, please note that I use the flexible gain controls by TA-A1ES (driving Be-tweeters) and A-S301 (driving super-tweeters) for compensating the high-Fq hearing abilities/slight-disabilities of audience(s) including myself as shared in the latter half of my post here and here in detail.
 
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