Review and Measurements of Okto DAC8 8Ch DAC & Amp

[JohnYang1997]

The noise at the speakers will be 2.6uV x closed loop amplifier gain (30x for example). So, the noise will be aprox. 80uV at the acoustic transducer. You can hear that noise only if you stick the ear by the tweeter. It is very weak level of the noise.

Yep, if there is no other attenuation along the signal chain.

 

[777]

Yep, if there is no other attenuation along the signal chain.

Most probably the filters are high-pass and low-pass type, beyound the audio-band, not inside, so the noise from audio-band will evolve directly proportional to the signal.

Anyway, there is an interesting thing to see if at tinny level of noise voltage, the electrical efficiency remain at the same high level, if not most of the current will be transformed in a heat into the coil, not in a mechanical motion due to inert mass and the magnetic losses.

 

[JohnYang1997]

Most probably the filters are high-pass and low-pass type, beyound the audio-band, not inside, so the noise from audio-band will evolve directly proportional to the signal.

Anyway, there is an interesting thing to see if at tinny level of noise voltage, the electrical efficiency remain at the same high level, if not most of the current will be transformed in a heat into the coil, not in a mechanical motion due to inert mass and the magnetic losses.

People ain't play at full volume right? I was just assuring we were talking about pure digital volume control. If there's a good preamp or the power amp has gain switch or it's an integrated, the noise will be suppressed along the audio signal.

 

[pos]

Choosing the proper amp and/or gain setting is the most important factor: most DAC have a wide enough dynamic range nowadays, so it is just a matter of aiming low enough for the noise (eg 10dB SPL at the listening position) and high enough for max peaks at 0dBFS (eg 110dB SPL at the listening position).
Active crossovers with compression drivers and horns complicate the matters as CD EQ will make you loose dynamic range by lowering the max achievable SPL while leaving the residual noise unchanged...

 

[gags11]

Active crossovers with compression drivers and horns complicate the matters as CD EQ will make you loose dynamic range by lowering the max achievable SPL while leaving the residual noise unchanged...

I am still trying to learn. Could you elaborate the above statement a little? Do you mean using active crossover will result in dynamic range loss in general, or do you mean digital active crossover with parametric equalization?

 

[pos]

When you apply EQ in a digital crossover you have to make sure the signal will not exceed 0dBFS, which mean you have to either only use negative EQs or lower the overall level after applying EQ.
The worst case scenario is constant directivity compensation in a horn system, where you will typically need to raise HF levels by some 10 to 15dB depending on the driver/horn combo. That means the signal around 1 or 2kHz will need to be 10dB to 15dB lower than in the top octave, ie -10dBFS to -15dBFS.

Here is the example of the JBL M2 EQ curve:

By doing so you effectively lower the maximum level you can output but keep the residual noise (noise floor) unchanged, which means you lowered the available dynamic range.
So now let's say you have 110dB of dynamic range in your DAC. With a passive system and a properly chosen amp gain (and assuming no other source of noise in the system, especially the amp) that means that you could possibly achieve, as an example, 0dB SPL at -inf dBFS (noise floor) and 110dB SPL at 0dBFS (max peak level).
But now if you need to apply 15dB of EQ you only have 95dB left, which means you have to either raise your noise floor or lower your max SPL.
And that example assumes a perfectly matched gain on the amp (ie sensitivity = 0dBFS, and an amp with good DNR too).

This is where DACs with very high DNR come in handy.

Of course in a real world scenario in the case of constant directivity compensation things are not that bad, because your system will never have to reproduce a 0dBFS signals in the top octave thanks to the power distribution of music, so you can allow your digital EQ to go past 0dB there...

 

[vitalii427]

When you apply EQ in a digital crossover you have to make sure the signal will not exceed 0dBFS, which mean you have to either only use negative EQs or lower the overall level after applying EQ.
The worst case scenario is constant directivity compensation in a horn system, where you will typically need to raise HF levels by some 10 to 15dB depending on the driver/horn combo. That means the signal around 1 or 2kHz will need to be 10dB to 15dB lower than in the top octave, ie -10dBFS to -15dBFS.

Here is the example of the JBL M2 EQ curve:

By doing so you effectively lower the maximum level you can output but keep the residual noise (noise floor) unchanged, which means you lowered the available dynamic range.
So now let's say you have 110dB of dynamic range in your DAC. With a passive system and a properly chosen amp gain (and assuming no other source of noise in the system, especially the amp) that means that you could possibly achieve, as an example, 0dB SPL at -inf dBFS (noise floor) and 110dB SPL at 0dBFS (max peak level).
But now if you need to apply 15dB of EQ you only have 95dB left, which means you have to either raise your noise floor or lower your max SPL.
And that example assumes a perfectly matched gain on the amp (ie sensitivity = 0dBFS, and an amp with good DNR too).

This is where DACs with very high DNR come in handy.

Of course in a real world scenario in the case of constant directivity compensation things are not that bad, because your system will never have to reproduce a 0dBFS signals in the top octave thanks to the power distribution of music, so you can allow your digital EQ to go past 0dB there...

Very informative, thanks!

 

[gags11]

When you apply EQ in a digital crossover you have to make sure the signal will not exceed 0dBFS, which mean you have to either only use negative EQs or lower the overall level after applying EQ.
The worst case scenario is constant directivity compensation in a horn system, where you will typically need to raise HF levels by some 10 to 15dB depending on the driver/horn combo. That means the signal around 1 or 2kHz will need to be 10dB to 15dB lower than in the top octave, ie -10dBFS to -15dBFS.

Here is the example of the JBL M2 EQ curve:

By doing so you effectively lower the maximum level you can output but keep the residual noise (noise floor) unchanged, which means you lowered the available dynamic range.
So now let's say you have 110dB of dynamic range in your DAC. With a passive system and a properly chosen amp gain (and assuming no other source of noise in the system, especially the amp) that means that you could possibly achieve, as an example, 0dB SPL at -inf dBFS (noise floor) and 110dB SPL at 0dBFS (max peak level).
But now if you need to apply 15dB of EQ you only have 95dB left, which means you have to either raise your noise floor or lower your max SPL.
And that example assumes a perfectly matched gain on the amp (ie sensitivity = 0dBFS, and an amp with good DNR too).

This is where DACs with very high DNR come in handy.

Of course in a real world scenario in the case of constant directivity compensation things are not that bad, because your system will never have to reproduce a 0dBFS signals in the top octave thanks to the power distribution of music, so you can allow your digital EQ to go past 0dB there...

Thank you for the explanation!
I do use an active system, albeit an analog active crossover from Marchand electronics. Now I do apply EQ in the bass frequencies in Roon, and decrease the signal by 4dBFS so Roon does not clip. But this is before the signal gets to the dac, so dynamic range in theory should not be affected.

However, I do use 18db attenuator boards within the analog active crossover for low and high frequency channels to match the midrange woofer, since it is considerably less efficient. Phil Marchand told be that this atenuation is a good thing, as it increases the SNR. Is that true? Will this affect dynamic range?

Thanks again!

PS: My understanding for your above explanation is that it applies to digital domain, where decreasing dBFS does not change the noise floor, ultimately resulting in less dynamic range. But decreasing db level in the analog domain, also decreases the noise floor, preserving the overall dynamic range.

 

[MZKM]

Just heard back from Okto regarding surround sound:

neither of them can decode Dolby/DTS. The USB input of both units is a
standard USB Audio Class 2.0, stereo in case of DAC8 DSP and 8-channel
in case of DAC8 PRO.

So, if you can feed it LPCM, then you can do 7.1 surround sound on the DAC8 Pro.

 

[Panelhead]

Can the unit be configured as a four channel dac with two channels paralleled for each? I use four channel now for mains and subs.
The volume controller build into dac chips is SOTA for multichannel balanced output applications. Analog solutions for eight channel balanced outputs would never deliver the tracking of the dac chip FREE volume control.

 

[maxidcx]

When it comes to DNR vs volume control and potential EQ, Pos explanation is clear and important, I would add that in home listening most of us will play with a volume range between -48db (night listening) to 0dbfs (critical listening session with hi dynamic music like classical), this is to be taken into account as well. the more DNR the better !
my personal approach is to use ultraslow noise amplifier typically the hypex Ncore (which are given between 13uVolt and 30uVolt depending on their preamp gain) and to reduce their gain by 12db by clamping one of the resistor in their preamp as per an application note available on hypex website.
So yes digital volume control is fine with Sabre but don't hesitate to make -12db or -24db switch somewhere in the analog domain or within the amplifier

 

[Panelhead]

I have several DACs with the digital volume control. The attenuation levels have varied, do not remember ever going higher than about -24dB. That was loud.
The speakers I use now are higher efficiency. Offset gain some in the crossover.
Using the digital controller at 0dB is normally extremely high. Most amplifiers would not like seeing the four volt potential input. Low gain amp would help. The Benchmark amp comes to mind.

 

[etc6849]

Great product! Are you considering a rack mount option for the DAC8 Pro unit?

Thanks!

There are two 8-channel inputs on the DAC8 PRO that can be switched between - USB and 4xAES/EBU. With USB input selected, it works as an 8 channel USB interface with one to one channel mapping, with AES/EBU selected, each AES/EBU input corresponds to one output channel pair. The lower price of the PRO version is due to the lack of Raspberry Pi and the streaming capabilities (as there is not much need for it in a pro/semi-pro environment).

 

[amirm]

Great product! Are you considering a rack mount option for the DAC8 Pro unit?

Thanks!

I either saw it here or on DIY forum that rack ears will be available.....

 

[etc6849]

Thanks! I am thinking about using 3 of them in my tri-amped 5.1 theater... I would use Acourate for FIR filtering and time alignment to replace three Xilica XD4080 units (each has 4 AES in, 8 analog out) driving 10 AHB2 amps (5 in mono, 5 in stereo) and 5 JL Audio subs.

Looks like the DAC8 Pro would work with the RME HDSPe AES-32 PCI-E card I am using (16 channels in and out). I'd either have to see if I can add another AES-32 card or just run the subs off their own DSP. I believe RME's TotalMix software should integrate the cards together.

My sound is so good, I am hesitant to mess with my setup, but the cost and performance of these 8 channel DACs is very tempting. They would have saved me money had they been out before.

I either saw it here or on DIY forum that rack ears will be available.....

 

[vitalii427]

Thanks! I am thinking about using 3 of them in my tri-amped 5.1 theater... I would use Acourate for FIR filtering and time alignment to replace three Xilica XD4080 units (each has 4 AES in, 8 analog out) driving 10 AHB2 amps (5 in mono, 5 in stereo) and 5 JL Audio subs.

Looks like the DAC8 Pro would work with the RME HDSPe AES-32 PCI-E card I am using (16 channels in and out). I'd either have to see if I can add another AES-32 card or just run the subs off their own DSP. I believe TotalMix should integrate the cards together.

My sound is so good, I am hesitant to mess with my setup, but the cost and performance of these 8 channel DACs is very tempting. They would have saved me money had they been out before.

Post few photos, pls!

 

[pos]

I either saw it here or on DIY forum that rack ears will be available.....

there: https://www.diyaudio.com/forums/ven...8-launch-tour-announcement-4.html#post5738383

 

[etc6849]

https://www.audiosciencereview.com/...nely-expensive-audio.4276/page-18#post-113479

Post few photos, pls!

 

[Sal1950]

https://www.audiosciencereview.com/...nely-expensive-audio.4276/page-18#post-113479

I'd forgotten your post there, that thread was going in so many directions at once. LOL
Anyway that's an incredible multich rig, congrat's on it's design and execution!

 

[Laserjock]

WiSA.

Also, if you do not insist on wireless, I am using a miniDSP U-DIO8 to send AES3 to the Dutch&Dutch 8c.

Much time with the D&D yet?
I found the read on them very interesting.

Chris