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SMSL SU-8 Version 2 Balanced DAC

Toku

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I got a pair of Genelec 8330s recently to hook up to this DAC with an XLR splitter. However I'm struggling a bit with the output levels. At max output there is a lot of distortion/clipping. I have to reduce the DAC down to like 25 before the distortion goes away. Any idea why? The 8330s are rated up to 24dbU (14 vRMS) input and this DAC supposedly outputs a max of 4 vRMS.
The first is the unit of dBu
0dBu = 0.775V / 600Ω. 1mW
This is the standard used by professional audio and DTM.
Originally consumer audio should use this standard, but there is currently no standard.

From the Genelec 8330s specification table
Maximum instantaneous sound pressure level > 104 SPL
RMS maximum sound pressure level > 96dB SPL
Proper input level -6dBu (100dB SPL @ 1m)
Max Input level (Maximum distortion rate, amplifier is not destroyed) 24dBu (12.4V) DTM may be used with the amplifier saturated in some cases.

Therefore, it is desirable to set the input level to -6dBu or less so that Audiophile can be used in the range where the sound is not distorted.
 
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svenz

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100db SPL at -6dBU is just the input sensitivity... not the max input the speakers can take. It works fine at max as long as I don't split the output. Why the split doesn't work is still puzzling to me.
 

svenz

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I'm just going to get a SPDIF->AES cable and use the digital inputs. I wanted less cables but it didn't work out :). I wonder if anyone else has had luck using XLR splitters on these to feed active speakers.
 

Toku

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I'm just going to get a SPDIF->AES cable and use the digital inputs. I wanted less cables but it didn't work out :). I wonder if anyone else has had luck using XLR splitters on these to feed active speakers.
I read your post again.

It turns out that the XLR output of SU-8 is connected to HPA and Genelec 8330s at the same time using an XLR splitter.
Perhaps both HPA and Genelec 8330s use a semiconductor electronic balance circuit like SU-8.
This is my experience of working in the design of professional audio equipment for broadcasters a long time ago.
If a semiconductor electronic balance circuit is connected in parallel to several different devices, interference may occur between the devices. In particular, the balance operation may be disrupted by potential differences, resulting in abnormal operation.
I think the abnormal behavior of your device is the same cause. Therefore, in this case, it is to make either one connection.
In my old design, I disliked this phenomenon and the balance circuit always adopted the transformer system.
 

dewdude

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I want to make a minor update to this DSD clicking.

Yes...apparently this SU-8 is emitting a very faint "pop" in the right channel just as music starts. It took me a while to realize it was happening every time. It's probably enhanced by the fact I use something to delay DSD audio to give the DAC time to initialize to a new stream.
 

Toku

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I want to make a minor update to this DSD clicking.

Yes...apparently this SU-8 is emitting a very faint "pop" in the right channel just as music starts. It took me a while to realize it was happening every time. It's probably enhanced by the fact I use something to delay DSD audio to give the DAC time to initialize to a new stream.
The SU-8 (Ver 1) had a fade-in operation at the start of playback to prevent pop noise when switching songs.
A post at that time can be found in Review & Measurements of SMSL SU-8 DAC # 139.
Was this fade-in operation deleted in SU-8 (Ver 2)?

#139
https://www.audiosciencereview.com/...ments-of-smsl-su-8-dac.3778/page-7#post-96976
 

dewdude

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So before I responded further...I ran a bunch of various tests, some of them not "real world" just to make sure it's not some kind of odd configuration of mine causing things.

There does not appear to be a "fade-in" since tracks that have audio that starts immediately doesn't have anything cut-off. Not only do I have a few vinyl rips I did in DSD...but I also transcoded some tracks from PCM to DSD just for the test.

Now what I notice with the DAC...and it was the same with the Sabaj Da3 I had....was a noticeable delay whenever the unit has to resync to a new sample rate. So if I play a bunch of 44.1khz stuff everything is fine; but switch to 96khz or DSD...there is a delay. I can tell the unit is "muting" something during this reinitalization; I hear a *very* quick fade as the audio comes up and it sounds like one channel will "double key" it's muting. This is part of the reason I run this ASIO DSD Transcoder; it's transition width just sends empty DSD data for an amount of time before sending the real data; I was getting annoyed by the audio starting a fraction of a second in to the music. My DAC seems to default to some random sample rate when I stop playing DSD data in Foobar. Last night it was defaulting to dsd128...today after booting the dac it's wanting to default to dsd256. This doesn't happen on PCM.

Anyway...the popping with DSD. I thought it may have been a result of reinitalization and the transcoder; but last night I set Foobar to transcode DSD to the DSD128 it was defaulting to so there was no need to reinitalize. There is a very very faint click when the thing starts and when I change tracks without the transcoder; but I have a blower motor on a heater that causes a much louder pop through the amplifier itself.

If I'm looking for it, I'll hear it. If I'm playing an album and it's essentially just one long stream of DSD, then there's no noise during a track change.

Compared to my original M8 the whole resync delay is a minor issue. I don't remember having "late starts" when the M8 had to resync; but for me, I'm not even that annoyed about it and can figure out ways around it.

The firmware on my unit is 1.6 if that means anything. If I can get my Korg working I'll try to record samples of the DSD click and the "muting" oddity I hear occasionally.
 

Celty

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Naedtrax

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Has anyone had "pausing" issues with their su-8? I've gone through 3 so far and the same issue is apparent on all 3 and they are all different firmware but latest build. Swap to 2 other dacs and no issue but then when listening for about 1hr with the su-8 it starts to drop out every 10 seconds or so for half a second.
 

AnikBanic

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Has anyone had "pausing" issues with their su-8? I've gone through 3 so far and the same issue is apparent on all 3 and they are all different firmware but latest build. Swap to 2 other dacs and no issue but then when listening for about 1hr with the su-8 it starts to drop out every 10 seconds or so for half a second.
Yo my one pauses when switching bitrate, that's the only cause for concern.
 

bikecrazy

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Hi All, great informative forum.

Just ordered the SU8 from Shenzen. I am still not sure about the digital vol control reducing clarity etc : /.

Planning to use the SU 8 to feed a yaqin mc 5881 tube amp or a SMSL sp200 head amp.

Question is do i leave the DAC volume at full ? or will it cause clipping from my tube amp / smsl sp200 ?

What vol level is recommended for the SU8 by experienced users here ?

Thanks !!
 

Celty

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Hi All, great informative forum.

Just ordered the SU8 from Shenzen. I am still not sure about the digital vol control reducing clarity etc : /.

Planning to use the SU 8 to feed a yaqin mc 5881 tube amp or a SMSL sp200 head amp.

Question is do i leave the DAC volume at full ? or will it cause clipping from my tube amp / smsl sp200 ?

What vol level is recommended for the SU8 by experienced users here ?

Thanks !!
With the Monolith 887, I set the volume of the amp at about 1 or 2 o'clock, then adjust final volume with the SU-8 remote.
 

MRphotography

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2) Yes, old version shouldn't be stocked anywhere anymore, so you should be safe there.

As to 1), "best" is pretty relative but it is a good DAC for sure. The Grace SDAC balanced is a better value contender for a cheap balanced DAC, but in that review's conclusion, amir recommends the SU-8 if you want more customization like selecting the digital filter, etc.

However, in the SP200 thread the SMSL engineer actually explains the single ended pathway is slightly less noisy than the balanced one. It is an amplifier optimized for single ended input and output. I quote: "the performance of the RCA input is the best.". Balanced will help with possible ground loops but that's it, the single ended input is slightly better aside from that.

Therefore, I'd simply recommend a DAC like the Topping D50s or upcoming single-ended E1DA 9038d, connect to SP200 via some good RCAs (or 3.5mm to RCA for the not-yet-released E1DA) and call it quits.

I haven't seen that thread how much less noisey is the RCA connection on the SMSL 2000 vs the Balanced cable...Seems crazy considering balanced is supposed to be superior generally correct? This is making me rethink my SMSL 8V2 purchase, the only reason I got it was for the balanced, now I am reading that the RCA's actually have better performance?
 

JohnYang1997

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I haven't seen that thread how much less noisey is the RCA connection on the SMSL 2000 vs the Balanced cable...Seems crazy considering balanced is supposed to be superior generally correct? This is making me rethink my SMSL 8V2 purchase, the only reason I got it was for the balanced, now I am reading that the RCA's actually have better performance?
Single end signal path doesn't require a single resistor in the signal path. Meaning the noise can be only limited by the opamp itself. Which can be extremely low 0.3uV or lower.
However if using balanced input, the differential signal needs differential amplifier that sums the signal. It needs at least some resistors in the signal path. Also CMRR is directly related to the resistor values. Too low resistance also adds loading to the front end(whether up stream device or buffer opamp in the device itself). Higher resistance = higher noise. So in the end the noise can well be around 1uV.
Fortunately, the volume control is after the input stage so the noise in real use cases will be limited by the output stage. So not really much difference in real world. Let alone the audibility of the noise minimal.
 

MRphotography

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Single end signal path doesn't require a single resistor in the signal path. Meaning the noise can be only limited by the opamp itself. Which can be extremely low 0.3uV or lower.
However if using balanced input, the differential signal needs differential amplifier that sums the signal. It needs at least some resistors in the signal path. Also CMRR is directly related to the resistor values. Too low resistance also adds loading to the front end(whether up stream device or buffer opamp in the device itself). Higher resistance = higher noise. So in the end the noise can well be around 1uV.
Fortunately, the volume control is after the input stage so the noise in real use cases will be limited by the output stage. So not really much difference in real world. Let alone the audibility of the noise minimal.


Thank you for clarifying, I did notice that balanced XLR cable to my headphones has slightly less volume at the same settings on the amp and DAC, which the specs from the SMSL engineer confirm. Maybe I am just overthinking all of this. I just wish I wouldn't have dumped the money into getting balanced cables for headphones, DAC, amp, etc if they really don't make any difference... The price of all these XLR cables adds up fast. After the price match, the SMSL Su8 was 170 dollars so I know the value is good. I just wonder if I wouldn't have bothered with an XLR balanced DAC how much money I could have saved on the DAC and the cables, seems like a couple of hundred bucks, and if I am understanding what you are saying is correct, that couple hundred dollars registers inaudible quality increases from connecting to the THXSMSL 2000 through RCA channel.
 

Veri

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Thank you for clarifying, I did notice that balanced XLR cable to my headphones has slightly less volume at the same settings on the amp and DAC, which the specs from the SMSL engineer confirm. Maybe I am just overthinking all of this. I just wish I wouldn't have dumped the money into getting balanced cables for headphones, DAC, amp, etc if they really don't make any difference...

XLR is pro standard, with excellent common mode rejection to prevent noise and ground loops. To use it in a consumer setting is quite honestly, for most people and in most situations, overkill.

I mean you will have the best connection. It's all good. But yeah, balanced is pretty overhyped on forums and reviews. RCA will often do the job just fine.
 

MRphotography

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XLR is pro standard, with excellent common mode rejection to prevent noise and ground loops. To use it in a consumer setting is quite honestly, for most people and in most situations, overkill.

I mean you will have the best connection. It's all good. But yeah, balanced is pretty overhyped on forums and reviews. RCA will often do the job just fine.


I mean my DAC is hooked up via XLR cables to the amp and then my Turntable is hooked up via RCA so there is a convenience factor to having both connections without having to use a splitter, but what seems like complete hogwash are these balanced headphone cables, I listened for the last hour back and forth between my stock headphone cables and the XLR balanced headphone cables and heard zero difference actually the XLR cables had less volume do to the design of the amp. Maybe if this little THX was a truly balanced amp I would hear a difference. I think I will keep the XLR cables between the amp and the DAC and just return the XLR headphone cables. I did notice some issues with the volume nob tonight on my Amp unfortunately and it will not increase volume from 3-5.... I can use my DAC as a workaround to control volume but maybe it's worth swapping it out for one that doesn't have this issue unless they all suffer from it. They guys at APOS Audio have been great, so I have no doubt they will take care of me.
 
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