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SMSL PO100 PRO Review (Digital Interface)

Nice device )
Teardown see attached file
 

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I could re-do few tests ouf of some MBs, but my experience so far : most are not clean.


You really should use the USB-B input instead. ;) It's clean, bitperfect, and supports RME Remote.
Thanks for the kind reply.
My PC (MB MSI X570-A pro) is 15 M away from my ADI-2 pro.
Optical SPDIF was the simplest solution.
I used a AUDIOPHONICS XMOS U208 (that I still have),
then I saw there was already an optical output from the MB, and used it candidly to simplify the setup, eliminating the need to pass through USB circuits ... but then using the Realtek chip (ALC1220, not so old...). There lies the question.
Should have to test, but not equipped.
Regards,
 
be interesting to see how good the average toslink output on desktop pc motherboards is, i avoided it for years but buggy usb drivers forced me into using it. Haven't noticed anything wrong with it anecdotally, that's running out of full-sized desktops with pretty powerful gpus in them.
 
Thanks for the kind reply.
My PC (MB MSI X570-A pro) is 15 M away from my ADI-2 pro.
Optical SPDIF was the simplest solution.
I used a AUDIOPHONICS XMOS U208 (that I still have),
then I saw there was already an optical output from the MB, and used it candidly to simplify the setup, eliminating the need to pass through USB circuits ... but then using the Realtek chip (ALC1220, not so old...). There lies the question.
Should have to test, but not equipped.
Regards,

15m with a plastic optic fiber generates jitter (24.3ns RMS - SNR jitter @20 kHz : 50 dB if the calculation is correct), and you’re at least 10x below threshold of audibility. Full data integrity though.
What’s more important is that you use WASAPI Exclusive (if you’re on Windows) from the player.
And also send the native bit depth of your files to the interface, to prevent resampling by the software. This could generate more distortion than the mentioned jitter.
 
Also the Schiit Eitr 2:


For what it's worth, I stopped looking for a digital interface with AES EBU output after I realized that I could use a cheaper and more easily obtained device like the PO100 Pro plus a cheap coax to AES cable.

I still wouldn't mind using a USB to AES converter instead - but I'm holding out for an inexpensive, auto-switching digital converter that has more than one input so I can easily plug in USB from my computer music server, plus SPDIF coax from my disc transport, with a single AES EBU out to go into my active speakers. I don't want to sidetrack the thread as we've collectively discussed this elsewhere before, but unless you want to spend $500 or more and end up with a bunch of features you don't need plus an extra remote, there's just nothing on the market that gives you multiple digital inputs including USB, and also a digital output (let alone AES EBU output specifically). You can only get digital switcher/converters that don't include a USB input, or DACs with multiple digital inputs including USB but only analogue outputs. It's weird.
thanks ill see if i can find some measurements for that one . im going from optical or usb to aes so I dont think ill get a cable for that thankyou though
 
I use a SMSL po100 pro to feed my SMSL DO400 with a i2s signal from my ROON server..(linux NUC/ROCK) Not that i notice any diffrence when running direct USB from ROON server to SMSL do400... but since i have it i use it.. is a great little device, Thanks for a super good review..!!!
 
Hello, absolutely not an expert here. I'm experiencing ground loop (electrical?) noise caused by my NUC feeding my SMSL D200 through USB cable. Would I solve this by connecting the DAC via optical instead through the SMSL PO100 PRO? The NUC only has USB type A output.
 
I had one and it worked great. Unfortunately it had a short life -- it died after a few months.
 
I have a PO 100 non-pro and it's been working decently well, the issue I have sometimes is with Windows' USB handling. Sometimes, after too long of a period without a restart or a bad wake up from sleep, or any other hiccup, I get a lot more audio drop outs and blips on the PO100 than on the other audio outputs.

Nothing that a restart won't fix, but irritating nevertheless.
 
I don't know about the PO100 Pro, but I've not had good luck with the non-pro PO100, neither the first nor second revision. They were on all the time and both flaked out in different ways after a couple of years. I was thinking of trying the Pro in the hope that it would last longer. The price is a bit above what I consider appropriate for something disposable, however, so I've been reluctant. Justified or not, I've begun to develop a prejudice against SMSL.
 
Thanks, @NTTY ! :)

Yep, that was indeed just a couple of weeks before I managed to get my ADI-2/4 PRO SE. Would have done it differently after that. ;)

... I since kept the PO100 PRO as my reference (AKA perfectly transparent) DDC for any measurements in digital domain.
Hello Everyone,

This is a review and measurements of the SMSL PO100 PRO USB Digital Interface.

View attachment 534009

It was already reviewed almost 4 years ago by @VintageFlanker, but only in analog domain, via an external DAC. I'll be testing in digital domain mainly.

This device is digitally "perfect" and its limits are the Software and drivers you'll be using with it, not its "hardware" performances. So this review will be more kind of an opportunity to learn more about PCM digital signal, rather than just a series of standardized measurement graphs, which would be boring because all state of the art.

A Digital to Digital Converter acts as a bridge between a computer and whatever device with digital inputs (DAC, preamp, integrated amp, powered speakers, etc...) you wish to use.

There are multiple business cases for this device, and mine is to use it to send high quality digital signal to a device under test. So it will be my digital generator, main advantage being that I can powered it from the PC directly.

It is very important for me to ensure that the digital output from this SMSL is of very high quality and unaltered, if possible, or that I know about the possible flaws and how to mitigate them.

User experience

It can't be simpler to use, I did not have to install a driver (Windows 11) although SMSL site says I have to, and I guess this is because I already installed a SMSL compatible driver in the past.
There are two buttons to select UAC-1 (for backward compatibility) or AUC-2, and one to change the pin allocation of the I2S (HDMI connector) which I did not use.


SMSL PO100 PRO - Measurements (Digital)

To measure the PO100 PRO, I used a Motu Ultralite Mk5 pro interface which has digital inputs (coax and Toslink). The software is REW, as always for me.

So the setup is simple, Computer (REW) -> SMSL PO100 PRO -> Toslink 3m cable -> Motu Mk5 -> Computer (REW).

The Motu requires I use its ASIO driver in REW, while the SMSL requires the Java one, so I had two instances of REW, one acting as the Generator and the other as the Analyzer. That means I can't run sweeps, but you'll see it's not an issue.

Let's start with the standard 1kHz test tone at 0dBFS (24bits data) without dither, the sampling rate is 44.1kHz:

View attachment 534011

Look at the y-axis that goes way below (-190dBr!) what you usually see with Amir's review. This is because we don't have analog noise here, it's all in digital domain.

Of course you see some grass at the bottom of the graph and that what we call "quantization noise". These are due to rounding errors when sampling the continuous signal into a digital one. Since this is 24bits, we have "only" 2^24 = 16'777'216 values (integers) to represent all possible values of an analog voltage swing. And sometimes the ideal value would need to be in between two possible integer values out of the 16M+ available. When that happens, it generates an inevitable error, and that translates into that type of distorsion, or better called "quantization noise".

Additional interesting fact is that these rounding errors can accumulate and show themselves as low level random noise or as some sort of distorsion that you see above.

As a proof, let me run the same test, but with a sine tone that is a prime number, instead of 1kHz. Here you go with a 997Hz test tone @0BFS:

View attachment 534014

See the difference? This time you see a low level noise that appears random. It hurts less you eyes. But that is the same reason, it’s because of rounding errors, quantization noise. Why does it spread better than 1kHz? This is because a 1kHz sine generates much less PCM unique subcodes than the 997Hz one. Having more unique PCM 24bits codes means the rounding errors are very variable and so the resulting distortion that they generate better spreads and shows as kind of random noise.

Actually, this is an interesting fact because it can influence the value of what is called "Spurious Free Dynamic Range" (SFDR). The SFDR is the ratio of the amplitude of the main output signal (the fundamental) to the amplitude of the next largest unwanted distortion or noise component (the "spurious" tone or spur) (Analog Devices).

When the PCM subcodes are limited, we are likely to see a decrease of the SFDR. This is of no importance for reproducing music, but it is for me as wanting to use this device as a high precision generator. Look at the below:

View attachment 534015

From that, you see an interest to go for the 997Hz test tone. Well, ok, this is not an issue in digital domain, since both SFDR are way better than the best DAC out there. But let's do the same with 16bits PCM:

View attachment 534037

Ouch, you get the interest, right?

Back to 24bits... The AES standard recommends to use dither (for that reason I guess) when generating test signals. Let's do that with the 1kHz test tone:

View attachment 534039

Yes, nice, but we lost one bit of resolution because that is what the dither noise consumes. So the dither (low level random noise) indeed allows for a better spreading of the quantization noise, at the expense of limiting the potential full resolution. Again, no big deal with 24 bits since all DACs and ADCs will be limited way before that in 24bits. But it is a different view in 16bits. And this is the reason why I use a 999.91Hz test tone for my tests of CD Players, because it generates a lot of unique PCM subcodes in very short time, spreading well the quantization noise and thus offering the best SFDR we can expect in 16bits PCM.

With all of the above, maybe you've learned something new, or not. But I guess you get that the digital output of the SMSL PO100 PRO is very precise since it is the one I used for all measurements.

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We can still chalenge it, at least on jitter perspective, though. This is the standard J-test at 24bits (44.1kHz):

View attachment 534046

Blue trace is from the PO100 and red one is from the WAV file. They are the same meaning no jitter. Actually, you also never see that trace so clean from Amir's review, and that is because the low level signal is hidden by the analog noise of the DAC. Here is the same jitter test with the Topping D50III that is an excellent DAC:

View attachment 534047

See how the analog noise covers the low level digital component of the Jitter Test?

And this is an opportunity to remember what the jitter test is made of. There are two components:
  1. The High-Frequency Component (The Carrier): The primary signal is a high-frequency tone that sits exactly at one-quarter of the sampling frequency (fs/4).
  2. The Low-Frequency Component (The Stressor): The second part is a low-frequency square wave designed to manipulate the Least Significant Bit (LSB), which is the smallest possible bit value in the digital word.
The second part is a bit more complicated to understand, so let me detail that. The PCM code is 2's complement, same as with computers. That means it is a signed interger code, that represents the swing of analogue voltage values (eg from -1V to 1V).

Since 2^24 = 16'777'216, the truth table of the PCM code is the below (24bits):


PCM CODESigned Decimal value
01111111111111111111111+8'388'607
......
00000000000000000000001+1
000000000000000000000000
11111111111111111111111-1
......
10000000000000000000000-8'388'608

This is a signed code where the first bit (called Most Significant Bit - MSB) is the bit of the sign. It is the one that has the most significant weight. The last bit of the code is the Low Significant Bit (LSB) because it is the one representing the smallest possible value (1/8388608).

Note that the code is asymmetric because the 0 value is considered positive. So we have one less value to represent the positive swing compared to the negative one.

Note also that from -1 to 0, all bits of the code will flip. This is what the Jitter Test leverages as a "stressor". So, the low Frequency component of the Jitter Test is a square wave that uses only the negative 1LSB and so the low level square forces a constant flip as below:

00000000000000000000000 : 0
11111111111111111111111 : -1


The frequency of the square is 1/192fs to align the SPDIF (or AES3) format that is based on blocks of 192 frames.

----

Now, let me have some additional fun with what Stereophile was calling the 3DC test. This is a test were only the two lowest negative and positive levels are stimulated. If you look again at the table the I previously showed, it means we create a sine that invoques only 3 PCM codes, as per the below:

00000000000000000000001 : +1
00000000000000000000000 : 0
11111111111111111111111 : -1


Not only this is the smallest symmetrical possible signal in 24 bits PCM, it is again generating a regular flip of all bits from 0 to 1. Very first R2R DACs did not like that, reason for the test. Here, we don't test a DAC as we stay in digital domain. But if the interface would be modifying the digital content, we would see a different code.

The digital level of that signal (in dBFS) is calculated by the formula 20log(2/16'777'216)=-138.47dBFS

To test that the code is unchanged, we simply look and the result in time domain. Since this signal is a constant change between only three levels, you get the same in time domain:

View attachment 534061

With only the lowest bit (LSB) used, we get three "DC" levels, of constant amplitude since in it impossible to represent a sine with only one bit and so only 3 level -1,0,+1 (half a period for the 0-cross). What is good with this test is that any modification on a single bit in the code would generate a massive error in the amplitude showing itself immediately. Here we get what we want, ie bit perfect.
The ringing you see is due to the Gibbs phenomenon.

I zoomed at max on REW's scope, but it's still low amplitude because that signal is extremely low and something we can't see in analog domain. The same with 16bits PCM is the below:

View attachment 534069

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Last and not least, as I use for my regular reviews of CD Players, my ultimate proof of "perfect" digital output is when I reuse the intersample overs test at 5512.50Hz, with a phase shift of 67.5°. This signal generates an overshoot of +0.69dB. And so, if the signal would be modified before being sent over SPDIF (by an ASRC for instance), it would show either a reduction of amplitude or we'd see some sort of saturation/increase of noise/distorsion. So, here we go (note this is a 16bits test):

View attachment 534063

And yeah, we get what we want. No distortion and the dashboard shows +0.69dBFS as expected. No ASRC on the digital path here. SNR is only 95.6dB because there is rectangle dither in this test file (consuming half of a bit out of the 16 available).

----

EDIT: I forgot to add that the clock deviation is lower than what I can measure, and that means better than 0.2ppm. I’ll add the corresponding trace later.


SMSL PO100 PRO feeding a DAC

To finalize this test, I still wanted to check the analog output of a DAC fed by the SMSM PO100 PRO. I used the Topping D50III that Amir reviewed.

This is the result at 24 bits with the standard 1kHz test tone (with Dither):

View attachment 534064

Note that I changed the y-axis to max -170dBr instead of -200dBr you saw before.
This is an excellent result (123.4dB SINAD), a little better than what Amir measured in his review (same bandwidth of analysis to include noise), I got less noise but a little more distortion. The SMSL PO100 PRO is far from its maximum but the results out of the Topping D50III puts it where it is: at the top of Amir's DACs ranking.


Conclusion

No surprises that this is a flawless device, but now you know and can say why ;)

With such precision, the potential issues might come from the Software and associated setup that you use.

Enjoy your WE!

Flo
Excellent little devices. I use one in each of my systems so that I can run lossless streaming from my cellphone or iPad to the optical input of either my Topping E70 or SMSL DL200. They've always worked perfectly.
 
It better be good at that because that is why I bought it. I’ll report if I have issues.
Be very careful with the coax, if you move it around a bit don't move it with a cable, I've seen 3 go out with friends (1 was mine) I think it's just a weak solder joints on the coax to the board, as we all could move it around and audio would play. Got the Eit2 and extremely happy and easy warranty.
 
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Be very careful with the coax, if you move it around a bit don't move it with a cable, I've seen 3 go out with friends (1 was mine) I think it's just a weak solder joints on the coax to the board, as we all could move it around and audio would play. Got the Eit2 and extremely happy and easy warranty.

Thanks to NTTY and Amir for posting this.

Appreciate Jason bringing up the coax cable concern of a potential weak point.
A legitimate one with my own coax cables being somewhat stiff and the SMSL being a light device with small footprint.
I also own the similar Douk Pro version of a DCC which has a better stronger case than the SMSL, but I am still trying to solve the occasional intermittent noise bursts from the unit
 
I have a PO 100 non-pro and it's been working decently well, the issue I have sometimes is with Windows' USB handling. Sometimes, after too long of a period without a restart or a bad wake up from sleep, or any other hiccup, I get a lot more audio drop outs and blips on the PO100 than on the other audio outputs.
Nothing that a restart won't fix, but irritating nevertheless.
If you are using Win11 and new-ish MoBo, there are a few power/efficiency settings which do/may impact a few USB performance criteria.
Such as:
*MoBo global UEFI PowerPlans (e.g.. ErP-ready),
*HW 'USB Selective Suspend' (DisableSelectiveSuspend=1)
*SW policies enforced at the (ncpa.cpl) USB Root Hub drivers (devmgmt.msc)
*Win11 'High Efficiency' power plan increases the likelihood of USB port suspensions. The hidden "Ultimate Performance" plan keeps all USB ports open and always energized...
For those who hate reboots...:rolleyes:
 
I finally found time to circle back and read this. Thanks @NTTY for your time performing all the reviews and adding in the explanations throughout this review (e.g., how the test signals are generated and their purpose)!
 
For some older DACs, such as the 2QUTE that I really like, this PO100 Pro can replace its USB connection to the PC, and the sound improvement is very noticeable. I think it's a worthwhile purchase, and I've incorporated it into my home speaker system.
 
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