SMSL PL200 Review (CD Player)

[NTTY]

At least as a CD player, the OPPO was reproducing the digital content of the CDA nearly unchanged, when measured from analog outputs, and that was a first in my tests. As of now, only the SMSL PL200 did as good.
So yes, an ESS or AK seem to be required to get there. Of course remains all other aspects of the drive and digital output, 24bits resolution, …

 

[NTTY]

I used it, and had a good time with it. Now I move to VMV T2 and I think T2 is even better although I did not do any measuring

The T2 is a transport, right? In which case it can only do as good as the PL200 which has a perfect digital output. The only room for improvement is the clock precision, but that would not make an audible difference.
I’ll be testing the PL200T soon, which offers similar functionality from the same drive and clock sync input too.

 

[Heat11]

The T2 is a transport, right? In which case it can only do as good as the PL200 which has a perfect digital output. The only room for improvement is the clock precision, but that would not make an audible difference.
I’ll be testing the PL200T soon, which offers similar functionality from the same drive and clock sync input too.

please do, im interested especially after the room correction comment from before. i use that with my wiim

 

[NTTY]

Sorry I didn’t get the relationship with room correction.

 

[Spyerx]

Been looking at this for a minute, and appreciate your review. I already have a good dac/streamer (eversolo A8) and for convenience it might be nice to use it with a singular input on my integrated amp. I'd gone digital a very very long time ago, but there is some nostalgia in my large CD collection. Coincidentally I have the same turntable (Gyrodec SE) and love that low resolution experience. And I like the top load aesthetics.

The difference between the transport only device and the full analog section model is about $160 US. Seems like I'd probalby have no use for the analog section? And just stick with the T model? I wouldn't use the clock sync. And I believe the PEQ/correction profiles of the A8 can be used by any of the digital inputs.

 

[multichannel_guy]

Is this the cheapest entry for a top loading player these days?

 

[gold-sunset]

I used to have a boombox with a CD player. I liked that the lid popped up when inserting or removing the CD but the PL200 just has a lid like a stew pot.

 

[NTTY]

The difference between the transport only device and the full analog section model is about $160 US. Seems like I'd probalby have no use for the analog section? And just stick with the T model? I wouldn't use the clock sync. And I believe the PEQ/correction profiles of the A8 can be used by any of the digital inputs.

I’d be going for the T, as the internal DAC would be clearly redundant in your case.

 

[LTig]

Sorry I didn’t get the relationship with room correction.

Room EQ is done in the digital domain, so it makes no sense to use the analog output of a CD player if a digital output is available.

 

[NTTY]

Ok understood. Review will come soon.

 

[Jazigo]

Hi everyone!

I trying to wrap my head around a subject that I realize I don't fully understand.

When @NTTY measures the XLR out, why do we see 98.7 db SNR?

A full 16bit (65,535) should produce a 4Vrms signal and a empty 16bit (0) should produce 0Vrms. Shouldn't we see the SNR of the DAC?

 

[Scytales]

Hi everyone!

I trying to wrap my head around a subject that I realize I don't fully understand.

When @NTTY measures the XLR out, why do we see 98.7 db SNR?

A full 16bit (65,535) should produce a 4Vrms signal and a empty 16bit (0) should produce 0Vrms. Shouldn't we see the SNR of the DAC?

I guess that you speak about that graph from message #1:

I understand that what is computed by the software is the noise (separated from the signal and the harmonic distortion) in the presence of the signal.

Therefore, it is more akin to a signal to quantization noise ratio than the signal to noise ratio that analogue measurement techniques have accustomed us to, ie a figure of merit obtained by measuring first the maximum output level of the signal and second the output noise level with the signal switched off. When digital sources are concerned, the latter measurement is made less relevant by the fact that many DAC chips actually mute their outputs when they are told that no signal is to be produced, which let only the analogue output stage noise floor to be measured and not the noise of the analogue circuit plus the noise produced by the DAC.

 

[Jazigo]

I guess that you speak about that graph from message #1:

I understand that what is computed by the software is the noise (separated from the signal and the harmonic distortion) in the presence of the signal.

Therefore, it is more akin to a signal to quantization noise ratio than the signal to noise ratio that analogue measurement techniques have accustomed us to, ie a figure of merit obtained by measuring first the maximum output level of the signal and second the output noise level with the signal switched off. When digital sources are concerned, the latter measurement is made less relevant by the fact that many DAC chips actually mute their outputs when they are told that no signal is to be produced, which let only the analogue output stage noise floor to be measured and not the noise of the analogue circuit plus the noise produced by the DAC.

I may be completely off with my thinking, but I would understand it if the ADC (the measurer) is doing so by 16bit.

If I play a platonic 1kHz sine wave and a ADC samples that with a 16 bit precision and compares that to a mathematical 1kHz sine wave, then we would see a SNR of 98.7 db.

But I thought that the DAC converted the 16bit data to 32 bit data and then oversampled the data, which in turn should give us a better analog output signal than the digital input signal.

Edit:

To be more precis on this:

My understanding of the internals of the DAC is that it converts 16 (and 24) bit data to 32 bit data.
After that step it does a source rate conversion, so that 44.1 kHz becomes 352.8 kHz (see picture)

In other words, I would expect to see the SNR of the DAC and not the source.

 

[NTTY]

To complement what @Scytales said about the signal to quantization noise which is the reason why you see that limitation:

Let’s assume the internal DAC of the CD players runs at more than 16bits, say 18bits to 32bits. It does not change what has been recorded on the CD, which is at 16bits. So that means, has you said, we get only 65’536 unique levels. This limitation means there will be rounding errors when converting the initial signal from the master (be it digital or analog) to 16bits. This creates "quantization errors" that generate a low level noise at around -98dB spread along all the audio bandwidth.

When converting the recorded signal on the CD back to analog, it is impossible for the DAC to differentiate the quantization errors from the musical content, because they are seen as the same. And so we get the same limitation.

 

[NTTY]

A full 16bit (65,535) should produce a 4Vrms signal and a empty 16bit (0) should produce 0Vrms. Shouldn't we see the SNR of the DAC?

One side note: the PCM code (2’s complementary) is signed. So a 4Vrms sine means it goes from -2V to +2V and each half get 32’768 samples. And since there is only one 0 (0000000000000000), then there are only 32’767 positive values and 32’768 negative values, hence an asymmetry. The first bit of the code (called Most Significant Bit - MSB) is the bit of the sign, 0 for positive and 1 for negative).

 

[EJ3]

I used to have a boombox with a CD player. I liked that the lid popped up when inserting or removing the CD but the PL200 just has a lid like a stew pot.

Since I am not going to be carrying it around (and I like to watch the disk spin), being able to set the stew pot lid aside is just fine by me. In fact, it makes me like it more. It's a BIG part of why I already purchased one.

 

[Jazigo]

To complement what @Scytales said about the signal to quantization noise which is the reason why you see that limitation:

Let’s assume the internal DAC of the CD players runs at more than 16bits, say 18bits to 32bits. It does not change what has been recorded on the CD, which is at 16bits. So that means, has you said, we get only 65’536 unique levels. This limitation means there will be rounding errors when converting the initial signal from the master (be it digital or analog) to 16bits. This creates "quantization errors" that generate a low level noise at around -98dB spread along all the audio bandwidth.

When converting the recorded signal on the CD back to analog, it is impossible for the DAC to differentiate the quantization errors from the musical content, because they are seen as the same. And so we get the same limitation.

Ok, I think I get it:

When we sample a perfect 1kHz sine wave with 16 bit data, we inherit a loss of precision. This loss is referred to as quantization error.

If we ask a perfect DAC to reconstruct this 1kHz sine wave from the 16 bit data, it will do so with the inherited quantization error.

When we measure this reconstructed 1kHz sine wave with the original 1kHz sine wave we sampled from, the "noise" in the signal will at best be the quantization error.


The thing that got me is that the DAC scale the 16 bit to 32 bit and upsample the data with inserted pcm values calculated from a FIR filter between values in the original file. I thought that these inserted values would remove some of the edginess of the original 16 bit, but since those values are already with a mismatch, then we an only reconstruct a close-to-perfect sine wave with a mismatch.

 

[NTTY]

All correct

The oversampling will indeed reduce the noise created by the conversion from Digital to Analog, but it will (should) not modify the initial digital signal that was recorded on the disc. And that one came indeed with some quantization noise.

 

[Jazigo]

All correct

@NTTY and @Scytales , thank you for bringing light to the subject.

 

[Kagenohi]

The T2 is a transport, right? In which case it can only do as good as the PL200 which has a perfect digital output. The only room for improvement is the clock precision, but that would not make an audible difference.
I’ll be testing the PL200T soon, which offers similar functionality from the same drive and clock sync input too.

In this situation, the DAC on my Denon 3000NE might be better than the one on the PL200. It's strange because when I use the digital output on my PL200, the sound is sometimes intermittent. As a result, I've always opted to use the analog output on my PL200. This could be causing the difference in sound quality.