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Teac VRDS-25X Review (CD Player)

NTTY

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Hello Everyone,

This is a review and detailed measurements of the Teac VRDS-25X Stereo CD Player.

TescVRDS25X_001.jpg


This is my second review of a "legendary" Teac VRDS player, after the VRDS-20 (I will need to update the measurements of this one with my latest Test CD).


Teac VRDS-25X - Presentation

This will be an epic review, maybe not for you as readers, but for me as a reviewer and writer, it is/was :p
Over the last weeks, I spend countless hours playing with this big baby, and I'll share as much as I can from this experience with you.

The Teac VRDS-25X was released at the end of the previous century, nearly 25 years ago. That is 6 years later than the VRDS-20, yet reusing the same VRDS mech.

The elements of interest are:
  • The VRDS (Vibration-free Rigid Disc-clamping System) mechanism of course! This is the version CMK-3.2 with resin (or polymer, I’m not sure) molded “bridge” over the (aluminum) clamper that is supposed to reduce vibrations of the disc when rotating. Below that is the Sony KSS-151A magnetic head, one of the most reliable, fastest head. And 25 years later, it runs flawlessly.
  • D/A architecture: Conversion is performed via two mono 20bits Analog Devices 1862N-J per channel. The AD1862 was already an "old" R2R architecture at the time and it is funny to see Teac going back to R2R conversion after the move to 1bit years before in the VRDS-20. The AD1862N-J is considered as one of the best R2R DAC of the time, and this is the "J" version in use here, which offers lower distortion by few dB compared to the standard version. This DAC, as the BurrBrown of the same period, offers an optional TRIM adjustment to improve distortion. As opposed to what I thought the VRDS-25X does not allow playing with the trimming. But because it uses two DACs per channel, the Teac offers an adjustment mechanism to reduce high-level even-order distortion by balancing complementary DAC levels, as @AnalogSteph identified.
  • 8x oversampling: The AD1862N-J are seconded by the 20bits oversampling filter SM5843A from NPC, which can run at up to 8x. It offers a choice of sharp and low roll-off, the former being is use as per my measurements.
  • Digital inputs: The Teac VRDS-25 in this version "X" offers the possibility to use its DAC from an external source, which is cool and was not often offered at the time.
  • Asynchronous Sample Rate Converter: All digital data are first fed into an Analog Devices 1893JN. This is a low cost ASRC version of the AD1890. It outputs only 16bits data, instead of 20bits for the AD1890. It means the documented performance is down from -106dB THD+N in the AD1890 to -96dB in the AD1893 :rolleyes: The main function of the ASRC is to simplify interfacing with digital sources. It is curious to need it in a CD-Player, but it could be explain by the digital inputs allowing to use this player as a DAC with other sampling rates.
Talking about outputs/inputs, this is the back of the player:

TescVRDS25X_002.jpg


We find RCA and XLR outputs with our regular digital S/PDIF outputs. And less classic, we have the chance to use the player as a DAC with a coax and optical input. They will sync up to 48kHz only, as per my measurements.

As you saw, the look is classic, far from the previous VRDS-20. The build quality is nothing less than impressive starting with a weight of 36.4lbs (16.5kg). Want to look inside? Here you go:

TescVRDS25X_003.jpg


You might have seen from the first picture that the top cover is divided in three sections. Well now you know why. The top cover comes in three pieces and shows three distinct sections:
  • On the right, the power supply with the massive transformer and power PCB for the Servo card (hidden on that photo).
  • In the middle the beautiful VRDS drive and behind it the power PCB for the Audio section.
  • Left side shows the analog to digital conversion card.
No fancy copper chassis here, but all of that looks really well done and like the look of this audio board:

TescVRDS25X_007.jpg


I almost forgot, this player comes with decoupling feet, this is one:

TescVRDS25X_008.jpg


No funny adjustable feet like on the VRDS-20, but a much less practical and probably refined audiophile concept of two pieces feet really painful to use. That said, it works well. Look at the below measurement of THD without (top) and with (bottom) the decoupling feet:

Teac VRDS 25X_Fun.jpg


This is a massive 24dB/4bits improvement in THD/ENOB!

Don't you believe me? Really? You're right, I'm kidding, of course :p This is when I played with the distortion adjustment option, and more to come on that soon (below). And so no, sorry, the decoupling feet are of no influence, or if they are, it's below the idle noise of the player ;)

Other than that, and same as with the VRDS-20, the CD Player is a delight to use as it is so fast to go back and forth, skip one or multiple tracks.


Teac VRDS-25X - Measurements (Analog Out)

The below measurements are consistent with those I described when reviewing the Onkyo C-733. So over time, this will help comparing the devices I reviewed.

The Teac VRDS-25 outputs a 1.78Vrsm from its RCA outputs and 8dB more from XLR with near 4.5Vrms! RCA and XLR showed the same performances. There was a channel imbalance of around 0.05dB (good). The single-ended outputs invert absolute polarity; balanced outputs are non-inverting.

The performances of this player are nearly identical from RCA and XLR outputs, which is cool, and by default I'll show measurements from RCA unless otherwise noted.

Here you go with my standard 999.91Hz sine @0dBFS (without dither) from the Test CD (RCA out):

Teac VRDS 25X_999.91Hz_0dBFS_RCA_LR.jpg


Right and left channels are shown but only one gest evaluated in the dashboard. Both channel are very close, though, and it's not always the case. THD sits at -94.6dB, which is good for a CD player at full scale, and with R2R type conversion. The plot and legend shows H2 level for both channels (-97.5dBr and -98.5dBr).

But can we do better? Maybe...

As I wrote before, the Teac offers to adjust the even-order distortion between the two DACs per channel. There are two plots to use in a sequence to adjust the distortion while playing a full scale tone. Have a look at the below picture that I made in B&W to ease identification of the respective plots:

TescVRDS25X_006.jpg


The adjustment procedure indicates to set S101 (circled in green) switch n°3 to "ON". It was already done... And then first adjust the distortion with those plots circled in red and then in blue, in that sequence, for each channel. I was wondering if it was done from factory and/or if some drift in time needed to be compensated.

The adjustment requires near live monitoring of the FFT, which REW allows when using a relatively low FFT length (128k for 44.1kHz test tone being played in my case).

Short story long, I was able to slightly improve things as you can see below:

Teac VRDS 25X_999.91Hz_0dBFS_RCA_LR_THDAdjust.jpg


Overall 3dB less distortion, not impressive but not bad. We are now very close to -100dB distorsion. If you compare the two measurements, you'll see that even harmonics improved.

This being done, let's continue to the same view @-6dBFS:

Teac VRDS 25X_999.91Hz_-6dBFS_RCA_LR_THDAdjust.jpg


The distortion is a little higher with level decreasing, which is often the case with R2R type conversion. But it's still very good here. It also means very good consistency in performances when close to full scale (understand with “hot” CD Masters).

You probably already noticed that this is a quiet CD player, with minimum power supply–related spuriae in its output (below -120dBr at 50Hz with full digital scale):

Teac VRDS 25X_PowerSupply_RCA.jpg


This is again RCA outputs. XLR do a little better because 100Hz and 150Hz harmonics are absent, but it won't change anything.

Next is the bandwidth (now measured from a long term average of periodic white noise):

Teac VRDS 25X_FR.jpg


Not completely flat, but this is zoomed (0.1dB per division) so we see -0.2dB to +0.1dB variation which is reasonable. Also, note the minimum ringing of the reconstruction filter. By the way, you can also appreciate the 0.05dB imbalance from the two channels.

Talking oversampling filter, this is below a view of its behavior (from white noise) and together with dual tones 18kHz+20kHz (AES17):

Teac VRDS 25X_Filter.jpg


This looks like the slow roll-off filter of the SM5843 as we see the alias of 18kHz being attenuated by 80dB (documented >=77dB in the SM5843 datasheet).
But since there is an ASRC at the input of this player, it too has a digital filter to perform the upsampling/downsampling in band limited. So we wee here the combination of the two.
No noise shaping technique shows here, though, this is good old R2R conversion with "standard" oversampling after all.

Multitone (1/10 decade) shows a happy CD player, not having issue to clear 16bits of data:

Teac VRDS 25X_MT_RCA.jpg


This is the Jitter test:

Teac VRDS 25X_JTest_RCA.jpg


This is an overlay of Analog (bleu) and Digital outputs (red) of the Teac. Too bad to see two side skirts (blue) and far from the fundamental (4kHz). Hopefully, they are low in level and will remain hidden when playing music.

Started with the Teac VRDS-20 review, and on your request + support to get it done (more here), I'm adding now an "intersample-overs" test which intends to identify the behavior of the digital filtering and DAC when it come to process near clipping signals. Because of the oversampling, there might be interpolated data that go above 0dBFS and would saturate (clip) the DAC and therefore the output. And this effect shows through distorsion (THD+N measurement up to 96kHz):


Intersample-overs tests
Bandwidth of the THD+N measurements is 20Hz - 96kHz
5512.5 Hz sine,
Peak = +0.69dBFS
7350 Hz sine,
Peak = +1.25dBFS
11025 Hz sine,
Peak = +3.0dBFS
Teac VRDS-25X-30.2dB-24.2dB-27.9dB
Yamaha CD-1 (Non-Oversampling CD Player)-86.4dB-84.9dB-78.3dB
Onkyo C-733-88.3dB-40.4dB-21.2dB
Denon DCD-900NE-34.3dB-27.1dB-19.1dB

I kept some references and will keep the same for other reviews, so you can quickly compare. The results of the Teac VRDS-25X mean the oversampling filter does not have headroom to process intersample-overs and is therefore clipping where it's most likely to happen. The Yamaha CD-1 shines here because it's old enough not to have an oversampling filter.

In reality, the situation is a little worse than the numbers show here. To illustrate the issue, let me show you first this 7350Hz ISO test with the Onkyo C-733:

1731511291233.png


We can see the saturation showing via multiple odd harmonics of 7350Hz. This is the beginning of a square, in other words: clipping. The calculated THD is -40.4dB. This test shows that the oversampling filter is saturated here and can't recompose the samples that go over 0dBFS. Not only that, but the filter became nearly inactive as show all high level harmonics after 20kHz. This means the oversampling filter is overloaded and no longer filters.

Now, the same view from the Teac:

1731511716052.png


Ouch! Not only the clipping is more evident as calculated (THD = -25.0dB) but there is a massive number of additional distortion all over the place and up to -50dB in audio band (below 5kHz). Similar to the case of the Onkyo, the oversampling filter barely filters anything above 20khz.

But why that difference?

It can be explained by the presence of the Asynchronous Sample Rate Converter (the AD1893) that precedes the Oversampling filter and feeds it with digital data. This ASRC converts digital input data of different sampling rates into a unique sampling rate output (I think 48kHz) as shown below (from AD1893 Datasheet):

1731512453916.png


This ASRC itself is not immune to inter-sample overs. Indeed, to convert from one rate to another, it will massively increase the number of samples (as shown above), filtering what's to keep in audio band and down-sample to desired sampling rate (probably 48kHz, again). This effect of increasing the number of samples between the original samples could potentially generate some that go over 0dBFS, and would saturate the ASRC.

Is it the case? Well, yes, and it is documented in the datasheet of the AD1893 in these words:

"Clipping
Under certain rare input conditions, it is possible for theAD1893 to produce a clipped output sample. This situation is best comprehended by employing the interpolation/decimation model. If two consecutive samples happened to have full-scale amplitudes (representing the peak of a full-scale sine wave, for example), the interpolated sample (or samples) between these two samples might have an amplitude greater than full scale."


Analog Device also mentions that "Clipping can also arise due to the pre-echo and post-echo Gibbs phenomena of the FIR filter, when presented with a full-scale step input" and so let me show you that in time-domain. For that, I used an unfiltered digitally created square tone at 1002.27Hz, one with a 3.01dBFS peak headroom to give space for the ASRC and oversampling filter to process it, and another with 0dBFS peak.

This is the one with 3.01dB headroom:

1731513736863.png


We can see the linear phase filter creating these riggings, typical to Gibbs phenomenon due to band limited restriction of the Audio CD, and implying proper filtering.

Now the same with 0dB headroom:

1731513838124.png


You can see that we barely have any riggings here, which means that the digital filtering is overloaded, with the consequence to be defeated. Some would be tempted to think that the second view is a better representation of a square, but it's not what we want to see in band limited digital audio.

Now, add to that the fact that we have two digital filters in action here. One in the ASRC, since it is mandatory to perform filtering before downsampling, and then we have the one of the standard oversampling filter. But both are overloaded and defeated. Why make it simple?

Sorry for that long digression but I thought this was an interesting finding to share.


To come back to the standard measurements, let's continue with the good old 3DC measurement that Stereophile was often using as a proof of low noise DAC. It is from an undithered 997Hz sine at -90.31dBFS. With 16bits, the signal should appear (on a scope) as the 3DC levels of the smallest sign magnitude digital signal (XLR out):

Teac VRDS 25X_3DC_XLR_MSBAdjust.jpg


This is a good representation of the 3 levels, which means the Teac has a good tangible resolution with undithered data.

Other measurements (not shown):
  • IMD AES-17 DFD "Analog" (18kHz & 20kHz 1:1) : -87.8dB
  • IMD AES-17 DFD "Digital" (17'987Hz & 19'997Hz 1:1) : -83.7dB
  • Dynamic Range : 96.5dB
  • Crosstalk: not measurable (ASRC/OSF/DACs shut down)
  • Pitch Error : 19'997.07Hz (19'997Hz requested) ie +0.0005%
Last but not least, I like to have a look a the THD vs Frequency when using a -12dBFS signal. This has proven to me to be a key differentiator, especially when I'm reviewing an old CD Player using R2R conversion (case here). Here are the results with the Teac VRDS-25X (Left and Right analog Channels shown, with the Onkyo C-733 as a reference) :

Teac VRDS 25X_THDvsFreq.jpg


The flatness of the THD across frequency is unusual, I never saw that before. Would it be the cause of the ASRC, again?
The ASRC AD1893 was the low cost one of Analog Devices. And in the datasheet, we find the below THD vs Frequency measurement (@0dBFS, not -12dBFS like mine):

1731518353844.png


Looks familiar? Well, we found who's guilty here (again).


Teac VRDS-25X - Measurements (Optical Out)

I've seen several of you reviewing CD players using their digital outputs, in case the results could be improved from an external DAC. And since this is a VRDS drive, I guess you want to know if it delivers.

First the digital output is as what we expect it to be, perfect (999.91Hz @0dBFS without dither):

Teac VRDS 25X_999.91Hz_0dBFS_Opti.jpg


And I like to put the 3DC level test as well in this case (mainly because I don't have a bit perfect feature on my Motu like we find on an RME):

Teac VRDS 25X_3DC_1kHz_Opti.jpg


Again, the signal is untouched and we see only the Gibbs phenomenon due to band limited in CD Audio.

So, no worries, it'll be a perfect transport.


Teac VRDS-25X - Measurements (Optical in / RCA Out)

The Teac VRDS-25X offers two digital inputs, which was rare at the time, and so allows to use it as a converter.

Of course, it is obsolete today as it will synchronise only up to 48khz. But I tested it anyways.

Here below is an overlay of 999.91Hz @0dBFS (without dither) with 44.1kHz and 48kHz sampling rate:

Teac VRDS 25X_999.91Hz_0dBFS_OptiIn_44vs48k.jpg


The performances remain the same as when reading from CD (good news for the VRDS). That said, we see a little more disruption with 48kHz input and the bottom of the fundamental. And that is because Jitter is of a different nature with that sampling rate:

Teac VRDS 25X_JTest_48k_OptiIn.jpg


Now the side bands are very close to the fundamental yet still at very low level, so not a concern, but not perfect.


Conclusion

To be honest, I think I measured the max performance of the ASRC which equips this player, and more than anything else. Its THD+N is given at -96dB as I wrote in the presentation. I had a little more here which means that all components surrounding the ASRC perform better. If only Teac had used the AD1890 (20bits, -106dB THD+N) instead of the low cost AD1893 (16bits), maybe we would have seen one of the best implementation of the AD1862 DAC, which was one of the best R2R type DAC of the time.

Nevertheless, this is a magnificent CD Player which delivers. The build quality is impressive, everywhere I looked. The intersample overs is an issue I documented for fun here, but I suppose it will remain hidden as would show only with too hot CD masters, which should be avoided anyways, and not only for that sole reason.

I can't say I'm disappointed by the Teac VRDS-25X, but I would have liked the same without the ASRC. Maybe that was to ease the implementation of digital input at different sampling rates, but well, hell is paved of good intentions.

At the end of the day, this is the one of the ultimate R2R type conversion in a CD player I’ve measured. Thanks to Teac for that.

I hope you enjoyed the review. I prepared it for days, and I'm sure I'll have to update it again multiple times. But hey, that was a nice device to put under heavy testing :p

Cheers
————
Flo
 
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WOW. What a beautiful beast that obviously still would every audiophile proud today.

Many thanks for the review!
 
Talking to myself here, but from the discussion in the thread about inter-sample overs testing, I was wondering if this Teac was overloading at the ASRC as I suspect (maybe again in the oversampling filter), or at the analog output stage, or both, and how to create a test for that, if that is possible.

Fact is that when I test resistance to ISO with other CD players, I see them going above their initial max voltage at the output because of the clipping. It’s not the case with the Teac, with which I saw a small variance, like less than a dB even though the output shows clear clipping.

If the digital filter would be overloaded, then we would see it defeated at the output and see typical aliases of the conversion.

Maybe a square at various levels from -3dBFS to 0dBFS, as @restorer-john suggested me once, would reveal more information when looking at wide band (beyond 20kHz).

And @AnalogSteph if you read me, your various levels test tone 11025kHz could be a good candidate too. I did not put it into practice yet.

If you have ideas, I’m interested!

And in the meantime, a little illustration with a square 1002.27Hz @0dBFS from my usual Onkyo C-733 (top) and the same from the Teac (bottom):

1731757110946.png


We can indeed see that the oversampling filter of the Onkyo is defeated as it shows the standard aliases of a non-oversampled non-filtered DAC (more info in AD tutorial MT-017 : Oversampling Interpolating DACs).

The Teac draws a different picture, but not conclusive to me o_O
 
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Talking to myself here, but from the discussion in the thread about inter-sample overs testing, I was wondering if this Teac was overloading at the ASRC as I suspect (maybe again in the oversampling filter), or at the analog output stage, or both, and how to create a test for that, if that is possible.
The sheer volume of seemingly random anharmonics @ 5512.5 Hz / 7350 Hz is a very good indication of aliasing. You would never seen an analog stage doing that, this has to be happening in the digital domain.

It could still be the digital filter, though we haven't seen any behaving like that either. You would have to test another player that is using a "barefoot" SM5843AP to definitely rule it out... unfortunately this is not the most common chip. Your best bet on the used market might be an Onkyo DX-7711, DX-7510 or DX-7051, or a DX-6890 if you fancy a classic big boy Integra.

BTW, I wouldn't adjust DAC trim with a full-scale sine, where you might end up partially compensating for analog distortion if you're unlucky. I might use maybe -20 dBFS to -60 dBFS.
 
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Thanks @AnalogSteph, true for the typical aliases created with the Onkyo, this sure comes from an overloaded filter in digital domain.

The Teac hosts two band limited oversampling filters, after all. Maybe what we see is simply a subsequent overload. This is something we could potentially simulate in Audacity: 1) create a square at full scale, 2) oversample 8x, 3) downsample to 44.1kHz, 4) oversample again at 8x, 5) save file and analyze it to check for similarities.

Else I need to find one of these Onkyo, thanks for the list.

BTW I perform these tests with enough headroom in the interface, not to overload its ADC, indeed. I got caught the first time I went for them :)
 
It could still be the digital filter, though we haven't seen any behaving like that either. You would have to test another player that is using a "barefoot" SM5843AP to definitely rule it out... unfortunately this is not the most common chip. Your best bet on the used market might be an Onkyo DX-7711, DX-7510 or DX-7051, or a DX-6890 if you fancy a classic big boy Integra.

BTW, I wouldn't adjust DAC trim with a full-scale sine, where you might end up partially compensating for analog distortion if you're unlucky. I might use maybe -20 dBFS to -60 dBFS.

I've got a few Integra Onkyos with the NPC OS filters here (DX-7xx series). Cant recall which NPC but it's an 8x.

Agree with MSB trim at -60dBFS if the D/A is a single R2R per channel. This player has 4 DACs- how are they arranged- differential/parallel? MSB trimming methods vary in that case.
 
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BTW, I wouldn't adjust DAC trim with a full-scale sine, where you might end up partially compensating for analog distortion if you're unlucky. I might use maybe -20 dBFS to -60 dBFS.
That’s the procedure (0dBFS) from the Teac service manual. But if I try at -60dBFS, distorsion is buried into noise, there’s nothing I can adjust with 16bits data input. The AD1862 says to use -60dBFS signal but I suppose they assume 20bits data input.
 
I've got a few Integra Onkyos with the NPC OS filters here (DX-7xx series). Cant recall which NPC but it's an 8x.
Those are hard to find!
Agree with MSB trim at -60dBFS if the D/A is a single R2R per channel. This player has 4 DACs- how are they arranged- differential/parallel? MSB trimming methods vary in that case.
That’s probably the reason. There’s a gate array between the oversampling filter and the DACs. It feeds the AD1862 from 4 distinct data outputs, I think one inverted per pair of DACs.
 
That’s the procedure (0dBFS) from the Teac service manual. But if I try at -60dBFS, distorsion is buried into noise, there’s nothing I can adjust with 16bits data input. The AD1862 says to use -60dBFS signal but I suppose they assume 20bits data input.
I had a look: The SM distortion adjustment procedure is aiming to reduce high-level even-order distortion by balancing complementary DAC levels as well as possible, so doing it at 0 dBFS makes sense. This procedure actually has nothing to do with DAC low-level nonlinearity, an adjustment of which is not implemented in this player - DAC pins 3 and 14 are not connected to anything. You would have to retrofit the required parts first, flying lead style.

Having two DACs out of phase per channel would make tweaking low-level nonlinearity a bit tricky as well. You'd probably have to take turns adjusting one and then the other to bring even and odd harmonics down. If push comes to shove, temporarily short opamp U114A (U214A) pin 3 to ground by jumpering over R122 + V101 (R222 + V201), optimize things on U111 (U211) like that, then remove the jumper and proceed to U112 (U212). Make sure the "normal" distortion adjustment has been carried out first.
 
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BTW, have you noticed any effect of setting S101 #3 back to off? The corresponding digital input pin on U110 seems to be labeled DCUT, but since this is a custom programmed micro we obviously don't know what exactly it does. It may be related to the DC offset adjustment (V102/V202), maybe it turns on an internal highpass so that you can do that without being disturbed by any DC in the audio signal. Normally this unit ought to be flat right down to DC (on account of zero coupling capacitors in the audio path), but your measurements show about -0.2 dB at 20 Hz.

Incidentally, you may be seeing less than usual output from the RCAs due to relatively high output impedance of 1.1 kOhms and resulting interaction with Ultralite input impedance. The combination of 1k + 100R looks a bit bodgy to me, not sure what they were trying to achieve there. It's fairly high by CD player standards either way.

EDIT: One of your fellow countrymen was ripping out the AD1893 on one of these 20 years ago:
He also ended up removing the uPD65031C, noting:
I’ve omitted UPD65031 as well and settled for balanced D/A conversion by inverting one DATA line for each channel.
UPD65031 did some kind of signal conversion, maybe something similar to what is implemented in PCM1702 or PCM1704. Once I measured the output of one AD1862 (with UPD65031 still in its place). It measured perfectly even at low levels, but at signal levels higher than –1dB, even order harmonics increased to a very high niveau of about –35dB. There were no odd order harmonics at all. I guess the difference amp that followed the I/V stage has cancelled all even order harmonics.
Without UPD65031 guaranteeing low low level distortions, I adjusted every AD1862 as recommended in the datasheet. It’s great to see disappearing every distortion peak below the noise floor when adjusting the trim potis. With UPD65031 in its place trimming of AD1862 was actually useless.
Possibly this chip does some shaped dither à la PMD100 to tame DAC nonlinearity.

Also:
Before I removed AD1893 I tried to trim all four D/A converter chips (AD1862) for lowest low level distortion as described in the datasheet. I couldn't measure any distortions because the noise floor was at abaut -92dB. I could hardly identify a -80dB signal on my screen. Not very good for such an expensive CD player. Without AD1893 noise performance is at least 20dB better.
It goes without saying that an operation like that is not for the faint of heart.
 
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I worry with such in depth 'tweaking' threads like this, that some numpty out there, would attempt to 'improve' a player like this and twiddle with the presets on the dac board without the faintest effin' idea what they were doing! I've witnessed this, so it's not without precedent.

Having said the above, well done on another great review.
 
I worry with such in depth 'tweaking' threads like this, that some numpty out there, would attempt to 'improve' a player like this and twiddle with the presets on the dac board without the faintest effin' idea what they were doing! I've witnessed this, so it's not without precedent.

I say let them mess up whatever they want. It makes for "non-working"/"untested" bargains on eBay for people who know what they are doing. ;)

It used to be tuners where geezers would start twisting pots and ferrite slugs, often entirely destroying them in the process of "tweaking".
 
BTW, have you noticed any effect of setting S101 #3 back to off?
Nope, it was set to on when I opened it and I did not think about putting to off. I’ll try.

The corresponding digital input pin on U110 seems to be labeled DCUT, but since this is a custom programmed micro we obviously don't know what exactly it does. It may be related to the DC offset adjustment (V102/V202), maybe it turns on an internal highpass so that you can do that without being disturbed by any DC in the audio signal.
Good point again.
Normally this unit ought to be flat right down to DC (on account of zero coupling capacitors in the audio path), but your measurements show about -0.2 dB at 20 Hz.
I’ll measure again. The Service Manual says to verify the bandwidth is in +-0.5dB with a sweep from 20Hz to 20kHz. But I think I can check down to 1Hz.
Incidentally, you may be seeing less than usual output from the RCAs due to relatively high output impedance of 1.1 kOhms and resulting interaction with Ultralite input impedance. The combination of 1k + 100R looks a bit bodgy to me, not sure what they were trying to achieve there. It's fairly high by CD player standards either way.
Indeed.
I use line in at the back of the Motu but I can try the Mic in front. With TRS input I get 1Mohm input impedance from unbalanced line. I’ll give a try.
Thanks!
 
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The combination of 1k + 100R looks a bit bodgy to me, not sure what they were trying to achieve there.

They realized driving the 2114 through 100R to ground (when the output switch was 'off'- XLR 'on') was a bad idea, so they added 1k to not imbalance the XLR outs too much. The design is a mess.
 
BTW, have you noticed any effect of setting S101 #3 back to off? The corresponding digital input pin on U110 seems to be labeled DCUT, but since this is a custom programmed micro we obviously don't know what exactly it does. It may be related to the DC offset adjustment (V102/V202), maybe it turns on an internal highpass so that you can do that without being disturbed by any DC in the audio signal. Normally this unit ought to be flat right down to DC (on account of zero coupling capacitors in the audio path), but your measurements show about -0.2 dB at 20 Hz.
Tested and no change in bandwidth (nor distortion, of course).
 
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