Sunday 12 May 2013

MEASUREMENTS: TEAC UD-501 DSD Performance (Part 3)

Okay, this is the 3rd (and likely) last part of my TEAC UD-501 review.

This will be the second time I measure a true DSD device. The first time was about a month ago when I checked out the beta firmware for the Oppo BDP-105 at a friend's house. Unfortunately, although we seemed to be able to play some DSD128 samples from 2L properly in DFF format, I was unable to play the DSD128 encoded test signals properly, so this will be the first time on this blog I can show the improvements between DSD64 and DSD128.

The setup is the same as with my PCM tests.
MacBook Pro --> shielded USB --> TEAC UD-501 --> 6' shielded RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop

DSD playback software: JRiver Mac alpha 18.0.177.

I used the same KORG AudioGate 2.3.1 DSD-encoded test signals as I did previously with the Oppo tests. Basically, I took the highest resolution 24/192 synthetic test signal produced by RightMark 6.2.5, ran them through AudioGate to produce the equivalent DSD64 and DSD128 versions to test. Of course, doing this involves a transcoding step so the result at best is that of the PCM signal minus small losses due to the transcoding using AudioGate. These tests will not demonstrate the best that the TEAC (or whichever DSD device tested) can do, but rather, hopefully a reasonable approximation. Doing this also implies the possibility that different conversion software could produce different results.

As you likely are aware, 1-bit quantization results it lots of noise. DSD deals with this by virtue of the high sample rate - DSD64 at 2.8MHz and DSD128 at 5.6MHz along with noise shaping to shift the noise up into the ultrasonic parts of the spectrum. By doing this, DSD is capable of high signal-to-noise ratio in the audible spectrum rivaling that of 24-bit PCM. However, this noise floor is not flat like for PCM, but rather will gradually increase higher up in the spectrum and then escalate rather quickly by 20kHz for DSD64.

As I showed in the Oppo test, when you look at something simple like a 1kHz sine wave through DSD64, you can actually make out this high-frequency noise. Here is how it looks coming out of the RCA output from the TEAC:

DSD64 1kHz sine wave at -6dBFS:

DSD128 1kHz sine wave at -6dBFS:

PCM 16/44 SHARP digital filter, 1kHz sine wave:

As you can see, the DSD128 and PCM (with digital interpolation filter) waveforms look nice and smooth compared to the DSD64 output. Good "concrete" example of the improvement with DSD128.

As I mentioned, noise shaping will move the excess noise up into the ultrasonic spectrum. Due to concerns that this ultrasonic signal may cause some amps to oscillate, an analogue lowpass filter is used to remove some of this noise. The finite impulse response (FIR) filter is selectable for the TEAC based on the options in the PCM1795 datasheet:
FIR1: fc = 185kHz, gain = -6.6dB
FIR2: fc = 90kHz, gain = +0.3dB (default)
FIR3: fc = 85kHz, gain = -1.5dB
FIR4: fc = 94kHz, gain = -3.3dB

Now before one gets overly impressed, realize that these filters operate in the ultrasonic range... That is, you're not going to hear the difference other than the potential gains or attenuation as it may affect the audible frequencies. However, those gain values are very audible indeed with FIR2 sounding clearly loudest and FIR1 softest (again, like for the PCM filters, just a turn of the knob allows you to instantaneously A-B the difference on the TEAC).

DSD64 (1-bit, 2.8MHz):

Let's get to it then, RightMark 6.2.5 results playing the DSD64 transcoded 24/192 test tone:

The first 4 columns are the TEAC with FIR filters 1-4. The fifth is the TEAC playing the original PCM test with the SHARP filter (best measuring of the three). Finally the last column is the Oppo BDP-105 playing from a USB stick.

Notice the default FIR2 filter performed the best.

Here's what the frequency response looks like:
Notice how little difference there is between the 4 FIR filters! I believe this is to be expected since the analogue filters as I mentioned are operating way in the ultrasonic spectrum and differences are seen only above 20kHz. In comparison, the PCM test behaves appropriately (yellow), and the Oppo (red) interestingly has a filter that measurably starts rolling off by 10kHz (no biggie, still only down by -0.7dB at 20kHz).

Noise Level:


It's quite evident from the graphs above just how noisy DSD64 is above 20kHz. The 24/192 PCM test signal (yellow) is cleaner beyond 20kHz. The noise characteristics of the TEAC and Oppo are about the same with a hint that the Oppo's analogue lowpass filter is stronger.

DSD128 (1-bit, 5.6MHz):

Finally, I get to have a look at what DSD128 measures like. Theoretically, 1-bit sampling going from 64x to 128x should allow the machine to push the noise an octave higher. Therefore, we should see the noise start ascending from 40kHz onwards. Likewise, the SNR in the audio spectrum should improve as well but this would be already beyond the E-MU's ability to measure.

First, the Summary:
Hmmm, nice. No deterioration in noise or dynamic range compared to DSD64 (like I said, these should be even better at DSD128 but will need a better measurement device). Again, we see the fluctuations in measured values between FIR1-4. The 5th column again is the native PCM 24/192 SHARP filter result.

Frequency Response:
Interesting, once we hit DSD128, they're all essentially identical. This too could be just a limitation of the E-MU hardware and RightMark software.



In both the graphs above, we see that DSD128 indeed does push the noise floor out to ~40kHz as predicted and from there, the level rises quite significantly. Although the noise level isn't as high as DSD64 by 100kHz, it is still significantly higher than with PCM (yellow).

Dunn J-Test:

As I've mentioned before with the Oppo tests, you cannot apply the J-Test to DSD and expect to stimulate jitter since this was designed for PCM with 16/44kHz or 24/48kHz sampling rates in mind; specifically for digital S/PDIF or AES/EBU interfaces between transport and DAC. Nonetheless, here's what they look like through the TEAC in DSD64 (DSD128 looks about the same).

16-bit J-Test:

24-bit J-Test:
Looks perfect (as it should with DSD).


Well everyone, there you have it. DSD64 and DSD128 off the TEAC UD-501. Contrary to the French Qobuz review where the authors have suggested that DSD was being converted to PCM internally, the test results here are consistent with direct DSD decoding. For a comparison, look at the results with the Pioneer DV-588A where internally 24/88 PCM conversion was being done on DSD64 (look at the unusual noise spectrum, PCM-type frequency response, and J-Test showing the jitter pattern found in PCM).

The TEAC UD-501 performed essentially the same as the Oppo BDP-105 from what I see here. This is good and provides a nice comparison with the level of performance out of the SABRE32 DAC. Again, this level of performance certainly is consistent with the subjective listening of DSD64 and DSD128 material I reported earlier.

About those FIR filter settings. There really is no difference in sound other than the differences in gain from my listening. I'd just stick with FIR2 unless there's a reason you would want to attenuate the signal (for example, FIR1 with -6.6dB allowed me to measure the TEAC through the XLR output without clipping the E-MU 0404USB - results are good BTW and shows the benefits of balanced interconnects).

Considering the ultrasonic characteristics of these filters for a moment, FIR1 with a cutoff frequency (fc) of 185kHz basically will allow all the DSD noise to pass through unattenuated. So, if you figure you have an amplifier that can handle all that ultrasonic noise feel free to give this setting a try... It's like the old "custom" filter setting of the first Sony SCD-1 (vs. the stronger "standard" filter) where some audiophiles felt the sound became more "open" and "airy" with the weaker filter. FIR3 is the strongest filter out of the 4. Personally I'm happy with FIR2 with its fc of 90kHz. Look at the PCM1795 datasheet for some nice graphs for these filters.

Benefits of DSD128 over DSD64 are clear in the measurements - you can see it in the cleaner sine wave above, and it pushes the ultrasonic noise out to ~40kHz. Subjectively, it clearly has the potential to sound fantastic if the recording is up to par. I guess we'll see in the days ahead just how much commercially available material gets released...

Bottom line: For the price, the TEAC UD-501 DAC offers up a lot of value and fantastic set of features. Sonically, I believe this DAC can easily trade punches with the best out there - whether PCM or DSD.

Enough with testing... Time to enjoy the music!

Before I end off, I just want to make a general comment & plea about the state of DSD computer audio now that I've got a chance to try it out.

The DFF and DSF file formats are inadequate. Compared to FLAC, APE, or ALAC, these DSD file formats feel geriatric! Seriously, PLEASE get the file format right.

Firstly, we need good tagging features - all the more important for DSD since much of the excellent material consists of classical music where it's important to document conductor, orchestra, composers, title, year of performance and composition, etc... Please let me be able to use something universal like the excellent Mp3tag to manage all my PCM and DSD files.

Secondly, a standard DSD file format NEEDS lossless compression. DSD is extremely compressible - using DST with DFF files, I regularly see compression ratios >2.5:1 losslessly, getting up to 3:1 in some tracks. This becomes even more useful for DSD128 where the space savings are very substantial. By doing this, DSD64 can be compressed to file sizes overall smaller than 24/88 encoded with FLAC with equivalent (some would say better) sound quality... I'd certainly be happy with that! Lossless compression would also save file transfer times and cost of storage for the music producer, distributor, and of course consumer. Seriously, what other modern hi-resolution media format doesn't allow for at least lossless compression?

Over the months, I have heard DSD apologists talk about how you don't need compression or native tagging because "hard drives are cheap" and "JRiver can tag with its internal database". Sorry... That's not good enough. This is a foundational matter and will impact future generations of products, so it's important to get it done properly instead of rely on work-arounds.

Please guys, now that you've gotten together to define DoP and manufacturers to make these DAC's, lets get it done right with a standard DSD format that's fully capable, preferably "free" as in "open". Maybe some enterprising coder like Josh Coalson of FLAC fame can apply their expertise!

Thanks in advance. ;-)


  1. Those level differences for the various FIR settings do they originate from the applied algorithm or are they deliberate to 'envoke' subjective differences between those 4 settings (so you actually hear something 'changes' when selected) ?

  2. Good question Frans.

    I wondered as well of all the potential options, why TI/Burr Brown chose these 4 analogue filters to implement. Notice that among the PCM179x line of "Advanced Segment" DAC's, there is variability to the analogue filters. For example, the PCM1792 looks like this:
    FIR1 - fc=185kHz / -6.6dB
    FIR2 - fc=77kHz / -6dB
    FIR3 - fc=85kHz / -1.5dB
    FIR4 - fc=94kHz / -3.3dB

    I presume it's not just for psychological effect since most DAC's will probably just choose one of them; very cool that this DAC has all the options open so the user can have a listen to each one. Maybe the engineers from Burr Brown can chime in :-)

  3. so can it decode apple lossless?

  4. The BB1795 converts to PCM, there no mentioning of DSD clock.