Hi-Res THD(+N) vs. Output Level Measurements (ESS "HyperStream" vs. AKM vs. TI/Burr-Brown). And a bonus R-2R!

Notice last time as I ended off the post, I showed what I think is an interesting "high resolution" graph of THD(+N) vs. Output Level for the Topping D10 Balanced which uses the ESS ES9038Q2M chip. This was spurred on after some discussions on glitches and anomalies one might see due to the "HyperStream" architecture of the ESS chip.

These days, other than the occasional fully multibit or discrete R-2R DACs, the vast majority of what we're using are multibit/multilevel sigma-delta devices. This includes the brands I have listed in the upper graphic; Asahi Kasei Microdevices (AKM), ESS Technology, and Burr-Brown (which was acquired by Texas Instruments in 2000). We'll also talk about the Philips later. ;-)

Today, let's have a look at "high res" THD(+N) vs. Output graphs (XLR output where possible to keep noise as low as possible) comparing different DACs from these companies...

Notes on DAC DSD (1-bit PDM) measurements going forward...

In the early 2000's, we witnessed the battle over hi-res audio in the form of SACD vs. DVD-A. SACD, the brainchild of Sony, utilized a 1-bit Pulse Density Modulation (PDM) method they called DSD (an advertising term) whereas DVD-A had the ability to store up to 24-bit, 192kHz Pulse Code Modulated (PCM) digital audio data (multichannel up to 24/96).

From the beginning, there were concerns about this push towards 1-bit systems into the consumer space along with claims that 1-bit PDM should form some kind of archival foundation for music. There were critics include Lipshitz and Vanderkooy - see their paper "Why Professional 1-Bit Sigma-Delta Conversion is a Bad Idea" from the September 2000 AES. And the next year in May 2001, they followed up with "Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications". Even Bob Stuart chimed in on the unsuitability of DSD for "high-resolution audio" back in 2004. This is no surprise since Meridian was firmly with DVD-A including developing the MLP compression system which subsequently has been licensed by Dolby and renamed TrueHD; it looks like Dolby and Meridian had an arrangement dating back even to 1998.

These concerns around fidelity and the unsuitability of 1-bit PDM as an editable format in audio production are why in the professional world, we see audio recorded and edited in 24/352.8 "DXD" and Sony's own "DSD-Wide" (8-bit/2.8MHz) instead of DSD64/1-bit "DSD-Narrow".

While this was playing out in the academic/professional arena, the advertising industry including the "mainstream audiophile media" championed DSD and published all kinds of flowery words suggesting how it sounded "more natural", or "analogue-like" compared to PCM. While I don't think we can put an exact date on when DVD-Audio officially died as a viable commercial product, I think by 2005 it was quite clear that hi-res physical formats were not going to be mainstream and DVD-A did not have the number of titles available compared to SACD. My sense is that the hybrid-SACD feature with both DSD and CD-compatible layers was a major differentiating factor that has resulted in still a trickle of SACDs released these days.

I'm bring this stuff up now as an extension to the discussions around SoX-DSD and the Philips Test SACD articles last year during my series on the Topping D90SE review because I've been thinking about how best to standardize the DSD test signals I use when testing. Different DACs tend to handle DSD playback differently and I wanted to make sure that my test signal parameters are at least somewhat in line with the music encoded on an SACD or maybe DSD128 download these days.

Revisiting the TEAC UD-501 DAC (2013): THD(+N), DSD output, Sweeps, Jitter, 1/10 Decade Multitone 32. (And on SSD transition, Corsair MX500 SSD, and Hi-Fi+ closes comments...)

After Bennet/Dtmer Hk's comment on this previous post, I thought it might be interesting to revisit my longtime friend, the TEAC UD-501 DAC that I bought new when it came out in 2013. My preview and measurements (Part 2 PCM, Part 3 DSD) are still online of course.

I still remember doing those measurements in my previous home - it feels so long ago! ;-)

These days with the E1DA Cosmos ADC and RME ADI-2 Pro FS available for measurements, let's take a trip down memory lane at what has been over the years a reference DAC for me. I think the TEAC UD-501 is a special device that ushered in for many audiophiles a cost-effective, high resolution, well-built, reliable, low jitter, asynchronous USB audio interface which I suspect has influenced the design and performance of other DACs over the years.

Spec-wise, it was one of the first to catch my interest with the ability to handle PCM 384kHz and DSD128. The high quality metal case looks serious. It's based on a dual-mono TI/Burr-Brown PCM1795 configuration with dual toroidal linear power supplies. There are a number of PCM and DSD filters including the ability to turn off the 8x oversampling and allow "NOS" (Non-OverSampling) mode. The quad JRC MUSES8920 opamps they used were also another "feature" talking point back then. (Quite a complete specs page here for the TEAC.)

MEASUREMENTS: Does "Burn-In" / "Break-In" Happen for Audio DACs?

Jazz at Lincoln Center Orchestra with Wynton Marsalis did a great job last Saturday night at the Chan Center here in Vancouver. A friend once told me "Jazz should be seen as much as heard!" I think he's right. Always great to watch artistry in the making and correlate the sounds heard with how it was done. It's also a good opportunity to check out the acoustics in a moderate sized venue with minimal amplification of a 15-piece jazz band. Puts into perspective the dynamics and detail of what one hears in the home system.

Some writers seem to idealize live performances, but IMO, more often than not, what I hear on the home system is clearly better - cleaner, more defined, often much more enjoyable assuming the recording was a good one. The best performance and best seat of the house every night! Of course a studio record could never (and is not supposed to) capture the live ambiance or variation that a live performance can provide. That too is often a good thing as I reminisce on some concerts I've been to sitting beside rather annoying concert-goers. :-)

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I realized something the other night after measuring the Belkin PureAV PF60. I have been measuring my TEAC UD-501 DAC numerous times throughout its lifetime with me. I bought it back in early May 2013 and it has been in use for measurements, headphone listening in the evenings, and as part of my media room since then. I've run all manners of signals through it from standard PCM to high-resolution PCM to DSD64/128. It has been on for days playing "background" tunes as well as much more attentive "serious" listening through numerous albums of genres from jazz, to pop, to blues, to hard rock, to classical...

As of this writing in early March, I've had this DAC for 10+ months. I'll very conservatively estimate that I've put on >300 hours of actual audio through it (not just time turned on). I made sure when I first bought it that I would measure it within the first couple hours of use so that one day (now), I can go back and do a comparison to see if any kind of significant "break-in" can be demonstrated.

Note that the test set-up isn't exactly the same... Components are different like the playback computer, the RCA cable, and I'm in a different house as well. But the setup is comparable:

Win 8 PC --> shielded USB --> TEAC UD-501 --> shielded RCA --> E-MU 0404USB --> Shielded USB --> Win7/8 measurement PC

Results:

So, without further ado, here it is at 3 time points - within 2 hours of use, around 200 hours last year when I moved home, and just about 2 weeks ago with about 300 hours of use...

Summary

Frequency Response

Noise Level

THD

IMD

I figure there's no point measuring at lower resolution than 24/96 for something like this. You will note that the stereo crosstalk has a 4dB spread from highest to lowest (remember, we're talking down at -90dB here). The reason is simple. These are different RCA cables. At "<2 hours" and "~200 hours", I was using a 3' length of RCA cable whereas the ">300 hours" measurement was done with a 6' cable due to the inconvenience of a short cable in the current set-up. As I showed in the RCA analogue interconnect test, length of cable makes a significant difference with stereo crosstalk measurements (shorter is better) for the standard zip-cord type I'm using. This could also be the reason for the slightly lower noise floor especially notable on the THD and IMD graphs above (again, remember we're looking at the -120dB level here!). Otherwise, I see no evidence here of a significant change that would be audible.

Conclusion:

These measurements suggest that there really is no such thing as audible "break-in" for purely electronic devices like DACs within a reasonable period of time. People hearing the "effect" on a regular basis are more likely to be changing their psychological expectations over time than the device actually changing sonic character. I guess change could be happening within the first 2 hours but one almost never hear people claim this; for the most part, people recommend something like 100+ hours. Logically, if anything, electronic devices deteriorate in time as components (like capacitors) get old and connectors oxidize.

Sonic change for mechanical devices like speakers would make much more sense... InnerFidelity had an article about change in the sound of the AKG Q701 headphones over time (65 hours). The measured differences in those graphs are much more than what I'm demonstrating here.

In summary... I wouldn't be worried about "burn-in" with purely electronic devices. If you like the sound at the start, great. If not, maybe give it some time for your ears/brain to adjust and see if you like it then. Sure, keep the device on, blast some hard rock or play a burn-in CD if you feel this helps.

If there's no difference with complex electronic devices like the DAC, it'd be quite unreasonable to expect to hear a difference with totally passive "components" (eg. wire/cable burn-in). I find it suspicious that some companies like this one would claim cables needing 400-500 hours (17+ days straight!) to break-in! The more cynical side of me wonders if there is a benefit for companies to do this because it gives them a "grace period" to tell customers to wait. Furthermore the message itself promotes expectation bias towards improvement in time... "No worries! Give it some time to really sound it's best, Mr. Audiophile!"

Subjectively, I cannot say I've ever thought I could hear burn-in. The TEAC sounded good to me from the start and I'd be foolish to claim with certainty any difference at this point almost a year down the road unless of course there were some kind of night-and-day change (which there obviously hasn't been).

An observation - why is it that "burn-in" essentially always results in a reportedly better / smoother / less harsh sound? How does the component "know" that it should go in the right direction? It's not like the electronic component is functioning like the cells of the body where there's a homeostatic mechanism directing the 'healing' towards optimal functioning... Unless of course, we are dealing with a biological mechanism - the ears & brain. ;-)

Perhaps the standard measurements I present here are unable to capture whatever change there's supposed to be. As usual, I propose to those who are certain that burn-in happens to present any links to information or data to support this belief.

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Listening tonight:
Sonny Rollins Way Out West - just got reacquainted with this old 1957 jazz recording. A fantastic vintage recording from the golden age of analogue done with the tube Ampex 350 tape recorder. This was chosen as the first CD release by Mobile Fidelity back in the mid-1980's (I think 1984). There have been many reissues of this over the years and I think the highest resolution one would be the Analogue Productions SACD from 2002.

Enjoy the music everyone...

MEASUREMENTS: Digital Filters and Impulse Response... (TEAC UD-501)

By now, I suspect that most of us have read reviews and comments about the different types of digital filters available in many of the modern DAC's these days. A number of years ago - 2006 to be precise - I remember reading in Stereophile about these filter types and first saw the impulse response graphs in this article by Keith Howard. At that time, I remember being perplexed by all the variation of filter types possible but remembered how "cool" it seemed that here is another factor beyond jitter which may differentiate the "digital sound" from analogue... Furthermore, those graphs looked a little scary - Wow! Look at all that ringing!

Skip forward a few years and we see the introduction of new DAC's with various filter options available. I remember taking notice with the introduction of the Meridian 808.2 around 2009 with their "apodizing", minimal phase filter. In time, folks on forums would talk about how detrimental the whole "pre-ringing" would be with various manufacturers following the minimum phase filter design as selling points. Of course, in life, almost nothing is that simple; as if simply getting rid of one "issue" (like pre-ringing) would lead to joy and happiness without some price to pay. In terms of filter types, the price to pay includes a combination of early frequency roll-off and potential aliasing products.

As I have shown previously, the TEAC UD-501 has interesting properties including 2 digital filters based on the Burr-Brown/TI PCM1795 (SHARP and SLOW), as well as an OFF mode which results in removal of the oversampling filter and hence a "NOS" mode of operation. Today, let us have a look at the impulse response graphs, and consider issues like aliasing...  I even found another little surprise from this DAC!

First consider the sine wave frequency roll-off I previously found - this is why NOS DAC's sound a bit rolled off on the top end:

I. SHARP (Linear Phase) filter.

Let us start with the SHARP filter because this is the standard filter implemented in most DACs where you do not have a choice of settings. As you can see from the frequency graph above, this looks like the "classic" brick wall, and as you have no doubt seen, the impulse graph looks "classic" as well:

I played back a 16/44 "impulse" file and "recorded" it with my E-MU 0404USB at 24/192 to obtain that image above from Adobe Audition if you're wondering. Notice the positive deviation (up) which correlates with the phase of the impulse file therefore the TEAC maintained absolute phase. This is the standard "linear phase" filter with moderate amounts of pre and post-ringing. Here are graphs using white-noise to demonstrate the steep filter roll-off around the Nyquist frequency.

16/44 - White noise:

24/96 - White noise:

And here is what a 19kHz -10dB sine wave looks like with this filter measured up to ~100kHz (similar to John Atkinson's recent measurement changes in Stereophile which he discussed in the April 2013 issue, page 180):

We see a number of harmonics present with the largest being up at 38kHz (-90dB amplitude so obviously inaudible) representing one octave above the primary signal at 19kHz.

Although there is pre-ringing in the impulse response - hence lower temporal resolution, the good thing about the linear phase filter is that audible phase distortions are minimized. Also linear filters tend to measure very well compared to other implementations mainly because of the high frequency resolution which is what we usually look at with our FFT measures like frequency response, intermodulation distortion measurements, etc.

II. SLOW (Linear Phase) filter.

Next, let us consider the SLOW filter with a more gradual and earlier top end roll-off. I believe in some circles, the term "apodizing" may be used to refer to these filters although other places seem to also associate this term with minimal phase filters - if there's a simple consensus definition of "apodizing" as it refers to digital audio, please help clarify this for me. Here's the impulse response:

Nice...  It's still a symmetrical profile so it's a linear phase filter with low phase distortion. The (supposedly bad) pre-ringing is significantly reduced, hence better temporal characteristics, but here's the price one pays:

16/44 - White noise:

24/96 - White noise:

Notice that in both graphs, the roll-off doesn't hit the noise floor until well beyond the Nyquist frequency; we're looking at close to 35kHz for 16/44 and almost 75kHz with 24/96! No doubt this will result in aliasing distortions (the 19kHz -10dB graph):

There you go. Notice that high amplitude 25kHz aliasing distortion showing up now almost up to the level of the 19kHz primary.

So, basically we see a compromise with the SLOW filter...  Allow more aliasing, but also significantly decreasing the duration of the pre-ringing in the impulse response plot.

III. Digital filter OFF ("NOS").

The last of the TEAC UD-501 standard settings is of course with the digital filter OFF - the "NOS mode".

Impulse response:

Wow... No pre-ringing. It's essentially a square wave representing the duration of a single 16/44 sample. But in order to achieve this of course, the price to pay is tremendous:

16/44 - White noise:

24/96 - White noise:

Without any filtering, all the aliasing distortions get through. NOS DACs are said to sound "dirty" for this reason although of course many folks subjectively find the sound pleasing. Behold the 19kHz -10dB signal and all the aliasing and harmonics that gets through:

"Impressive"... Again, the price to pay for essentially having "perfect" impulse response with zero pre- or post-ringing. I would say that these effects are the most audible. So for those folks who really advocate for the "NOS sound", I guess it just means they prefer noisy, inaccurate reproduction with numerous harmonics and aliasing distortions. (I would love to see how these graphs look like on those expensive Audio Note NOS DAC's!)

IV. Minimal Phase filter!

Surprise! The TEAC UD-501 also has a minimal phase filter mode. This one is implemented by the 24/192 upsampling setting coupled with the digital filter turned OFF. Here's the impulse response:

As you can see, with a minimal phase filter, pre-ringing is not an issue. However, all that "energy" has been shifted into post-ringing. In fact, given the long ripple trail (about 4ms compared to SHARP filter 0.8ms start to finish of impulse), the absolute temporal resolution is the worst with this filter. However, since many believe that post-ringing is more "natural" and furthermore masked by the primary signal, its presence isn't particularly problematic (according to them). Here's the rest of the graphs:

16/44 - White noise:

24/96 - White noise:

Using an onboard ASRC chip, the upsampling algorithm does a excellent job achieving a sharp frequency roll off to prevent signals going beyond Nyquist.

Here's the 19kHz -10dB signal:

Very nice - even cleaner than the SHARP filter (as one would predict given the lower temporal resolution). Some harmonics present but no aliasing distortion.

What's the price to pay for minimal phase filters? The post-ringing duration is quite long in this implementation. Also, given the non-linear characteristic, there may be audible phase distortion.

V. So what?

Yup... Now you know what the graphs look like and the options available on the TEAC DAC (and of course variants of these filters with other DACs)...

So what?

Objectively, I think it can be said that the fact that NOS DAC's don't just plain sound awful really speaks to just how forgiving our ears are!

Subjectively, as one who aims for accuracy in waveform reproduction, I've found that my "favourite" remains the standard linear phase (SHARP) filter both based on what I hear and intellectual satisfaction. I've never been a fan of the NOS sound - the rolled off highs and slight "dirtiness" to the sound quality robs it of resolution IMO (I presume this is secondary to the high aliasing distortion). I will however give a thumbs up to the TEAC engineers for the minimum phase upsampling with digital filter turned OFF. Some audiophiles have commented that in many DAC chips, it's actually not the pre-ringing that is an issue, but rather the processing of the digital filters themselves and taking out these built-in filters improves the sound quality... At least there's that opportunity on the TEAC - I'll have to listen a bit more...

One final thing. Consider this, just how audible is pre-ringing anyways? Realize that music isn't an impulse so these graphs are artificial exacerbations of the "ringing", just like testing with square waves generally do not show perfect instantaneous transitions in real life equipment. But even if I zoom into that SHARP impulse response (note I purposely inverted this impulse), this is what I see:

We have 7 waves over 60 samples, measured at 192kHz. This means we're looking at low amplitude pre-ringing at around the Nyquist frequency ~22kHz (remember 16/44 impulse signal). Hmmm, what kind of human physiology has the capability to hear that lead up to the primary impulse wave? Has anyone ever proven this is audible in music? Any references to controlled trials? Even the golden ears at Stereophile seemed unclear about the significance of these filters in that 2006 article before folks started claiming pre-ringing was "bad" (they were even listening to maximal phase filters with lots of pre-ringing and didn't dislike that nor show clear preference to the minimal phase filter).

As usual, if anyone has information/data/results about the audibility of these digital filters, please drop a note!

Time to go listen to  those 2013 Eagles remasters now...

(PS: Looking at the oversampling function of this TEAC DAC, I was hoping ASUS could have done something similar with the XONAR Essence One as I noted a few months ago - instead they made a gimp upsampler which rolled off way too early! Unfortunate.)

Addendum: As suggested by Shoddy in the comments - if I just have a look at the pre-ringing waveform from the SHARP (linear phase) filter, boost the levels and check the FFT; here's what it looks like:

Indeed, most of the energy of the pre-ringing waveform seems to be clustering around the Nyquist frequency and measures about 10 dB louder than what I assume is low frequency noise...

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.

Setup:

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:

THD:

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.

Noise:

THD:

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).

Summary:

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!

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NB:

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. ;-)

MEASUREMENTS: TEAC UD-501 PCM Performance (Part 2) [Updated June 23, 2013]

Okay folks, let us continue with the TEAC evaluation... First, we need to look at the PCM performance of this DAC. Although DSD may be the "hot" feature of DAC's these days, PCM remains the most important digital encoding method. A good DAC MUST perform well with PCM music.

General setup:

MacBook Pro (Decibel bit-perfect) --> shielded USB --> TEAC UD-501 --> shielded 6' RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop

The TEAC has quite a "hot" XLR output and unfortunately clips the E-MU so I was unable to get an accurate reading without using volume attenuation (rest assured it does look very good, dynamic range is probably about another 6dB better that what I got with the RCA 24-bit tests; beyond the resolution of the E-MU).

A look at a 24/44 1kHz square wave at 0dBFS using SHARP digital filter. RCA peak voltage at 2.8V. Nice wave morphology. Yellow is right channel, blue is left; channel balance looks good.

PCM 16/44:

The most common digital sampling rate is of course "good" old 16/44 Red Book format. These days, any DAC worth it's salt MUST perform close to ideal at 16/44...

Summary (RightMark 6.2.5):

As you can see, the UD-501 was measured with the 3 digital filters (DF's - OFF, SLOW, SHARP). Comparison was made with the Logitech Transporter (ethernet), Touch (ethernet), and Oppo BDP-105 Blu-Ray player (USB). Unless explicitly adjusted, vast majority of DAC's utilize some form of the SHARP filter by default at least for 16/44. Overall, you see that 16-bit audio is absolutely no problem for any of these devices.

Here's the frequency response of the devices:

The most obvious thing to see here is that the Touch rolls off on the low end by 1dB compared to the others, and the digital filter "OFF" setting of the UD-501 rolls off quite early starting around 5-6kHz...  Let us focus for a second just on the TEAC:

Interesting; the SLOW and SHARP settings are pretty self explanatory (the Transporter has similar settings). The OFF setting results in significant roll off even earlier and by about 18kHz, the OFF setting is about -2.4dB, compared to SLOW setting at -1.7dB, and SHARP at -0.1dB.

Hmmm, where have we seen that kind of filter "OFF" curve before? Oh yeah, the old TDA1543 :-). Here ya go:

What does this mean? Yup, the TEAC can function in "NOS [NonOverSampling] mode" with the digital interpolation filter turned off! Behold, stair-stepped NOS waveforms out of a modern DAC with DSD capabilities:

Digital filter OFF:

Digital filter SHARP:

That's really quite a trick from the TEAC!

Noise Level:

All pretty equivalent with fantastic level of functioning. Note the 60Hz powerline hum visible.

Stereo Crosstalk:

Very close; worst being the SB Touch. Same 6' shielded RCA cable used in all tests.

PCM 24/96:

24/96 is the "sweet spot" for high-resolution PCM DAC's these days. Certainly in DAC's I have previously tested, once you go above 24/96, there's often deterioration in the dynamic range. Furthermore, I do not believe there is any scientific evidence to suggest human ears can experience sound beyond the resolution encompassed by 24/96 so it's important that a true "high resolution" DAC be able to demonstrate adequate performance at this level.

Here's the "big board" summary with a number of devices tested (you probably need to click on the image for more comfortable reading):

As you can see, for comparison I've thrown in results from the E-MU 0404USB itself, Logitech Touch, ASUS Essence One, Logitech Transporter, and Oppo BDP-105. Obviously every one of these devices is capable of >16-bit resolution showing improved dynamic range beyond the 16-bit test above. "Top tier" devices are the TEAC, Transporter, Oppo; each measuring beyond 110dB potential dynamic range using the RCA output - essentially at the limit of the E-MU's abilities (the Essence One & E-MU would also be on this list when using balanced XLR or TRS cables respectively).

Frequency Response:

Since the graph got too busy, I removed the ASUS and Oppo - they basically look like the Transporter in terms of frequency response. Again, the Touch drops a dB down near 20kHz.

Here's the graph with just the TEAC settings plus the old TDA1543 NOS:

Notice again the similarity of the TEAC's digital filter "OFF" setting and the TDA1543 NOS DAC in the high-frequency end.

THD:

Interesting increase in high frequency noise with the digital filter "OFF". Looks like unfiltered delta-sigma noise shaping coming through?

PCM 24/192:

Next, one more step up in sampling rate:

For interest, I threw in the Logitech Touch with EDO plugin --> coaxial --> AUNE X1 DAC. Notice how well this combination measures! The AUNE X1 is only a $200 DAC and the combination produces very respectable measurements (and sound very good IMO). In comparison, I am disappointed in the Essence One going from 24/96 to 24/192. The TEAC and Oppo really hang in there with essentially identical results compared to 24/96 - great to see!

Frequency Response:

As I mentioned in the MUSE TDA1543 measurements, one way to improve NOS DAC performance is to feed it with higher sampling rate data...  In doing so, you get closer to the performance of oversampling interpolation filters. You see this here - the higher the sampling rate, the closer the digital filter "OFF" curve gets to the "SLOW" and "SHARP" settings (in fact, you see in the next section, they become identical).

There's that early roll off with the ASUS Essence One previously measured.

Noise Level:

Essence One getting a bit noisy at high sample rate compared to the others (realize it still has >100dB dynamic range though). Again, we see quite a bit of high frequency noise with the digital filter "OFF".

PCM 24/384 (more than DXD [352.8 kHz]!):

This is a "pseudo-test" actually. The fact is that the E-MU 0404USB is incapable of digitizing at 384kHz so what I did was upsample the 24/192 test signal using SoX so see if running the TEAC at the higher sampling rate will cause a measurable loss in the analogue output dynamic range or worsen noise characteristics within the measurable capability of the E-MU.

Summary:

First 3 columns were measurements done with the TEAC running at 24/384 with various digital filter settings. The last column is the "SHARP" filter measured at 192kHz. There may have been very subtle loss in dynamic range. Some or even all of this could be due to the upsampling conversion algorithm. In any case, the measurements look excellent and it seems indeed the TEAC is able to maintain low noise even at the extreme sampling rate of 24-bit & 384kHz!

Frequency Response:

Note how the NOS-like digital filter "OFF" setting is identical to the other settings now. Basically, sampling at 384kHz is like 8x oversampling of a 44kHz signal (2x oversampling of 192kHz).

Jitter:

As usual, let us look at some FFT's from the Dunn J-Test. For simplicity, I'll just show the spectra from the SHARP filter setting.

USB input (16-bit and 24-bit spectra):

Coaxial input using CM6631A USB to S/PDIF:

TosLink input using CM6631A USB to S/PDIF:

TosLink input again fed by CM6631A with *24/192 upsampling*:

The reason I didn't bother showing any results from hardware upsampling to 24/192 in the tables above was because the numbers and graphs looked essentially unchanged. However, there is one situation where upsampling makes sense... The same reason Benchmark chose to use ASRC (Asynchronous Sample Rate Conversion) for the DAC1 and DAC2 - jitter reduction. Although by no means high, the sidebands are more pronounced using coaxial and TosLink interfaces. The sideband peaks around the primary signal clearly were reduced with 24/192 upsampling using the TosLink input. As usual, whether anyone can actually hear this difference in properly controlled testing is another matter!

Summary of PCM Results:

TEAC has created a machine which objectively compares very well to some other excellently measuring devices like the Logitech Transporter and Oppo BDP-105. It's great to see that even operating at the extremely high DXD-level sampling rates, noise level remains low and dynamic rage appears preserved.

What I found surprising was the option to allow the digital filter to be turned "OFF"; I don't recall any reviewers spending much time on this (even the AudioStream review just glossed through this and didn't comment on the sound). This setting puts the DAC into a "NOS mode" where digital interpolation is suspended - this appears novel especially in a device with low-jitter asynchronous USB interface and a true 24-bit (err... ok, 32-bit as if that makes a difference) DAC... In general NOS DACs these days are still based on obsolete decades-old DAC chips like the Philips TDA154x (16-bit) or Analog Devices AD1865 (18-bit) which tend to perform poorly on measurements. Although personally I am not a big fan of the roll-off and aliasing distortion, some have commented on subjective improvement by taking out the digital oversampling filter, so I definitely consider it a positive that TEAC offers this option for anyone to try (in real time with instantaneous A-B'ing no less just by turning the knob)! I can certainly see this option useful to tone down some of the overly "bright" digititis-inducing recordings. Looking at my pop CD collection, an example where this was demonstrable was Jason Donovan's disco-inspired Too Many Broken Hearts from Ten Good Reasons (first pressing, 1989) where the OFF setting was more tolerable after 3 minutes :-). As a compromise, the SLOW filter may be reasonable.

As I mentioned at the beginning, PCM remains the cornerstone of digital audio. These TEAC UD-501 results suggest that nothing has been sacrificed in terms of performance in the PCM domain. Note that the ASUS Essence One is also based on the PCM1795 chip in dual-mono configuration but doesn't measure as well, highlighting the importance of the electronics around it like the analogue output stage, power supply and USB/coaxial/TosLink interface circuitry affecting the final output quality. One thing I wish the TEAC had from the ASUS is the beefier headphone amp though.

Bottom line: these results are consistent with the excellent subjective sound quality described in the previous UD-501 blog post. I would happily present some kind of award if it meant anything :-).