PROTOCOL: The DiffMaker Audio Composite (DMAC) Test.

Up to now, I have been using primarily a combination of RightMark along with the Dunn J-Test for my audio measurements. IMO, the standard procedure I've used thus far isn't bad and can already detect many anomalies in the hardware tested so far within the limits of the test system (ie. using my E-MU 0404USB as the ADC).

In the days ahead, I am going to start doing some Audio DiffMaker tests where appropriate; another freely available tool for the audiophile tester to find out what works, what doesn't, and to identify the difference. If you have not already guessed, some of my motivation in doing these tests is not only to feed my own curiosity, but also to encourage others to understand the tests and technology - hopefully in time elevate the knowledge base rather than unquestioned acceptance of many senseless audiophile myths out there.

If you peruse the DiffMaker site, it's quite obvious what this program does. It basically takes two recordings of the audio (presumably under 2 conditions or with different hardware), inverts one of them, and applies it to the other to see if the signals "null" each other out. The "magic" of course is in the algorithm used to align the samples in terms of time (including sample rate drift), and signal amplitude. If the recordings are identical, there should be a complete null where the result is silence. The program will create the "null" WAV file to review (very useful) and spit out a number representing the amount of "audio energy" left in the resulting null'ed audio file - expressed as dB's. The program calls this the "Correlated Null Depth". The higher this value, the more correlated the 2 samples are (ie. the "closer" they sound).

The beauty of this method is that one is free to use any audio input signal - freed from the need to remain bound to synthetic test tones which thus far I have been using. The main limitation so far with this software I have seen appears to be memory limits I've run into with long audio segments, it also takes a fair bit of computation to get the results. With my 6GB Windows 8 x64 laptop and DiffMaker 3.22 (September 2008), once I go beyond ~35 seconds 24/96 audio, the program runs into an error condition - presumably memory issues. Fair enough, I think 35 seconds is adequate to allow a decent comparison.

After a bit of consideration, I decided to create a "composite" audio test signal that I hope represents a reasonable survey of real music that is also challenging enough for a high-end audio system to reproduce.  For fun, I've called this audio track the "DiffMaker Audio Composite" (DMAC) Test which I think would be a reasonable test to apply to future evaluations I post on the blog. The DMAC consists of the following 4 tracks - all downsampled to 24/44kHz. Why you may ask? Simply because most digital music exists as 44kHz so it's important that this sampling rate be done right, and it is believed by many that 24-bit depth is the major factor lending improvement to hi-res audio quality. The tracks:

Rebecca Pidgeon - "Spanish Harlem" 3:02-3:11 (The Raven, 1994) - 9 seconds taken from the 2009 Bob Katz 15th Anniversary Edition at 24/88. Well known to most audiophiles as a vocal test track... Shakers in the background and such... Good evaluation of the mids.

The Prodigy - "Smack My Bitch Up" 2:13-2:22 (Fat Of The Land, 1997) - 9 seconds of loud and clipped techno/electronica. I applied -2dB to the track to allow extra headroom for the ADC without clipping. Low dynamic range, but intense bass. An example of "modern" mastering efforts. Taken from the CD 16/44.

Rachel Podger & Brecon Baroque - "Concerto In G Minor, BWV 1056: Presto" 00:02-0:10 (J.S. Bach: Violin Concertos, 2010, Channel Classics SACD to 24/88) - 8 seconds of lovely string classical work - good mid-range to highs, nice "microdynamics".

Pink Floyd - "Time" 00:06-00:10 (Dark Side Of The Moon, 1973) - 4 seconds of bells & chimes taken from the start of this track. Quite a lot of high-frequency content, detail in the sound, and channel separation. I used the 2011 24/96 Immersion Box Set remaster.

Interspersed between each track are dual bursts of 0.1s 1kHz tone at -4dBFS interspersed with 0.1s silence. This serves as a "beacon" for DiffMaker's alignment algorithm. The trickiest part of this test is temporal alignment and doing this has significantly improved the consistency of the results for me.

DMAC Waveform:

Vital stats for the 35 second test track:
DR9 (thanks in a large part to the loud compressed Prodigy track). Peak volume: -1.37 / -1.46 dB. Average RMS Power: -27.1 / -26.66 dB.

As with any proposed test, first thing to do is some form of validation.

I. Reliability

Setup: MacBook Pro Decibel --> shielded USB --> TEAC UD-501 (SHARP filter) --> shielded  RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop

Although the DMAC track is 16/44, it was measured back at 24/96 where the E-MU 0404USB functioned optimally. I also turned ON compensation for sample rate drift. The rest of the settings are as per default.

Here are 15 runs with the DMAC track played back through my TEAC UD-501 looking at the reported "correlated null depth" as an objective measure by the program. I also had a look at the null waveforms to ensure there were no obvious technical issues. The runs were spaced out over 24-hours to capture changes in conditions that may be present over the course of the day, temperature variation, electrical condition, and how long the DAC and ADC had been turned on in order to get a sense of the error range. Interestingly, from what I can tell, the result seemed to vary with ambient temperature. Trials 4-8 were done in mid-day with temperatures going up to ~30 degrees Celsius where I did the tests. Of course, maybe other factors like electrical noise and powerline quality may have a hand in the variation during that time of the day. In general, since I do most of my testing in the evenings, those lower results serve as a reasonable lower extreme for this test. (BTW: I turned the WiFi off on the computers if anyone thinks that makes a difference.)

As you see, there is a range of results (mean = 80.74/79.66, standard dev = 3.88 / 3.89). Remember that because we are measuring the analogue output from the DAC, there will be some noise in the signal - this is an inevitable property of analogue signals especially since I'm re-digitizing it back with the ADC to measure.

II. Validity

Given the error range above, is it good enough to detect very small changes?

Let's try to measure the following conditions:

1. Adobe Audition 3 Graphic EQ boost of +0.3dB at 16kHz with another EQ boost of +0.3dB at 5kHz. The 16kHz change should be inaudible, and the 5kHz adjustment likewise should be inaudible except maybe to the best young golden ears. I was unable to ABX this EQ change using the Sennheiser HD800 + TEAC UD-501.

2. TEAC UD-501 digital filter set to SLOW. This involves a high frequency roll-off starting north of 15kHz. May be detectable to those with excellent high-frequency hearing but I think for the vast majority of us, this difference is unlikely to pass an ABX test.

3. TEAC UD-501 digital filter set to OFF. This is of course the "NOS" mode for the TEAC. I can quite readily hear the difference in an A-B test. Should not be a problem for the DMAC protocol.

Reminder of the TEAC filter frequency response curves:

Result of test conditions 1-3:

Not bad. Note that I only did 5 runs of each test condition (vs. 15 runs for the DMAC Reference). The Graphic EQ test and especially the "NOS" mode demonstrated significant variance from the Reference results. Setting the digital filter to SLOW hinted at lower correlation depth but remained within the range for the Reference tests suggesting that the DMAC protocol was unable to differentiate this condition (not surprising by the way since musical content drops off significantly up at 15+kHz where the SLOW roll-off operates).

4. Changes due to MP3 encoding. We know lossy encoding changes the bit-perfect nature of the signal. We know ~320kbps is audibly very subtle (as per the test that kicked off this blog). We know that lower bit rates will result  in more sonic degradation. Can the DMAC test differentiate MP3 from the lossless and further discriminate different bit rates using LAME 3.99.5 (3 runs each condition)?

Nice, it looks like indeed we can! Good correlation between decrease in "correlated null depth" (increasing variance) and lower bitrate for MP3 encoding. The machine isn't fooled by MP3 algorithms :-).

Of course there are other things I can do to demonstrate the validity of this test to show variance... I've done a few other things like varying degrees of EQ changes to demonstrate the correlation which I won't bore you with here.

Summary:

As you can see, it looks like the DMAC Test is quite reliable and can be shown to discriminate differences in audio even down to levels that are very unlikely to be heard by human listeners with the E-MU 0404USB as a measurement device.

A word about tests like this and audibility. Remember that humans listen with a powerful psychoacoustic "filter". The ear has significant physiological limitations. For example, we are sensitive especially to the 1-5kHz audio spectrum and quickly lose sensitivity to frequencies higher up - have a look at the Fletcher-Munson curves. Secondly, psychoacoustic effects like simultaneous and temporal masking renders certain details inaudible. This is part of the "magic" of lossy encoding algorithms - allowing software to throw out quite a lot of data/details yet maintaining excellent audio quality. (Interestingly, the DiffMaker program does have an "ARM-468 weighted energy" setting which may be closer to human perception but I have thus far not tried it yet.)

The results of tests like this one I believe can be used for correlation of the sonic output to demonstrate variance between signals (which is of course the intent of the software developers). However, because the machine does not have the psychoacoustic mechanism of humans, the results can never directly correlate with what is being heard subjectively. A good example is the similar score between the digital filter OFF (NOS) condition and MP3 192kbps. They both score around 50dB in "correlated null depth", but I would argue the MP3 encoding changes the sound significantly less than removing the digital filter (ie. the effect from a NOS DAC). In an AB test, I can detect a "dulling" of the high frequencies on tracks like the Prodigy sample with the digital filter turned off whereas the MP3 sounds less 'colored'.

One more thing about using the "Correlated Null Depth" value. What I'm showing here is all based on the measurements off my equipment using the E-MU 0404USB, TEAC UD-501 DAC, and procedure/settings I'm using. This means it's only useful for my test purposes and cannot be generalized otherwise. The measured value itself of course will fluctuate and time-to-time, I'm going to need to readjust the reference score based on hardware changes.

I look forward to incorporating this test with the others in the days ahead...

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Addendum: Curious to see the difference between Reference null and what happens without a digital filter (ie. "NOS mode" on the TEAC)?

The following is what a high quality null WAV output looks like (~85dB) - "Spectral Frequency View" where the X-axis is time and Y-axis is frequency with the color representing amplitude at that specific frequency (blue/dark = low amount, red/bright = high):

Here's the TEAC UD-501 in "NOS mode" with digital filter turned off:

Impressive amount of variance. Also note the amount of high frequency content being recorded above 20kHz without the filter in place!

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)

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).

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 :-).

INITIAL IMPRESSIONS: TEAC UD-501 USB PCM & DSD DAC (Part 1)

This guy arrived at my doorstep on May 7. Over the next week or so, I'll just build up this blog with TEAC UD-501 information as I gain experience with the unit.

Initial Impressions & The Basics:
By now, you would likely have seen the specifications sheet on this device if you've been researching.

It came relatively well packed in the box. I paid the current market price ~$850USD. Standard styrofoam protectors to withstand bumps and thick plastic bag around the unit itself. Inside the box is just a standard decent IEC power connector, an instruction pamphlet I didn't even look at and a really unimpressive thin zip-cord RCA cable :-).

The unit itself IMO looks great as do the line of "Reference Series" gear - utilitarian in terms of knob and display placement with a hint of the TEAC heritage with "pro" gear given the side metal handle bars - looks like rack-mount gear. Remember that TEAC [Tokyo Electro Acoustic Company] Audio is in the same family as Esoteric (consumer audiophile) and TASCAM (pro audio); depending on how you look at it, I guess it's either an upscale TASCAM without all the plastic or 'baby' Esoteric without as much of the mass and audiophile aesthetics.

The weight is quoted as 9lbs and it certainly feels substantial. It's about the size of an A4 (letter) sheet of paper (front "handles" poke a bit forward) and 3 inches high. The construction is metal all around with a nice brushed metal texture so there's no shiny bits - nice. Knobs feel very stable and responsive. The headphone knob on the right rotates smoothly and the MENU button feels authoritative when pressed (unlike the front buttons for the ASUS Essence One - just one of those subjective look-and-feel things which adds to a positive impression).

The organic electroluminescence (essentially OLED) display is easy to read, has 3 brightness settings and an "OFF" setting. I like the amber color which is non-distracting and I made sure I set the default to the dimmest setting. Great also that the amber LED for input selection isn't too bright and certainly less distracting than the Essence One's blue LEDs (not a big deal for me but I know many folks get bothered by this).

Other than to get more detailed descriptions of the menu options, the manual is quite unnecessary - it's really easy to operate... Basically push the MENU button to toggle between options, turn the left knob to change selections, that's really it. In looking over the menu selections, one cannot help but think that the TEAC engineers basically took the TI/Burr Brown PCM1795 DAC chip, looked at the datasheet - considered the undocumented modes, and created a machine that took advantage of everything this DAC chip can do! Here are the main options:

1. PCM Upconversion to 24/192 - presumably could help reduce jitter.

2. PCM1795 digital filters: SHARP, SLOW, and OFF - hadn't seen the OFF option before; an interesting mode which I believe was intended to allow the DAC chip to be mated to an external filter.

3. DSD Analogue FIR filters: FIR1 to FIR4 - I'll discuss more about this when I present the DSD measurements.

4. Analogue output: either RCA, XLR pin 2 hot, or pin 3 hot. Cannot output both RCA + XLR.

5. Simultaneous headphones + analogue line out: ON or OFF.

6. USB input power - powers off the USB port if another input being used - not sure the reason for this, actually, just power saving I guess?

7. Setting mode display: ON / OFF for the display to show if upconversion is happening, PCM / DSD, sampling rate... Very cool. I leave this ON.

8. LCD dimmer - 3 levels & OFF.

If you look at the PCM1795 datasheet, you see that it's documented to be a 32/192 part and can do DSD64 (2.8MHz) conversion. Perhaps a little known fact is that this DAC chip is capable of 32/384 PCM and DSD128 (5.6MHz) as "undocumented" features which the TEAC designers obviously capitalized on. Note that the ASUS Essence One also uses the PCM1795 and "symmetrically upsamples" to 24/352 or 24/384 depending on whether the input sampling rate is a multiple of 44kHz or 48kHz.

So far, the Windows driver 1.02 seems quite stable. No problems with ASIO PCM using foobar2000, and DoP bit-stream support through JRiver 18.0 works well for DSD. The current TEAC HR Player 1.0.0.4 (small basic music player, "portable" so no install) works to play back DSD and can stream using either DoP or "native" ASIO 2.1. If you have DST lossless compressed DSD audio, the TEAC player doesn't seem to handle these but they're fine with JRiver.

On the Mac side (MacBook Pro with Mountain Lion), it uses the standard USB Audio 2.0 driver so nothing to install. I have used both Decibel for PCM playback and the "alpha" JRiver 18.0 for Mac works essentially the same as the Windows version for DoP support.

Subjective Sound Quality:
So far most of my testing has been with the Sennheiser HD800 pictured above. I'm just going to put on my "subjective reviewer" hat for a moment...

The headphone amp sounds good. It's not powerful - rated at 100mW into 32 ohms but it drives the HD800 loud enough including some relatively soft classical test tracks I had. The amp could easily drive these headphones to ear-splitting levels with the usual commercial rock/pop/jazz/country tracks. The AKG Q701's are a bit more difficult to drive so I would avoid using these with softer classical selections with the TEAC. It's quite clear that the ASUS Essence One has a significantly more power headphone section in comparison. (Of course if you're a big time head-fi fan, TEAC would want to interest you in the HA-501 headphone amp.)

So far, I have no complaints of the sound. PCM performance is excellent. For example, a test track I often use to weed out poor systems is Tyler Bates' "To Victory" from the 300 (2007) soundtrack. It's recorded "hot" and dynamically compressed and the cacophony of sounds tends to get muddled very easily on a poor system. This track was reproduced excellently with this DAC (I also find the emotional response - that sense of dread - conveyed by this track a good personal gauge).

On the Kodo track "Niji No Nagori" off the Tsutsumi (2000) album, there's a nice build up of multi-layered drums, flute, vocals, culminating in a woman singing with clapping, percussion, and male backgrounds around 5:00. The drums sounded dynamic and "full". Bass went deep with the HD800; and thanks to the "speed" of these HD800's, it sounded precise. Again, excellent performance and I would certainly rate this DAC+headphone amp highly.

Currently, I don't have much DSD music collected yet but have ripped a number of my SACD's which I know are either DSD sourced or high-resolution analogue in origin - no PCM or worse Red Book-sourced DSD for me like in this review, thanks.

Albums heard or tracks sampled: old analogue sourced Nat King Cole's The Very Thought Of You (Analogue Productions 2010), Pink Floyd's The Dark Side Of The Moon (2003 remaster), Michael Jackson' Thriller (1999 remaster), Al Di Meola et al. Friday Night In San Francisco (1997 remaster), Miles Davis' Kind Of Blue (2007 Japanese SACD). They sound good overall...  Limitations of the analogue source quite evident with obvious limited noise floor on most of these. The 80's sound of Thriller is pretty dated but I think the SACD version is the best sounding 'pressing' I've come across...

Modern DSD sourced SACD's: Erich Kunzel's Tchaikovsky 1812 (Telarc 2001), John Hiatt's Master of Disaster (2005), Jorma Kaukonen's Blue Country Heart (2002), Rachel Podger's Bach Violin Concertos (2010), Stuttgarter Kammerorchester' Die Rohre (Tacet 2003). Nice, clean, great sense of space especially the Stuttgarter and Rachel Podger SACD's.

There's very little DSD128 content out there as far as I am aware...  However, I downloaded a few of the samples from 2L. They sound excellent but since they're sourced from DXD (24/352), I could also download those massive files (1GB for 10 minutes!) and play them PCM direct and be even "closer" to the performance :-). Seriously folks, I think this would be a real waste of disk space!

You may be asking - is there anything "special" about the sound of DSD - especially after I penned this piece on DSD? Well, honestly, it's hard to say... Really hard to do any kind of direct comparison since the foobar200 ABX tool doesn't work for this, and the switch from PCM to DSD results in a soft 'click' sound as well as a brief delay...  Furthermore, volume levels aren't exactly matched. All I can say is the music just sound good whichever format :-). I don't think DSD is "needed" for good sound, but it's nice to be able to play back the music in whatever the original format was without transcoding.

I'll be back this weekend with some PCM measurement results...

Links to the objective evaluations:

PCM Evaluation (Part 2)

DSD Performance (Part 3)

MEASUREMENTS: Power Cables for Low Power Audio.

We are told - "everything makes a difference!"

Expensive power cables are an example of taking this principle more than likely to the extreme - well into the territory of the neurotic obsessive-compulsive. Some audiophiles claim there are very significant differences to be found by replacing standard cables like the common IEC connector varieties between the mains and one's gear. DIY plans are available on the Internet, and of course many enterprising companies have produced all kinds of cables to satiate those "believers". Like other cable claims, it's difficult to determine what scientific / engineering theory could account for these beliefs. While there could be some justification to use of heavy duty power cables for high-powered amps with dedicated circuits for example (very rare for home audio), why would someone need fancy cables for devices like DAC's or CD players where internally the AC is converted to low voltage and current DC to power the electronics? Furthermore, we all know that the electricity supplying our gear is connected by hundreds of miles of plain old non-"audiophile approved" copper cables of various diameter and quality.

In order to look for tiny differences, I'm going to try using various power cords with the ASUS Essence One DAC (note that my DAC is slightly modded with all LM4562 op-amps)... Let's see if there are any differences looking at the analogue output and changes to the J-Test jitter spectrum.

First, as usual, I had a look into my closet of cables to see what I have. Here are today's selection:

Cable A:
No nonsense generic freebie 6' cable that came with my old Antec computer power supply. Has the brand name "LINETEK" stamped on the connector.

Cable B:
Notice the green dot on the plug. That means this is a higher quality "hospital grade" cable. Also 6', but it's about 25% thicker, and twice the weight of Cable A. Strain relief is fantastic. The metal wall plug prongs are more substantial and the ground prong is solid metal instead of hollow like for Cable A. Presumably the thicker diameter indicates better shielding. I know this particular brand of cable is being used in the local hospital's ICU department. If this cable fails during use, patients could die...

Cable C:
I looked around to see what was the absolute WORST power cable I could come up with. Here it is - total 56' long. Using Cable A, I connected it to a 50' yellow outdoors cable I used over the Christmas holidays for the outdoor lights. In fact, this cable has been used for this purpose for the last 5 Christmases at least, so it's been exposed to the dirt, rain and snow. The metal prongs in fact look worn and oxidized. In fact, this is so nasty that I took a picture of it out on the deck since my wife refused to have it indoors for more than a few minutes for testing :-). I tested it connected to the DAC pretty much looking like this tangled mess. Unless you think the last 6' of generic power cable can make a difference, the "performance" of this cable should unequivocally "sound"/perform terribly.

Gear Setup:
I used a variant of the usual testing setup:
Win 8 laptop --> shielded USB --> CM6631A asynchronous USB to SPDIF --> Acoustic Research 6' TosLink --> ASUS Essence One (*connected to wall outlet by test cable*) --> 6' XLR cables --> E-MU 0404USB --> shielded USB --> Win 8 laptop

Note that I decided to use the CM6631A device for USB input and TosLink out (previously tested) instead of the native Essence One USB because I actually found less jitter this way. I noticed that the Essence One's USB input has a fair amount of low level jitter artifacts - not sure if it's a result of the CM6631 (non-A) chipset or the drivers in this configuration.

Analogue Measurements (RightMark Audio Analyzer 6.2.5, 24/96):

Summary:

Pretty much identical...  Very small differences within the error range for each "run" of the test.

Frequency Response:

Noise floor:

THD:

Stereo Crosstalk:

As you can see, there's nothing here to differentiate the analogue measurements from the DAC using the different power cables.

Jitter Analysis (Dunn J-Test - 16-bit and 24-bit variants):

Cable A - 6' generic:

Cable B - 6' Hospital Grade:

Cable C - 56' - 50' outdoors corroded prongs + 6' Cable A:

Again - no real difference folks. Not really that one expects any difference since it's unlikely that the DAC's internal timing circuitry could be affected by the AC input. Note that with the Essence One, we can actually see the 24-big jitter modulation pattern due to the very low noise floor below -140dB.

Conclusion:

As usual, I listened to the audio output using the poorest cable configuration after I ran these tests and as I ponder what to write for the blog entry. Indeed, the sound was fine.

Tonight, I was listening to the Erich Kunzel & Cincinnati Pops' rendition of Tchaikovsky's 1812 Overture (Telarc 2001, SACD digitally ripped & converted to 24/88) with the Essence One powered by the nasty 56' length to the wall socket. It sounded good. By ~14 minutes into the track, we hear a multi-textured climax with church bells, choirs, brass, percussion and of course cannons. The complex mix was reproduced very well and rendered nicely with my AKG Q701 headphones - plenty of dynamics being pumped out into the Essence One's headphone amp.

Sure, it's possible that "everything makes a difference!" As in most things in life, the wise man needs to ponder the claims a little further to divine the truth. At least when it comes to power cables, I think the wise man can comfortably walk away from such claims of audible differences and realize that a decent IEC cable is all that's needed - at least for low power devices like a DAC.

As is my usual policy, I do not bother measuring high-priced cables - partly because I don't have any at home - but these posts are not about pointing fingers at specific companies. Rather I hope the measurements and comments stimulate thought. Note that I have "heard" expensive power cables over the years so am well aware of their "performance". As usual, drop me a note if you have good evidence to show otherwise...