ANALYSIS: DSD Conversion Impulse & Spectral Display... Another few images for completeness!

A cool example of the "Spectral Frequency" display - from Adobe.

Okay guys, I actually wrote the text that follows 2 months ago before the Digital Filters Test. As happens sometimes, a post can get lost in the "draft" bin and later found. It refers back to the DSD-to-PCM analysis series from April:
ANALYSIS: DSD-to-PCM Conversion 2015 - Windows & Mac OS X
ANALYSIS: DSD-to-PCM 2015 - foobar SACD Plug-In, AuI ConverteR, noise & impulse response...

Consider this as part 3 of the 'trilogy' for this week as I continue to work on the results of the recently-closed Digital Filters Test.

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I was reminded recently by Wombat in his post here that we can have a look at the spectral frequency display as well when assessing sonic data in the audio editor. Also, Mnyb talked about the low-pass filtering in SACD players and what "standards" were used. Well, I don't know about the formal standards, but the first Sony SCD-1 SACD player back in 1999 had a defeatable analogue filter placed at 50kHz and according to the DSD Wiki, all SACD players were supposed to include this "optional" filter. I know a few people on audiophile forums suggest they subjectively preferred the filter be turned off.

ANALYSIS: DSD-to-PCM 2015 - foobar SACD Plug-In, AuI ConverteR, noise & impulse response...

Noise characteristics of PCM vs. DSD - image found here.

In my post last week looking at the various DSD-to-PCM converters, Solderdude Frans made a good suggestion... Let's have a look at the newer SACD plugin which has superseded the DSDIFF plug-in as the converter of choice these days for foobar. Also, it was suggested by Yuri Korzunov, the author of AuI ConverteR 48x44 to have a look at his converter package as well.

ANALYSIS: A Comparison of DSD Encoders & Decoders (KORG AudioGate, JRiver MC, Weiss Saracon)

Hello guys & gals, I've seen the question asked of comparing various DSD conversion programs on message boards over the years but have never seen someone try to "compare and contrast" with objective analysis. Let's at least give it a try here. I don't promise unequivocal answers, but hopefully a decent stab at it :-).

Remember that any conversion between DSD to PCM is a "lossy" process. Therefore, it is of course preferable to keep PCM sourced recordings in PCM and DSD likewise if possible. There will be some compromise in the accuracy each time conversion happens. Even though the bitrates for DSD64 and 24/96 PCM may be similar, the modulation technique used to represent the resultant sound wave is different (as per the above image). The question of course is how much difference and if it's quantifiable.

This question of conversion is important because as I have discussed before, many if not most DSD releases have gone through some kind of conversion for the flexibility and ease of editing in the PCM domain. The most blatant examples of conversion are the ones sourced from 44/48kHz material, but I'm sure many others are from 96/192kHz origin but they would not be easy to differentiate from a DSD original.

I. Procedure:

I do not have access to DSD recording gear but I can convert recorded PCM to DSD and back again to see what the conversion process does. For example, the PCM test signal from RightMark Audio Analyzer can be sent through the conversion process and we can see what happens to it to get an idea of the amount of degradation. For these tests, I chose to use the 24/96 test signal which I feel is a very reasonable hi-resolution specification exceeding DSD64 in a number of resolution domains. I know that 88kHz may be better as an integer multiple of the 2.8MHz sample rate but I figure these days in a high resolution studio, 24/96 is probably the standard and is common as high-resolution HDTracks and Blu-Ray audio releases.

Here then is the general procedure:
- Take the 24/96 RightMark test signal.
- Convert the PCM to DSD using the various encoders.
- Reconvert the DSD file back into 24/96 PCM using each program.
- Analyse effects of the 2-way conversion and differences between the programs.

I decided to use 3 commonly available conversion programs for this test - something free, something a consumer can afford, and finally the professional "standard" made available to me thanks to a friend who runs a studio. This will result in a total of 9 final 24/96 WAV files to "measure" with the RightMark software (3 PCM-to-DSD encoding x 3 DSD-to-PCM conversion). The 3 software programs used for conversion are:

1. KORG AudioGate 2.3.3. This software is available free. All it takes to run conversions is access to your Twitter account so the software tweets each time a conversion takes place. Small price to pay for the ability to do the conversion I suppose. I used the default DSD encoding (DSDIFF / Stereo Interleaved / 2.8MHz / 1-bit) and decoding to PCM (WAV / Stereo Interleaved / 96kHz / 24-bit) parameters. I noticed that AudioGate will apply a +6dB gain with the DSD to PCM conversion (-6dB DSD is equivalent to 0dBFS PCM, not uncommonly this standard is not followed and +6dB gain can result in clipping).

2. JRiver Media Center 19.0.117. I've used this program before to test the PCM-to-DSD conversion playback last year. You can also save the resultant PCM --> DSD and the converse DSD --> PCM conversion files as well. The DSD --> PCM conversion happens in 24/352.8 so I used the best resampler I have - iZotope RX 3 - to convert back to 24/96 for final analysis using a steep filter at 48kHz. There is also no +6dB gain applied so the default volume of the PCM output file is softer than with AudioGate and Saracon at default settings.

3. Weiss Saracon 01.61-27. The standard DSD <--> PCM conversion package used by a number of places like Channel Classics, many HDTracks releases, Pentatone... Again, I just used the default settings for conversion to DSD (dff, CRFB 8th Order, 0 gain, 2.8224MHz, Auto channel mode, Smart Interleave, Enable Stabilizer). Likewise the conversion back to PCM was with default settings (WAV, 24-bit fixed point, TPDF dither, 96.0kHz, +6 dB gain, Smart Interleave).

II. Result:

As usual, I'm going to present the data as summary charts to start. There are 3 DSD encoders and the same 3 can be used to decode DSD, so let's just present them organized by the encoder used. When I say something like "AudioGate then JRiver", I'm referring to the use of AudioGate as the DSD encoder, then using JRiver to do the conversion back to high-resolution PCM to be analyzed by RightMark (remember, for JRiver's case, I also used iZotope RX 3 to resample from 24/352 --> 24/96).

AudioGate as DSD encoder.

JRiver as DSD encoder.

Weiss Saracon as DSD encoder.

As you can see, the first column in each table is the 24/96 RightMark PCM test signal with no conversion done. These would be the ideal numbers if one could measure a perfect DAC/ADC setup or in this case the results of perfect conversion.

The rest of the columns reflect what happens to the 24/96 PCM test signal as it goes through the DSD conversion and decoding steps. Remember that RightMark is analyzing the audible 20Hz to 20kHz spectrum only. As you know, DSD64 conversion adds quite a lot of ultrasonic noise if left unfiltered and this would result in some poor noise levels and lower dynamic range if frequencies >20kHz were analysed.

Indeed, various amounts of distortion and imperfections can be seen. On the whole, it's far from bad though. At worst, the cumulative noise level is still down below -120dB and dynamic range >120dB with each of these encoder/decoder pairs.

Comparatively, you can see the free KORG AudioGate encoder table above seemed to have the worst results in terms of noise level irrespective of what other software was used to convert back to PCM. This is followed by JRiver and then Saracon puts out some very fine numbers.

There's a similar tendency when comparing the DSD-to-PCM decoder used. In general, the JRiver and Saracon DSD-to-PCM conversions (columns 3 & 4) resulted in better measurements of noise level, and dynamic range than AudioGate (column 2).

Let's now have a look at some individual graphs to see what's going on - here's using AudioGate to encode PCM-to-DSD:

Frequency Response

Notice the different software used to convert DSD back to PCM all have different low pass filters. As expected, PCM (white) is flat all the way to 48kHz. AudioGate (green) uses a very weak filter and is only attenuated by <1dB at 48kHz, followed by Saracon. JRiver at the default "Safe" setting has a steep 24kHz 48dB/octave slope applied as noted here (you can change this if you want up to 30kHz cutoff, 50kHz cutoff, or filter turned OFF).

Noise Level

There's deviance from the PCM noise floor using AudioGate DSD conversion as you can see. AudioGate is more noisy at converting PCM to DSD than the other programs (as will be evidenced later). The noise floor also isn't as smooth as the others (interesting notch at 10kHz and 20kHz).

In comparison, let's have a look at the JRiver PCM-to-DSD encoding:

Frequency Response

Noise Level

The frequency response curves are similar to the AudioGate DSD encoding representing the respective low-pass filter settings of the DSD-to-PCM converters. The main difference is with the noise level. As you can see, JRiver as DSD encoder is able to maintain a very clean noise floor essentially equivalent to 24-bit PCM until about 13kHz before rising - and this low noise floor is maintained by JRiver and Saracon when reconverting back to PCM. The AudioGate DSD-to-PCM conversion in comparison has a higher noise floor throughout the audible spectrum - perhaps a higher level of dithering is being applied?

Finally, let's look at Saracon used as the PCM-to-DSD encoder:

Frequency Response

Noise Level

A clean PCM-like noise floor all the way to 20kHz is achievable after going through Saracon DSD encoding but this quickly increases thereafter. Again, AudioGate conversion to PCM results in a higher noise floor which I speculate is due to stronger dithering.

III. Conclusion:

Since DSD <--> PCM isn't a straightforward process (like say resampling in PCM), as expected, at a "microscopic level", conversion software does make a difference in resolution.

What is much harder to quantify is audibility. Those frequency response, noise floor, distortion, crosstalk results are all below what I believe are human thresholds of audibility and overall there is minimal change to the 24/96 PCM original signal within the audible frequency range. Remember, the results I show here are with both conversion to DSD and back again to PCM, not just a single conversion step. Yet, I have seen commenters on-line insisting that the conversion results in audible deterioration in sound (even with just a single step like DSD --> PCM).

Looking at these 3 software programs, we can say with some certainty from an objective perspective that Saracon PCM-to-DSD transcoding maintains the lowest noise floor from 20Hz to 20kHz. JRiver is also very good in this respect, while AudioGate's results are less accurate but obviously still very good and of questionable audible significance given that the difference is still below the measured noise floor of all except maybe the very best DACs.

Of course, one has to pay big bucks for Saracon compared to the free AudioGate software!

As for DSD-to-PCM conversion, the main difference appears to be where each program has decided to put the low-pass filter to remove DSD's ultrasonic noise. Of the 3, JRiver has the most conservative low-pass filter at 24kHz (with small notable effect beginning around 20kHz) by default. Saracon allows a bit more to pass through up to around 30kHz, and AudioGate allows essentially everything to pass through up to 48kHz with 24/96 sampling. The only other difference seems to be a stronger dithering algorithm (I'm guessing here) with AudioGate such that the noise floor is marginally higher than the others. Again, we're looking at differences way way down in the noise floor so it really should not be an issue. I think the real question is where you think the low-pass filter should be set for DSD64 material (ie. at what point is recorded ultrasonic signal drowned out by noise and not worth keeping?)

From what I see here, I'm quite happy that Saracon is used in most commercial releases I've come across for DSD-to-PCM conversion. Within the 20Hz to 20kHz audible spectrum, it does appear to be the best even though I highly doubt one could go wrong with any of these. Just remember that the steep low-pass filter in Saracon means there's nothing above ~40kHz and therefore no point buying a Saracon DSD converted file above 96kHz (88kHz is all that's needed).

Over the years, I've listen to original DSD and compared to PCM conversions at 24/88 using Saracon and AudioGate output level matched as best I could (using the TEAC UD-501, never tried formal ABX or blinding). IMO, it's tough to assess since you can't instantaneously switch from DSD to PCM. The PCM converted files sound good to me and I would not hesitate to archive the DSD64 library as 24/88. Whatever difference has always been subtle at best (despite claims from the DSD faithful that somehow DSD sounds much better). I suppose it's possible that different DAC devices could also sound different depending on PCM or DSD input.

Has anyone out there done an ABX or other controlled listening test with DSD-to-PCM conversion? Would love to hear of your experience and preference... 

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Rant of the week...
In the high-fidelity audio world we've often discussed the ills of severe dynamic range compression (DRC). I'm just going to go on my soapbox for a couple minutes and complain also about the ills of DRC for soundtracks these days... Notice how LOUD TV shows have become lately? A couple years ago, I tried watching NBC's Hannibal. Not only was the pacing terrible, meant for folks with ADHD, but the audio was so annoyingly grating that I could not tolerate more than 3 episodes. (I don't know if the series improved after those 3 episodes...)

More recently, I've become annoyed by the recent Cosmos: A Spacetime Odyssey hosted by Neil DeGrasse Tyson playing on Fox and National Geographic Channel. I mean... COME ON PEOPLE! This is a science program. This is a documentary (with some science fiction entertainment thrown in). WHY DOES IT HAVE TO BE SO LOUD? It's like there's no subtlety left... No opportunity to whisper... No opportunity to wonder... No opportunity to enjoy the eye-candy of some excellent CGI graphics without the blaring of some "majestic" soundtrack through many parts of the show. Aren't the ideas being presented supposed to be what it's all about? But yet at times, the narration gets muddled by the background audio.

While I can still enjoy Cosmos 2014 with my kids for the topical presentation, I'm left wondering how much better it could have been to allow the dialogue to take center stage and the background soundtrack to accentuate the emotional impact instead of being ridiculously front-and-center as if I'm supposed to watch this program on a tiny smartphone screen on the subway (maybe that's the target audience!). As usual, it's hard to know who to blame - is it the sound engineers working on this series behind the mixing console or the folks manning the TV station transmitting the signal running it through their compressor? Unfortunate.

I'll end with a quote from Carl Sagan. Certainly worth contemplating when reading comments posted on the Internet in general... (Not just as audiophiles.)

"We live in a society exquisitely dependent on science and technology, in which hardly anyone knows anything about science and technology." Carl Sagan (1989) [good article BTW]

I wonder what Mr. Sagan would think about the current state of affairs regarding the level of understanding of science in our society today. I suspect if he were still alive (he died in 1996), he'd be impressed by the access to information and interconnectedness we have these days through the Internet. That's not necessarily saying a lot though about the level of understanding.

Still a great read after all these years... Originally published 1980.

MEASUREMENTS: PCM to DSD Upsampling Effects (JRiver MC19 Beta).

We're continuing to see a push into the DSD domain with renewed talk of music release as digital downloads requiring the purchase of a DSD DAC to natively play (eg. recent Acoustic Sounds DSD releases). For now, I have already voiced some concerns about DSD including practical issues like the gross limitations of the file format itself. I demonstrated the noise characteristics for both DSD64 and DSD128 in my TEAC measurements. Furthermore, I have shown that there already exists many SACD's appearing essentially to be upsampled PCM from standard sampling rates of 44/48 kHz (remember, it's almost a given that any music we listen to created in the last 20 years has at some point been through a PCM stage except specified pure analogue recordings or those specifically recorded in DSD with minimal processing).

Some writers have voiced that even the process of upsampling PCM to DSD will imbue the music with some of DSD's beneficial properties, is this true? If so, what happens?

Well, thanks to ongoing advancement in the computer audio world, we can now easily have a way to listen to our PCM music as a converted DSD stream... Enter JRiver Media Center 19 and it's ability to stream PCM --> DSD64/128 in realtime to a compatible DAC. [Note that this should also be possible using ASIOProxy from foobar - not tried personally.] This will allow an easy way for everyone to listen for themselves what happens to the sound either as the original PCM or transcoded to DSD with the assurance that we're comparing "apples-to-apples" with the same mastering.

First, as has been my custom, let's start with some objective measurements to see what the DSD encoding does to test signals.

I. Objective Measurements

General Setup:
AMD A10-5800K HTPC Win8 x64 running JRiver 19.0.37 (ASIOProxy workaround for DSD128) -> shielded USB -> TEAC UD-501 -> shielded 6' RCA -> E-MU 0404USB -> shielded USB -> Win8 laptop

TEAC DAC settings:
     - PCM: "SHARP" BB PCM1795 filter, no upsampling (default)
     - DSD: FIR3 analogue filter (closest volume match to PCM output)

JRiver Media Center 19.0.37 setting:
     - ASIO buffer set to "minimum hardware size" since someone suggested it sounded better :-) - no stuttering encountered playing music.

First, let's have a look at the 1kHz SQUARE WAVE off the digital oscilloscope (24/44 source):

PCM:

DSD64 realtime conversion:

DSD128 realtime conversion:

As you can see, volume is about the same with the DSD FIR3 filter on the TEAC vs. PCM SHARP; both output ~2.85V peak with the square wave. What is also very obvious is how clean the PCM is vs. DSD. Notice the extra high frequency noise for both the DSD64 and DSD128 traces, with the DSD128 clearly less noisy. No surprise, right? If you've looked at the objective results from DSD here and elsewhere, this is pretty "normal" for DSD.

IMPULSE RESPONSE (16/44 PCM impulse):
PCM SHARP filter:

DSD64 realtime upsampling:

DSD128 realtime upsampling:

Noise is again very evident in this "zoomed in" impulse response measured at 24/192 especially from the DSD64 process. Although impulse response graphs can be excellent with MHz sampling rate (this is often a "talking point" in the DSD/Sony ad literature over the years), when resampling PCM to DSD, we're still hampered by the PCM signal's original sampling rate (eg. 44kHz). Evidently JRiver uses a typical linear phase reconstruction filter; hence the symmetrical pre- and post-ringing.

DUNN J-TEST:
PCM:

16-bit

24-bit

DSD64 realtime upsampling:

16-bit

24-bit

DSD128 realtime upsampling:

16-bit

24-bit

No meaningful differences between PCM and the DSD realtime conversion. This also means no evidence of worsened jitter with all that extra processing converting PCM to DSD in the computer at least based on the spectral output of this test (typically, this computer's AMD CPU utilization went from <5% with PCM to ~15% for DSD upsampling). DSD64 is obviously of enough resolution to accurately demonstrate the jitter modulation signal in the LSB for the 16-bit test.

RightMark:

Calculations done in the AUDIBLE SPECTRUM (20-20kHz).

Frequency Response

Noise

THD

These graphs of upsampled 24/192 test tones echo the results in the TEAC DSD Measurements back in May. I used KORG Audiogate to convert PCM to DSD back in May and it looks like the mathematical process in both JRiver and Audiogate are of similar precision. Within the audible spectrum, the PCM and DSD measurements are all very similar. A good indication of high precision in the conversion process. What is again evident is the noise once we get above 20kHz especially with DSD64.

II. Subjective Experience

As per the premise of this blog being "more objective", I'm not going to write pages on that which is experienced for oneself. However, I'll put down a few thoughts for consideration...

Listening gear:
     - Headphones: Sennheiser HD800 off TEAC DAC
     - Speaker system: TEAC UD-501 --> Simaudio Moon i3.3 --> Paradigm Signature S8 v3 (standard OFC cables)

The DSD conversion process through this TEAC DAC does change the electrical output as seen by the objective measurements above. This alone means that it's real compared to the identical measurements found with different bitperfect software and digital cables previously reported.

There does seem to be a change in the perceived detail of the sound subjectively through the gear I listed... Note that I'm taking the liberty here to not subject myself to a blind test so I fully admit that I could be wrong on this :-). Furthermore the fact that since it's not an instantaneous 'flip', echoic memory is prone to be unreliable. With these caveats, my current feeling is that both DSD64 and DSD128 conversion adds a potentially euphonic characteristic to the sound. No, IMO, it's not a dramatic difference when listening volume is controlled. [For those using the TEAC DAC, remember that the default FIR2 filter for DSD is louder than PCM by ~2.5dB - this could of course be misconstrued as sounding "better" for DSD.]

What do I hear? As I mentioned in my previous post on getting DSD128 upsampling working on the TEAC, I think the sound is less "etched". There's a pleasant subtle added smoothness to the transients. I think many may describe this as being less fatiguing, maybe less of the "digital glare". I couldn't specifically put a preference on DSD64 vs. DSD128 but knowing the ultrasonic effects, it wouldn't take much to convince me that DSD128 is better since the ultrasonic noise is further away from the audible spectrum. However, if you believe that the noise itself creates euphonia, it's also conceivable that DSD128 would sound closer to PCM than DSD64. Maybe.

I listened to a few standard 16/44 albums in DSD128 like a first pressing Michael Jackson Thriller, the well recorded Al Di Meola Winter Nights, and Suzanne Vega's Solitude Standing. They all sounded great. Like I said, marginally smoother than PCM. I think poorly recorded harsh albums may benefit even more - for example Alan Silvestri's The Avengers score is mastered in "modern" overcompressed fashion with DR9 average dynamic range (not good for an instrumental soundtrack IMO). DSD128 upsampling seemed to make it more listenable for longer duration.

Vinyl rips (24/96) of Tracy Chapman's Fast Car and Whitney Houston's One Moment In Time sounded very nice as well... "Extra" analogue from digital from vinyl :-). Again, the inability to instantaneously switch between PCM-to-DSD makes it hard to A-B compare reliably.

Unfortunately I did not take a screenshot of the phase measurements, but it looked good. Listening to phase-effect tracks such as those encoded in Q-Sound like Def Leppard's Rock On (David Essex remake off Yeah!), and Roger Waters' Too Much Rope (off Amused To Death) nicely created the impression of spatial surround and depth. Whether that sense of depth is any better with the DSD upsampling is of course debatable.

III. Conclusion

1. PCM to DSD upconversion is a DSP process. The signal output is measurably different.

2. Noise shaping pushes the DSD quantization noise into the ultrasonic frequencies as expected. In DSD64 it rises above the noise floor almost right at 20kHz, and in DSD128 it starts around 40kHz. (I vote for pushing it up to 40kHz as less likely to cause distortion through the amp & speakers.)

3. Pre- and post-ringing is similar to standard PCM with upsampling using MC19's algorithm so this would not explain any audible differences.

4. The algorithm used in JRiver MC19 does a good job with maintaining classic measurement parameters like frequency response, dynamic range, and distortion from 20-20kHz  - basically this means the math is as expected and fits the DSD output profile. Results are similar to the KORG AudioGate software converting PCM-to-DSD.

I can't help but wonder if what's happening here is like tube amps and analogue playback (eg. vinyl). Objectively the DSD conversion adds distortion but the anomalies are not perceived as objectionable and in some material, the added noise and imprecision actually makes it sound less "sterile", "clinical", more "real" (conversely being in an anechoic chamber is disturbingly unreal due to the profound silence). It would make sense to me that some people could prefer DSD64 over DSD128 upconversion since DSD64 will give you more of that distortion. Even though the noise is ultrasonic in nature as measured off the DAC, nonlinearities in the playback system like your headphones and speakers (perhaps certain amps as well) could create audible intermodulation. Maybe for certain music, this could be especially beneficial.

Out of curiosity... For anyone out there with the EMM Labs DAC2X which upsamples to DSD128, it'd be great to have a look at what the measurements are from that unit! With all the positive press about how this DAC sounds (ahem... $15.5K), I have yet to see any measurements... I wonder what a 16/44 impulse response looks like for example to see if it bears similarity to what JRiver is doing. How about the ultrasonic noise with DSD64 & analogue filter strength? Does the EMM upsampling process for PCM result in similar frequency response pattern?

In any case, give this PCM-to-DSD process a try at some point when you can. If nothing else, at least to say you've experienced it... See if you can perceive a difference and/or judge if it's beneficial for yourself.

Tonight's music: Valery Gergiev & LSO's Mahler Symphony No. 9 (2011). Nice recent classical recording available in SACD format as well.

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You might notice that I turned on AdSense on the blog.

As I have said in the past, my intention for this blog has never been about making money. I have no formal relationship with any company so have no sales incentive and am not interested in making this some kind "publication" other than what it has been - a blog about my own journey in audiophilia with a bent towards finding answers using empirical/objective means. That remains my main interest.

Nonetheless, it's trivial to "flip" the AdSense switch. I would have no idea what Google tries to market to you, and trust that the layout won't be distracting (I've switched off some questionable types of ads like for dating sites or of a sexual nature). If it gets me a few bucks for my digital downloads for what I do as a hobby anyway, I'll be happy with that!

Best regards...

HOWTO: Getting JRiver MC19 PCM to 2xDSD (DSD128) upsampling working on TEAC UD-501...

A quick post here for those trying to get beta JRiver MC19's 2xDSD (DSD128) upsampling working on the TEAC UD-501. [Currently I'm using beta MC19.0.37.]

See the thread here where the discussion was started - thanks for putting attention on this InflatableMouse. Apparent there are some buffer issues with the TEAC driver and native ASIO, and at this point MC19 isn't supporting a 2xDSD upsampling with DoP option. Here's a workaround:

1. Download and install ASIOProxyInstall-0.6.5 from the SourceForge link here.
2. Go into MC19 and set audio device as "foo_dsd_asio [ASIO]", Bitstreaming as "Yes (DSD over PCM (DOP))". Should look like this:

3. Now lets set up ASIOProxy itself. Go into that "Device settings..." tab. Look at the "Tools" section and click "Open Driver Control Panel...". You'll see this pop up:

Make the settings as above especially with Fs as DSD128.

4. Finally, go into "DSD & output format..." in MC19 and set output format to "2xDSD in native format":

There you go.

Should now be listening to all PCM music upsampled to DSD128 on the TEAC. Native DSD files will be bitstreamed in their respective DSD64 or DSD128 forms direct to the DAC without MC19 processing like volume control.

Using an AMD A10-5800K processor, I'm seeing CPU use peaking at ~20% and usually 10-15%  even with upsampling 24/192 music. Not bad!

Hopefully TEAC and JRiver can come up with a solution for native ASIO streaming or support of DSD128 upsampling in DoP in the future...

BTW: I just got back from holidays a little jet lagged so haven't played with this much yet. However, on my system, upsampling to DSD64/128 certainly sounds different than PCM and I can certainly see the appeal - there seems to be more weight to the bass and the sound isn't as "etched".  There is a bit of level difference between DSD & PCM playback so I will need to listen more back and forth to see which setting I prefer.