Sunday, 3 May 2015

MEASUREMENTS: Corning USB 3 Optical Cable, Ground Loops, and Noise.

Folks, remember, the "Linear Phase vs. Minimum Phase Digital Filter Test" is still running as I post this! Please give it a try and report back in the survey. I want data and hopefully an opportunity to report back on what the readership perceives (or not perceive).


I. Introduction

Indeed, "noise" can be an issue with audio systems. There's little worse that can happen to an audiophile after spending hundreds or even thousands of dollars on new equipment, plugging everything in as per "best practice", getting ready to play your first piece of music, and finding that for some reason, noise has seeped through to disturb the expectation of pristine, clean background silence from which music can blossom...

Before proceeding further, remember that noise can sometimes be subtle, and very importantly, ambient noise must be low in your sound room. It's quite possible that slight hum and low-level noise may not be significant from a normal listening distance if the ambient noise level is high (something we urban dwellers especially do have to consider!).

It is not surprising that the more complex our audio system gets, the more potential there is to experience analogue noise as an issue. Sure, we can have digital anomalies ("digital noise" would be meaningless in this context) that cause errors in the transmission through poor cables as I have previously demonstrated, and I guess one could have so severe analogue noise as to cause digital errors (I suspect very rare and likely due to poor equipment, device failure, or extreme conditions). But those anomalies have characteristic "digital" traits as I previously discussed. What we normally think of as noise is the analogue stuff... Characteristic humming at a certain (often mains resonant 50/60Hz) frequency, constant random "white" noise like a poorly tuned radio, or episodic blips and bursts associated with intermittent electrical circuit activity (EMI from computer activity, hard drive access, intentional transmission from cell phone placed too close to equipment, etc.).

If we look at the "big picture" of how noise contaminates audio systems, we can see two major mechanisms:

1. Noise from without: RF/EMI radiation inducted noise because you put an inadequately shielded device or cable close to a source of radiation like a high power computer. Just take an old AM radio and put it close to one of these radiating devices to "hear" the noise. In this case, we have to either get better shielding (no need for fancy cables or expensive ones necessarily), reduce the chance of a cable from acting like an antenna in receiving the radiation (eg. ferrite beads), and/or move sensitive devices like your DAC or pre-amp away from the source of radiation. There are standards for allowable RF radiation from consumer devices so generally this isn't a huge problem except in confined spaces. The inverse square law; placing devices away from the radiating source is you friend. (My father used to run a 1kW Ham radio transmitter in the basement in my high school years with a large antenna in the backyard - now THAT was how I learned the importance of shielding!)

2. Noise from within: Noise conducted within the electrical circuit / cable originating from the equipment itself. High quality audio demands low noise floor so a good piece of audio equipment should not be a major source of noise itself (this is of course why everyone who does measurements always include evaluation of the noise floor). These days, the device that is most likely to add electrical noise to the system is of course our computer in the sound room (hopefully your computer isn't adding much acoustic noise with loud fans). In modern systems with a DAC, conduction through the USB cable is therefore commonly suspect. Instead of just transmitting the electrical "pulses" down a digital cable, the conductor will also carry with it the unintended electrical signal (noise). If it does not get filtered out, noise becomes amplified and transduced in our speakers/headphones where we can hear the unintended distortion. It is this form of noise which can be extremely difficult to deal with since shielding and fancy metallurgical properties will likely not make any difference irregardless of expense. In other words, noisy electrical currents do not care whether it's travelling down copper, silver, gold, or unobtanium once it's in the conduction path. For these situations, the ideal is galvanic isolation which "breaks" the electrical contact but allows only the intended signals to cross - this is more difficult and costly to solve. Although I'm focusing on USB cables here, there are of course many potential sources of conducted noise such as pollution through the power system (eg. washing machine, fridge, light dimmer switches, etc.) and any electrical interface can be affected in varying degrees (coaxial S/PDIF, HDMI, Thunderbolt, DVI, DisplayPort, SATA, ethernet, etc...)

II. The Situation...

Now I've got a little situation here which I think can provide a nice "case study" of what could happen, and how we can remedy the issues.

Here's a picture of my setup in the soundroom:

As you can see, I have a number of pieces to this system obviously interconnected with all the wires in the back. The Paradigm Signature S8 speakers are powered by the two Emotiva XPA-1L monoblock amps sitting on the floor with my Technics SL-1200M3D turntable in the center. Signals going to the monoblocks are from the center Emotiva XSP-1 preamp (turned off with the amber LED visible mid-center rack).

I have a center channel on the rack (Paradigm Signature C3) so this is a surround system (5.1) with the rear channels behind out of view and the subwoofer (Paradigm Signature SUB 1) on the left beside the left S8. Surround processing duties are done by the Onkyo TX-NR1009 receiver situated on the bottom rack (between the turntable and Emotiva XSP-1) connected to my computer through HDMI.

Stereo audio is quite straightforward, I have the TEAC UD-501 DAC (top-left rack) and Squeezebox Transporter (mid-right rack) sending signals to the XSP-1 preamp. The Transporter is just a streaming device so it gets data from the wired ethernet network (which as I have shown is generally immune from issues). The TEAC UD-501 however is connected by USB to the computer to the far left of the image (with the white Logitech keyboard on top beside a GIK Acoustics sound panel at the first reflection point).

I use generic (various Mogami and Monoprice) balanced XLR interconnects throughout the system for both the TEAC and Transporter, as well as the preamp to the monoblocks. The noise floor in stereo mode is measurably very low and I really could not wish for any better! The only times I bother with single-ended interconnects in this system is with the Technics turntable to the XSP-1 pre-amp phono input (which sounds fantastic) and when I connect the Onkyo to the Emotiva XSP-1 preamp when I listen to surround sound through the "Home Theater Bypass" input (which as you will see was problematic) .

As much as I love the Emotiva XSP-1 pre-amplifier, there was one issue I ran into from a sound quality perspective early on. It was exquisitely sensitive to noise in the "Home Theater Bypass" analogue inputs. This is significant only when I enter surround sound mode where the output from the Onkyo is sent to the pre-amp for the subwoofer and two front channels (unbalanced signal). The signal would be polluted by both a hum as well as computer-associated EMI noise. So... How does one remove this nasty noise?

III. Ground Loop Noise (Hum)

First, let's talk about audible low frequency hum. Most of the time, this is due to ground loops. Here's a nice summary from Associated Power Technologies of what happens. Basically, when a number of electrical components are interconnected, the shielding ties the ground potentials of the equipment together. There would not be an issue if the world were perfect and every single outlet potential was exactly the same. Unfortunately, it isn't perfect and variations in grounding point potentials will lead to small currents injecting noise into the system. Typically, this ground loop "hum" will either be 50/60Hz correlating with the mains AC frequency of your local standard or a resonant multiple (eg. 100/120, 150/180, 200/240Hz, etc.).

In my system, with the highly sensitive "Home Theater Bypass" analogue input, here's what it sounds like:


After making the video, I realized the little camera microphone is having a hard time picking up that low hum (maybe even filtering it out?) - trust me, although faint, it was audible in my room... In fact, I can record it and analyze the spectrum to see this:

There it is, the 60Hz ground loop with resonance up at 120Hz and 180Hz. Like I said, it's faint but I'm a perfectionist and I could hear it in late night listening sessions when the world is asleep :-).

Because the ground loop is caused by differing ground potentials, the easiest ways to deal with this problem is to connect all the devices to the same ground tap. This could be the same power outlet or in this case, I'm connecting them all to the same Belkin PureAV PF60 power conditioner I have been using in the system for many years. Thankfully, this fixed the issue for me as per this second video:


As you can see, I also show another "trick of the trade" if needed in the video... The "cheater plug". Basically, an inexpensive <$5.00 adapter plug that disconnects the ground pin hence removing potential for ground loops through that pin (and also I may add removing electrical noise from the computer/power supply through the ground connector). As I said in the video, be aware of potential safety issues. Also, it may or may not work depending on your computer power supply. Consider using a Ground Fault Circuit Interrupter (GFCI) like this one if you intend to do this longer term. As you can see, they're not too expensive (~$20). An even more permanent and esthetically pleasing method would be just to replace the wall socket with something like this. Isolation transformers would be another option but these tend to be more expensive. Another potential device would be the Hum-X - I've never used one but have heard good things.

With the ground loop inaudible, the spectrum now looks like this:

Beautiful humless noise floor :-)!

IV. USB Cable Conducting Electronic Circuit Noise

In a way, what I showed above is a bit artificial... The truth is, in order to just demonstrate the hum problem to show you guys that issue, you may have suspected that in the video above, I had already isolated out what is actually even more problematic - electrical noise from the computer circuitry itself, transmitted in the wiring causing distortion where there is susceptibility such as the analogue "Home Theater Bypass" input of the XSP-1. In the video below, I plugged in a typical shielded USB cable to my TEAC DAC directly to show you what the interference sounds like:


Nasty eh? As you can hear, the pre-amp analogue input is picking up all kinds of computer noise and this is getting amplified and sent to the speakers. You can hear JRiver starting up for example; likely a composite of noises from SSD data transfer and program execution. There's also a distracting high pitched whine which actually the camera picked up very well (it sounds louder in the video than hearing it live in the room). This is the infamous 8kHz "packet noise" originating from the high-speed USB "microframe" rate (1kHz for "full"/slow speed, see here under "Frames and Microframes"). And yes, it can be measured:

As you can see from the data to the left, where I put the cursor (8kHz), there is a -92dB peak measured by my E-MU 0404USB. We'll look at this again a little later.

What to do about this? USB noise isolation of course!

To be honest, I have no idea if specific USB audiophile products work. For example, here's a "review" of products from Schiit, iFi, and UltraFi, without any objective context (ie. measurements of a situation in which the device worked). From reading the article I have no idea what has improved sonically other than rather generic and subjective opinions, so how is the consumer supposed to know what to buy? Since I live in Canada and haven't seen any dealers with these units to borrow and test, I really do not want the hassle of ordering and returning if it doesn't actually work for me.

So, I figured, let's just do this logically myself. We basically need to isolate noise from two places: 1.) the serial data pins of the USB cable (D+ and D-), as well as 2.) minimizing noise from the power pins (+ & -).

For the data pins, the best way I can think of is to use an optical technique. This is similar to using a TosLink cable instead of coaxial for S/PDIF back in the "old days" for galvanic isolation. And recently, Corning has come out with this interesting USB 3 Optical Cable (~$100):

As you can see, it's quite long at 10m (33') and meant to be used as a USB extender. However, it is capable of USB 3 (5Gbps!) speeds and so can be repurposed in the future for other duties if I don't need it for my sound system. Also, since it's capable of very high speed, there should be no issue with transmitting DSD128, DoP, and 24/384 PCM to the hi-res USB DAC. Remember that this can be an issue with some USB isolation devices like the Firestone GreenKey which goes up to 24/96 only.

It's important to realize that the optical USB cable does not provide 5V USB power (and thus noise on the voltage pins will not be transmitted) and so it will not work directly with devices that need it to operate (for example USB stick DACs like the AudioQuest Dragonfly or AudioEngine D3). Because my TEAC UD-501 required the USB 5V to be present for the interface, the straightforward way to do this is therefore using a powered USB hub to provide the 5V. I decided to try a spare USB 3 hub I had along with some cables to connect to the DAC:

Nothing special, just one of these generic USB 3 hubs from a local store. I suspect that USB 3 hubs may be a bit better as electrical tolerances probably are superior to standard USB 2 devices in order to accommodate the higher transfer speed.

So... Did the combination of removing ground loop hum and using the optical USB cable (plus powered hub) work? Yes:


Here's how the FFT spectrum looks now (specifically focused at the 8kHz spike):

Compared to the spectrum above, although the technique did not completely remove the 8kHz "packet noise", it is now down to about -120dB, plus the overall noise floor is down another 5-10dB. Basically, there has been a >20dB suppression of the packet noise which has now rendered it inaudible at normal listening levels. Also, no more audible "processing noise" when I open or close programs like JRiver, or do other tasks on the computer.

Remember, the issues I'm showing in this post were always just when the XSP-1 preamp was in the analogue "Home Theater Bypass" surround mode which was susceptible to noise. Normally, I never have a hum or hear electrical noise from the TEAC DAC output directly or through playback in stereo mode from the XSP-1.

For completeness, here are some measurements of the TEAC UD-501 using RightMark Audio Analyzer 6.4.1 PRO comparing the 33' Corning USB 3 optical cable with direct 10' USB 2 wire cable. As usual I'm measuring off my E-MU 0404USB, at "high resolution" 24/96:

No difference; and a few of the graphs if you care for the details (just click to enlarge):
Frequency Response
Noise Level
THD + N
Stereo Crosstalk
Here's the Dunn J-Test with the Corning optical USB 3 vs. standard USB 2 cable if you're worried about potential for jitter:


I remain unconcerned about jitter with modern asynchronous USB interfaces (not that I was ever really worried even in the S/PDIF days about audible distortion caused by jitter even though measurable).

V. Conclusion

I hope this little discussion and demonstration using my set-up helps as a practical way to understand noise in a high-fidelity audio system. As expected, suggestions here may or may not work in your situation but I suspect it will help in many cases if you're running into trouble with noise. I have already received E-mail from folks with the Emotiva XSP-1 who have experienced the same noise issue with the analogue "Home Theater Bypass" input.

I repeat. I have never run into these noise issues with the direct DAC output! It is only with the analogue bypass mode on this pre-amp that I hear the noise. Here then is a good example of how the "system" as a whole can be susceptible to analogue noise... The digital transmission itself is fine and in fact the DAC output does not show the 8kHz contaminant nor ground loop hum. And if I did not set up a multichannel system with the XSP-1 preamp, I would never have realized there could be a problem.

A few conclusions then:
1. If you hear a hum; usually low frequency 50/60Hz (+ resonant multiples), think ground loops resulting from the AC power configuration. First thing to do is to reconfigure the power cables and see if you can connect the devices as close to each other as possible. Using a common power bar/power conditioning device worked for me. Even if you don't use a "cheater plug", it's worth having just to temporarily diagnose a problem by "lifting" the ground pin off various devices to have an idea of the "culprit" device. They're cheap. (Another link for those who want to read more about ground loops.)

2. Yes folks, computers are noisy! Indeed, the analogue noise can be transmitted down cables like the USB cable and cause audible interference. I demonstrated how the Corning Optical USB cable can be used to minimize the noise originating from the computer. It's logical. It can be measured. No magic.

3. Despite the use of the optical cable, the 8kHz high-speed USB "microframe packet noise" is still present but much reduced in my "case study". Not audible anymore, so that's good enough. It is possible that this residual signal is originating from the generic USB hub (PHYsical transceiver chipset) I'm using to provide the 5V power. It is possible that a device like the Uptone Audio Regen could result in a better, lower noise outcome. Another possible solution could be a high quality USB card like the oft-mentioned but expensive SOtM product. These solutions need to be objectively tested however and I suggest evaluation using a set-up like what I have here where there really is a noise problem rather than folks just plugging them into a system where there might not have been an audible issue to begin with!

4. Objective measurements of analogue output from my DAC using the optical USB cable shows no difference compared to a shielded 10' USB 2 wire cable. Likewise, no difference in the J-Test FFT to suggest audible jitter anomaly at 16-bit/44kHz or 24-bit/48kHz. Rest easy if you have one of these 10m (33') cables to connect your audiophile DAC. I have been using the Corning optical cable for almost 2 months now and it has worked flawlessly with DSD64/128, and PCM all the way to 24/384 with the TEAC DAC. Subjectively it sounds no different from any other functioning USB cable I have tried.

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For those who may have followed this blog for awhile, you may recognize that this is not the first time I wrote about a device that reduced noise with the "Home Theater Bypass" mode of the XSP-1 preamp. More than a year ago, I bought an inexpensive USB to Ethernet extension device that actually did the trick. I did not measure the 8kHz level with that but I don't believe it attenuated the noise as well as the optical cable shown in this post. It's still a fine solution for my problem until this optical USB 3 cable caught my eye!

If you look at the measurements in that previous post, you will see that I could not find a difference in the DAC analogue output there either between a direct USB feed compared with going through many feet of ethernet cable. Again, just more evidence that a good DAC is likely not going to sound any different between USB cables even when the data is processed through transceivers to convert to ethernet or optical fibres as in today's post. Subjectively I can't say with any certainty that I have heard differences.

I have been planning to write this piece for about a month now so I think it's interesting that this post just came out the other day. There are some interesting links in there, but I'm ultimately unclear of the practical intent other than to remind us that noise can cause problems and can be transmitted through a digital cable. Sure, it's possible. Let us consider this concluding paragraph for a moment:
"One approach is to just shrug this information off based on the belief that EMI and RF noise levels in our home and in our network-connected hi-fi are just not worth worrying about. There's no experience like no experience. Another approach is to try different solutions including different digital cables and see if they make a difference. There's no experience like experience."
Seriously, are those the only approaches? Why so black or white!? May I suggest a third approach, and IMO best approach... Become "more objective" in the analysis. "Experience" is nice but fully made valuable when paired with understanding of WHY something works (or not) especially as regards to scientifically engineered devices. To not explore why something is better than another (eg. Hmmm, it seems my system could be noisy. I tried and I think this fancy cable sounds awesome compared to that fancy one... Don't really know why... Just try it and experience for yourself!) is akin to superficially "shrugging off" an opportunity to learn; to understanding oneself and the technology behind higher fidelity. Nobody needs to enter this endeavour essentially blind using just trial-and-error! Why not try a bit of objectivism and confirm what is "heard"? Yes, develop experience but through understanding the system and showing others in what way a device (like the optical USB cable or if one believes in a $166/m USB cable) provides value. Demonstrate the actual problem and potential magnitude of the problem so the reader can evaluate with some confidence that they should give something a try because of a demonstrable deficiency that was improved rather than vague testimony. That IMO is what a review should be about, especially for passive devices like cables which as we know can be quite a financial "investment"! (I hear the term "accountability journalism" being tossed about; not sure what to make of this for the majority of audiophile "reporting" these days.)

One final and I hope obvious comment on this noise "issue". Don't worry unless you hear a problem! I've read comments over the years from some and you'd think unless someone invested in thousands of dollars of power conditioning, dedicated electrical lines, expensive cables, esoteric devices (cable lifters?) and removed every switching power supply within sight of the system, one would never achieve a system free from noisy distractions! Nonsense. I bet that the vast majority of setups employing reasonable equipment, decent cables, connected together in a logical fashion are just great as is. It's the systems with numerous components hooked up (like mine with both a pre-amp and surround processor switching between modes of operation with some vulnerability to noise), or maybe highly sensitive systems like very low voltage phono cables, step-up transformers, and phono-preamps that are most susceptible.

Okay. I hope this post has been useful. Let me know of corrections you would suggest and other interesting links. I'm sure there are many things missed and other ways to control noise I have not mentioned; remember, it is a "case study" so what I discussed here works for me and I bet the techniques (like using the optical USB 3 cable) will work for many out there also.

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I recently watched an interesting documentary from BBC - "When Albums Ruled The World" (2013). Lovely archive footage and historical review:


The sun's out and the Springtime temperatures are gorgeous on the West Coast... Time to head off...

Enjoy the music and please don't forget the ongoing Digital Filters Blind Test! Get me your results and tell your high-resolution audiophile friends to try! Come, experience...

Sunday, 26 April 2015

INTERNET BLIND TEST: Linear vs. Minimum Phase Upsampling Filters


I. Introduction

Ladies and gentlemen, audiophiles of all ages; you have probably seen pictures like the ones above many times in the magazines, posted online, etc. You know that many DACs these days feature the ability to adjust the way the antialiasing reconstruction filters are "tuned". I remember reading about and becoming fascinated by the different filters back in 2006 with Keith Howard's article on Stereophile. But it wasn't until around 2009 that I noticed Meridian started using minimum phase settings in their equipment like the 808.2 Signature Reference CD player. (I don't know if many other consumer audio devices used a form of the minimum phase filter before 2009; perhaps not as well advertised as Meridian?)

As you can see by the images above, when we send a 16/44 "impulse" through these reconstruction filters, we create different forms of high-frequency ringing. Linear phase filters do not result in phase distortion. However, the "issue" we are to note looking at the typical linear phase impulse response is the symmetrical ringing as seen above. It has been suggested that the pre-ringing ("pre-echo") is damaging to the sound and "unnatural". As a result, minimum phase filter algorithms became staples of companies like the aforementioned Meridian and later on Ayre with their well known whitepaper on their filter choice (I see this is also used in the recent PonoPlayer). Realize however that in removing the pre-echo, minimum phase filters will result in high frequency phase shift compared to the linear phase filter as shown here (from the Infinite Wave website looking at SoX 14.4 VHQ Minimum Phase vs. Linear phase):
Phase shift with minimum phase filter.
For completeness: Notice how sharp the SoX filters are which results in the ringing above. Both linear and minimum phase settings show a sharp "brick wall" frequency response like this.

As discussed on this blog and elsewhere, many have criticized the 44.1kHz sampling rate chosen as the CD standard decades ago. The Nyquist frequency of 22.05kHz is said to be too close to the audible frequency typically noted as going up to ~20kHz. Could this "ringing" at 22.05kHz cause audible effects? Can the linear phase "pre-echo" lead to a less realistic sound? Can the minimum phase filter setting sound "better" because we've removed the pre-echo (the post-echo ringing supposedly masked and less likely to cause audible effects - post-masking) despite the phase shift?

Remember that the technical discussion of digital filters can get complicated, often with no actual data around audibility to support positions. Over the years, special player software such has HQPlayer have been released with upsampling algorithm tweaking as a major feature. However, I hope that in using the open-source SoX algorithm with basic parameters that are not specially selected to reduce the "ringing", this test can provide some insight into audibility in the "real-world" with your help.

Here's your chance to listen/test for yourself and let me know what you hear!

II. Test Procedure

As usual, I have put the test files to download off my FTP server and thanks again to Ingemar at Privatebits, the file can be downloaded there as well. Warning: File size of 130MB, "Linear-vs-Minimum-Phase.zip".

Download off Archimago's FTP (digitalfilterstest.dyndns.org) - Login: "filter", Password: "test"

Download off Privatebits

As you see in the ZIP package, there are six FLAC files; each ~1 minute in length named:

A big thank you to Juergen for providing the first two excellent sounding recordings (Mandolin and GrandPiano) as 24/44 WAV files. As Juergen described when he provided the samples, the recordings were done in pure stereo using 2 cardioid condenser microphones, equivalence stereophony technique. No dynamic compression, no limiting, no reverb or other artificial processing. The music was originally recorded at 24/88 and downsampled with Reaper. I'll let Juergen describe the recordings specifically:

Mandolin concerto: "recorded live in a church. So lots of room information and a bit longer decay in the bass section. But fine and nice details."

GrandPiano: "And here is a photo showing the microphones, where they are placed, during the recording of the Steinway Grand D Piano. This is the only Grand D in Europe that is not painted with piano lacquer, but only with some thin matte black, in order to have a more raw sound. This is the Piano where Oscar Peterson played on, on some of his recordings. This piano has a more 'wooden' sound and not so 'polished' as the Grand D in concert houses."



The third sample is the title track from Mighty Sam McClain's "Give It Up To Love" (AudioQuest, 1992); a very well recorded blues album from an artist who I believe is well known among audiophile circles. This is certainly one of those "demo CD's" worth showing off a hi-fidelity system with. Original source is 16/44 CD rip.

As usual, I use these samples under the principle of "fair use" for the purpose of research and commentary.

I took ~1 minute of these three 24/44 or 16/44 recordings and using SoX, upsampled them to 24/176.4 with either the linear or minimum phase upsampling algorithm, re-naming one file A and another B. Note that A and B are not the same for the three samples, so if I thought "Mandolin A" was the linear phase filter, this does not necessarily mean "GrandPiano A" was also the linear phase output.

Here's the DR measurement result of the test samples:

This confirms that the average amplitude is not significantly different between samples A and B. Slight differences in peak amplitudes (please do not think that this correlates with the type of filtering used!). As you can see, each track maintains excellent dynamic range with DR 12-14.

Your job:
1. Make sure you have the proper pre-requisites:
     - High resolution DAC that can play 24-bit, 176.4kHz sampling rate. (NOTE: Most 192kHz-capable DACs should have no problem.)
     - Make sure your player / DAC is not resampling the audio (eg. don't play on a Squeezebox system that has a maximum of 96kHz for example).
     - Turn off all DSP processing like room-correction and EQ for the computer and if you're using a receiver / processor make sure it's "direct" audio, no Audyssey processing and the like.
     - If you're using playback software like JPlay where you might not be able to verify that it's not adding any processing, I recommend just playing this with foobar and bit-perfect ASIO or 24-bit WASAPI to your DAC.
     - Feel free to use tools like the newest ABX Comparator in foobar. Just make sure again the music is not being resampled and being sent out bit-perfect to the DAC.
     - Recommended to listen with speakers to recreate the soundstage. However, headphones may provide better temporal resolution with no room distortion- your pick, just let me know which!

2. Have a good listen to the A and B samples. Consider which of the 2 samples you liked better - that is, sounding more "real", "lifelike" considering these are natural recordings in "live" space. Which did you think had better sounding transients (better temporal resolution)? Did you hear or suspect you heard any artificial "pre-echo" that would suggest the linear phase filter? Did you hear any high-frequency distortion?

3. Fill out my survey and tell me what you heard. As usual, the data will be collected anonymously but cookies have been turned on to limit submissions from any one machine. Note that in previous blind tests, I have had people contact me asking me what they entered because they forgot. In such cases, it would of course be easier if one could leave some kind of easy-to-find identifier or just make sure you include detail I can easily search for later.

The survey can be found here:

Data collected will include:
- Demographics: gender, age range, which continent you're from
- For each piece of music, which one sounded better / more natural / more realistic? There will also be a choice for "sounds the same" if you don't hear any difference.
- Tell me how confident you are about your choice.
- Describe your audio system - seriously folks, this does NOT need to be some $$$$ system! (Checkbox for headphones vs. speakers for the main system used to evaluate.)
- Subjective impression for what you heard. You might also want to comment if you have special listening training here.
- Identification of whether you're a musician / audio engineer / audiophile writer for cohort analysis if possible.

For the sake of consistency and obtaining a good data set, I will insist that most of the questions must be answered. Thanks!

III. You have 2 months.

Hopefully there will be no rush since the samples are only 1 minute each. The survey will close on JUNE 25, 2015. Make sure to get all your responses in by that time!

Remember, this is a listening test! Please refrain from using an audio editor and your eyes :-). Also remember there is no right or wrong; I just want to get an idea if there is a significant preference out there or if you believe you hear a difference. Also, please do not bias others in discussions... I think it's cool to talk about what you heard or did not hear so perhaps other can learn to perceive the difference but let's not specifically talk about file A or B, and even if you know which is which, let's be courteous and keep this as a "blind" test for others until all data collection is complete. I will of course reveal all when I summarize the results in June or early July.

Feel free to advertise this test among friends / family / enemies. Also, feel free to put this up on the various audiophile websites, Facebook, wherever you know high-resolution audiophiles may reside (remember, these are high-resolution 24/176 test samples). The more testers the better to improve statistical power.

Much appreciated!

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I've been getting into some newer music lately. A reader switched me on to Butch Walker's stuff recently so I'm working through his discography. Good "new" rock! Enjoying his 2011 album The Spade. I just wish the recording quality was more natural and less dynamically compressed.

With the weather getting better here in Vancouver, it's time to also get outdoors with the family and enjoy other hobbies like photography; perfect time to focus on other things while data is getting collected! Have fun and let me know in the comments if you run into troubles or have other questions.

Enjoy the music...

Monday, 20 April 2015

PERSPECTIVE: Poll - Upgraded USB and Sound?

I enjoy visiting the Steve Hoffman Forums for perspective. I think it's probably one of the most balanced places to hang out at. Of course, as in any audio forum, there will be a number of heated arguments here and there, but overall, it's great to see a place where heterogeneous music lovers of various experience levels and beliefs can congregate, share tips, and get advice... It's certainly one of the more vibrant audio communities out there!

Recently there was an interesting poll on whether USB cable upgrades can improve a system's sound quality. Titled "The Great USB cable debate poll" (full disclosure, I took part and posted a comment as well), it ran from February 22 to March 8th before it was closed. In total, it received 415 votes, gathered 36 pages of responses and the outcome was:

About 1/4 felt that upgraded USB cables make a difference, and 3/4 did not. Surprised?

Of course, any poll must be viewed in the context of the respondents. This one was conducted in the "Audio Hardware" forum and like I said, I think the denizens of this site does capture a broader group of audiophiles and music lovers than most places; from pros to reviewers to the general music enthusiast. Topics range from people posting pictures of their audio room, to discussion of LPs and turntables, to opinions on the latest DACs, computer servers, speakers, headphones...

I certainly do not expect we should put full faith in something or other based on general consensus. In areas where we are passionate about, of course we should figure things out ourselves and come to our own conclusions... Once awhile, it is nice however to see a poll such as this to get a sense of what the "cohort" thinks. As an audiophile who reads the usual magazines, one might get the impression that there's almost 100% acceptance that upgrading USB cables for sound quality is a "given" within this hobby. Likewise, some might pigeon-hole all "audiophiles" as "audiophools" spending all kinds of money on questionable claims. IMO both perspectives would be inaccurate.

I think it's fair to recognize that most audio lovers are reasonable people even though sometimes it seems like the most fringe and outlandish voices appear to be the mouthpieces of this hobby.

Hmmm, I wonder what the result would be if we asked "Can upgraded ethernet cables at various price points improve the sound of your system?"

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Okay, just a quick post here today as I'm working behind the scenes to get another blind test going... More information in the next week or so I hope!

Have a great week.

Saturday, 11 April 2015

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.

I. DSD-to-PCM - foobar SACD plug-in & AuI ConverteR

So, I downloaded the newest SACD plug-in currently at version 0.7.7 dated 2015/03/16. I deleted the DSDIFF plug-in from my computer so there are no interactions, and installed the new files.

Notice that the SACD plugin has a configuration panel for settings:

Because this plug-in does not directly output to 24/96, I figure let's try with the highest output (352.8kHz) and I will use the best samplerate converter (SRC) I have (the excellent iZotope RX 4) to bring it down to 24/96 for analysis as before. Here are the settings I used in iZotope RX 4:
Sharpest "max" filter for 24/96 in iZotope RX 4, linear phase without suppression of pre-ringing - nothing fancy...

The other parameter we can play with in the SACD plug-in is the DSD2PCM mathematics setting. By default, it's the standard fixed-point integer mode. Let us also analyze the result from the highest precision mode "Multistage (Double Precision)".

As well, I downloaded the AuI ConverteR 48x44 software. The current demo is version 4.1.20. Other than setting the output for WAV 24/96, I left the rest of the settings to default.

Using the exact same procedure as last week, here's a summary of what I got:

Interesting... It looks like the SACD plug-in is actually about the same as the old DSDIFF 1.4 (<1dB difference) in terms of noise level and dynamic range. Notice just like last week's results from XLD, that going from fixed-point to double floating-point calculations made no difference here.

AuI ConverteR resulted in some impressive numbers! Let's see what it's doing in detail...


As you can see, AuI ConverteR is using a sharp "brick-wall" filter right at ~20kHz to remove essentially everything after 20kHz. As such, have a look at what it does with the noise profile:

Wow. That is an impressively sharp, precise filter at 20kHz! I can approximate that effect with iZotope RX 4's EQ plug-in with a low-pass at 20kHz, high Q of 25 or so (not shown) but AuI ConverteR looks even cleaner with less noise floor irregularity.

That little bit of high-frequency "rippling" with DSDIFF is probably a result of the resampling algorithm. Otherwise, foobar DSDIFF and the newer SACD plug-ins appear very similar.

Basically, this is what we can say at this point...

1. foobar SACD plug-in works about the same as the old DSDIFF plug-in. I would not be surprised if the algorithm (DSD2PCM) is essentially the same if we look "under the hood".

2. The AuI ConverteR software puts up some impressive numbers. This is done with a very strong low-pass filter. If you feel there is no need to retain frequencies >20kHz, then this will clearly get the job done.

II. All that noise!

But wait, there's more! SACD Plug-in also has a 30kHz lowpass mode - "Direct - (Double Precision, 30kHz LF)". Hmmm, I wonder how that looks?

Engaging the 30kHz lowpass mode really resulted in a step down in calculated accuracy. Here's a look at the graphs:

Indeed, the 30kHz low-pass filter is doing the job (yellow).


We can see the effect of that 30kHz filter on the noise floor... Certainly not the prettiest filter out there! Realize that although the differences are there in these graphs with a synthetic test signal, we're talking about noise down below -150dB (below 20kHz). It's just not an issue in terms of audibility.

Now, let us see if we can do it better by using iZotope RX 4 to do the 30kHz low-pass filtering instead of the algorithm used by SACD plug-in. Here's a simple setting:

Low-pass filter at 30kHz as seen, Q = 5.0 (not too steep), linear-phase FIR with FFT size of 32k.

VoilĂ :

This is what a good low-pass filter can do for the results. As you can see, when we use the 30kHz iZotope low-pass filtering, the calculated noise level drops substantially on the RightMark analysis... This also tells us that the reason Saracon and JRiver measured so well is because they implement very good quality noise filtering algorithms beyond just the DSD-to-PCM calculations.

The SACD Plug-in's DSD2PCM algorithm is excellent, and when we pair it up with iZotope's SRC and 30kHz low-pass filtering, we get some fantastic results easily on par to what Saracon does:



III. Impulse Response and DSD-to-PCM Converters

Despite the inherent noise in DSD, we can drop overall noise levels substantially with a good low-pass filter. In fact, since a picture is worth a thousand words, this is what a 15kHz (-12dBFS) sine wave looks like comparing the unfiltered foobar SACD plug-in output at 352kHz with the 30kHz iZotope RX 4 low-pass filtering (again, this is with Saracon as the encoding software for PCM-to-DSD):

This is what all that extra high-frequency noise looks like in DSD when you don't filter it out at all. Notice that DSD128 is significantly less noisy. The question is, just how much noise reduction should we actually do? (You can also see the noise through an analogue oscilloscope - as shown here.)

As noted by Juergen in the comments to the previous post, there is this matter about time-domain behaviour as well which can be skewed as we apply various filters.

Let's see what a 24/96 impulse looks like after going through the DSD encoder [Saracon] and most of the decoders I looked at (DSD-to-PCM converter output set to 24/352 for each, AudioGate's max was 192kHz):
(Click to enlarge.)
In the top left panel, this is what a 0dBFS 24/96 "impulse" would look like with a typical linear-phase oversampling interpolation showing symmetrical pre- and post-ringing. Even though Adobe Audition renders the interpolation, the actual PCM data itself is a simple, single "pulse" (see Addendum below for screenshots using Audacity).

When I convert this waveform to DSD64 and DSD128 with Saracon and then back to PCM with the foobar SACD plug-in to 24/352 unfiltered (retaining all that ultrasonic noise), you get the 2nd and 3rd left images. Notice again the amount of noise in the signal and again, we see the superiority of DSD128. From a time domain perspective, the SACD conversion process is excellent. The shape and timing of the impulse would be completely retained since the 2.8224 MHz sampling rate of DSD64 provides ~29 samples within each 96kHz time period.

When we use iZotope RX 4 with 30kHz low-pass filtering (4th left image from the top), the "impulse" amplitude is significantly reduced and we see the corresponding ringing pattern as the high frequency noise is removed and no longer obstructing the picture.

AudioGate and Saracon both look very similar. Both use linear phase filters with characteristic symmetrical pre- and post-ringing. Whereas AudioGate allows high frequencies through (and thus well formed impulse), we see the effect of Saracon's filter (pre- and post-ringing ~30kHz).  JRiver looks like it uses an intermediate phase filter (with 24kHz or 30kHz low-pass) which minimizes but does not remove pre-ringing. Comparatively, we see that DSD Master is using a form of minimum phase filter that removes the pre-ringing but the post-ringing is augmented.

AuI ConverteR is an interesting case. As we saw above with the RightMark tests, it implements a very sharp ~20kHz low-pass filter. This impulse response looks to be linear phase with accentuated pre- and post-ringing due to the sharpness of the filter; the "price" to pay I suppose.

I'll leave you to decide how you feel about this information and whether you think the relative time domain effects resulting from implementation of the filters are audible. Back in 2013 I had a listen to some filter settings off the TEAC DAC and had difficulty noticing much of a difference; again here, I listen and fail to convince myself that I have any clear preferences among the converters including using ABX Comparator. So far I'm using headphones (Sennheiser HD800 + TEAC UD501 DAC, ASIO driver playing DSD64 converted to 24/352kHz) so perhaps I need to try again with the speaker system. You guys up for an internet "blind" test to see if there's a preference towards linear phase vs. minimum phase upsampling???

IV. Conclusion

I hope we can appreciate the compromises we face with DSD to PCM conversion. How much noise can we tolerate from the 1-bit quantization when we move the signal to PCM? What's the best frequency to set a low-pass filter assuming one believes it's necessary? What parameters should we use to filter (minimum / linear / intermediate phase, sharp vs. gradual roll-off...)? What's the best sampling rate to spit out the PCM data (eg. do we need to produce >96kHz files if we roll-off before 48kHz)?

As I noted last week, I really am not convinced that these differences are audible beyond volume level changes and whether the ultrasonic noise causes problems for one's audio system (eg. intermodulation distortions, interaction with tweeter ultrasonic peaks, and other non-linearities). This is why I don't think there's any point in "crowning" any software package as being superior. Although it's interesting to demonstrate and experiment with, I suspect this is all rather obsessive academic results of interest to audio geeks :-).

If I had to choose, I remain partial to Saracon and JRiver because of the excellent results from the low-pass filtering used by default with those programs; one-step easy conversion using very reasonable parameters. As you can see, I can get similar results with the foobar SACD plug-in creating 24/352 output, and running that through iZotope RX 4 with high-precision samplerate conversion and low-pass filtering indicating that the underlying free DSD-to-PCM algorithm works well. AuI ConverteR is interesting in that the default setting I looked at resulted in a very clean output so long as one does not feel there is any need for >20kHz signals to be retained nor concerned about the effect on the impulse tracing.

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You can perhaps imagine, after "penning" these last 2 posts, I'm pretty well done with talking about DSD for awhile. The most interesting question for me currently as suggested by the discussions with the previous post is this whole notion of just how much significance we should place on resolution in the time-domain irrespective of audible frequencies.

If it is significant (I hesitant to use the word "important" since that should be obvious by now if it is the case), then how much is enough? Should we take research like this paper by Kunchur (2008) seriously? Or is it possible that for practical purposes, it doesn't really matter that much when we're listening to real music as opposed to test signals? In any case, I have a strong suspicion that we will be revisiting this in the days ahead since this seems like an area that will be brought out when Meridian's MQA becomes available as I suspect they will emphasize time parameters, digital filter types, and samplerate given their apparent satisfaction with 16-bit resolution.

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Finally, it has come to my attention that there was much unhappiness regarding a recent blog post on the importance of noise (here also) in digital audio reproduction to the point of using speculation to support an underlying belief that expensive ethernet cables could somehow impart beneficial effects (as you know, I found no evidence of significance in my testing with various types of ethernet cables). As usual, no empirical data or real-life examples were provided and support came from more testimony from the like-minded and some links that are at best tangential to high-fidelity audio. It looks like bans from commenting were issued for what seems like rather fair statements calling out the obvious lack of substance. I guess that's how people not felt to be "true believers in the audiophile experience" are dealt with. IMO, this is unfortunate behaviour for a site reporting on mature audio computing technology.

There is much that can be said, argued and refuted in that article, but I think for most reasonable audiophiles it's rather obvious and many excellent points can be found in the comments... What is of relevance to this blog entry is that if one believes that expensive ethernet cabling can reduce noise in the "system" (in a way that appears difficult for these people to produce empirical evidence for), why would any audiophile who subscribes to this theory even want to listen to DSD64 (where the noise is obviously demonstrable and a potential cause of distortion)? Or even consider DSD64 superior to 24/192 at times? Would it not be just as likely that some folks actually like the ultrasonic noise and what it actually is doing through the system? Perhaps similar to how some tube-lovers talk about certain types of distortion being unobjectionable? In fact, back in late 2013, I posted on my impressions with realtime PCM-to-DSD transcoding with JRiver 19 and felt that DSD64 did impart a subtle change to the sound. I wouldn't say that I felt the sonic difference compelled me to convert all my PCM files to DSD for listening, but it was an interesting effect. Maybe that's why some people would prefer a DAC that purposely converts PCM to DSD like the PS Audio DirectStream DAC (the signal is purposely downsampled to DSD128, and then only noise filtered by 80kHz according to this review).

I hope you enjoyed this exploration into the world of DSD (again)... I got a few projects piled up to work on so might not be around as much for the next couple weeks. I'll also be in Boston in the next little while so if anyone has a recommendation on music store I should check out near downtown, let me know!

I'm also thoroughly enjoying David Byrne's book How Music Works (2012, with 2013 update) - check it out for entertaining reading!

Enjoy the music folks :-).

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Addendum:
Note that Adobe Audition renders the PCM data with a linear phase interpolation filter. Here are renderings of some impulse waveforms using Audacity which does not do the fancy interpolation for reference:

Friday, 3 April 2015

ANALYSIS: DSD-to-PCM Conversion 2015 - Windows & Mac OS X

Impulse Response: One of the talking points from back in the day as a selling point for DSD... Yup! DSD can better reproduce a 0.000003 second "click". Source: Merging Technologies

I. Preamble

It is amazing how quickly another year has passed. About this time last year, I posted the first comparison of DSD Encoders and Decoders "shoot-out" of sorts comparing Weiss Saracon 01.61-27, KORG AudioGate 2.3.3 and JRiver 19.0.117 in terms of quality - both encoding and decoding fidelity using the RightMark Audio Analyzer software. The idea was to determine which of the three created DSD files from an original 24/96 PCM test signal and then decoded it back to 24/96 in a way where there was as little change in terms of distortion, flat frequency response, and lowest amount of added noise.

Perhaps not surprisingly, the expensive Weiss Saracon software sets the standard as the most consistent DSD encoder that resulted in the best output once decoded. The differences in decoding capability appeared to be very minor (questionable audibility between the 3) but objectively, both Saracon and JRiver 19 were on par and the free AudioGate 2 somewhat "noisier" in terms of the PCM output (I speculated this was due to stronger dithering algorithm).

Well, another year has passed in terms of software upgrades to DSD decoding and I was interested to compare the decoding capabilities as of late. We have brand new versions of JRiver and AudioGate now, plus I didn't get to test foobar with the DSDIFF plugin last year. Plus we now have DSD decoding on the Mac OS X available with XLD and commercially with DSD Master.

To maintain an "apples to apples" comparison as best I can, I will use the Saracon 01.61-27 encoded DSD file of a 24/96 test signal with each of these decoders. I think this is fair given that Saracon produced excellent results last time as an encoder plus it is an "industry standard" used in many professional studios.

Let us have a look at the decoders I will be testing:

1. foobar with DSDIFF 1.4 plugin [Windows]. foobar2000 does not need any introduction as a freely available Windows music player. It's stable. Sounds fantastic. Is bitperfect. Is extremely feature-rich. And as witnessed by the myriad plugins like DSDIFF, very extensible. The DSDIFF plugin has been around as version 1.4 since 2011 by kode54. Decoding was set to 24/96 PCM, WAV output.

2. KORG AudioGate 3.0.2 [Windows]. AudioGate got an upgrade in June 2014 to version 3. It's no longer "free" like before where you can do conversions after just allowing Tweets from your account. I'm going to test with the output set at 24/96, "High Quality", no dithering. Let's see if the noise floor is better than AudioGate 2.

3. JRiver Media Center 20.0.87 [Windows used, Mac and Linux also available]. Upgrade from version 19 to 20 has introduced some new features (one of which I'm most interested in I'll talk about another time). Don't know if DSD decoding has changed... We shall see! For the test this year, I decided to stay with the default "safe" 24kHz low-pass filter like last year. Remember, you do have a choice: go to Tools --> Options... --> Advanced --> ... Configure input plug-in ... --> DSD input plug-in...

4. X Lossless Decoder (XLD) 20141129 [Mac OS X]. Freely available audio tool for Mac users. As of the 2014/11/09 release, it has the capability of decoding DSD files. Let's see how this newcomer compares! Again, decoding target was 24/96, I used higher quality "SoX VHQ Linear" resampling. Since the options allow easy adjustment of parameters like decimation and quantization, I will test both the default (8:1 decimation, 24-bit integer quantization) as well as higher quality (8:1 decimation, 32-bit floating point). [Decimation correlates to the output samplerate, so 8:1 for DSD64 is 352.8kHz output which then gets resampled by SoX to 96kHz.]

5. DSD Master 1.0  [Mac OS X]. Thank you to Richard at BitPerfect Sound Inc. for letting me give this program a spin (US$29.99 on the AppStore)! As per the company namesake, these are the same guys who brought you BitPerfect for the Mac. I actually downloaded this program in late 2014, unfortunately it took me awhile to get to the testing... After migrating to Windows in the last 2 years, I just haven't been using the Mac nearly as much. As with the other programs, I've set DSD Master to do all processing back to 24/96. Write as a WAV file. No gain applied. I see DSD Master can handle up to DSD256. One interesting feature is the DSD Hybrid mode where the DSD data is retained along with PCM conversion - large file sizes to be expected, and you will need BitPerfect 2.0 to play through iTunes to a DSD DAC.

For completeness, I will also post up the results for Weiss Saracon 01.61-27 which I obtained last year as the standard for comparison. (I borrowed Saracon to test last year so have not kept up as to whether there have been updated versions since.)

II. Results

Here are the summary results (to keep the tables smaller, I separated the Windows and Mac software used):
Windows DSD Conversion
Mac DSD Conversion
Remember that the RightMark software is calculating these results based on the audible 20Hz to 20kHz spectrum. Ultrasonic components therefore will be ignored in the numerical result so factors like different low-pass filtering settings will not show up here unless it clearly encroaches into the audible range. The first column is for an untouched copy of the original 24/96 PCM test signal - this represents the ideal / "perfect" result possible. The second column is the results from Saracon - same numbers as last year for comparison with a professional piece of software (with professional price tag).

If we compare the software updates, we see that there has been a substantial change with KORG AudioGate 3 compared to version 2 in terms of noise level; I'm seeing about a 10dB improvement compared to last year. If my suspicion is correct, perhaps they were using a stronger dithering algorithm back in version 2. The result is now much more in line with the other conversion packages.

As for JRiver 20 vs. 19, there has been little change; about 1-2dB difference. Note that JRiver posted fantastic numbers last year already so the slight improvement is actually very impressive!

Let's now look at the newcomers to this round-up...

Foobar DSDIFF did okay overall. Good performance and about the same as AudioGate when decoding the test signal. Hey, it's free :-).

XLD on the Mac provided good results as well. As you can see comparing the 24-bit integer to 32-bit floating point calculations, there was no difference. Any difference between 24-bit and 32-bit processing would be below the precision of the final 24/96 WAV file and RightMark's calculations. Therefore, if you use this program, you might as well stick with 24-bit integer calculations as it's faster.

Finally, we have the DSD Master software. Excellent quality output converting DSD64 to 24/96. Very low noise level and high dynamic range essentially the same as the much more expensive Weiss Saracon. (Remember, this is all relative since we're talking about noise levels below -130dB!)

Let's have a look at the graphs:
Windows DSD Conversion - Frequency Response
Mac DSD Conversion - Frequency Response
As you can see, JRiver has the strongest low-pass filter of all the programs (24kHz, 48dB/octave by default) followed by Saracon. Foobar DSDIFF plugin rolls off a little earlier and finally AudioGate looks like it lets most/all of the noise through. [Remember, in principle, it is a good idea to keep the low-pass filter to reduce potential for intermodulation distortion on playback and reduce any risk of damage from excess high-frequency noise. It also reduces the file size.]

On the Mac, I see that neither XLD nor DSD Master perform much low-pass filtering - at least not within 48kHz.

Here are the noise level graphs:
Windows DSD Conversion - Noise Level. Slight irregularity in the DSDIFF noise floor at high frequencies.
Mac DSD Conversion - Noise Level [note XLD tracings exactly the same so only the purple 32-bit tracing showing up]
No surprises around the amount of ultrasonic noise resulting from the noise-shaping required with 1-bit DSD quantization. As usual, with DSD64, the noise starts right around 20kHz using Saracon as encoder (remember that some encoders like AudioGate 2 last year, the noise floor isn't as flat all the way to 20kHz). We know that Saracon conversion back to PCM deals with this ultrasonic noise using a strong filter which essentially suppresses everything by 40kHz (hence 24/88 is good enough with Saracon). In a similar fashion, the default 24kHz low-pass filter with JRiver does a good job keeping ultrasonic noise down. As you can see, the others - foobar DSDIFF plugin, KORG AudioGate, XLD, and DSD Master are all allowing the high frequency noise through at default settings.

III. Conclusions

As I suggested last year, I believe that DSD --> PCM conversion is transparent. I'm measuring the distortion added by both PCM (24/96) --> DSD64 [via Saracon] as well as DSD64 --> PCM (24/96) steps and as you can see, there is nothing showing up of concern. Fidelity is maintained beyond any DAC's analogue output I am aware of except for all that ultrasonic stuff peaking at about -85dB if filtering is not applied. Speaker system / headphone playback would add more distortion than this (at least within the audible frequencies).

You might be asking - what about subjective listening?

Well, I did convert Jorma Kaukonen's "Blue Railroad Train" (from the SACD Blue Country Heart, 2002, recorded in DSD) with each converter for a listen with the TEAC UD-501 DAC (reviewed here) through my Emotiva XSP-1 preamp, XPA-1L monoblocks in Class A bias mode, and Paradigm Signature S8v3 speakers, connected with all balanced interconnects. Once I made sure the levels were matched (for example, by default JRiver did not perform +6dB gain, where as AudioGate does), I could not confidently differentiate the output from each converter. It's hard to do a blinded comparison between DSD and PCM because the DAC emits a faint "click" each time there is a switch between the formats to let me know something has changed, furthermore, DSD playback isn't at exactly the same level (not to mention the DAC boldly declares "PCM" or "DSD" on the front LCD panel). The recording sounded beautiful, "crystal clear" - excellent whether in the form of a direct DSD playback (native ASIO) or converted to 24/96 PCM. Let's just say that if I walked in the room, I would have no trouble thoroughly enjoying the music produced through these converters...

Bottom line: No need to worry about the sonic output from any of these converters IMO. Conversion algorithms and software look mature with little difference between them. The only significant choice is whether you want to have a low-pass filter in the conversion process (I do so I'd prefer Saracon or JRiver). I know some people claim they can hear qualitative differences between conversion programs beyond just level differences... Maybe. I'd certainly be impressed if anyone can show positive controlled, blinded listening test results given the minute changes I see/hear while doing these tests!

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So guys, what do you think about the state of DSD these days? It really looks like the "push" has fizzled lately... Other than more DACs supporting native DSD playback, there seems to be little news out there. Anyone actually buying many DSD downloads?

As I wrote back in April 2013 (On SACD & DSD audio...), there are many factors working against DSD audio if the goal is to expand beyond just a small audio-geek niche format. It appears my concerns around the need for a modern file format that provides full tagging and data compression persists...

Psssstttt... Coders... Want to be famous? Pull together some code to create an open source ID3 taggable compression CODEC for DSD (.fdac? Free DSD Audio Codec - how about just .dac format). Get the guys at JRiver and foobar2000 to support it, and make sure it runs in Linux for music servers. I bet this format would become widely used among the guys ripping their SACD's and those who rip LPs into DSD! Make sure to support compression of DSD128+ as well just in case hi-res DSD becomes available (as far as I know the old Philips ProTECH DST Encoder could only handle DSD64)...

Until next time, have a wonderful April and hope you're enjoying the music!