Thursday, 27 March 2014

MEASUREMENTS: Another Look - Audioengine D3 clipping at 100% volume.

The ability to interact with you guys over the last year or so running this blog has been vastly educational for me! Whether data gathering with the MP3 test last year or comments and suggestions with each post... In general I do try to keep up with comments if I can but after 80+ posts now, I apologize if there are some comments I've missed along the way.

The post today is thanks to the keen eye of "Solderdude Frans" and his comments & suggestions to the previous post on the Audioengine D3. As per Addendum 2 in that post, indeed there is indication that a 0dBFS signal is clipping with the D3. I had missed it due to the fact that I was looking at the square waveform without considering the possibility of clipping contributing to the shape observed. Also while calibrating for the RightMark tests, RightMark and E-MU were flashing red for clipping and I thought it was that the amplitude was too high for the E-MU 0404USB rather than considering the possibility that the clipping was actually from the D3 output itself.

So, as suggested by Frans, this is what a 0dBFS 1kHz sine wave looks like with no volume attenuation (ie. 100%):

Yup, the peaks are being clipped!

The point where the clipping ends is with the hardware volume control turned down to 92% (93% almost was good enough) in the Windows 8.1 control panel (software volume level in foobar stays at 100%):

I re-ran the RightMark measurements ignoring the warning about clipping to see what happens to the measurement results:

As expected, there's a deterioration to the measured THD and IMD results going from 92% to 100% with clipping - still low using the RightMark methodology but more than 10-fold worsening with the clipping in place.

Interesting... Like the first sample AudioQuest Dragonfly reported in Stereophile awhile back, it looks like one needs to back off the Audioengine D3's volume setting a bit in order to avoid clipping. In the case of my sample, pulling down the Windows volume control to 92% did the trick.

Well, this fact obviously blemishes my impression of this little USB DAC as an accurate audio device. In practice, it's unlikely I will push the volumes to 100% but I wonder if this was by design. I find it hard to believe that the engineers did not check the clipping characteristics. Rather, my suspicion is that the engineers felt it was OK to allow the signal to clip a bit to increase the perceived amplification level. In a noisy environment, the extra amount of amplification of low level signal at the expense of distortion might be a reasonable trade-off that may not be too noticeable (possibly no problem at all if the music is soft and rarely hits 0dBFS) - of course, this would not be a "high fidelity" practice.


Seems like there are some ruffled feathers around the recent "Geek Out vs. others" measurements published by Light Harmonic. Good to see that the Geek Out is being designed with good measurements in mind; encouraging. I can see how other manufacturers can be a bit unhappy about all this... Although it's probably wise to have a 3rd party perform the tests rather than a direct LH release, I suppose if it gets the manufacturers thinking about better engineering and competition in this regard, that's a good thing. Curious that there was no frequency response measurement. I wonder if at 44kHz, the Geek Out does measure like a NOS DAC with some early roll-off in the high frequencies.

Ho! Larry somehow believes that "It's proven wrong to assume people could hear only below 20KHz" (see the comment section of that article). Ok. I suppose Ashihara's paper from 2006 could be used to argue this. But we're talking SPLs around 80 dB at 20kHz for most subjects as the threshold of hearing pure tones and there's quite a significant jump between 18 to 20kHz. Threshold for hearing pure tones isn't exactly evidence that this is beneficial for real music! Furthermore, if you look at the test subjects, they're aged 18 to 33 and the majority of these are young women! Good that hi-res can satisfy the golden-eared audiophile lady in our midst... Unfortunately it's not going to do much for the old boys I suspect ;-).

Friday, 21 March 2014

MEASUREMENTS: Audioengine D3 USB DAC / Headphone Amp Reviewed.

In the last couple of years, we have seen a proliferation of small sized USB DACs. Devices small enough for laptop-sized portability aimed at the headphone user who wants a bit more power to drive better 'cans' and provide improved sonics than what's available through the laptop's phono jack.

I guess it must have begun with the AudioQuest Dragonfly back in 2012. Currently in the 2nd incarnation as version 1.2. Version 1.0 got quite a positive review in Stereophile among other magazines which has helped propel this class of device into audiophile acceptability (if not some mainstream approval?). With the Kickstarter success of the Geek (Out) last year, these devices continued to gather press.

To create a device like this can be difficult given the dependence on the USB port for power. I've certainly experienced first hand the noise pollution from plugging my TEAC DAC into the computer USB port and this being picked up by my Emotiva XSP-1 preamp's analogue passthrough. That was why I bought the USB-to-ethernet cable extender to provide some noise isolation that thankfully worked.

So, a few weeks ago as I was perusing the local computer store, I ran into a sale on the Audioengine D3 USB DAC. I figure, what the heck - at less than $180CAD, it'll give me something to play with and if it sounds good, maybe it'll accompany me on overseas trips with a good pair of headphones.

I'm not new to the Audioengine brand. I've been using their A2 powered speakers on my desktop for at least 2 years now and can certainly vouch for the build quality. As you can see from the picture above, the D3 package comes with the DAC itself which has 2 LEDs - white for power, and a blue one that lights up with 88 & 96kHz audio. It also comes with a good quality phono adaptor for 6.3mm(1/4")-to-3.5mm conversion. There's also a functional grey fabric pouch for the DAC - useful to prevent it from scratching other things due to the "metal injection molded" aluminum case which makes the device feel quite solid in the hand.

I made it a point to take a picture of the text on the side of the box as well. As you can see, Audioengine likes to drop some hints on what's inside... The DAC is the AKM4396 (same as my Squeezebox Transporter), and the audio amplifier is the LME49726 opamp. The 2Vrms output level is good as standard line level RCA also. Asynchronous USB communication rounds out the specs. About the only concern I had was the 10-ohm output impedance on the device (see Tech Specs). This suggests that the DAC would best be suited for 80+ohm headphones (see here for details).

One of the things I always check with headphone outputs is whether it's loud enough with my AKG Q701 as pictured above (the Audioengine A2 speaker can be seen also). Happily, it does indeed power the Q701 reasonably well. I can listen to dynamic music without needing to push the volume to 100% which I'll speak more about later.  Also in that picture you see the 2 LEDs lit up. The inner one is power, the outer slightly bluer one is for >48kHz audio.

One observation is that because of the aluminum case, this little USB DAC can get pretty warm after awhile. Not enough to burn fingers of course, but mildly uncomfortable if you put it in a shirt pocket for example while on the go (cools down quickly though). Make sure to use the little supplied pouch!

One last thing before we get to some measurements. This little DAC does not require a driver in Windows or with the Mac (alas, I do not have a Linux machine handy). There's no ASIO driver for that TI1020B asynchronous USB chip inside and audio will be sent out based on the Windows Mixer settings by default. This is also how you control the volume. Remember that if you don't want Windows to resample, make sure to go in and manually change the sample rate settings in the "Sound" panel. As with most of these Windows driverless mini-DACs, the maximum samplerate is 24/96 and no DSD support (like the Dragonfly - I assume the Geek Out will require drivers for DSD and >24/96). IMO this is not a problem. More important than DSD and 24/192 for me is that it can handle 88kHz natively for my DSD-to-PCM conversions (which this does).

There is one other way to get a bit-perfect stream without ASIO in Windows - use WASAPI (Windows Vista onwards). The foobar WASAPI component worked well for me and I used that for all the measurements with the Windows machines. WASAPI will do the samplerate switch automatically with this device.

I. Objective Measurements:

A. The Basics
As usual, let's start with getting some charts and numbers out there for the product so we can get a sense of what we can expect from the sound.

First, the digital oscilloscope displaying the 0dBFS 1kHz square wave (connected to ASUS Taichi Ultrabook):

That's some nice looking square waves! (See Addendum 2 - the shape here looks better due to clipping.) Essentially no overshoot and no ringing. Remarkably precise channel balance which handily beats the ASUS Essence One. Gotta say that I'm quite impressed by this graph since this is "just" a small USB DAC. A healthy 2.83V peak (which would correlate to a 2Vrms sine wave).

Impulse response (16/44) off Asus Taichi Ultrabook:

Standard symmetrical linear phase filter used. Typical of sharp roll-off filters as seen in a previous post. Absolute polarity is maintained.

-90.3dB 16-bit & 24-bit Signal (off ASUS Taichi Ultrabook):
Have a look at the "Waveform Peeping" page for some comparisons. Better-than-90dB dynamic range resolution. Enough resolution to demonstrate the -90.3dB square waveforms created by flipping the least significant bit of a 16-bit signal. It's not as clean/precise as my full-fledged desktop DACs like the Essence One or TEAC UD-501. The waveform appearance is reminiscent of the Squeezebox Touch but better in that there's no DC shift and the channel balance even at such low amplitude is better. Again, not bad at all for USB power off a laptop/ultrabook.

B. RightMark 6.3.0 (newest free version now with ASIO capability)

UPDATE: See the latest update demonstrating clippling at 100% volume with this device! So long as you keep volume setting down a few clicks (~92% in Windows), the following test results apply.

So far so good. It's time to grab some RightMark results to see how it fares in terms of distortion, dynamic range, noise level, etc.

Since this is a portable device which is dependent on the computer for power, I think one of the most important questions to answer is how well it functions between different machines. So far, I have not seen any reports looking at the variability between machines in the objective tests with these small USB DACs. With that goal in mind, I'll be providing results with a number of different computers.

Test machine + AudioEngine D3 --> shielded phono-to-RCA cable --> E-MU 0404USB --> shielded USB --> Win 7 laptop running RightMark 6.3.0

Summary of the machines tested (the laptops were described here in more detail although OS was updated in some cases):

1. ASUS Taichi 21 Ultrabook (early 2013): Intel "Ivy Bridge" i5-3317U (1.7GHz dual), 4GB, Windows 8.1 x64, 128GB mSATA SSD, USB3 port, foobar + WASAPI component

2. Apple MacBook Pro, 17" (early 2008): Intel Core 2 Duo (2.6GHz dual), 6GB, OS X 10.8.2 "Mountain Lion", USB2 port, 240GB SATA SSD, Decibel 1.2.11 player

3. Apple MacBook Pro, 15" (mid 2009): Intel Core 2 Duo (2.26GHz dual), 8GB, OS X 10.9.2 "Mavericks", USB2 port, WD 640GB SATA HD, iTunes 11.1.5 (AIFF files)

4. HTPC: Intel "Haswell" Pentium G3220 (3GHz underclocked to 2.5GHz & undervolted, dual core), 400W Seasonic fanless power supply, 8GB, Windows 8.1 x64, USB3 port, 240GB SATA SSD, foobar + WASAPI component

5. Server: AMD "Trinity" A10-5800K APU (3.8GHz quad), 700W Antec power supply, 16GB, Windows Server 2012 R2, 128GB SSD + bunch of HD's inside! USB3 port, foobar + WASAPI component

6. Workstation: Intel "Ivy Bridge" i7-3770K (3.5GHz quad), 800W Antec power supply, 16GB, Windows 8.1, 240GB SSD + 2 HD's. USB3 port, foobar + WASAPI or + JPlay 5.2B (newest 6/12/2013 trial, Kernel Streaming/ULTRAstream/DirectLink).

Except where stated with the MacBook 15", all the audio files were encoded in FLAC. The D3 was connected either directly to the laptop or the desktop machine's motherboard USB connector (ie. no hub in between).

For those unfamiliar with computer hardware, the above may seem too technical. I just want to make sure I covered the variables thoroughly. Let's just say that this is a pretty decent range of machines from laptops to desktops and the CPUs range from relatively slow by today's standards (Pentium G3220) all the way to a reasonably fast Intel i7. Macs are a bit older as I've transitioned away from Apple in the last few years for work. I focused on USB3 ports where available as the standard these days. I'll throw in both OS X "Mountain Lion" and "Mavericks", Windows Server 2012 R2, as well as the JPlay software in one of the tests using the most extreme settings.

Summary of this USB DAC connected to each computer:

Frequency Response

Noise Level
As expected, these days, any advertised high-resolution DAC should have absolutely no issue with standard resolution 16/44 signals. We're basically looking at ideal and identical results here with this little USB dongle on all the computer hardware platforms. Clearly, from the frequency response, the AKM4396's standard "fast roll-off" filter is being used instead of the "slow roll-off" option for the chip (you can see the effect here with the Transporter).

Now we move to high-resolution:

From the numbers it's looking really good across the board. No surprises with any of the machines and minimal inter-test variability.

Frequency Response

Noise Level


Notice the use of JPlay for one of the measurements with the i7 Workstation setup. As with previous testing, there's no evidence of difference even with what I believe are very high settings - ULTRAstream / DirectLink. Also the MacBook Pro 15" used iTunes and AIFF uncompressed files rather than the standard FLAC with the others - again, no difference.

Here's the stereo crosstalk graph. Remember that this is highly dependent on the analogue interconnect cable used so doesn't really say much about the DAC per se. In this case, I'm using a short 3' length.

C. Jitter
Okay, to finish off, let's see if the J-Test shows any significant sidebands.

Some minor variation in noise between the machines (some of which I'm sure contributed by the E-MU 0404USB) but it's all rather low level and of no concern. Jitter modulation pattern easily visible in the 16-bit test. As usual, these J-Test graphs are meant for analysis of SPDIF interfaces. Asynchronous USB converters almost universally do not demonstrate issues. As I have said before, I doubt jitter is audible these days given how good even basic DACs are in this respect. I would seriously like to see evidence to suggest a need for stuff like "femto clocks" suggesting any difference that would be audible!

II. Comparisons:

One of the fun things about using the same test kit over the last while is that I've got a little database of objective results to make relative comparisons. Here's how the D3 stands compared to my other units (24/96 only, there's no point comparing 16/44):

The first 2 columns are the Audioengine D3 playing off my ASUS Taichi Ultrabook either on AC power or unplugged with the battery. Notice there is no difference. Some folks feel that running off battery power results in less noise; I certainly did not see that here.

A reminder - the better stereo crosstalk values for the D3 are because of shorter analogue interconnect cable (3' vs. 6').

look at the graphs:
Frequency Response

Noise Level



As you can see, the outlier here is the Squeezebox Touch with the "poorest" performance (realize though that the performance is still excellent!). The SB Touch rolls off the bass a little earlier than the others (-1dB by 20Hz) and with the highest noise level.

Although buried in the various overlaid graphs, the D3 noise level is remarkably low around 60Hz (mains fundamental frequency).

The Audioengine D3 fits at an intermediate level between the SB Touch and the rest of the high-performance DACs. The noise level is just a hair above the larger full-sized DACs. There's also more ultrasonic roll-off which I believe is inconsequential.

III. Subjective Evaluation:

I spent about 2 weeks listening with this headphone amp / DAC combination in the evenings while doing some other work. It sounds very good. Headphones used in the evaluation include my JVC HA-FXC51B IEM, Sony MDR-V6, AKG Q701, and Sennheiser HD800. With an output impedance of 10-ohms off the Audioengine D3, it's recommended to have >80-ohm headphones to maintain a relatively flat frequency response. Of the group, the worse impedance match would be the JVC IEM at 16-ohms. Despite this, the sound was quite nice if a bit bass shy with an accentuated mid-range that worked OK for some pop and rock recordings I was listening to (Donald Fagen's The Nightfly, Cat Stevens Tea For The Tillerman). I took the ultrabook, D3 DAC, and these IEMs around one weekend while my son was doing his sports lessons and it did not seem too cumbersome.

The sound was good with the 63-ohm Sony V6. These 'phones have plenty of bass and I had a chance to listen to a few vinyl rips. For example on Falco's Rock Me Amadeus (1985 Canadian Release) vinyl rip, you can hear the limitations of the digital sampling used in those days with an elevated background noise during the vocal overlay in the 1st half of the song. Daft Punk's Random Access Memories vinyl rip sounds very good as well with some surface noise reduction.

As I mentioned above, this little headphone amp can power the 63-ohm AKG Q701's reasonably well but I do push them up to ~80% with softer well recorded albums. For example, Rachel Podger's Guardian Angel (Channel Classics 24/96) sounds fantastic through these headphones. The open design of the AKG's are great for classical music transparency. I also find them very comfortable for longer listening sessions. Timbre and resonance of the violin in the recording space were well rendered... Enough resolution to easily hear the performer's occasional toe tap and breath sounds :-). As a side note, I do not recommend downloading the 24/192 versions off Channel Classics since the material was sourced from DSD64 and converted with Weiss Sararon - there's a steep lowpass filter in place and there's no audio above ~40kHz so 24/96 is more than enough. I also listened to the large choral arrangement of Mahler's Symphony No.8 ("Symphony of a Thousand") (Antoni Wit & Warsaw Philharmonic Orchestra, 2011 Blu-Ray rip). Comfortable listening level for this louder album was at ~65%. Lovely conveyance of the intermix of solo and choral voices; tons of detail to be unearthed in this recent Naxos recording of the 8th. [BTW, the multichannel mix is excellent as well!]

I spent a couple nights listening to the Sennheiser HD800 with this DAC. These are 300-ohm headphones so should be good impedance-wise and due to its sensitivity, can be driven to an uncomfortably loud volume (for me anyway). The thing I appreciate about the HD800 is that they're good at pretty much anything! The bass is tight without accentuation and the highs sound clean and dynamic. I keep wanting to use the term "precise" about the HD800. They can be very unforgiving of poor recordings; especially those with accentuated trebles (it's worth trying out the "Anaxilus mod"). First up was some Guns N' Roses - Appetite For Destruction (1997 MFSL release). Very good definition and detail even with loud and "busy" tracks like "Welcome To The Jungle". Not exactly high-fidelity but can still be used to judge quality of reproduction especially in teasing out the individual parts and making sure the sound doesn't become "muddy". Moving on to another memory of the 1980's, INXS' Kick (2002 Rhino remaster) sounded very good as well... I've always enjoyed the intro to "Devil Inside" with the synth-percussion and electric guitar as we get into Michael Hutchence's moderately reverbed vocals in this "classic" INXS lineup at their peak.

Moving on to the audiophile female vocalist genre... About 13 years ago, I was in Chengdu, China for some travelling when I came across a copy of Cai Qin's (蔡琴) "Golden Voice" (金片子 壹) album. This is one of the best recordings I have heard. If all CD's were recorded like this, I don't know who would even consider the need for "high-resolution". Needless to say, the rendition out through the Audioengine D3 and Sennheiser HD800 was fantastic. Simple instrumentation with the voice beautifully detailed consisting of classic Chinese songs originally written back in the early half of the 20th Century. I suspect many audiophiles would call this DAC/headphone combination "analytical". I just call it brutally honest; what I personally look for in true high-fidelity.

Finally, I hooked this little DAC using phono-to-RCA plugs to the Audioengine A2 powered speakers in the picture above to have a listen... My daughter really wanted to hear the Frozen soundtrack. Yup, sounds good - all that Disney high production value. Alas, although I really enjoy Idina Menzel's "Let It Go", I can never shake off the vision of Elphaba in my head! Overall, it sounds as good as my usual desktop DAC (ASUS Essence One) connected to the Audioengine A2 in terms of resolution and soundstage (within the limits of these small desktop speakers of course). So far I have not hooked this up to my main sound system downstairs (inconvenient re-routing of analogue cables). I suspect it will sound very good as well.

IV. Conclusions:

1. I was pleasantly surprised by the objective measurements! For something this small, the measured results are almost as good as the better DACs I've measured. All this driven off a USB port. Impressive.

2. Great to see that a USB DAC is capable of essentially identical performance off a number of different computer setups. The machines are of various age, make (Intel and AMD), both USB2 and USB3 ports used. Despite all the hoopla around fears of poor power quality or potential of electrical noise, this little device has demonstrated to me that these fears are unnecessary - or at least can be dealt with. It sounds clean, the noise floor is excellent. No evidence of jitter issues with the asynchronous USB interface.

3. Following from item 2, although previously demonstrated, again in these tests there were no significant differences in the sonic output with this DAC between computers running different OSes (Windows 8.1, Server 2012 R2, OS X "Mountain Lion", "Mavericks"), and playback software (foobar, JPlay, iTunes, Decibel). Likewise, I remain unconvinced that jitter is affected by anything other than the hardware interface itself. A good DAC and bit perfect is all that is necessary for quality playback as far as I can tell these days.

4. Specifically regarding the Audioengine D3, remember to keep the fabric pouch handy so as to avoid scratching other things packed with this stick due to the aluminum construction and sharp angles. Also, it does get warm after awhile. However, on the plus side, the construction is very solid! It might not have the color-changing Dragonfly, but the little blue LED is good enough to know when sampling rates have gone >48kHz. At 2.5" long (slightly longer than the Dragonfly), it does stick out from the computer a little bit but is generally unobtrusive.

5. Yes. Subjectively, the Audioengine D3 sounds very good and has enough power to drive my AKG Q701 - this is better than my TEAC UD-501. Most of my subjective headphone listening was done with the ASUS Taichi Ultrabook and desktop i7 workstation. I did not notice any difference in sound despite the different underlying computer hardware.

I think the proliferation of small high performance USB DAC / headphone amps is a good thing. I appreciate the recent Tom's Hardware article on computer audio suggesting that there's generally no need for anything better than the ubiquitous sound chip in the computer. That's probably also true. There's generally no need for better sound especially on-the-go where the ambient noise is high. However, "native" sonic capabilities of laptops can vary as I previously demonstrated. What a small device like this brings to the table are more power for demanding headphones and uniformly good sound quality off the USB port for whatever computer one owns. When I'm on the road with a good pair of headphones, this device will do nicely for listening in the quiet evening in a hotel...

Time will tell whether mini USB DACs like these will maintain much popularity however. For audiophiles it's not all that inconvenient to carry or plug in, but I'm not sure if this would continue to capture the interest of the mainstream market (assuming the interest in devices like the Dragonfly and Geek Out represents mainstream acceptance). In time, it's possible that laptops will on average achieve better sound quality which would negate the need for this class of device. We've already seen quality improvement with support of high sample rates and 24-bits. Remember the days of noisy audio where you can hear the electrical interference from spinning disk drives or busy CPUs? Or when everything was resampled to 48kHz? Thank goodness standards have improved.

I hope this review provides a good glimpse into how the Audioengine D3 measures and sounds. Certainly it has given me some perspective into this class of device as new ones come out. I 'hear' that the Geek Out is shipping in small quantities. I like that it has 2 outputs (one with <1-ohm impedance) and can deliver more power. Excitement is certainly there and things can get a little silly when this happens (as witnessed by this hilarious video). For the more powerful 1W model, I wonder what heat production and battery drain will be like with portable devices (supposedly Class A?). I'm most curious about how the crossfeed algorithm sounds. To date I have not found a headphone crossfeed DSP I've ever really liked. To call the feature "Awesomifier" I think could lead to the butt of many jokes if it actually doesn't end up "awesomifying" anything. Also, I saw on a video that it runs in non-oversampling mode which is certainly possible given the use of the PCM1795 DAC like I showed in these TEAC UD-501 measurements. I don't know if running this DAC in NOS mode (digital filters off) is wise for something like this. We should be seeing some reviews soon I hope...

OK, enough audio geekiness for now! It's Spring Break for the kids. Time for family R&R in the week ahead. :-)

Enjoy the tunes dear readers...

Around Christmas time I wrote a little opinion piece on the next gen video game machines and how I decided to just stick with the PC. Looks like indeed some of the new multiplatform games are probably best played on good ol' Windows... Wasted a few hours on this over the last few days :-)

Big push for Titanfall on the XBOX One but I think it's pretty clear that the Windows version has better visuals (true 1080P), better frame rates, and excellent gameplay even on a modestly powered machine.

Addendum 2:
Managed to whip out the oscilloscope for a quick look...
-3dBFS 1kHz square wave, 44kHz
Slight Gibbs undershoot and overshoot seen, so clipping indeed likely contributing to the idealized appearance of the 0dBFS plot above. Nonetheless, still looks really clean and balanced compared to others...

Addendum: (January 30, 2016)
Just did my own measurements of output impedance using 1kHz signal and 20-ohm load, maximum volume 92% to avoid clipping. I get an effective output impedance of 11.5-ohms. Close to the rated 10-ohm spec.

Tuesday, 18 March 2014

MUSINGS: "The CD can be said to be an analog disc, just like the LP" (UHF Magazine, 2014)

(On occasion, I will refer to copyright works under the principle of "fair use" for the purpose of commentary. This is one of those posts.)

I don't get a chance to hang out at the local large bookstore chain (Chapters) very often. When I do, I will wander over to the magazine rack and see what's on offer for the month. Here in Canada, there is one (and I think the only) magazine catered to audiophiles published in the country - UHF Magazine (Ultra High Fidelity) from Montreal. Here is Issue #94 (March? 2014):

So it was with interest to see an article on "A critical look at the Compact Disc for music delivery" referred to on the front cover written by Mr. Paul Bergman titled "Inside The Compact Disc". I figure it might be interesting to scan the article quickly to see if there's anything new to be gleaned... Alas, within a couple of minutes of scanning page 25, this paragraph just jumped out at me from the article:
To save space, individual binary digits are not stored. The length of a pit corresponds to the number of "one" samples before a "zero" is encountered. The length of a pit or land is of course an analog value, and for that reason the CD can be said to be an analog disc, just like an LP.
Wow... Ponderous man... So a CD is a material thing. All material properties have variability - the length of a "land", the depth of a "pit", the transition between "lands" and "pits". So what? As long as it's within the margin of error within specification, a one is a one and a zero is a zero. No problem. It's not like the length of a pit is going to end up with any other interpretation than either 1 or 0 by the decoding circuitry. It's binary; there is no 0.5. It's digital. The author even talks about the use of Reed-Solomon coding for error correction and mentions interpolation when errors are detected.

Here's another nugget from the same article (p. 26):
Though the CD is not, as I have explained, a purely digital medium, some true digital media do exist. Reference Recordings, for instance, uses a data DVD to distribute its 24-bit 176.4kHz music files...
So data DVDs are "pure digital" and (audio) CDs are not? Does the author not know that "lands" and "pits" also underlie the physical structure of a DVD as well (higher track density of course)?

I assume that the author would consider CD-ROM "pure digital" since it too contains data (as if digital audio is not data) and DVD evolved from it?

Presumably the author doesn't understand that the only difference between the "Red Book" (1980) CD format and CD-ROM "Yellow Book" (1988) is in the data structure. The older Red Book format is much simpler - an inner lead-in segment which contains the TOC (Table of Contents) that gets read when you first insert the disk. When you select the track, the player will look for the time code corresponding to the start of the track and start playing music through to the end of the track sequentially. (The data stored in the TOC is contained in the ".cue" sheets in CD rips.)

Yellow Book and all that followed, leading to the DVD-Books and modern Blu-Ray Disc formats have a much more complicated structure which contains an actual file system; typically ISO9660 or UDF. This adds the extra layer of complexity and features like file names, permissions, directories, opportunity to append/delete/move for rewritable media, etc... Error correction is also much better at the expense of some storage space. I can only assume the author of that article somehow associates these characteristics with "pure digital".

How strange... In 2014, a "technology" magazine would even declare that the Audio CD is somehow not a "pure" or "true" digital medium. I guess the editors must have missed this! Hopefully the readership can send a few "Letters to the Editor" to set this straight.

Well, it's only been about 32 years since the reflective polycarbonate disks came out...

Perhaps the author is taking to heart the title "The Analogue Compact Disc" from Robert Harley writing for Stereophile a little too strongly! That article was from 1994 and speculated about jitter rather than questioning the digital-ness of an audio CD. Even in that article, I'm concerned about this declaration: "It's becoming incontrovertible that CDs containing the same 1s and 0s produce varying levels of sound quality." Incontrovertible. Really?

Friday, 14 March 2014

Recalled Article on Vinyl Rips and High DR...

Hey guys, I "recalled" an article I posted this AM on why DR increases with vinyl rips based on RIAA EQ processing.

The reason being I think small changes like +2-4dB of DR in compressed recordings using the RIAA EQ are very much based on the resolution of the emphasis and de-emphasis filters. JR_Audio made a comment on this and I had a look at the accuracy of the RIAA curves I made. Indeed, mathematically essentially perfect curves <+/-0.1dB add very little DR extension, maybe only 1dB whereas "looser" accuracy like +/-0.5dB through the audio band could give +2dB with the DR Meter with some idiosyncrasies depending on the song and which frequencies the RIAA inaccuracies lie.

The problem is that these are simulations done on a computer with accuracy of 32-bits. As I said in the original article:
In real life, things are more complicated. Even without a significantly better source master, for the final vinyl mix, small changes can be done to the signal such as mixing all the bass frequencies up to 100Hz into mono or taming of the high notes along with the RIAA curve. The results might be more pleasing or euphonic (see this article on mixing for vinyl and how much work potentially needs to be done). On playback, other physical factors are also involved such as the tracking force of the cartridge/tonearm or idiosyncrasies of the cartridge and accuracy of the phono pre-amp. Also, post-processing such as noise/click/pop reduction would add another variable for the results from "needle drops". If you look at a number of different vinyl rips, it's quite common to see slight DR variation depending on the equipment and technique used (typically another DR +/-1dB is not uncommon).

Indeed, I noticed this afternoon playing with Audition just how easy it was to artificially inflate DR values! For example, mixing all low frequencies <100Hz as mono. I played around with this using Lorde's Royals song using a lowpass filter to isolate those low frequencies and then mix them back into the highpassed upper portions as mono... Even though the sound wasn't significantly different when volume matched, it was not difficult at all to get a DR8 original 24/48 HDTracks version up to DR13. As I noted in the original article, I believe this is just the result of removing peak limits and clipped portions, allowing the calculations to extend these portions with the DSP operations.

Given the effect I saw, ultimately I think there's no conclusion one can draw between what's measured on a vinyl rip and knowledge/proof of whether truly new masters are being used unless specifically told about it!

I guess after trying this for awhile, I realized just how sensitive the DR Meter can be. Ultimately it is useful as a tool to explore the average dynamic range of albums and especially pick out very poor masters with strong peak limiting. It's also useful to determine if 2 pressings are exactly the same.

Anyhow... I might review this later if anything clears up for me in the days ahead; just really hard to post something unless good conclusions can be reached.

I'll leave you with the last bit of the post however...

Parting example:
One example of a vinyl remaster that's truly the "definitive edition" is Stadium Arcadium by the Red Hot Chili Peppers. As usual for Vlado Meller the CD's mastering engineer, the dynamic range compressor was stuck at "11" resulting in a sick album with an average DR5. The vinyl release had proper mastering treatment by Steve Hoffman resulting in a DR12 average. That's the kind of difference one wants to see between CD and vinyl to be sure of truly better vinyl source material.

Have a good weekend everyone. Now go enjoy the music and the weekend... Careful about the Pono droppings this week. :-)

Speaking of Pono, check out all the new "limited edition" Pono players you can get on Kickstarter today with one's favourite band named on the unit! Wow, looks like they're going all out on moving as many virtual units as possible sight-unseen except for a few pictures and ear-unheard... It'll be interesting to see what the final product reception will be like when shipped. (I'm also very curious about the objective measurements!)

Friday, 7 March 2014

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

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

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


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

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

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

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


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

Frequency Response
Noise Level

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


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

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

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

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

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

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

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


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

Enjoy the music everyone...

Sunday, 2 March 2014

FOLLOW-UP: Anomalies in Beck's "Morning Phase" (HDTracks 24/96).

After I posted the last note on "High Resolution Audio Expectations", I got a few "tips" to have a look at some tracks on Morning Phase in more detail... So I got my friend who informed me about the DR6 from HDTracks to send me some pictures:

Track 4 - "Say Goodbye"
Spectral frequency display:

Looks like there's almost nothing after 22kHz but low level noise. Essentially a 44kHz "upsample" for many of the notes. Actually, my friend says a number of tracks he looked at is like this; for example tracks 3 and 5 also (I don't think he checked every track). Since this is a multi-tracked recording with synthesizers and various studio effects, this is actually not surprising - many synth/pop/rock albums are like this. Many samplers and DSPs operate in the 44/48kHz domain so what's laid down is "limited" and there's just no 'genuine' 96kHz sound available.

Track 10 - "Phase"
Now this is interesting:

Track 11 - "Turn Away"
And this:

My word... It looks like tracks 10 & 11 are sourced from some kind of lossy original! Notice the characteristic low-pass from about 16kHz! Might as well be the output from LAME 320kbps (on second thought, 320kbps usually retains up to a full 20kHz with the psychoacoustic model so we're likely looking at 192-256kbps).

Now I'm definitely going to be purchasing the CD rather than any high-resolution download (assuming I want it... Haven't listened to any samples yet). I don't know if folks have checked the CD or if the Qobuz version is any better.

Seriously, HDTracks... Do you guys ever look at these files for quality control purposes before declaring the album fit for high-resolution and charging folks $18USD? I know you claim to just sell what the label gives you, but isn't it a bit disingenuous to be calling much of this album "Audiophile 96kHz/24bit"?

ADDENDUM (2014-03-05):
Given the publicity Beck's album received a week ago and now the revelation of the quality issue, I do hope this serves as a meaningful wake-up call to HDTracks specifically and the audiophile download community as a whole.

If HDTracks sticks its head in the sand on this, it would be telling that their commitment is clearly not to their customers and they certainly cannot maintain any quality control to assure their product is "HD" in any sense of the acronym. Upsampled 44/48kHz is bad enough, but lossy sold as "HD" is just ludicrous!

Thankfully, I did not buy this album from HDTracks... I suppose if I did, I'd be asking for a refund ASAP or at least demanding a higher quality download.

Saturday, 1 March 2014

MUSINGS: High-Resolution Audio (HRA) Expectations (A Critical Review)...

I guess an LP can still be considered "high fidelity" by some in 2014... But doubtful it should ever be called "high-resolution"!

Like DSD / SACD, high-resolution PCM (24-bit, 88kHz+) in the form of DVD-V (up to 24/96) and DVD-A has been widely available for more than 10 years already ("rebadged" recently as HRA for "High Resolution Audio"). DVD-V's with 24/96 audio tracks could be easily ripped back in the early 2000's, and by early 2007, the DVD-A copy protection was overcome allowing easy DVD-A ripping and evaluation of the sonic data up to 24/192 2.0 and 24/96 5.1.

In 2010, out of curiosity, I ran a little foobar ABX trial using the equipment I had back then to see if I could tell the difference between 24/96 and 16/44 and posted this on Audio Asylum. The conclusion... I didn't think I could. (I've included that post below as Appendix A for completeness.)

As you know from previous posts, I'm not a DSD fanboy. It does have some limitations compared to PCM; the technological limitations themselves in terms of high frequency noise due to noise shaping, non-uniform noise floor as a result, lack of opportunity to run DSP algorithms, and the current file format implementations make it cumbersome. In business, the opportunity to differentiate oneself from another provides the opportunity to sell the item as "new", "different" and "better" and indeed DSD provides many talking points and sales opportunities. As I wrote in the PCM-to-DSD article, there are certainly some audible differences DSD processing imparts and this change can be perceived as euphonic even though the actual underlying resolution is no better. It's a bit of a philosophical point: is it better to listen to something played back accurately? Or should we aim for euphonia even though it clearly adds something (distortion) to the signal that was not there originally? In the case of the PCM-to-DSD algorithm, clearly ultrasonic frequencies are being added to the signal as demonstrated by the measurements. As I have opined a number of times in the past, my preference is for accuracy; this is my definition of high fidelity and I generally feel that PCM retains accuracy better than DSD64 and with DSD128 obviously capable of better accuracy than DSD64. (Of course, if a recording began life as DSD and wasn't tampered with, that's as good as it gets and no point losing precision going to PCM...)

Let's consider in this post what it would take to overcome the "limits" of 16/44 in a home component system. Remember that headphones are a much simpler case and for a lot less money, one should be able to better speaker systems - but of course the presentation of soundstage would be much different. (In a way, this post will be similar to a previous one I linked to in What Hi-Fi? but I hope to be more thorough and more realistically critical of the whole endeavour.)

I. The High-Resolution Hardware Needed.

A. Good enough DAC?
Let's start with the DAC since unless it can reproduce analogue waveforms of high dynamic range and the full frequency spectrum, there's no point proceeding. I have already posted measurements throughout this site and demonstrated these qualities in many DACs already. I do not believe it is difficult to pick out DACs which measure well. A DAC capable of >16-bit dynamic range is not hard to find; almost any decent unit today should do (even the electronics-packed Squeezebox Touch from years back can do this). Furthermore, it doesn't have to be expensive; that old AUNE X1 DAC (<$200) from at least a couple years back can easily convey the measurable benefits of hi-resolution PCM up to 24/192. If you have a look at the Stereophile measurements page for modern DACs, it's easy to see that the state-of-the-art DACs all essentially measure in an ideal fashion these days for 16-bits with the best achieving a noise floor around 21-bits with high-resolution material. Talk about 32/48/64-bits and such is nonsense in terms of DAC output resolution. High bit-depth would be beneficial for accuracy of complex DSP calculations like in the studio environment with numerous Pro Tools effects. Although I haven't heard all the "best" DACs, I have heard a number of good ones and feel that differences are minor and most likely inaudible in volume-controlled blind testing despite reviewers' comments. Measurements can show slight differences with impulse response, the occasional jitter difference. No biggie.

B. Good enough pre-amp, amp, speakers?
If you look at the measurements in Stereophile under the pre-amp section, it's not unexpected that vacuum tube preamps generally measure poorer in terms of unweighted audioband SNR; typically around 75-85dB with a 1V signal including some very expensive models. In comparison, good solid state preamps should be able to achieve around 100+dB with a similar test. In general, you'll find this difference with any vacuum tube vs. solid state comparison... As per the philosophical question posed above, vacuum tubes are less accurate (older obsolete technology) but can be euphonic (and nostalgic) for some people.

In my system, I've already shown that the Emotiva XSP-1 pre-amp and Onkyo TX-NR1009 receiver are capable of passing through an analogue signal of easily >100dB dynamic range from the TEAC UD-501 DAC. Unfortunately the old Denon AVR-3802 was incapable of this, therefore, it would not be a device I'd use in the high resolution audio chain. Measurements are needed to know whether the gear is good enough for high-resolution audio.

The amplifier and speakers are even harder to quantify... For high resolution dynamics, we need the amplifier to be able to produce enough power to drive the speaker to create >96dB dynamics in the room! Likewise the speaker will need to be able to handle the power to recreate the full dynamic range without distorting to a significant degree in order to benefit from the resolution afforded by >16-bits! This is tough.

That's just the dynamic range. What about the frequency response from a speaker? Assuming that ultrasonics have some kind of beneficial effect (dubious), most excellent speakers can reproduce frequencies beyond 20kHz. Few speakers have a "super tweeter" to reproduce those frequencies up to 40+kHz flat however. In fact, many listeners would prefer a slightly rolled off response by 20kHz as per the Brüel and Kjær "house curve". I know I can't hear much above 16kHz so would find it hard to feel any need to spend money on accurate ultrasonic frequency reproduction.

Success in conveying the high resolution audio signal also depends on...

C. Good enough room?
It goes without saying that to achieve a dynamic range over 96dB, we need a quiet room; the main point of my previous post about the importance of "silence". Assuming you have amps and speakers capable of it, to better the dynamic range afforded by the 16-bit CD format in a very quiet 20dB(A) listening room, we need to be able to achieve dynamic transients >116dB SPL. In my stereo system consisting of 250W (continuous power) monoblocks feeding quite sensitive 92dB/W/m speakers, placed near a wall, the theoretical maximum SPL at my sweet spot 11-feet away is "only" 111.5dB with the amp already contributing 0.05% distortion (check out the Collins' Cinema SPL calculator to do your own calculations). Accounting for some headroom, I can maybe get up to 115dB for dynamic transients in music. Remember, due to logarithmic properties, even if I doubled the amplifier power to 500W, the SPL would only increase by 3dB to 114.5dB (maybe getting close to 118dB transient peaks). I suspect the speakers would be significantly distorting the sound at these loud volumes, not to mention we're well into hearing damage levels with repeated exposure.

Furthermore, in-room frequency response, reverberation time, speaker placement effects are also significant... I believe all of this would be of greater effect than what is afforded by the extra resolution.

As you can see, we've got a problem here already from the hardware setup perspective in arguing for high resolution audio of >16-bit dynamic range and >22kHz frequency response.

[Note that I haven't even mentioned dithering and noise-shaping increases the perceived 16-bit signal's dynamic range even further at reasonable volumes, so in fact, it's not as "easy" as the 96dB quoted above. Read this iZotope Ozone guide for more details.]

D. Good enough ears/brain?
Serious golden ears (platinum ears?) are mandatory I believe :-).


II. The High-Resolution Software Needed.

When we buy our music, how do we know that all the complex bits and pieces were done well enough to ensure quality? Furthermore, when we go on a web site like HDTracks, Qobuz, 2L, Channel Classics, etc... how do we know it's worth downloading 24/48, 24/88, 24/96, 24/175, 24/192, DXD, DSD64, DSD128?

We don't.

Sadly, I think the state of affairs right now for many audiophiles (promoted by music web sites, distributors and those with financial/advertising revenue interests) is akin to the megapixel race in digital cameras. Thankfully, that megapixel race seems to have died down significantly as consumers have gotten wiser to the fact that quality of each pixel is important. The JPEG image from a cellphone's 12 megapixels is way inferior to that from an SLR with nice lens at 12 megapixels (even though JPEG itself is lossy)! But in audio, there's still the impression that bigger numbers are somehow supposed to be better without qualifying where the source comes from. 192kHz is better than 96kHz. DSD, the "super audio" format beginning at 2.8MHz sampling rate is believed to be even better (by some).  I have yet to see a professional reviewer prefer the sound of a 24/96 over 24/192 when both versions are available - is it because he has hearing higher than 48kHz? Do all DACs sound better running at 192kHz? Or is it just bias from the numbers game? (I suggest it's just the latter.)

Of course, in real life it's never as simple as bigger numbers being better... As consumers we usually have no insight into the hardware used like the microphones or how the ADC was accomplished (remember the technical limits of the studio equipment!). Complex studio "processing" for multi-tracked projects, and the myriad of DSP plug-ins require expert sound engineers to ensure quality with each step. I would be remiss to bring up concerns around decisions to use dynamic range compression (eg. Loudness War) which could have artistic merit but which also lowers the ultimate fidelity of the original recorded signal and diminishes the number of bits of dynamic range required to fully encode the sound. As a result, many audiophiles including myself would regard some of the first CD releases back in the 1980's to about mid-1990's as superior to subsequent remasters using heavy handed processing despite the fact that ADCs back in those days would have been inferior. For example just look at the mess UMG did in 2010 with the Rolling Stones remasters among many recent cases.

To some extent we could make decisions based on the reputation of certain companies. Audiophile remastered releases by Analogue Productions, Audio Fidelity, Mobile Fidelity can generally be accepted as the best remasterings using better equipment and maintaining higher standards of audio quality whether it's standard CD, SACD, or potentially high-resolution PCM. Some mastering engineers like Barry Diament, Kevin Gray, and Steve Hoffman are among those who have made a name for themselves as mastering to a higher audiophile standard (as opposed to this guy and his ruinous work on RHCP's Californication and I'm With You). Realize of course that remastering demands that the original source material to be of high resolution quality if one is to show the benefits of the high-resolution format over CD - this is debatable for even the best analogue tape. Likewise, certain record labels like 2L, AIX/iTrax, Channel Classics, Naim, Blue Coast, Reference Recordings produce superb sounding original recordings catered to the audiophile segment with a strong reputation to maintain.

However, we are witnessing the rise of the high-resolution internet store fronts. Places like HDTracks, Qobuz, Acoustic Sounds, Native DSD Music, Linn seem to be resellers of high-resolution format files (PCM and/or DSD) from some of the major/larger record labels. Quality control issues have been discussed over the years as obvious "upsampling" (taking a standard 44kHz and resampling it to 96kHz to sell for example) have shown up on places like HDTracks which really does a disservice to the music buying public. Like the list of SACD's appearing to be nothing more than upsampled PCM, we do not know the origin of many of these "high resolution" albums. I've certainly run into my share of questionable releases that appear to be blatant standard resolution upsampled music or looking like standard 44/48kHz PCM data run through an analogue board and re-recorded with low-level high-frequency noise on a 24/192 release. The former is easy to spot, but the latter is hard to prove!

As audiophiles and audio enthusiasts, we can of course look beyond just the hyped numbers and acronyms like 24-bits, 96kHz, DSD... I suggest doing a search on places like the new Computer Audiophile Music Review segment (good job Chris on the new feature) and see if folks have "measured" the release for the following characteristics to look for a "true" high resolution master as best we can as consumers:

A. Absence of steep low pass 'filtering' suggesting upsampling.
Good examples are all over my list of suspected upsampled SACDs. Note that in the case of DSD, since there's rising high frequency noise, the discontinuity exists as a steep cliff around 22kHz. You can also see a good example of this in the HDTracks release of Graceland 24/96 as per this thread on Computer Audiophile. Because upsampling of PCM is done within the digital domain, it results in the "brick wall" abrupt transition like this:
Blatant example of 24/96 upsampled to 24/192 using "Blue Train", John Coltrane. No frequencies above 48kHz at all...
Since this is too obvious, and analogue tape always has high frequency noise, let's just add a little bit of low-level -100dB white noise above 48kHz which would be inaudible:
Upsampled track with -100dB spatialized white noise mixed in to look like there's something >48kHz.
There, you'd never know this was just an upsample by looking at the spectrum (but can perhaps guess if looking at the spectrum in realtime playback).

Here's what that track in true 24/192 looks like, but as shown above, it's really easy to conceal a 24/96 upsample:
The real 24/192 "Blue Train" off the 2001 Classic Records HDAD release.
On a side note, one experiment to try is to take a high pass filter of everything above 30kHz on a 24/192 file and bring it down in tone into the audible range and have a listen... I rarely hear anything correlating with the underlying music - this is a big reason why I rarely bother with 24/192.

B. High average dynamic range.
What is the point of going above 16-bits if the mastering is compressed to hell and back? The opportunity to easily measure the DR "Dynamic Range" value using the foobar plugin (use the newer version found in the DR Database) has been extremely useful. This value is calculated by looking at the difference between peak volume vs. RMS "average" volume over chunks of time throughout the music. A higher number correlates to the presence of loudness peaks which of course corresponds to a more dynamic sound. Real life sounds dynamic! You can also check the Dynamic Range Database before purchasing to see if the title is in there.

Although there is no simple rule about this, the DR value can give you insight into whether the high resolution version of an album is the same mastering as the CD release and what difference exists. Consider the example of Lorde's Pure Heroine...

CD Release:

Here's HDTracks 24/48 "Studio Master":

In this example, the HDTracks release looks (thankfully) a little bit better than the CD - this is actually somewhat rare with modern pop/rock recordings since I find the majority are the exact same master with the same DR value. Generally we're seeing a 1dB improvement in average dynamic range with the high-res version. The problem is, we're still at a DR of 7dB only! Furthermore, there are no nuances to the peaks with every track pushed up and limited to 0dB (not really an issue here since the music is clearly synthetic but this would be very bad to see with acoustic music or orchestral music like some soundtracks such as the recent DR7 The Dark Knight Rises).

For perspective, back in the late 1980's and early 1990's, the average pop and rock album had average dynamic range around 10-13dB. A well recorded classical or jazz album with all the dynamic transients that come with natural recordings should average around 13-16dB.

Personally, my audio "New Year's resolution" for 2014 is to avoided purchasing *any* high-resolution album without at least a value of DR10; and I feel this is very conservative already (I probably should advance this to DR12). The rationale is simple...  Recordings with low dynamic range and peaks pushed to ~0dB are loud and do not require us to turn up the volume to listen at reasonable levels - typically I play DR7 tracks at something like -35dB through my system. This type of recording does not utilize anywhere near the limits of a 16-bit medium like the CD nor that provided by my home audio system, much less demand the 24-bit dynamic range of high-resolution downloads.

As an aside, be very cautious in using the DR Meter for vinyl rips! Just because the values tend to be higher doesn't mean that it's due to the LP having better mastering. Have at look at Ian Shepherd's video here (essential viewing IMO):

In that example, with pops and clicks removed, even though the DR score is higher with vinyl, the original source is the same and we can speculate in a later post as to why this might be the case. Remember that vinyl rips remain plagued with the limitations of the LP source including surface noise, subtle wow/flutter, inner groove distortion, idiosyncrasies of the playback gear, etc. They can and do still sound great especially with some judicious post-processing, but not true high resolution like a good quality pure digital recording.

Nonetheless we see on DR Database someone's vinyl rip of Pure Heroine:

DR10... Not sure if that is an actual better master or just the effect of transfer to vinyl (as per Ian Shepherd's example). With only a 3dB increase compared to the HDTracks version, I do not believe we're looking at a true remaster effort here.

III. Conclusion - Expectations for High-Resolution Audio?

So, let's try to answer the question posed above of what to expect from high resolution downloads in conclusion... I think the answer is simple: Not much if anything compared to a technically good 16/44 version or CD of the same mastering.

From a hardware perspective, digital audio technology has advanced to the point of easily overcoming the 16/44 "limitations" within the DACs. The problem revolves around the ability of the analogue side (especially the speakers and room) and ultimately, the perceptual limitation of being human. Apart from listening in unreasonable ways (eg. listening to fade outs at extreme volume levels comparing undithered 16-bits vs. 24-bits), I don't know of any controlled study suggesting an audible difference between standard and high-resolution music.

From the software perspective, we are faced with  just how well the recording was done and mastered in the first place. I would gladly listen to an MP3 192kbps of a well recorded/mastered album than a poor DSD128 or 24/192. For folks to place an emphasis on the value of 24-bits, 88+kHz, and DSD appears to be ultimately a fallacy. The most egregious examples of these are the pseudo-high-res albums like the upsampled SACDs or upsampled PCMs sold under the high resolution moniker. But as Mark Waldrep ("Dr. AIX") has been talking about for years, true high-resolution demands that the source be high resolution... The hardware reproduction challenges are difficult enough, but if the source is anything other than a pristine (digital) production, the potential resolution of 24/96 and above would never be utilized (assuming anyone can even hear it!).

I often criticize Neil Young and his comments on these pages (and I see many others do as well) simply because he hypes up the 24/192 number in a way personifying the "megapixel race" of the audio world (I'm sure he's a decent fellow and all, and I certainly respect his artistry... But just watch this cringe-worthy "Dive Into Media" video from 2012 to see what I mean.). I think it's quite clear that unless his Pono brings something truly new like getting the record labels to provide good quality, dynamic, remasters, there is essentially no hope of Pono hi-res sounding any different than what we have so far with HDTracks or Qobuz. The 16-bit, 44kHz resolution "container" was never a significant "problem".

In the years since my post in 2010 (Appendix A), my opinion really (perhaps to my surprise) has not changed. Even with better equipment like newer DACs, better headphones (Sennheiser HD800), a much improved room, separate components, better speakers. Taken together - the challenges of hardware and software - I suspect that high-resolution PCM (like DSD) doesn't have much future in "making money" for the companies other than as it is today... Niche products with a small target audience. Unless the software companies somehow decide that the default digital resolution is greater than 16/44 and provides it as such, I don't see why the average person would pay more for what essentially is of little (if any) benefit in the consumer environment. The thought of making high-resolution lossless downloads default is unlikely given that the majority still just buys music from iTunes which isn't even lossless to begin with!

Now on a personal note, I am mindful that hobbies in general are not meant to be rational pursuits; they're emotional endeavors for the sake of pleasure and fulfilment of some form. We (audiophiles) probably spend more time than needed obsessing over all manners of "trivial" things from what to buy to whether a cable is "good enough". Most people in this world would look upon the perseverative audiophile as a form of obsessive-compulsive neurotic but so be it. So are car collectors, stamp collectors, wine connoisseurs, watch aficionados, etc. In this context as an emotional pursuit, I'm actually OK with still collecting my favourite music in 24/96 or DSD if that's "as good as it gets". It is about perfectionism. There still could be something wrong with brickwalling at 22kHz and 16-bits do not challenge the objective capabilities of my DAC - perhaps some day I'll hear the difference assuming there are some merits to the recording being "high-resolution". Nonetheless, to buy a recording which is just upsampled or has severely squashed dynamics offering no potential benefits at all to the collector is just plain foolish. As a "more objective" audiophile, there are limits to perfectionism which I would be unwilling to cross based on the scientific data.

For those who insist that they can hear high-resolution audio, I would love to see comments below... I would highly encourage doing a controlled test like the ABX below for yourself with your own high-resolution music. Let me know if you ABX successfully!

Recently, I've been listening to some Larry Carlton. Excellent stuff especially some of the older recordings from the early/mid 1980's - have a listen to Discovery from 1987. No argument from me that 16/44 sounds fantastic when done well!

Audio low point of the week? A friend notified me that Beck's recent album Morning Phase scores DR6 from the HDTracks 24/96 "Studio Master" (I see the Computer Audiophile just put up an article on this). Same as CD. As per my New Year's resolution above, this is one album I'll just pick up at the local CD vendor for $10.

As usual... Enjoy the music. I've promised my wife I won't be buying another copy of "Kind Of Blue" - even in High-Resolution Audio 32/384 :-).


APPENDIX A: Old post from Audio Asylum in 2010

Test with 24/96 vs. 16/44 LITTLE TO NO DIFFERENCE.
As a "data point" and discussion for those who have been singing the praises of hi-res audio downloads, I did a test recently.
Recently got the very versatile E-Mu 0404 USB (AKM AK4396 DAC) to play around with on my computer (quad core 2.8GHz, 8G RAM, low DCP latency, Win7, through USB of course). With some recent high definition downloads / DVD-A source:

Rebecca Pidgeon - The Raven: "Spanish Harlem" (24/88 Bob Katz 15th Anniversary Ed)

Carol Kidd - Dreamsville: "When I Dream (2008)" (24/96 Linn Studio Master)

Laurence Juber - Guitar Noir: "Guitar Noir" (24/96 AIX DVD-A rip)

Took these FLAC/WAV files, down sampled in Adobe Audition to 16/44 (no dither, no noise shaping) then resampled back up to 24/96. Verified that frequencies all truncated to 22kHz. Then listened to them with Foobar 2000 ABX comparator using the E-Mu ASIO output plugin. This allows me to A-B on-the-fly and do some "blind" ABX'ing.
Listened with headphones: Audio Technica ATH-M50, Etymotics ER-4B.

With this setup, I figure I've removed all variables except for sample rate change - same mastering, same DAC running at same sample rate.
Results: Essentially NO DIFFERENCE between the native 24/96(88) and 16/44. Blind ABX results NO SIGNIFICANT DIFFERENCE. When I do the rapid A-B switch in the middle of a song, I thought there MAY have been slightly more smoothness/openness in the high-def version but this could just be placebo and the improvement was MAYBE 5%.

At 38 years old, very few loud concert experiences, I don't think I have 'tin ears' (hey my wife thinks I have better ability to pick out music in noisy environments so I guess it's at least as good as some females :-). 
My conclusions:
1. Either my equipment sucks or these samples suck and there's alot more but I need to fork up more $$$$.
2. Or high-def cannot be well appreciated with headphones.
3. Or the upsampling back from 16/44 --> 24/96 somehow reconstitutes the sound.
4. Or, there's really not much difference.
5. At this point I'd probably spend a few more dollars to buy a high-def download (maybe at most $5-10 more if it's something I like) when given the option but not expect significantly more revelation in the sound. 
I've listened to good SACD as well and like them but there's no way to do tests like this. I didn't bother with 24/192 material since I figured most improvement should come from this first step up 44 --> 96. Anyone else done such tests for themselves?

For those interested in the audibility of high resolution audio, consider reviewing the results of the 24-bit vs. 16-bit Blind Test conducted shortly after this article was written.

Also potentially of interest: A few words about Pono... And what is "the finest digital copy"? (when John Hamm was still CEO... if only for a few months)