INTERNET TEST: 24-bit vs. 16-bit Audio - Can you hear the difference?

24-bits vs. 16-bits Audio - A Visual Analogy?

Those that have been reading this blog for awhile will recall that these pages started with an MP3 vs. Lossless test that was posted here back in 2012. That was kinda fun :-).

As you know, the "High-Resolution Audio" movement is on. A major cornerstone of this is the belief that in the PCM world, 24-bit audio resolution imparts clear audible benefits (as opposed to the standard 16-bits for CD resolution). Indeed, most decent DACs these days are capable of measuring >16-bit dynamic range. Clearly, a push is being made for the subjective "virtues" of 24-bit audio for your headphones, into your home and into your car. The 8-bits of difference between 16-bits and 24-bits of PCM data provides 48dB of extra dynamic range or 256x the number of values to represent each PCM sample! Fantastic advertising buzz. The 24-bit "container" is certainly capable of significantly higher resolution.

Here's the question... Can you hear the difference? Evidently many people believe the difference is audibly significant.

So, you've tried "crowd funding" (and you're probably at this point still waiting for a product), it's time for another round of "crowd testing", folks! Here's what you do to participate:

1. Download the "24-bit Audio Test.zip" file:

WARNING: The test file is big (a taste of the storage demands for those who have not downloaded high-resolution audio). Approximately 200MB for a total of 6 musical samples at 24/96 lasting less than 12 minutes with fully tagged FLAC lossless compression.

Get the file from my FTP server:
ftp://24bit-test.dyndns.org
Login = 24bit
Pass = test

Please have patience if the server load is high... Anyone able to help out with this, please drop me a note below!

Alternate download sites:

Thanks Ingemar:
http://www.privatebits.net/24-bit_Audio_Test_(Hi-Res_24-96-FLAC_2014).zip 

From Uploaded.net: http://ul.to/fqziwlfs
From FilePost.com: https://filepost.com/files/m5b3em11/24-bit_Audio_Test_(Hi-Res_24-96,_FLAC,_2014).zip/

2. Extract that ZIP to wherever you want for playback (computer folder, music archive, server, etc.)

Located within are 3 musical pieces in 24/96 FLAC, each piece with Sample A or Sample B versions. One of them (A or B) is the 24-bit original and the other contains a dithered 16-bit version which has been converted back to 24/96 so your DAC will basically be playing back the same bit/samplerate when you switch between tracks.

The samples are all classical pieces but with variation in instrumentation, vocals, and dynamics. Realize that it's not easy to find good high-resolution audio where the music is recorded and mixed to the highest standards with known provenance. Classical music as a genre is where some of the best recordings can be found.

Here are the tracks:

1. Eugène Bozza - la Voie Triomphale (performed by The Staff Band of the Norwegian Armed Forces): A well recorded orchestral track originally in DXD (32/352.8) freely downloadable as a sample from 2L here. In the interest of download size, I extracted 2 minutes from the track. It features good dynamic range with a DR13.

2. Vivaldi - Recitative and Aria from Cantata RV 679, "Che giova il sospirar, povero core" (performed by Tone Wik & Barokkanerne) - String orchestra with female vocals. Also DXD-recorded and available as a free download from 2L here. Again, I only extracted 2 minutes from the track. DR14 for this track.

3. Goldberg Variations BWV 988 - Aria (performed by Kimiko Ishizaka) as taken from the freely available Open Goldberg Variations 24/96 release. The recording was done at Teldex Studio in Berlin using the Bösendorfer 290 Imperial CEUS concert grand piano. Simple instrumentation for those who love and appreciate the sound of the piano. It's also a much slower piece which provides an opportunity to listen to the decay quality. Low-level spatial room acoustics also easily heard on this recording. Measured dynamic range is a reasonable DR12.

I've included the DR printout from foobar to demonstrate that peak and average volumes are identical for Samples A & B for each piece.

** Note that I'm using the samples under the principle of "fair use" for the purpose of education, research and commentary. I have no ties with Industry and have no financial gain by organizing this. As noted above, these tracks were selected based on excellent quality of the recording and I would highly recommend visiting the source links to sample more high-quality audio tracks in the high-resolution delivery formats (the 2L download page also includes DXD, multichannel and DSD samples).

3. Listen and compare Sample A with Sample B.

Can you hear a difference? Can you tell which one was the original 24-bit audio and which was dithered down to 16-bits?

Needless to say, you need to make sure your DAC / computer / streamer / etc. is capable of >16-bit performance!

I encourage the use of tools like foobar's ABX comparator to switch quickly between A & B. Make note of which Sample for each piece of music you believe is the original 24-bit track. Presumably, since the 24-bit version is higher resolution than the 16-bit one, there could/should be improved transparency, ambiance, definition, smoother decay, etc... For example, if you think the 24-bit sample is A for Bossa, B for Vivaldi, and A for Goldberg; keep track of that and make note of how confident you feel about your selection.

4. Make your voice heard on my survey form!

Go to the survey here:
http://freeonlinesurveys.com/s.asp?sid=k70bh4nrjagkj06462640

I have 14 questions (15th optional) - it should not take long to fill out as most are multiple-choice tick boxes. As expected, I want to know whether you think A or B is the 24-bit track. I also want to know if you're guessing or confident - if you spent time listening, make it count by visiting the survey even if you don't think you can tell a difference. A 'negative' report is just as important as a 'positive' one. I also want to get a sense of your age, gender, tiny bit on musical and technical experience. Also, equipment used, and approximate cost of the audio gear. All data is anonymous and I have no access to your IP address (the web site will keep track of what country you're submitting the survey from - let's make this an international effort!).

I'm going to be busy with work and other responsibilities for the next while so this is the perfect time to gather some data. As I did with the MP3 test, I'll summarize the information and demonstrate the conclusions of this survey once it closes. I think about 2 months is adequate time for everyone who wants to get involved to have a good listen. Therefore, I will close the survey around June 20, 2014 - fill in the survey before that time! Note that IP filtering is ON so only 1 response from each IP please.

Finally, remember to relax, take your time, and have fun with this... Enjoy the music and see if one version "speaks" to you more than the other in terms of sonic quality. Feel free to share this test with friends / family / music lovers / audio reviewers / audio forums / enemies. Also, please try not to whip out the audio editor before doing the listening and completing the survey! Honesty is extremely important for an open "naturalistic" survey like this one. If you know which is which, please do not share it with others so as not to bias the results. Feel free to leave a comment down below if you run into any problems or need help. Thanks...

A few guys got to "beta-test" this little project and I want to thank 'Wombat' and 'Mnyb' especially for providing technical suggestions and detailed feedback on the Squeezebox Audiophile forum before this went "live"!

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I've been enjoying some Ladysmith Black Mambazo over the last few evenings at home. As usual, with some of the more recent CDs I wish the dynamic range were higher especially for multilayered vocal music like this. I'll be sure to update when I get stuff like the pre-ordered Geek Out in my hands over the next while or if I've got some burning thoughts to share :-).

Enjoy the music... Happy Easter.

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UPDATE (April 24, 2014): ~1 week into the test...

Great to see the responses so far! Just with the FTP site, >120 uploads of the file already. I of course do not expect that many responses to the survey yet since there's plenty of time to listen, but I am impressed by the quality of responses so far and the variety of gear people are using. Clearly folks are not using "junk" gear to assess and the majority are spending time listing out for me what they use and subjective impressions. All will be revealed in time of course...

Keep the responses coming!

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UPDATE (April 30, 2014):

>50 responses now. Need more for better statistical power; especially in analyzing each test piece. Expanded the advertisement to the Audio Circle "HiRez Music Circle".

May 2, 2014:
Formal invitation extended to WiredState and Steve Hoffman forum.
May 10, 2014:
Invitation to Pink Fish Media forum.
May 17, 2014:
Invitation to Head-Fi forum.

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

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

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

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

I. Procedure:

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

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

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

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

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

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

II. Result:

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

AudioGate as DSD encoder.

JRiver as DSD encoder.

Weiss Saracon as DSD encoder.

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

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

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

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

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

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

Frequency Response

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

Noise Level

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

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

Frequency Response

Noise Level

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

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

Frequency Response

Noise Level

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

III. Conclusion:

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

MEASUREMENTS: Nexus 7 to Audioengine D3 (A "Kinda Portable" Audiophile Playback)

Okay, for fun, I thought I'd grab a few measurements of something... Somewhat... "Portable" :-)

What you have here is my Nexus 7 tablet connected to an "on the go" cable (5" male microUSB to female standard USB, off eBay - pack of 3 for $10) --> Audioengine D3 DAC/amp --> Sennheiser HD800. Unfortunately, The D3 would not power up consistently when plugged into the Nexus 5 smartphone since that would have been even more portable! Looks like the D3 demanded more power than the Nexus 5's USB port could deliver.

I got USB Audio Player Pro software for the Android in order to get the USB DAC working. Unfortunately USB Audio Class 1/2 devices are not supported by Android by default. You can see the basic interface above (I happened to be playing some Bruno Mars Unorthodox Jukebox). It recognized the D3 without a hitch. It sounds the same to me connected to the HD800 like the other machines I tested the Audioengine D3 with last time so no need going into any subjective evaluation here.

I was more interested in whether the objective data showed any difference between this "mobile" set-up compared to the laptops / desktops.

RightMark Results:

Remember the clipping at 100% with the Audioengine D3. All the measurements are done with hardware volume attenuated to 92%.

16/44:

Frequency Response

Noise Level

THD

IMD

24/96:

Frequency Response

Noise Level

THD

IMD

The first column in the summary tables is the Nexus 7 + Audioengine D3. The second column is the ASUS Taichi laptop connected to the Audioengine D3. Finally the 3rd column is the Nexus 7 natively without using the external USB DAC.

As you can see, there is no substantial difference whether the ASUS laptop/ultrabook or Nexus 7 was used in either the 16/44 or 24/96 test case. The DAC determines the final audio output, not the source "transport" device.

The Audioengine D3 is substantially better as a DAC of course. As you can see, the Nexus 7 + D3 easily outclasses the native Nexus 7 audio output off the headphone jack. It has a flatter frequency response with lower noise floor measurable even at 16/44. The difference is more evident with a 24-bit audio signal... The 24/96 frequency response demonstrates that the native Nexus 7 in fact is incapable of 96kHz sample rate.

Jitter:

Slight difference between the 2 J-Test measurements. The noise floor seems a little bit higher on the whole with the Nexus 7 when I ran this test making the peaks such as the 16-bit modulation pattern less obvious. Also a little cleaner around the 24-bit 12kHz primary tone with the Nexus 7. The differences are down around the -120dB level and would not be audible IMO.

Conclusion:

Okay... So this set-up isn't exactly a portable device. But it does demonstrate that you can get excellent sound out of a small Android device with the USB DAC that is objectively equivalent to using a standard computer.

Truth be told, I don't need "audiophile quality" sound in a portable device. I can't remember the last time I listened to my smartphone or iPod somewhere quiet with expensive headphones. On-the-go, convenience trumps everything else IMO so there'd be no way I'd bother with full sized headphones. If I did bring full-sized cans around on a train/plane/subway, it certainly would not be the expensive high-resolution open design headphones!

Day to day, I have my Nexus 5 phone with me. There are a few albums saved on the phone as MP3 or FLAC when I want to listen. Otherwise there are countless apps to listen to Internet radio and music streaming services which is what I listen to most. The days of the isolated non-networked portable music player are long over for me and probably the vast majority of music lovers.

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For fun I decided to jump on the Geek Out bandwagon for a spin and see how it goes.

Looks like we're starting to see some user reviews coming out now, the first more formal one I see being this one at Part-Time Audiophile. So far it looks encouraging. As expected there are a couple of comments about the heat production of the 1W model. No comment about how it handles DSD and it looks like those guys are Mac-centric, so not clear how it works out in the Windows world (ie. drivers). For what it is, I find purely subjective reviews interesting to read but I would prefer something a little more than the usual "drive by shooting" ;-).

After a bit of humming and hawing, I decided to go for a blue "Super Geek" ($250, 720mW, 3.4Vrms peak) model as the best compromise for my case. My rationale is simple... The amplitude difference between 720mW to 1W is only 1.4dB (as compared to 2dB between 450mW to 720mW). I'm still concerned about heat production since this is a class A design which sucks full power all the time... Assuming the enclosure design for heat dissipation is the same, the lower power model should drop the running temperature a few degrees. I can see myself using this as a line-level DAC if it measures really well and tap the headphone amp feature only on occasion (the Audioengine D3 appears lighter and smaller for travel). As a 'standard' USB DAC, this also means it'll likely be "on" all the time so I'd rather not have something too warm sitting on my table. Furthermore, with a laptop computer, an inefficient class A device means more power drainage. The $50 saved is trivial for this hobby and not really an issue.

Anyhow, once I get this, I'll let you know some results... It will be interesting to see how this compares to the TEAC UD-501 which has the same DAC chip and similar feature set (NOS, DXD, DSD, etc...). Hopefully it doesn't take too long to ship out - my understanding is that only the 1000mW "Super-Duper Geek" model is released so far.

Juergen mentioned having a look at the HpW-Works software package for jitter analysis. Might just do that although I remain unconvinced it makes any audible difference in 2014 especially with asynchronous USB. I have yet to see a good example of a decent modern piece of equipment where jitter can be shown as the culprit for impairing the sound quality.

Tonight's music:
Kodo - Mondo Head - Although I generally prefer the more traditional sound of Tsutsumi, this one not only sounds nice in stereo but fantastic in multichannel off the SACD!

Enjoy the tunes everyone!

MUSINGS: On Experts, Experience, and Opinions...

So last night, instead of going to bed early as I was supposed to, I decided to have a look at this interview of Allen Sides from Ocean Way Recording on TWiT.TV. As usual, Scott Wilkinson does a fantastic job with the interview and takes questions from the audience.

Obviously, Mr. Sides is a man of many years of experience and can speak authoritatively on MANY topics related to audio hardware, studio production, and historical anecdotes based on those years.

But some things bothered me. Around 16:00 there was talk about DSD: "somehow between making the recording and that SACD it doesn't sound quite as good as it should". Really? By 17:00, there's discussion about CD copies, different stampers sounding different, then an anecdote on Mariah Carey and how the pressed CD sounded bad compared to the reference. Okay... Maybe... How about someone ripping the disks and comparing the data integrity and talking about that? Given that Mariah was married to Tommy Mottola until 1998, this anecdote is now at least 16 years old so any hope of forensic assessment is long gone - does it still apply as a generalization these days?

Things get really bizarre by 26:00 - "I have never been able to even make a copy of a CD that sounds as good as the CD I started with... It always sounds worse". As you can see, Mr. Wilkinson was perplexed and commented that "it's almost like generational loss in analogue" that Mr. Sides is referring to. "Multiple degenerations"? Please...

I wished Mr. Wilkinson would have done a follow-up question like - "What if you bit-perfectly ripped that original CD to a computer - does it sound the same then?" "How about if you copy to a different hard drive, does it sound different?"

It's also clear that there are some limits to Mr. Sides' knowledge/experience which most of us in the hobby world would have no difficulty discussing. (34:30) Q: "What's your idea of FLAC encoding?" A: "I'm really not that familiar with it." Fair enough, a person cannot know everything.

Although I'm not that old at this point in my life, I have learned some things in my "travels" both personally and professionally. One which I hold dear is that no matter how much we can respect and trust the "experts" for their lived experience and knowledge, they (like us) are all just human. And as humans we all have idiosyncrasies and biases. In this case, I don't think it's much of a stretch for any of us who have spent hours on our audio systems, used EAC or dBpoweramp for ripping to ensure bit-perfect copies, to stand up with good confidence and tell Mr. Sides that he's just plain wrong about not being able to make a copy of a CD that sounds identical. The fact is that in more than 30 years of the existence of the CD format, there has been no evidence of this when variables are controlled for (eg. ensuring that bit perfect copies were achieved, the rater was blinded, etc.). If indeed "generational losses" were possible with digital, this would already be a well known fact and there'd be no uncertainty whatsoever! Moreover, if this were fact, it would change significantly how we deal with accuracy of our digital data (hey... how can I be sure that those numbers in my bank account are accurate?!). Experts can provide educated opinions, but ultimately they are just opinions and not necessarily fact. The same goes for his strange comment about SACD not sounding as good as it should between studio and the physical disk. What is more likely, that the digital data somehow mysteriously changed over time or that his own psychological expectations changed as the memory of the live studio event consolidated? (Assuming of course that the data wasn't altered by some mastering engineer along the way.)

In this world, there are many mysteries yet to be discovered and likely much we as a species will never know. But digital audio systems which are inventions of the human mind based on mathematical constructs and technologically engineered devices (like the CD) that ultimately changes the physical world (sound waves) were not produced by serendipity. I do not feel it's good enough that we should just shrug our shoulders and declare some experiences to have enduring truth like parts of this interview. That surrender to logic leads us into the realm of "anything is possible!" and ultimately the slippery slope on the path to "snake oil". This is especially significant in the impact on those already obsessive-compulsive and perfectionistic (ahem, like many audiophiles). Maintaining an objective approach hopefully allows a counterbalance to this. An opportunity to take a step back and question the things which seem to make no sense. An opportunity to explore reality with techniques and at times instruments of greater sensitivity than that which we are endowed with within this mortal shell. Although the human mind is the best "instrument" to perceive the beauty of music, accuracy of the reproduction chain is a different matter and can be detected by the use of objective techniques with obviously greater sensitivity.

It's fun listening to interviews like this and I certainly felt it was time well spent nonetheless.

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.

Conclusion:
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.

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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 ;-).

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.

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...

Addendum:
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?

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. :-)

Addendum:
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!)

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

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

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

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

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

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

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

Results:

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

Summary

Frequency Response

Noise Level

THD

IMD

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

Conclusion:

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

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

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

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

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

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

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

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

Enjoy the music everyone...

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.