MUSINGS: It's Vinyl Time!

Hey there everyone... The 24-bit vs. 16-bit audio survey has now surpassed the 100 responses "milestone" which I unofficially set as the minimum I wanted to achieve for this test. Thanks to all who have already submitted to the test results and please keep them coming - the more results the better the statistical power... You have until June 20th. Still looking for some Russian input to fill out the international effort :-).

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Over the last few weeks, I finally bothered my father enough to drag out the old turntable for a spin. It's a "vintage" Sony PS-T15 from the early 1980's - he bought it with an old Sony receiver, tape deck and Sansui floorstander speakers in 1981. Just a couple years before the CD juggernaut started rolling. I essentially grew up with CD as the primary music collection medium although did play some LP in those early years (along with a collection of cassettes). It's a "middle-of-the-road" model from Sony that's supposedly a step up from the "L" series but below the Sony "audiophile grade" "X" series. Although old, the condition is still VG+ with the only blemishes being the top transparent plastic cover slightly scratched from the moves between homes and cities over the years. The direct drive motor still spins smoothly and quietly, and the tonearm looks and functions decently. Pitch control is functioning accurately. I suspect this works better than many current-model inexpensive consumer turntables.

As you can see, there's a long discontinued Audio-Technica AT70 MM cartridge in place. The stylus still looks good under a loupe.

I reacquainted myself with the turntable setup routine (here's a good AXPONA 2014 video from Michael Fremer - funny interchange around 20 minutes vs. CD). I've still got lots to learn about this but I did reseat the cartridge and cleaned the contacts, reapplied the counterweight and antiskating settings, realigned with a protractor (check out some free ones on Vinyl Engine), leveled the unit, etc. Good enough for now... Grabbed an inexpensive Spin Clean Starter Kit MKII, and went to work getting the gunk off some very old records in boxes which have not seen the light of day in decades!

I must admit, after having not handled LPs in at least 10 years and more than 20 years since using a turntable with any regularity, this has been fun... Without doubt, psychologically the act of setting up even something as simple as this "non-audiophile" gear taps into a different kind of mental "reward center". It's a sense of achievement brought about by the ritual. To hold in hand the 12" vinyl disk, and examine the full-sized LP cover brings with it a different kind of "joy" which for me isn't really about the sound quality, but rather caters to the "collecting" (hopefully not "hoarding"!) habit. Furthermore, the discipline of keeping the gear and vinyl clean and getting the albums organized intensifies a collector's obsessive-compulsive tendencies. This indeed all translates to a different experience which again can be pleasurable and memorable irrespective of the sonic quality.

My kids checking out the spinner - for them it's the first time seeing one of these in real life :-).
Listening to an old Teresa Teng pressing from 1980.

You might be wondering... What about the sound? Well, it's pretty good. Obviously this isn't high-end vinyl gear by any means, and I don't have many audiophile grade LPs on hand, but the sound is definitely enjoyable. Over the years, I have heard some excellent vinyl setups and can attest to fantastic imaging and clarity. However, as alluded to above, for me, vinyl isn't about audio quality being "accurate" or "high-resolution"; there are just too many variables and little objective frame of reference to accurately verify what is "high-fidelity" from a vinyl setup.

Despite the audiophile buzz, turntable & vinyl ownership is still very rare among my friends (late 30's to early 40's). I think many of us also recognize the "coolness" factor associated with being retro. This is heightened further by the positive sentiment perpetuated by folks like Mr. Fremer that somehow LP's in general sound "better" than digital (like this video where he also talks about CDs being "analog" at 43:30). Seriously, if it's only sound quality I'm after, I still think a well produced non-dynamically-compressed digital copy is superior.

In any event, there is fun to be had and that's good.

Now that I have the sound room and space, nothing wrong with putting together a little vinyl collection of cherished recordings. Besides, the stack of LPs could help with room acoustics... :-)

KoB 50th Anniversary box set - fitting 1st LP purchase in >20 years... And I swear this is the last time I spend another dollar on this album! Of course I said this >10 years ago when I bought the SACD... :-)

I was chatting with a friend the other day about maybe doing some "controlled" vinyl rips using my basic gear versus his high-end multi-kilobucks setup to demonstrate what $$$ buy in terms of sonic output. That could be fun and maybe another blind test to try out. I'll keep you guys posted :-). Let's get through the 24-bit vs. 16-bit survey first...

There just happens to be a couple of good used LP stores within 10 minutes of home, so I might check them out between driving the kids to events this weekend. Until next time, enjoy the tunes everyone...

Addendum (June 1, 2014):
I ended up returning that Kind Of Blue (50th Anniversary) Box I showed above. I noticed that Side B had quite a number of light surface abrasions and 2 strange "bumps" about 1cm in diameter. I've not seen these bumps with the black LPs I have... Disappointing since it's quite a nice package and essay book... Maybe I'll just grab the standard black vinyl release later.

MUSINGS: A few words about Pono... And what is "the finest digital copy"?

Remember everyone, we're not done with the 24-bit vs. 16-bit blind test yet! We're about 1/2 way through before I close off submissions to the survey site. Again, I thank everyone who has tried it thus far and all the awesome responses with detailed descriptions of the gear used and demographic information... Many audiophiles from around the globe have already voiced their "vote" and it's getting close to 100 submissions. I'm somewhat surprised that so far there hasn't been a submission from India, Russia or S. America!

In any case, one more month to go everyone!

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Up to this point, much has already been written about Pono from both supporters and detractors. As you know, the media show around Pono started anew at SXSW with the announcement in mid-March and concomitant Kickstarter project. They did quite well financially with the Kickstarter campaign so it's an encouraging start for opportunities ahead. A few details were clarified such as the designer of the PonoPlayer being done by Ayre (rather than Meridian as previously thought). But at the end of the day, it's really hard to pin down just what is going to be all that different in the Pono "ecosystem" or why a music lover would choose this over others (eg. HDtracks, Qobuz). Maybe it is as Neil Young said here; "There's nothing, there's nothing new. There is no new thing. It's just available. It's available to everybody". So high-resolution, lossless audio will be available to everybody - at a price... Indeed, that's not new.

So, after all the star-powered hoopla and idealistic catch phrases like "rescuing an art form" (seriously, what's that supposed to mean?), what we know is that Pono is primarily a music service that sells "high-resolution" audio tracks in FLAC much like HDTracks has been doing for years. They are selling the PonoPlayer designed by Ayre to get the ball rolling as a physical device to brand the service with the 'iconic Toblerone' form factor but there's no clear commitment to ongoing hardware development. There will be no DRM which is of course good for compatibility. Since many DACs and devices these days can already do high-resolution decoding and playback, the flexibility is very much appreciated. It's amazing that even the recent May 7 Pono-listening-gathering did not use a prototype Ayre player for demonstration; again, stressing the music software as the prime objective rather than the player hardware. As you know, a number of folks have signed up for the limited-edition Kickstarter autographed PonoPlayers which are nice for collectors and as novelty items.

In the last month, I see that the Pono CEO John Hamm has taken the lead on presentations at shows like AXPONA and demonstrations like the May 7th one above. This seems like a good move since he's clearer and obviously more in tune with the technology (see the AXPONA video) than NY. There's of course still a major sales pitch going on in the video which is understandable if not annoying at this point. For example, those video clips of musicians/producers with testimonies of being "blown away" by Neil's car stereo!? You have to wonder just what kind of music these millionaires have been listening to in the last few years. Their own CDs remastered loud and volume compressed down to DR5 is my guess; using MP3 as the scapegoat of course. Furthermore, the discussion still gets stuck around the technical "container" (24-bit, high sample rate, MP3 as "5% of the data",  etc...) rather than the elephant in the room; the actual quality of the source recording, mixing and mastering process. If you're using a massive shipping container, you better have lots of good stuff in there worth taking up space for!

Encouragingly, it's great to see questions being raised at the end of the presentation about dynamic range compression and some answers being provided; John Hamm addresses this around 25:30 in the video above. He states that there will be no special remastering or remixing although it sounds like they could exert some influence in selection of which mastering is sold... But he does say something intriguing: "The promise of Pono is that we will sell you and play on our PonoPlayers the finest digital copy of that song or album available in the world" (38:35). Elsewhere, we are told that if you buy a copy of an album and a "better" remastering comes up, you get an automatic upgrade... (Right... I'll believe it when I see it.)

Interesting... Consider the question then: How does one know which is the "finest digital copy"? First, it almost goes without saying that to answer that question for any album with remasters over the years requires subjective evaluation; much more complex than just throwing out a number like "192kHz!". Second, I suspect the process whereby that answer is arrived at could allow a fundamental shift in consumer expectations if actually accepted by the public, and implemented professionally. It could make a substantial dent to calling out that proverbial elephant in the room and setting us free from ridiculous over simplifications like focusing on just bitrate / bitdepth / samplerate or simply MP3 vs. lossless. Educating the general public on how to listen for a quality mastering job that encompasses the full dynamic range, natural sound (eg. judicious EQ), with fully detailed resolution would in itself be the most beneficial outcome of this whole "high resolution audio" endeavor even though it may have started with "objective" parameters like pushing 24-bit and 96+kHz... I trust that already many of us have experienced 24/96+ music which is either no better than an equivalent CD (have a look at most of the new pop releases on HDTracks like Beck's Morning Phase), or in some cases the "high-resolution remaster" sounds worse.

There are suggestions in these presentations that Pono somehow represents a "legacy" (1) that Neil Young wants to leave beyond his copious body of work already. There is certainly an opportunity. But I believe it's not going to come from geeky talk about sample rates and meaningless stats like "30x more data than MP3" with a graph to accompany the rhetoric as if this directly correlates to sound quality. It's going to come from choices made and promoted about not just the provenance (ie. where the audio recording came from including true hi-res recording practices - see Mark Waldrep's site), but the technical and artistic bit around adjudicating what actually sounds "best" and which of the versions of many recordings is "best" irrespective of the file type. Does the artist choose which version is best? Is it down to vital stats like dynamic range (eg. something like DR values)? Is it up to the record labels thinking that the most recent "anniversary" remaster is best? In Neil Young's ears we trust? Or a mindless adherence to 24-bits and 192kHz sampling rate?

No my friends, on the first day that the Pono Music site is up, I'm not really going to care whether the PonoPlayer sounds great (it better for essentially a no-frills unconnected music player at the >$300 price point), or if the website is user friendly (2), or how many albums are available to choose from... I mainly want to know whether they're selling the first digital release of Nirvana's Nevermind (16/44), the 1996 MFSL remaster (16/44), the HDTracks 2011 remaster at 24/96, or the Blu-Ray 2013 24/96. Nevermind is of course just one of many albums to keep an eye on for what Pono does... Pono's offering will speak volumes about their commitment and whether this "movement" has any steam - are they walking the talk about "good sound", stuck with fancy specification numbers, or just "nothing new"?

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(1) See around 7:00 into this Esquire interview.

(2) There's time to optimize and streamline the website. A nice website with full provenance data and ability for consumers to leave comments & reviews would be awesome of course.

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!