Saturday, 4 June 2016

MUSINGS / ANALYSIS: Is there any value to 176.4 and 192kHz Hi-Res audio files? A practical evaluation...



Check out this article from 1998 written by the founder of Earthworks, David E. Blackmer (1927-2002).
 
Although I believe some of the contents in the article above are debatable, in the years since 1998, high-resolution, high samplerate audio has of course become common-day reality for audiophiles. As I expressed years ago, I do like the 2xCD samplerates like 88 and 96kHz. But as a result of realizing that 176.4 and 192kHz songs were not being streamed properly with my Logitech Media Server with BrutefirDRC set-up described a few months back, I started asking myself, what is it we would be missing if these albums were downsampled to 88.2 and 96kHz?

Put another way, we could ask "Is there something musical about the highest octave in these 4x samplerate files?" This highest octave for 176.4kHz files would be audio containing 44kHz to 88kHz, and in 192kHz files from 48kHz to 96kHz.

Now ideally, if a file contains no audio information in the highest frequencies, we should see a very clean - low and flat - noise floor... Something like this:
Frequency FFT
Spectral Frequency Display
This is the track "Grande Sonate Op.25 I-Andante Largo" by Ricardo Gallén from the album Fernando Sor: Guitar Sonatas which was recorded in DSD256 and converted to a 24/176 PCM file. (You can get this album from NativeDSD.com.) If you watch in realtime the frequency FFT as the song plays, you'll notice that the music is contained up to about 30kHz; that is, the noise essentially stays flat from 30kHz to 88kHz on that graph. This can be easily seen on the "Spectral Frequency Display" in Adobe Audition where I also placed a 44kHz line since if we were to downsample by half, the natural target would be 88kHz. As you can see, there's basically no musical content at all in that highest octave.

Another example of a relatively clean noise floor would be something like this:
Frequency FFT
Spectral Frequency Display
This one is from 2L records, the beautifully recorded vocal track "Nato Canunt Omnia" from Psallat Ecclesia by Schola Solensis downloaded as 24/192 (itself downsampled from an original DXD 24/352). While the noise does creep higher from about 60kHz up, it remains low overall. Just like the track above however, if we watch the frequency spectrum in realtime during playback, we see that there is no apparent musical information above 40kHz or so; it's just noise which stays essentially static. Again, beyond the 2x samplerate of 96kHz (48kHz line), there's nothing in the uppermost octave that appears to represent any musical or ambient information.

More often than not however, you see files which are contaminated with noise peaks probably picked up in the analogue recording chain. Something like this:
Frequency FFT
Spectral Frequency Display

That was taken from the Linn 24/192 "Studio Master" release of Carol Kidd's album Tell Me Once Again (the track was "Moon River"). It sounds fine on playback of course but you can see that the ultrasonic spectrum is contaminated with peaks up to -68dB or so especially around the 60-70kHz regions; easily seen as a band across the Spectral Frequency Display. If you consider however that there does not appear to be anything but noise above 30kHz, one certainly should start wondering if it might not be best to just filter out all that extra crud since there is potential that intermodulation distortion could affect the audible frequencies in playback.

I started to look at just how many of these "very high-resolution" albums (that is, 176.4/192kHz albums) I had in my Logitech Media Server library. As expected, relatively few - only 5 24/176 albums and another 35 24/192 albums out of about 8,000 total albums (0.5%) excluding singles and vinyl rips. Many of these I collected back in 2008 to 2012 as high-resolution downloads off HDTracks or my own DVD-A rips purchased in the early 2000's. Remember that it was during that time frame when high-resolution audio started becoming available online and DVD-A Explorer came out for DVD-A ripping (~2008). Well, I loaded the first 14 albums into my audio editor and made notes about whether I thought there was musical content in the highest octave afforded by the higher sampling rate.
- Neil Young - American Stars 'N' Bars - 24/176 DVD-A rip was just an upsample, happily resampled to 88kHz
- Neil Young - On The Beach - 24/176 DVD-A rip was also just an upsample, brought down to 88kHz
- Neil Young - Archives Volume 1 - Blu-Ray 24/192 rips. I didn't go through every song but the ones I looked at all looked like 96kHz upsamples at best.
- Bill Evans Trio - Waltz For Debby - 2011 HDTracks 24/192. Easily resample down to 48kHz and dither to 16-bits without fear of losing anything. High noise floor on old recordings do not benefit from 24-bits.
- Bob Marley - Legend: The Best of Bob Marley & The Wailers - 2012 HDTracks 24/192 download, no content worth keeping so downsampled to 24/96.
- Fleetwood Mac - Tango In The Night - 2011 HDTracks 24/192 download - clearly not worth more than 16/48 due to high noise floor and no high frequency content.
- Fleetwood Mac - Tusk - 2011 HDTracks 24/192 download - like above, no point keeping >16/48.
- Neil Young - Harvest - 2002 DVD-A 24/192 rip - upsampled from 96kHz.
- Carly Simon - No Secrets - DVD-A 24/192 rip - awful DR9 remaster, just get the DR13 first press CD or maybe Audio Fidelity remaster I see is available.
- The Eagles - Hotel California - 2001 DVD-A rip - feel free to go down to 24/96.
- Cat Stevens - Tea For The Tillerman - 2012 HDTracks 24/192, happily downsample to 96kHz.
- Steely Dan - Everything Must Go - DVD-A 24/192 rip. Basically a 96kHz upsample.
- Muddy Waters - Folk Singer - 1999 Classic Records HDAD 24/192 - lovely recording but just noise >25kHz. In fact 16-bits would also be enough for a recording of this vintage with high noise floor.
- Antonio Forcione & Sabina Sciubba - Meet Me In London - 2011 Naim 24/192. Sounds great but even an "audiophile" release like this has a high noise floor and nothing but noise beyond 48kHz as well as some moderately strong high frequency tape bias signals.

... and so it goes; that was just the first 14. Even classical and acoustic music (from 2L and Linn) as suggested above in the graphs looking like genuine high-resolution captures at 192kHz could easily be downsampled to 96kHz and I don't think an audiophile should have concern about content loss. I know that we often joke on forum posts that maybe these very high resolution audio files would be preferred by cats and dogs, but honestly, I suspect one's dog and cat would prefer a downsample taking out all that noise and high-pitched continuous tones in the highest octave found in many of these albums (like the Carol Kidd tune demonstrated above)!

As I went through what I had, the only time I considered maybe 192kHz could capture more musical information was with Trondheimsolistene's In Folk Style 24/192 Blu-Ray rip where the track "Grieg Two Nordic Melodies Op. 63: II. Kulokk and Stabbelaten (Cow Call and Peasant Dance)" encroached up to 48kHz and maybe just barely surpasses. Obviously, any information going above 48kHz is of very low levels and unlikely audible given human hearing limitations, so it's all rather academic and perfectionistic.



Years ago, Monty at Xiph.org already laid out the technical case against 24/192. Some excellent points there, but even just looking at the realtime spectrum analyser while playing the music, one gets the sense that there's just nothing there to even bother with.

Finally, if you look at how we make recordings, it gets down to the basic fact that there are not many microphones with the capability for extended frequency response. For example, the recent Sennheiser MKH8000-series models can go up to 60kHz, but typically to around 50kHz. And as noted by Demian Martin in the comments of a previous post, we need to be careful about drop-off when recording off-axis (he gives the example of the B&K 4133 and off axis drop-off based on the data sheet page 7). Even for Earthworks' own demo material to show off their microphones' high frequency response, the wav audio download is presented as 16/44 files with obviously no content above 22kHz. Are they suggesting that higher microphone frequency response might improve tonality in the audible range?! If we look at the specs for a typical list of "recommended" studio microphones, the vast majority are rated to 15-20kHz; bottom line is that there's not much high frequency material being captured even if one were to argue that some instruments are capable of strong ultrasonic harmonics. Sure, there could be "life above 20kHz" based on a CalTech paper (you can still find it on Google cache), but how many studios actually capture this and of what fidelity? What I have seen suggests that there's little being retained whether by choice or from the limitations of the microphones used.

[On a side note, of course we do have microphone technologies that can record ultrasonics quite well. For example, in the biosciences, a number of research papers on animal vocalizations such as this paper on rat communication uses the Avisoft-UltraSoundGate CM16 which is capable of up to 200kHz within the limitations of the polar diagram. Great for bird chirps, rats, echolocating bats, and understanding porpoises; not quite necessary for music recordings as far as I can tell :-).]

If you're wondering what settings I use to downsample from 192kHz to 96kHz, the obvious answer is that it really doesn't matter with modern sampling rate converters since accuracy would be excellent. My favourite program for this is iZotope RX using a relatively gentle linear filter (compared to the typically steep 44kHz settings where "Filter steepness" would be around 30 to keep frequencies intact to 20kHz) beginning just beyond 40kHz and essentially free of aliasing:


Sounds great and works for me... Of course I could use an even gentler filter but I had this romantic notion that it would be nice to keep frequencies as flat as possible to 40kHz (two times the typical ideal top frequency response for human hearing at 20kHz) and to use linear phase setting ("Pre-ringing" at 1.00) to prevent phase shifting since we don't know if the DAC will further oversample with its own minimum phase algorithm at playback. Not that this likely matters being ultrasonic and all...

In summary: Yes, there is "life above 20kHz". But I see little if any sign of life above 44/48kHz (or 88/96kHz samplerate) in recordings out there. IMO, this is further justification for the music industry, if they choose to be serious about providing truly high resolution quality offerings to consider "keeping it simple" and standardize on 24/96 masterings.

In the comments, let me know what albums you've found to contain recorded harmonic frequencies benefiting from 176 and 192kHz samplerates... I'd love to have a list for demo and assessment purposes!

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Before closing off, remember that in this post I'm only reanalyzing my 176.4/192kHz files. In truth, remember that there are many 88.2/96kHz files out there that probably can just as well be downsampled to 44.1/48kHz simply due to lack of actual content. I've many times come across 96kHz files for example that just look like they may have been put through some kind of analogue mixer but the underlying musical content originated at 44.1/48kHz. Apart from noise picked up in the ultrasonic range, and unless on wants to argue that said ultrasonic noise has beneficial and audible effects, I'd be inclined to downsample these as well. This is of course the natural outcome when the music industry cannot standardize on a technically quality controlled product for what constitutes as "high resolution". Also, I believe many audiophiles are tempted to buy the "high resolution" file simply because of the "bigger is better" mindset in the absence of objective standards and analysis (is it any surprise that websites like HDTracks don't allow user review comments for folks to discuss the value of these downloads?).

Finally, remember that I'm not at all saying anything negative about the value of higher technical capabilities like 24-bits or >88/96kHz sample rates. There is of course a time and place where 24-bits resolution is essential (eg. in the studio for dynamic range overhead during production) and high sample rates used to optimize the mastering quality and archiving. It is important to however remember that the needs of the studio does not necessarily apply to the consumer home playback of the final product. Sure, there might be some psychological satisfaction in owning a "studio master" better than CD-resolution despite the technical standard of the content. I look around my collection and see albums like Miles Davis' Kind Of Blue in 24/88 and 24/96, or a copy of 24/96 Harry Belafonte's Belafonte At Carnegie Hall; both from 1959 and clearly neither of which have a low noise floor requiring 16-bits nor frequency extension beyond at the very most 48kHz samplerate. I also see Bob Dylan's Highway 61 Revisited (2015 MFSL SACD) in 24/88 from 1965 - again 16/48 would have been enough. But due to the "classic" status of albums like these, I'm OK with keeping them in a bigger bit "container"... Remember though that it's one thing to accept oneself as human and have idiosyncratic "values" but I would certainly not argue that these albums would sound any worse downsampled to 16/44! This is an example of what it means for me to be "more objective"; maintaining a foundation of decision-making and understanding based on the science, but at the same time allowing oneself a license to take joy in one's own subjective psychological idiosyncrasies.

Have a great week ahead. And of course I hope you're all enjoying the music.

Addendum (06-10-2016): Based on the discussion below with sk16 about Steve Wilson's production, a reader sent me some information about the recent album Hand. Cannot. Erase. (HDTracks 24/96, 2015).



DR11 is quite good for a modern album of course. The lower graphs are for the HDTracks 24/96 download of the first song "First Regret". As you can see there's nothing beyond 29kHz demonstrated on the FFT and Spectral Frequency Display. Based on what I see on playback, I believe resampling and dithering down to 16/48 would have been just fine while saving a good amount of storage space.

51 comments:

  1. Cue the ignorant "it's not about frequencies, it's about improved time-domain response" comments.

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  2. I have an album by Papa Bue’s Viking Jazz Band in 192kHz that has a lot of sound above 48kHz, especially when the trumpet is played.

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    1. Interesting Nils...

      What album is this and is this an old recording? I looked around and couldn't find any info.

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    2. This is the album: https://www.discogs.com/Papa-Bues-Viking-Jazz-Band-A-Song-Was-Born/release/5514331

      It's a "Direct-To-Disc" recording with outstanding sound quality. Track number two (Smiles) is my favorite track. It doesn't sound like the trumpet is coming from the speakers, it really sounds like the trumpet is there for real in your home.

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    3. Interesting... Are you listening to a vinyl rip to 24/192?

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    4. Yes I am. The ADC that have been used is: RME Fireface UC
      I can send you a sample if you are interested? It's a great demo-track if you want to surprise other people with the sound.

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  3. I don't know why record labels don't use software to create plausible higher frequency components in their higher res versions so that people can't tell they're upsampled, or to give the impression that their studio has used exemplary techniques to capture the ultrasonics.

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    1. They sometimes do. Thankfully it's quite rare.

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  4. What about time domain?

    Say 44.1k vs 88.2k, same bitdepth, you have 2x the info in time domain, less error for digital filter downstream.

    Btw, Shannon sampling theory dictates 2x minimal to reproduce band limited signal. However, the more the better for less distortion. Just another two cents.

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    1. As predicted...

      That's not how it works. Increasing the sample rate gains you nothing. Increasing the bit depth does.

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    2. Thanks for the note Sombody and Mans.

      Indeed, very important to have this discussion and hammer in the point that it does not work like that with modern DACs although the audiophile press will push this idea. I continue to see stuff like the "stair-stepped" digital diagrams and such which perpetuate this despite the fact that the vast majority of DACs out there are not compromised "NOS" devices.

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    3. @Mans & Archimago. In real life, lets have an example: you take 2 microphones. you take 1 adc and as source a tweeter. Now we generate 1 sine of half the sample rate that the adc uses (or: whatever the dac, later while playback commences uses, or simulates but let`s stick to Shannon and r2r NOS and say its 44k1). Now what is the output signal of that dac if the recorded signal was 22k05 and the acoustic phase difference between the microphones (because of the difference in distance from microphone to the acoustic source) is:
      - 0 degrees
      - 90 degrees
      I`m curious about the answer, because to me, this looks like a real problem..

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    4. sorry I was incomplete: the question should be: what are the output signals, for the left and for the right channel?

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  5. My comment seems to have been lost.

    Anyway, I have taken high sample rate files and done a steep filter removing everything below 20 khz. The reverse of the normal brickwall. That leaves only the ultrasonic material. I then slow it down to 25% of normal speed for playback. This is almost like upping your hearing to 80 khz. Even when you do this, there just isn't much up there to hear. It isn't loud at all. To think this, masked by the music, and well above your hearing will make a big difference becomes a clearly ludicrous idea at that point.

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    1. Yup. I've tried the same thing... High pass the stuff >20kHz, either slow it down or pitch shift down and have a listen. Most of the time noise and even if it's correlated to the underlying music, usually extremely low amplitude and hard to imagine how the human ear can can even perceive or how most speakers would even accurately reproduce the sound without "supertweeters"!

      I'm sure some folks will still comment on the idea that this stuff somehow adds to the "ambiance".

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    2. Those high frequencies aren't for us, not directly. They are for the DAC. They really grove on that stuff, and when the DAC is groving, it takes playback to a whole new level man. Just amazing!

      (ok, I'm not sure whether I'm about to chuckle at my own cleverness, or throw up in my mouth a little)

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    3. This is the most genius thing I've read in a long time about audibility of HiRes. Why didn't I think of this! Off to Audition! :D

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  6. Arch,
    Great article as usual. Well done.
    Have you ever analyzed any of Steven Wilson's recordings (solo and Porcupine Tree)? I regard Wilson as the gold standard of modern rock recording; I think others may as well. He has released on bluray at 5.1 24/96 for some years now. The two most recent releases, The Raven Who Refused to Sing and Hand.Cannot.Erase sound really, really good. Wondering if you have measured them?
    Wilson is also noteworthy for remastering Aqualung, King Crimson, Yes, XTC, and Tears for Fears in 5.1 2.4/96.
    Highly recommended.
    Best wishes...

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    1. Thanks sk16.

      Yes, a big fan of the Steve Wilson remasters. As far as I know he's quite reasonable with the remastering efforts and aims up to 96kHz I think. I don't have the recent albums like "Hand Cannot Erase"... I believe some of the older Porcupine Tree albums are 48kHz.

      Has he even released official 176/192kHz albums?

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    2. Hi Archimago

      Yes I do agree. I also appreciate DR11 for "modern" rock recording. And it is nice to see, that the 96k file is not upsampled 44k1 (or 48k) file, but native 96k. And also on my side, as I mentioned several times over the last years, 96k sample rate is sufficient, especially when used with slow roll off stopp band filters (and begin their work at 30 kHz and not at 0.45 x 96k, as most stopp band filters do).

      Juergen

      BTW: The former album "The Raven That Refused to Sing" was one of my favorite recordings in 2013.

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  7. "It is important to however remember that the needs of the studio does not necessarily apply to the consumer home playback of the final product".
    Well, that is only true for people running a NOS dac which surely isn't done by "the consumer" in any significan't numbers. What you're saying is just brick wall (or gentler) the original or upsampling material, then feed it to a dac that up samples itself. This is complete the opposite of what you'd be doing if you want less junk in and fooling around with your data. So this, IMHO, is more psychologically biased than not down sampling at all.

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    1. Not sure I understand.

      But the point also remains that if ultimately *there is nothing* above say a 96kHz samplerate, then why bother keeping the file at 192kHz? There is a very real practical saving in storage space which may not be a big deal with cheap storage these days, although it is significant for limited storage devices like mobile music players... (Though I honestly see little point in walking around pretending that "hi-res" music playing off headphones walking down the street or in the subway is of benefit.)

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  8. Well for mobile use it those 192k files are indeed big. The point I'm trying to make is that all dac chips convert all 44k1 to at least 8 times that much. Now if you have a NOS dac, that wouldn't be true of course, but not many have. Apart from the discussion if someone's audio chain is helped or not (in the nos case) with higher bit rates, it most certainly doesn't help your dac chip to feed it with modified (lower sample rate data) if what it does itself is always up sample (to at least 352.8KHz). I presume everyone here knows how dac chips work and that the first stage up sample process in a dac chip is, generally, the most important one. If not: usually the dac chip is either ladder or sigma delta. All of them work at way higher sample rates than what you feed them (44k1 and such, but also 192KHz is considered a too low native bit rate). So what a dac chip does is up sample 44k1 8 times. Most of the times it does this in steps. Like input frequency times 2, then one more time times 2 and the 3rd stage also 2. So you have an 8 times oversampling dac in the case of a ladder dac. A sigma delta dac is different, but let's just suffice with saying that 8 times is more likely to be 64 or even higher. What you do by removing the high sampling frequency, is making the dac chip have to work harder to get to that same sampling frequency. So it has to work harder (=more errors because of internal precision), and you add errors because of the downs ample process. This doesn't make sense to me. Sorry for ranting and its not my native language, nor am I a dac chip designer, so do take this with a small grain of salt: things might be slightly different in real life. The theory stands though!

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  9. To further cast doubt: the principle of sampling theory is that the length of time a sample is being taken of a signal must be very (almost infinitely) small. What happens if I sample at 44k1 and I feed it a 1KHz signal? It distorts. That's why a lot of dac manufacturers state their harmonic distortion plots at a integer division of the sample frequency. As in: Fs = 44.1KHz -> test signals = 1.027Hz (divided by 44). This helps understanding the fact that the higher the sample rate, the better the sampling theory works out. There are also hardly any oscilloscope manufacturers that sell oscilloscopes that comply to the bare minimum of the Shandon theory. They sample at least 3 to 5 times higher.It's being done because it's really hard to filter everything above half the sampling frequency with an analog filter that is ridiculously steep. Easier to do it less steep and up the sampling frequency. At least: nowadays that easy.

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    1. Go try measuring 1 khz distortion and see if your statement is true. I can save you the trouble having done it for several modern DACs. It isn't true. Therefore no evidence to cast doubt.

      BTW, it is Shannon not Shandon.

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    2. Dennis, have you ever used a pro measurement device? Tell me, what frequency do you use when generating a tone and what does it do (how does this relate to spectral purity) when using modern DAC chips (read: sigma delta)?

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  10. Re Wilson...yes the Porcupine Tree releases were 24/48. I believe the first 24/96 product was the live Anesthetize bluray.
    The first PT DVD-A releases were engineered by Eliot Scheiner. Wilson watched and decided he could do it better.
    Grace for Drowning was the first 24/96 product. I think he did his research and realized 96k was what was required as you write above.
    Good listening.

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  11. Well, I also in the past was kind of thinking that more could be better. Last year is optimized my audio chains a littel bit (although i have not much money to spend) and have to say that anything above 48k is overkill concerning audible frequencies. We can store our recordings at 96k to be safe, but for common usage 48k/44.1 is completely OK. The case with 16-24 bit is different, since theoretically the dynamic range could be perceived and morever 16bit requires dither which is artifical element, although inaudible under common conditions. 176-192 kHz is totally pointless, though.

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    1. You should always use dither when "bumping"to a lower bit depth. This is true for any change: from 32 to 24, and for 24 bits to 16 etc. It`s not only needed for 16 bits formats. Try varying the dithering algorithm and shaping options and tell me if you hear/can measure a difference or not. Long story short: why would you throw away bits if, when played back later, you artificially have to add those bits again? How can this be a smart thing to do?

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    2. Maybe yes, but dithering to 24 bit is not required in any recording since the distortions at 24 bit are very very low.

      I listened to various 16 bit dither and the best were either plain tpdf or naoki shibata ath curves - the low amplitude ones.

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    3. While potentially true that with a good enough quiet room and the right music (eg. the dynamics of a symphony), maybe >16-bit dynamic range could help especially if the music was recorded at very low levels... At least assuming a theoretically ideal ear.

      Reality of course is more complex and I suspect a decent 16-bit dithering would be more than good enough. The vast majority of music - essentially every analogue recording, most digital do not have noise floors below 16-bit anyways.


      Having said this, dithering is still "best practice" with bit depth reduction.

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    4. Dynamic range is one thing. The second one is that adding dither is artificial elements, although inaudible in most conditions. Given todays space possibilities, many users can allow to store 24 bit recordings without dithering. This also makes possible to better resample in the future to different rate in the future if required.

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    5. I repeat and correct my post: Dynamic range is one thing significant only for limited number of tracks. The second one is that adding dither during 24-16 bit reduction is artificial element, although inaudible in most conditions. Given todays space possibilities, many users can allow to store 24 bit recordings without dithering. This also makes possible to better resample in the future to different rate if required.

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  12. And one more thing. Whether the DAC oversamples or not is the matter of the DAC designer decision and they for sure can do that. But this does not say anything about storing/playback format, which should be fed to any compliant DAC available. And there, 48/44.1 is completely OK.

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    1. Honza, there are no audio DAC chips designers that choose whether or not they use oversampling. They all do (apart from the boutique NOS versions like Audio Note).
      It does say something as the recording -> playback chain is made a lot more complicated by doing your own downsampling. Remember: the DAC chip will always have to add the upsample step that you just so "smartly"removed. This is not a choice. For reference:
      http://www.analog.com/media/cn/training-seminars/tutorials/MT-017.pdf

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    2. Well i think for example many common conputer dacs like realtek alc's do not oversample. So it is really a designers choice.

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    3. Why do you think so? Their datasheets state otherwise.

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    4. Really ? I do not see very "deep" into realtek DAC technology but can you provide a link to datasheet where the oversampling is specified?

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    5. But anyway even if they do it is a matter of tehir onteranl construction, they should support 44.1/48 at the specified quality as they declare and not to require higher rates.

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    6. I have actually never measured a laptop or desktop motherboard with a non-oversampling DAC. It's a fair bet that practically every computer audio output these days oversample.

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    7. So the Realtek ALC's oversample? If so, can you provide a link to where it is documented? I have no problem in believing in it however would like to see some evidence.

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  13. And second more thing :), by expressing opinion to 16-24 bit I do not mean that 16 bit cannot be used well, actually the difference to 24 bit would be inaudible in 99 % tracks. But the bit rate could be more important than expanding frequency.

    Generally I would recommend to do and store studio recordings at 24/96 or 24/48, if space is not concern use 24/48 or 24/44.1 for playback, if space matters or CD is desired then with a peace of mind use redbooks 16/44.1. In either case our ears will not miss much, if anything :-]

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    1. Totally agree. There isn`t much quality difference between the 2 per se, yet saying it is totally irrelevant and can not be heard under any circumstance is a different thing. It`s been hard getting 16 bits 44k1 sound really good, doing higher rates is even more difficult..

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    2. it is not irrelevant technically but since content above 20k is irrelevant for human hearing, then 48k is plenty enough even for common playback filtering. 44.1 is a little bit on the edge but still manages well, the worst that can happen is very slight loss in the upper range which is not very significant anyway.

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    3. I agree with 48kHz. More and more of my vinyl rips these days are 16/48 simply as a natural downsampling of 24/96 (with dither). Any more bits IMO is wasteful.

      As you suggest, 48kHz provides a very reasonable transition band above 20kHz for filtering that I think if streaming services "insist" that we "need" hi-res streaming, they should consider just a "flat" 24/48 stream... Forget the MQA stuff with compromised bit-depth and ultrasonic reconstruction (which being ultrasonic is inaudible anyways!).

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    4. Yes, 24/48 format has great perspective :-), since it has no known practical limitations. As I wrote before, given todays space, I would go from 24/96 only to 24/48, not adding any dither and playback that way.

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  14. On the subject of dither, I did a little piece last year for a hifi magazine here in Australia demonstrating its effect by downsampling a short piece of music from 16 bits to 8 bits, undithered, dithered with vanilla noise, and dithered with shaped noise. Far easier to hear what's going on than when downsampling from 24 to 16. Piece is here: http://hifi-writer.com/wpblog/?page_id=4544

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    1. OK, I know about that. For me the most important thing is that adding dither is adding (although inaudible under most conditions) additional artifical noise, in some cases shaped. Given todays space available and nearly all DACs capable of 24 bit I do not see much reasons to dither to 16 bit except when redbook (CD) is required. In that case Naoki Shibata's ATH curves (I like ID=2 or ID=3 http://shibatch.sourceforge.net/ssrc/) are worth utilizing.

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  15. Whoops, should have added my name since I don't think I have a compatible 'Comment As" profile (and I hate anonymous commenting): Stephen Dawson

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  16. I guarantee you this will be the next 'talking point' from the hi-rez cheerleading squad:

    "A Meta-Analysis of High Resolution Audio Perceptual Evaluation"

    http://www.aes.org/e-lib/browse.cfm?elib=18296

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