A High-Resolution Audio Primer

January's Industry Update included a report on a scientific article presented at last year's AES meeting, in which the authors used test tones and a modest audio system (albeit in an anechoic chamber) to prove that listeners can discriminate between high-rez and CD-rez audio. This is important because scientific evidence of an audible difference between high-rez and CD-rez music is considered weak by some, even as anecdotal evidence grows stronger by the day.

The response to my Update—in the article's online comment thread and elsewhere on the web—was vigorous. Some accepted the result; others attacked it in varied if mostly predictable ways. In the real world, the noise floor will be too high to hear what they heard, said some. People don't listen to test tones, said others, so what difference does this result make? Here's the answer to that question:

It shows that high-rez and CD-rez audio are audibly different. If test tones sound different, then it's completely plausible—even likely—that music sounds different, too, although differences are likely to be harder to detect with complex stimuli like music. A preference for high-rez music becomes a more defensible position, from a scientific perspective.

As I pondered this, I recalled a recent paper I'd seen in the Journal of the Audio Engineering Society but hadn't yet read. "High Resolution Audio: A History and Perspective," which the AES has made available free online, does precisely what the title says: reviews the history of digital audio beyond CD-rez and frames the issue of high-rez audio's audible superiority on the basis of the available evidence.

If you're up to speed on the issue, little in the article will surprise you, but a few ideas are worth recounting.

Some of those who reject the idea that people can hear differences between CD resolution and higher sampling rates argue that, because people can't hear frequencies above CD's Nyquist frequency (22.05kHz), the presence or absence of those frequencies on an audio recording cannot possibly affect what people hear. But, as author Vicki Melchior explains, that's not really the point.

"A frequent misconception is that high data sampling rates assume the audibility of frequencies above 20kHz," writes Melchior, who has a PhD in biophysics from Yale, runs her own consultancy focused on audio DSP, and has served as vice-chair or chair of the AES Technical Committee on High Resolution Audio since its inception in 2000. The idea that direct perception of ultrasonic sound has "a role in normal audio listening has been rejected since about the late 1990s (due to lack of evidence) in favor of the ideas discussed in Secs. 4.4 and 4.3," Melchior writes. Section 4.3 is about changes in the dynamic range when the bit depth is changed.

Section 4.4, "Filtering and the Time Domain," gets to the crux of the issue: whether "filters with shorter impulse responses, gentler transition bands, and less ring"—all of which are facilitated by higher sampling rates—"might sound better than steep brickwalls." That possibility was suggested as early as 1984, by R. Lagadec and T.G. Stockham, in a paper establishing that pre-echo is clearly audible when fast and slow filters are compared (and when the cutoff frequency is well inside the audible range). "The steep roll-off filter produces a clearly [sic] ringing sound," Lagadec and Stockham wrote. "The effects which are present in such an exaggerated form in the experiments above also exist, though on a smaller scale, in today's digital audio systems. To which extent they are objectionable is still unknown, as the conceptual framework for such investigations has only now been established—a few years, unfortunately, after the promises of 'perfect digital sound' have first been uttered." Twenty-nine years later, we're just getting started.

Melchior concludes: "Current thought regarding the relationship of sonic transparency to high resolution is that ultrasonic frequencies are not involved in normal music listening, that the effects are not due mainly to lower hardware distortion, and that, besides dynamic range, the most likely issues are the anti-alias and anti-imaging filtering chains used to implement classic Shannon sampling." The main evidence that such filtering chains are responsible for the alleged sonic differences is the fact that "a broad group of design approaches that reduce time dispersion are reported to improve transparency compared to the sharp brick-walls typical of CD." "These filtering ideas," Melchior concludes, "are amenable to scientific testing but this has only recently begun."

Skeptics of high-rez audio, and of high-end audio in general, should note that Melchior's article, like many others in this field, cites, in addition scientific evidence—with its listening panels, blind tests, and rigorous statistics—a second kind of evidence: the judgments of recording engineers and other audio professionals. "[M]usic professionals with access to first generation data have widely reported subjectively better sound" with high-resolution audio, Melchior writes. In the world of perfectionist music production and consumption, such opinions constitute real evidence—not decisive, certainly, but worthy of consideration and respect. Given that the subjective judgments of those same people profoundly affect our industry's main focus—how recorded music sounds—how could it be otherwise?

jimtavegia's picture

Even my old ears can hear the improvement in 2496 and DSD in SACD discs. It is also clear that many enjoy the improvements in tape recording and playback from the increases in 7.5, 15, and 30 ips; in addition to LPs going from 33.3 to 45 rpm. Capturing more information is always better.

It is one thing to recreate a single tone sinewave with 16/44.1, but do do actual music with a wide range of frequencies, which may be adequate for some, there is proven results that a higher sample rate always captures more information more accurately.

I would also add that one must have an audio system capable of great accuracy and clarity to be able to discern the improvements, but it doesn't take $100K system to do it. Today's audio gear is excellent and affordable for anyone who cares about quality. I wish SACD had remained stronger in the market, but with USB DACS capable of DSD playback streams there are still many options out there.

Bogolu Haranath's picture

Wonder whether R2R DACs vs Delta-Sigma DACs, would make a difference for CD resolution? ..... EMM Labs DV2 DAC's impulse response (Fig.1), reviewed by Stereophile, shows no pre-ringing (pre-echo) or post-ringing (post-echo), with CD resolution :-) ........

jimtavegia's picture

Is certainly not high end...Yamaha S-1800 DVD/SACD player, 5 Sony DVP-NS 755 DVD/SACD players; and 2 Focusrite Scarlette 2nd Gen 2i2 USB I/O boxes good to up to 24/192 and I can hear the difference with any of them using my 72 year old ears. My son confiscated my Steinberg UR-22 24/192 interface for his Twitch on-line gaming audio I/O. He loves it using a SE Electronics V7 mic and Audio Technica ATH-40 cans.

I will admit that with most of the DACs today the good old CD can sound remarkable good, if mastered properly. The $300 CD players from today can run rings around the CD players of the 80's. I owned the first Technics player and was impressed mostly by the quiet backgrounds, but there was no doubt that the clarity (edginess) was there, and with an open mind one could see the potential. I still own my over 20 year old Sony DTC-690 DAT with the Pulse dac. I still think it sounds very good, and not just for its age. It recorded at 48 khz.

I bought a MidiMan Flying Calf so I could record at 44.1 khz into the Sony. I also bought the Flying Cow I/O card for my Win 98 Computer for getting it into my computer, They were 24 bit AKM devices.

I just entered the WayBack machine.

Bogolu Haranath's picture

It is possible, some of the newer model CD players could be using the minimum-phase filters (apodizing filters), to eliminate pre-ringing, like some of the examples I mentioned below :-) .......

Bogolu Haranath's picture

Hegel Mohican CD player ($5,000) uses minimum-phase filter (apodizing filter) which shows no pre-ringing (Fig.1) ....... HR and AD liked the SQ of the Hegel CD player :-) .......

Bogolu Haranath's picture

Another example ...... ATC CDA2 Mk2, reviewed by Stereophile, also uses minimum-phase filter and shows no pre-ringing (Fig.1) on CD resolution ....... KM liked the SQ of ATC :-) ........

Bogolu Haranath's picture

HR's reference HoloAudio DAC also uses R2R DAC and shows no ringing in impulse response measurements (compare Fig.1 and Fig.2, in measurements) :-) ........

Glotz's picture

The Chord Hugo M Scaler appears to bring much of the advantages of high-rez to CD/44.1 digital resolution sources. I'd like to hear JA1 chime in on the subject, as his review verifies the M Scaler adds missing data that the CD loses in creation.

glq's picture

The M Scaler and other upsampling filters do not, and cannot, "add missing data that the CD loses in creation", as you say. Any data lost in downsampling to the 44.1 kHz sample rate of CDs is gone. The only thing upsampling (upscaling) filters do, other than filtering, is to interpolate the existing 44.1kHz data up to the rate needed by the sigma delta modulator in the DAC.

Upsampling filters are always included internally in DAC chips, but those are normally a series of half-band filters. What custom upsampling/upscaling filters do is to bypass the chip filters and hand off upsampled data directly to the modulator. Accordingly, they can have a number of advantages: better design, greater accuracy, and lower noise by using, say, 64b floating point processors and dither.

Glotz's picture

Those were quotes made by Chord... so you would know better?

I do trust the manufacturer here. You are confusing upsampling with upscaling, in a very major way. This product's claims and performance differ drastically from an upsampling definition. Re-read the review if it's unclear how they differ.

But, since I Love Controversy..
Will someone from Stereophile review Chord Company’s ChordOhmic fluid, now released? It runs about $275 per small bottle and it is Sure to anger a ton of people! How delicious is that?!?

Glotz's picture

I was wishful thinking... JA1 stated that it doesn't add data at all. I am wrong, sir.

Thanks for your clarification.

brenro's picture

That the science of what the human ear can perceive is incomplete?

cjstone's picture

The link to the article, https://www.aes.org/e-lib/browse.cfm?elib=20455, 404s...

glq's picture


jeffhenning's picture

This is new?

After I got into digital audio recording 25 years ago, moving from the 16 bit audio card that I first bought to a 24 bit MOTU Firewire interface was not a small step up.

OK, the original Korg audio card was pretty noisy and a bit crusty sounding, but, even given that, the jump to 24 bit from 16 was very noticeable.

Then a couple years later, when I got a DBX Quantum digital dynamics processor that had wonderful 24/96 AD/DA convertors, the improvement in the sound was not as huge, but it still was very obvious.

From my experience:

• The 24 bits are way more important than the 96 kHz. The 256 times extra resolution offered by the 8 extra bits is huge as opposed to the 2 times extra offered by the higher sampling rate.

• Anything that can get the input and output filtering in the digital chain away from humans' audible range is a good thing

• Getting caught up in the method used is much less important than the end result for the user. I'm quite sure that just about any topology/methodology for digital conversion can be screwed up by poor design and parts.

• It always interesting to see tests done by JA of D/A convertors that offer different filtering methods. Not having heard any of them, I wouldn't mind giving them a spin to see which type of filter I preferred. My thought is that what works great for a CD might not be as great for 24/96 source and vice a versa.

I think that, at this point, it's very unlikely that we will ever see a D/A convertor that has both perfect phase and impulse response. We basically have perfect phase response nailed. Impulse? Not so much, but then begs the question...

...Forgetting whether it's attainable, is perfect impulse response necessary? The real life equivalent of the one-bit impulse used to test D/A converters do not exist. While it can be done digitally and offers insight, it is not analogous to any event we can hear let alone produce in air.

No musical instrument or speaker can create or reproduce a single sampling frequency cycle digital pulse. Our brains and ears are geared towards speech and music. A single pulse is most likely not audible to humans, but the distortions of the speaker rather than the signal itself (just a guess) will be.

It seems that every time I see the test of D/A's with multiple filters that the filters with the best impulse response have the most aliasing and distortion. What is the sweet spot between the two?

Final thought: We are really running out of space for any revolutionary improvement in digital audio. It has already exceeded the limitations of human hearing in the larger sense. With greater range in both frequency, dynamics and linearity than we can perceive, that's over.

It's been onto the micros for a while and those will offer much smaller returns as well as being much less discernible if at all.

JHL's picture

In every other component the results of very low energy storage and very fast recovery time is increasingly well-founded. It's also fundamentally more difficult to pull off, while at the same time whole camps have sprung up essentially denying it, which I emphatically disagree with.

I imagine the same is as true for front ends as it is all the way through to speaker systems. I know I've convincingly heard the essential benefits of very fast, linear recovery.

CG's picture

I have to wonder whether the overshoot and undershoot you often observe in amplifiers is audible in the very same way.

jimtavegia's picture

The extra bits from 16 bit to 24 bit are at the bottom, not the top. 0db is 0db whether at 16 bit or 24 bit. I think of it as a glass of water, once it is full (0db) it is full, nothing more to be captured. 6db times the bit rate is the dynamic range: 96db for 16 bit and 144db for 24 bit. Theoretical noise floor of the quietest transistor used to be -138db. Maybe it is better now?

The problem is not theoretical, but when you record and open up your mics in the venue we are all lucky to get -80 db on our meters with the inherent noise floor of the room and our mics. I have achieve -80db in my home studio with the HVAC off and my Rode NT-1As open (they have only 5 db of self noise) and I use some AC line noise filtering. The "room" is always the biggest problem.

I think of the sample rate like a loaf of bread, sideways seeing the slices. If I wanted to see (recreate) the uneven top of a new loaf more slices will give me a better tracking of the overall top shape of the loaf. I believe that this is why 96khz sounds better than 44.1 as we get a more accurate recreation of the waveform. In my wave form editor I can easily see a single tone sine wave recreated whether it be 44.1 or 96 khz; pretty easy to do. When we get to complex wave forms of music with many frequency tones piled on top of each other and at different loudness levels this is where a higher sample rate more accurately defines the music. You can see the change in shape of complex music if you streetch out the waveform in your editor. It is not the easy sinewave you saw with one tone.

192khz would be better if the DAC has the lowest jitter possible, but I can't hear it in my music recorded in my 6 track-Tascam DR-680 MK2 and it is due to both my decreased hearing acumen and the Tascam 680 AD. I have no idea what the jitter level of the Tascam is. (I'd love to get JA1 to measure it.) I believe that at these high sample rates low jitter matters.

I have not done it but I will bet that you could record a single, 500 hz tone at 320kbps and still see a sinewave in your audio editor. It cannot do or sound as good on complex music as 2496 or, with better hearing, 24/192 can. Just my 2 cents.

I read years ago that a well known recording studio in NYC had experimented with 16/96 and 16/88.2 recording and found they liked it as much as 24/96. It just left less work going back to redbook for their CD releases. They could not reach the theoretical bottom noise floor of 16 bit anyway.

I am ready to be schooled if I am wrong. At 72 I can be wrong.

fricc's picture

Hi Jim, DACs interpolate the samples before turning them into analog signals, otherwise your music will sound like a staircase :)

Good reconstruction filters use a sinc transfer function, which perfectly interpolates the signal, giving you a perfect reconstruction. Unless the DAC's filter is broken, there will be no difference in the reconstructed 20Hz-22kHz signal, regardless of the sampling frequency.

So, unless your hearing is as good as that of a bat, you won't be able to appreciate differences.

As for the bit depth, 24 bit will possibly give you an audible advantage to 16 bit. Our hearing can indeed be sensitive to signal differences of 118dB or so, which is quite a bit more than the 96dB that 16 bit allow for.

See the following for details about sampling theory and signal reconstruction:
- https://en.wikipedia.org/wiki/Digital-to-analog_converter
- https://en.wikipedia.org/wiki/Nyquist–Shannon_sampling_theorem

jimtavegia's picture

To sound decent, humming bird stuff for sure, but there is more to it than that, or all CD players and DACs would sound the same. Certainly redbook playback has improved in 40 years. And if we went backwards to MP3 quality many find THAT acceptable, even to 128, 256, and 320 kbps. I have been trying to listen to a steaming site and just could not enjoy it due to what is lost in MP3 quality. The best they do is 320kbps, for a Premium membership.

I can tell you that myself and many others can and do love 24/96 and can hear the improvement over redbook, without a doubt. I wish I could afford to get into what Chord is doing, I love the JA1 piece about the Chord UPsampler, but it is out of my league. Clearly people are not paying those prices for their DACs and not hearing the improvements, even without their up-sampler.

When I can hear the improvements with just my up to 24/192 $150 USB Scarlett and Steinberg boxes I can only dream what the Chord products can do. This year I will buy a 3rd gen Focusrite Clarett ($450)and run it Thunderbolt into my MacMini. I don't expect a night and day difference, but those I know who own them are not disappointed. And there is an improvement in the mic preamps as well for those who will use them. There are many folks spending good money on USB dacs for streaming, many way more than just the cost of a CD player.

It is all for the 1%'ers who care about hearing the most they can. I buy improvements that I can hear and see in my Sony Sound Forge. And often the mastering will overly compress, raise the gain and then apply sharp limiting just to be the loudest, and all the recording work will be lost.

Bogolu Haranath's picture

Denon DCD-1600NE, CD/SACD player ($1,200) has built-in upsampler :-) .......

JamieAngus's picture

Hi Fricc,
It seems simple doesn't it? Just use a sinc impulse response, which is equivalent to a "Brick-wall" low pass filter, for reconstruction. As you state, this works perfectly as proposed by Shannon.
Unfortunately, such filters are neither realisable or usable. (we would have to wait a long time for our audio!)
Can I direct you to the article "Modern Sampling: A Tutorial" from the same AES issue cited in this article. (It is also open access) which discusses this in some detail and suggests alternative ways of achieving perfect reconstruction of audio signals.

absolutepitch's picture

Your explanation of the additional 8 bits in 24-bit recording being at the bottom end appears to differ from another explanation I've heard, that of the bits being extra headroom above 0 dB. So in the former (your) explanation, the 0 dB level is maximum voltage recorded, say that is all 1's in the 16 bit word (11111111 11111111). In the latter explanation, it's the least significant 16 bits out of the total of 24 bits that are all 1's with the higher bits all 0's (00000000 11111111 11111111). Both these apparently assume the same voltage step between bits, and that 0 dB is at 11111111 11111111.

So why wouldn't setting 0 dB at 24-bit high (11111111 11111111 11111111) result in greater resolution than setting the 0 dB at 16-bit high (11111111 11111111), since there are 256x more steps in 24-bit word than 16-bit word for the same voltage swing (setting the voltage steps in 24-bit smaller than the steps set in 16-bit)? I'm aware that this description isn't entirely accurate, because the voltage swing is both positive and negative, but it gets the question asked relative to what you see on a recording "VU" meter, and the assumptions made for dynamic range being calculated from 6 dB step/bit.

jeffhenning's picture

The extra "bits" (in this case 8) subdivide the audio signal by a factor of 256 equally from top to bottom.

16 bit audio runs out of resolution at -96dB because it's reached its limit. Any signal below -96dB is too small to resolve. 16 bits gives you 65,536 possible levels. Since you have to render both positive and negative, that gets cut in half to 32, 768 levels for each side of the wave. The entire wave, though,

Also, there are a few other things to consider. Human hearing is logarithmic, not linear in respect to volume levels. Digital audio could be, but it is not at present. It's linear.

The reason you can measure further down into 24 bit audio is because that first of the 16 bits above 0 just got divided into 256 more bits and so did its 15 other brothers above it.

24 bit audio breaks the signal down into 16,777,216 discrete sample levels that are equal in size from 0dB to -infinity dB, but it's done using voltage not dB's.

cyborlink's picture

Although somewhat complex, I found the explanation by Hans Beekhuyzen to be very well presented and shows the logic in why we 'hear' or perceive a difference. The video starts at 10:59, although watching the full video many be of interest.



Archimago's picture

Remember guys, even for 16/44.1 CD-resolution digital, the resolution is NOT 1sec/44,1000 = 22.68μs like what Beekhuyzen says in that video. This has to be the most misunderstood, oft repeated inaccuracy in audiophilia.

The answer is closer to 110ps. That's PICOsecond folks, or 10^-12, not 10^-6.

Here's the math:

Ortofan's picture

... high-resolution recordings (as in greater than Red Book) is considered to be of significant benefit, then why isn't JA1 measuring speaker frequency response beyond 30kHz?
Likewise, why aren't most, if not all, speaker manufacturers incorporating super-tweeters into their products?

Suppose that your $200K audiophile speakers have a frequency response as shown in the graph in the following link:
How much "hi-res" do you have any chance of being able to hear?

tonykaz's picture

Of course, people will continue to choose higher quality in all things. ( almost all things, the world doesn't give one hoot about 33.3, no matter how beautiful the new stuff is )

All audio is now recorded in high resolution digital which played back thru a Mastering Studio's System shames a typical Audiophile System. ( Pro's vs. amateurs )

Recorded Sound Quality & Playback have made gigantic advances in these last 5 Decades, the Transportation Industry understands that Sound Quality will continue to advance for the foreseeable future.

I see inflation adjusted prices dropping with percievable Sound Quality increasing . ( at amazing rates )

I remain satisfied with 2011 Redbook quality levels but I'm an old geezer, buggy whip vintage sort of guy.

Tony on the Super Tuesday Trail

ps. a nice piece of Wire will yield greater improvements in Sound Quality than high res. with lesser wire.

Amphibica's picture

I have CDs that sound great. I have CDs that sound, well, bad. There are "audiophile" YouTube channels that sound amazing, while the rest of YouTube sounds dull

This makes me think that ENcoding (recording) is more important than the DEcoding (playback), or least the FORMAT in which the recording is preserved is not the issue

pat mcginty's picture

HD audio can present your speakers with both resolution and dynamic range that they cannot transduce.

Back in the 80s the arrival of CD raised the dynamic range bar for speaker makers from the 75dB necessary for LP to 103dB.  A huge jump. Many of us met the challenge with the existing technology, but barely. It was a stretch.

With HD, the bar has been raised again. And again by a huge amount. The available dynamic range now surpasses that of human hearing, 140dB, so it is effectively limitless. Let's say we choose to use 110dB or more, that's greater than four times CD's dynamic range. Good luck meeting that challenge with the already maxed-out conventional speaker technology. 

For the new level of detail resolution, conventional speakers stumble largely because of their signal-smearing, amp decoupling passive filters.

If your speakers barely do CD resolution and CD dynamic range, applying an HD signal won't do you much good.  Don't waste your money. 

Happily, the technologies for successfully transducing HD are at hand and are about to upset the whole darn audio apple cart :-)

Apesbrain's picture

"The 24 bits are way more important than the 96 kHz. The 256 times extra resolution offered by the 8 extra bits is huge as opposed to the 2 times extra offered by the higher sampling rate."

Where exactly does this myth come from? 24-bit does NOT offer more resolution than 16-bit. It offers more capacity (i.e. higher volume), but not more resolution. Think of it as pouring 14 oz. of water into a 16 oz. container vs. a 24 oz. container. There is NO CHANGE IN RESOLUTION! The analog waveform drawn from the data is exactly the same.

glq's picture

The reference is to amplitude resolution. The greater the bit depth, the lower the noise levels due to truncation and (required) dither. The result is higher SNR and greater precision in the digital word. The more precise digital word is a better representation of the original analog waveform, which is why it's higher resolution.

The process has limits for recorded music due to the noise present in the analog signal at the time of digitization.

Glotz's picture

One would be increasing mass, not volume.. The volumes would be the Same.

The higher-bit rate song would be longer otherwise.

And it would follow logically, that greater mass would equate to higher resolution. 24-bit does have more information, and therefore more resolution, than 16 bit.

jeffhenning's picture

In its most simplistic form using "0" as no voltage and "1" as the maximum that a digital converter can handle (0dB):

• From 0 to 1 volt, with 16 bits of resolution, that signal is broken into 65,536 equal values

• With 24 bits, the same signal is broken into 16,777,216 equal values

If you can't understand how that impacts the resolution of the audio signal, I really don't have anything left to say.

Bottom line: you do not know what you are talking about.

Do some research.

rt66indierock's picture

High resolution lacks one important thing to succeed. It doesn't have enough customers. I've said for several years to people in high end audio you will find high resolution audio a very difficult sell.

Fancy marketing reports, AES papers and endless promotion aren't moving the needle. Maybe the evidence isn't compelling to potential customers?

Dick James's picture

There are clear advantages to recording with 24-bits, but playback with 16-bits is plenty. There are no recordings with a 96 dB or higher dynamic range that require 24-bits for playback. It's just wasted disk space, streaming bandwidth, etc. The higher sampling rates help lower phase distortion (i.e., flatten group delay) in the audio frequency range because the filters involved can have a slower roll-off and that makes the higher rates worth while.

AJ's picture

The funniest part about this particular form of OCD (that the afflicted can't demonstrate on their own systems without peeking/mismatched voltages, etc, etc, etc, etc) is that those systems are still archaic stereo. In 2020!
A flawed, completely artificial studio construct, that "sampling rates" can't possibly fix.
But alas, missing the forest for the tree is what OCD is after all ;-).



AJ's picture

Audiophiles are now accepting the results of blind tests???

Jack L's picture


I am an audio fan for sure though I'd not be qualified as an audiophile due to my cheapie humble home-brew audios.

I do NOT believe in any blind tests as there are hardly any repeatable valid test results due to the room acoustics etc etc where the tests are carried out.

A strong case was established end last years by Audio Engineering Society, New York & Hiroshima City University, Japan in blind tests for
the audible difference in a lab condition: in an anechoic chamber, 7 young males/females (sharp ears), white noise encoded with 16bit 44,1KHz,96KHz & 192KHz. No music was used due to the inconsistency of the music contents which would invalidate the outcome.

The result came out promising: the young people there did hear the sonic difference of the white noise test signals encoded with different sampling rates.

So how can we untrained ears of generally more advanced age can detect the sonic difference CONSISTENTLY of test music in normal room conditions??? Too much uncontrolled conditions involved!

Listening is believing

Jack L

jdt1's picture

I am not an audio engineer or consider myself an audiophile. However, having been a musician, I believe I can knowledgeably comment on music reproduction. With the “I can hear hi-res and you can’t” argument, I have a simple solution. Let’s have all the CEO’s of the companies that will benefit the most if hi-res were to gain broad public acceptance, take a sound comparision test. Jay Z and Tidal can lead the way and prove once and for all that there really is a difference in sound reproduction quality between a standard CD and a hi-res file. Qobuz, Amazon (I have Ultra HD and you don’t) and HD Tracks CEO’s can all follow suit. Also, let’s have the editors of this and other quality industry publications take the test. An independent lab can conduct them and publish the results. What do you say Stereophile?

Jon T

jimtavegia's picture

validate the "Improvement" and benefit for those who care about higher sample rates. Here in my studio I can hear the benefits of 16/96 as well.

There is often the time that when I am leaving recordings in redbook as that is how I recorded them, as I don't have the best software or a Levinson to reduce higher quality files to redbook in a pristine way as JA1 can. That is how my clients listen anyway and not on esoteric gear.

When Sony and Philips designed the redbook standard and their first players the hardware they offered, often did not meet the full 16 bit standard of the "promise" of perfect sound for ever. Many players did not even recreate 12 or 14 bit performance. I can also recall some reviews of players of the last 5 or 10 years that did the same, some more than $1k in price.

The issue of "filters" and time domain performance and the uncovering of Jitter issues was key to finding ways to even meet the standard of 16/44.1. We have all read reviews of players costing thousands that had poor jitter rejection. Certainly DCS is one of the companies that leads the way, at a price, in trying to make redbook sound the best it can.

For way less money I can record 24/96 files with the most affordable gear available to anyone that cares, and then burn those files with Cirlinca or Sony DVD Architect to DVDs and listen to them in any of my DVD/SACD players and hear more. I do not hear that what HD Tracks, BlueCoast, eClassical, Primephonic, sells me as 2496 files and I burn them to DVDs as just snake oil or folks out to make a quick buck on audiophiles who mistakenly "think" they hear more. I don't need to do double blind tests and the differences are not so small as I am trying to trick my self into believing they "have" to be better, they do sound better.

I also don't believe that those who record in DSD are wasting their time in trying to capture more, as redbook is good enough for the masses. They spend great money on better and better microphones to capture more, and often are the same mics they used for redbook recordings they released years ago. Many of you who bought the Mozart K-622 Project that was released as a CD, SACD, and as nicely pressed LP should be able to determine which format you enjoyed the most. There was a Stereophile piece from August 1, 2004 That started me on this quest to find out how I could make better recordings. I printed out the whole article and have read it many times about the recording process. I grew to greatly respect both John Atkinson and Tony Faulkner for their work in making the best recordings they could and have followed them closely ever since. The article is long, but well worth the read and study. It was the starting point for me and the beginning of using many digital recording devices of the last 15 years. I would urge a read for those interested and those of us who have copies of the LP and the SACD and also listen to the CD layer have enjoyed them immensely. At that time I had two CD only players in my home that were over $1K in cost to be fair to all formats. Understanding now that much has changed in 16 years in the quality of digital recording and playback, especially in 24 bit PCM . Still Recording in DSD is an issue for a hobbyist as the cost to master is out of the question. Affridable recorders are out there for sure from Tascam's DA 3000, Sony's PCM D-100, and Korg's MR 2000S.

I have read the Chord M Up-Sampler review 3 times now and am amazed at what has been learned and applied, when more is to come to gain improvements in the future and those who I trust who do hear the differences/improvements of redbook. None of this comes without a cost of which, sadly, many do not care enough about as the for the masses redbook or less is enough, convenience reigns.

It is OK for us to disagree and I accept the comments for my "foolishness". I record and I listen and choose what I feel is the best sound. I don't live in a theoretical world, but I can listen to flawed recordings. I just choose not to for the most part.

jdt1's picture

The limitations of human hearing must be taken into account when discussing hi-res reproduction. Why pay for it if you cannot hear it?

Instead of a sine wave played in a sound chamber, I prefer learning what experts in the field believe and prove. Chris Montgomery is the well-known and respected creator of the Vorbis audio codec and the Ogg container format. He states, "The more that pseudoscience goes unchecked in the world at large, the harder it is for truth to overcome truthiness." If interested, you can read more from Mr. Montgomery at xiph.org.

David Harper's picture

Forgive me if you've already mentioned this but specifically what speakers are you listening to thru which you are so convinced that you can hear the sq difference between CD and 24/96? I can only assume that they are something better than dynamic drivers in wooden boxes which lack the resolving power to make this sq difference audible. Quad or M.L.Electrostats? Full-range ribbon drivers? Electrostatic headphones? And I trust also that you're sure that the difference you think you hear is not due to a difference in the quality of the recordings? Maybe the hi-res version was mastered differently? It is, I believe, pretty much common knowledge that the quality of the original recording and mastering eclipses completely any s.q. difference that bits and sampling rate can possibly have. I am no expert but I have listened carefully and compared a CD to a 24/192 bluray disc of Supertramp's "Crime of the Century"on my magnepan LRS ribbon speakers and try as I might I can discern no difference in s.q. I concede that your ears may be better than mine but all of this begs the question that if this difference in s.q. is so insignificant that I'm unable to hear it at all no matter how hard I try, what can possibly be the justification for hi-res other than the endless capacity of audiophiles to experience the placebo effect?

jimtavegia's picture

I have had a number of "wooded box" speakers here in the house over time, many at least in the class B Stereophile ratings, and some in the lower class A, and yet even Mr. Atkinson is using affordable bookshelf models and can hear the improvement. I still can hear it from my refurbished Large Advents and AR-58's, to newer bookshelf models.

I just download an 88.2 version of a favorite song of mine that I have in CD, LP, SACD from MoFi, and then the HD tracks version that is the 88.2 khz; They all have HF extension to 19khz, but differ greatly in mastering EQ across the band and all have LF extension into the 30hz range. The LP version is also problematic in it depends upon the phono cartridge and the phono preamp, yet I do not have esoteric models of either of those and I can hear the differences in all three formats quite easily, but what I prefer is the one that has the more flatter EQ mastering, and I am not going to tell you which is which, but it is clear that the "Mastering" really does matter greatly and I can pick out more from the higher rez versions. I was pleased to see that the mastering of all these formats did not kill HF extension. I am going to look at the FFT components of each of these and see what else I can find out about EQ distribution. A well mastered cd could possibly sound better than a poorly mastered SACD or high rez file. Mastering makes a world of difference.

I do wish I had a DAC in which I could change the filter setting as I do believe that the Chord M up-scaler has hit upon what the true "mountain to be climbed" is for high rez improvements, and even in improving standard CD playback. I believe that the jitter issue, for the most part, is solved and a none issue in most gear.

Comments after yours are a case in point with the Cambridge DAC Magic being able to change filters and improve sound. For under $500 it has gotten good reviews.

As someone who does recording in all formats I have great experience at hearing first generation high rez up to 24/192 from sessions when I come home. I do hear better up to 2496, 24192 in my Tascam DR-680 Is not as discernable and I would like to know why and will be talking to Tascam this week about why that might be. The differences in DSD/SACD are easier for me as it is a totally difference digital format with its own set of issues to overcome, and mastering gear costs are out of my league. My goal would be to buy a Tascam DA-3000 for location recording in 2 channel and then bring that master home and mix down to two channel in what ever format I need. SACD is my favorite and sorry it did not catch on. I can't afford a Sadie work station so I must stay in the lower rent district.

There is a great interview with revered engineer Tony Faulkner on youtube I would urge you to watch. When a Master speaks I listen and what he has to say is well worth the time. Every time I go out to record the "room" decides much about the final sound as does the ambient noise of the room that I cannot control. I was also pleased to hear that Mr. Faulkner was going to by a Tascam 680 for some of his film work. Nice to know I buy right once in a while. I also have 2 Tascam DR-40's that are portable 2496 SDHC card recorders as well.

David Harper's picture

So then I assume that you don't believe that the 50 year old technology of dynamic cone drivers in wooden box speakers (like your primitive Advents) does not approach the resolution and transparency of the newest electrostats and planar ribbon speakers? And if some stereophile writer says so then those speakers must be state-of-the-art in sound quality?

jimtavegia's picture

if anyone who owns speakers in Stereophile Class A cannot hear improvements in properly recorded and mastered high rez files. What I find off putting is that those with no recording experience or desire to give it a go as so bent on reading "articles" claiming high rez is all hogwash, yet never do any recording or testing on their own.

I hear what I hear, even with my own modest gear, as I have tried it, recorded with it, and find it an audible improvement over redbook and many lps, so I use it. I find it odd that those who can't are so upset with those of us who do hear an improvement. Once I buy a better dac for my transport I am sure it will be even better and will improve my enjoyment of my CD collection as a bonus.

Those of you who don't record have never heard 1st generation recorded music so I am not surprised at the angst over our positions. Sad.

I do find that my AKG K 701 headphones are very revealing and give me all the resolution that benefits my work. Certainly not the best, but I have no interest in getting on the merry go round of $2k headphones.

Bogolu Haranath's picture

Reviewing the Quad Artera Solus, HR said 'it sounds superior' on CDs ........ It costs less than the Chord M Scaler ....... Artera Solus is also an integrated amp which can drive bookshelf speakers ...... It has multiple filters and can decode hi-res digital :-) .........

jayg's picture

I had never paid much attention to the debate over DAC’s and hires-files. Then I got DACurious. I started doing research and reviewing the reviews. One piece of equipment emerged from the background noise, the Cambridge Audio DAC Magic Plus. I bit; I chewed; I swallowed my pride. The difference was transformative! And for all of $350 I got a DAC, a headphone amplifier and a preamp! If only I had done this earlier! The dynamic range, the silence, the air and resolution improved to a degree that still leaves me sometimes breathless. I wish I was selling these guys. I just listened to the CD, VANGELIS DIRECT. Whoa, what was that “instrument?” Literally, I did not hear it before, at least the presentation had changed so much that it’s character was completely changed. This was electronic music, not instrumental! Sign me up; beam me up Scotty! I have sampled it and I am sold on this brave new world that was hiding in plain sight all these years. I don’t have a super high end system but a thoughtfully curated one. John

Bogolu Haranath's picture

Cambridge Audio DAC Magic Plus was reviewed by Audio Science Review, among many other audio magazines .......

What filter are you using for your listening? :-) .......

jayg's picture

B.H. I listen mainly with linear but I do need to experiment more than I have. I am just so happy with it.

I would like these additional features.
1) I would like a remote. The unit supports Bluetooth through an optional Bluetooth antenna. I would think the firmware could be updated to allow this through an App.
2) I would like the ability to switch to a mono mode.

3) I should also say that I do a lot of streaming and sourcing from a CD player. I have not used any “hi-res” files, yet. There are at least a couple of hi-res formats that the unit does not support. Check out the Cambridge Audio website for more details about file compatibility.


eriks's picture

I've been saying this publicly for a while, with almost exactly the same time frame, ~ 15 years:

The question of whether hardware performance factors,possibly unidentified, as a function of sample rate selec-tively contribute to greater transparency at higher resolu-tions cannot be entirely eliminated.Numerous advances of the last 15 years in the design of hardware and processing improve quality at all reso-lutions. A few, of many, examples: improvements to the modulators used in data conversion affecting timing jitter,bit depths (for headroom), dither availability, noise shap-ing and noise floors; improved asynchronous sample rate conversion (which involves separate clocks and conversion of rates that are not integer multiples); and improved digi-tal interfaces and networks that isolate computer noise from sensitive DAC clocks, enabling better workstation monitor-ing as well as computer-based players. Converters currently list dynamic ranges up to∼122 dB (A/D) and 126–130 dB(D/A), which can benefit 24b signals.