There is no point to distributing music in 24-bit/192kHz format. - <AUTHOR> http://people.xiph.org/~xiphmont/demo/neil-young.html <comment><USER3> I must say I get rather irritated when people spend time worrying about dubious 'tweaker' methods to improve their audio, when the most under- performing component of most people's sound equipment also has the lowest- hanging fruit: The room itself.

When people ask me where they should spend money to improve the quality of their hi-fi or home theater system, in nearly every case my response will be something like "get a thicker rug" or "put something on this wall to absorb sound reflections, even if it's just a bookshelf."

Beyond that, I'd tend to say something like "stop being so paranoid about what you think you can't hear, and enjoy the damn music."

<comment><USER4> Agree one-hundred percent about the room (although the prescription isn't always as simple as "get a thicker rug" etc).

The other issue regarding high-frequency sound reproduction is that in most cases, the loudspeaker won't be outputting much beyond 22-25 kHz (assuming very good quality loudspeakers, cheap consumer grade units might struggle to hit a -6 dB point at 18 kHz) and even for the speakers that have usable output at that range, the directivity at those frequencies will be so narrow that your head will have to be locked in the perfect "sweet spot" to hear anything.

<comment><USER5> I don't really know much about this, but wouldn't the 22kHz sounds potentially create beats in the lower frequencies?

<comment><AUTHOR> Acoustic "beat tones" aren't "real" tones— you hear them because of non- linearies in the ear-brain system, but you have to hear the initial tones first. (Well, unless you're talking >>130dB SPL levels where the air starts becoming non-linear, but then lower frequency recording would capture it fine)

If you could hear subharmonic beats from ultrasonics then it would be _very_ easy to demonstrate, alas.

<comment><USER6> Curious, what does non-linear mean in this context?

<comment><USER7> IIRC, linearity is when you put a sound wave frequency into the medium (air) a some point, you can predict the frequency of the sound wave at some other place using a linear function - meaning that there is no distortion. Non- linear is when the physics of the medium starts screwing with that function.

</comment></comment></comment></comment></comment></comment><comment><USER8> There's a lot of scientific-sounded content in this, but unfortunately most of it couldn't be further from the truth. I'm an ex-audio engineer and studied digital and analog audio engineering; this has been debated to death over the last 15 years.

Digitally recording a triangle is the best example of why 48kHz is very limiting. The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 5kHz and of many very high-pitch harmonics. Most of these harmonics are above 20kHz. The harmonics are what makes it sound like a triangle, not the frequencies below 20kHz. This is why the triangle is one of the hardest instruments to digitally record. It always sounds like crap.

In theory, it's true that the human hear can't hear above ~18kHz, but it can hear the influence of the very high pitch harmonics on a lower frequency.

EDIT: here's more data backing what I said <http://www.cco.caltech.edu/~boyk/spectra/spectra.htm>

EDIT 2: typos, frequency mistake

<comment><USER9> The linked article was accurate. You are confused.

"I'm an ex-audio engineer"

Hard to believe.

"The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 10kHz"

That's a pretty high note - higher than the top key on the piano. But an "audio engineer" would know that.

"many very high-pitch harmonics"

Since the next harmonic after the fundamental would be at 20khz, which only young people can hear, and none of the others are audible to any human, I don't understand what you are talking about.

"Most of these harmonics are 20kHz."

OK, you don't either.

"it can hear the influence of the very high pitch harmonics on a lower frequency."

Sure....

<comment><USER10> You clearly have little to no musical background, and think that your basic math skills are a substitute. The overtones present in a cymbal or triangle are not straight multiples of the fundamental, they are chaotic, and are very important in determining the timbre. Anyone (and I mean that) can easily tell the difference between a cymbal with and without a low-pass filter with the threshold around 22kHz, because these "inaudible" frequencies are lost.

<comment><USER11> If anyone can hear it, then surely it must have been verified through a double-blind test. Can you provide a citation?

<comment><USER10> I don't know of any to point you to. They probably exist, but I haven't read them. Let me know if you stir some up.

</comment></comment></comment></comment></comment><comment><USER12> He raises a lot of valid points. However...

192 kHz is clearly overkill for listening. Not so for further editing of the data.

Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible.

44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there.

That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data. However, you would never do that. You would digitally process 192kHz all the way. Only your loudspeakers or ears would introduce a high-pass filter, and a rather bening (flat) one at that. There is certainly no aliasing going on there unless you resample (wrongly). Intermodulation distortion is not the fault of the sample rate.

I mayored in hearing technology. Calling 192/24 _worse_ than 44.1/16 is total BS. How useful it is is a different debate.

<comment><USER13> _Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible_

No, the difference is not audible at all. At 16 bits of depth on a normal low- level audio signal (~0.3 volts), we're talking about less than 0.000005 volts per amplitude step. This difference gets lost in the THD already at the DAC in your audio output stage. Then it gets lost again in the amplifier. And again in the cable to your speakers or headphones. And then it gets lost again in the speaker elements. What survives in a normal low-level audio signal is about 14 bits of resolution.

 _44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad. A bit more headroom is well needed there._

44.1khz IS a bad sampling rate for accurately reproducing anything except a triangle wave or square wave above 5khz.

<comment><USER14> Heh, it's funny to see this late-nineties debate get re-hashed here. Also kind of fun.

If it were true that there's no audible difference between 16 and 24 bit, companies like Alesis, Otari, ProTools, etc. wouldn't have spent the last 15 years ditching 16 bit like an old pair of smelly sneakers. (better metaphors welcome).

Seriously, anyone who has sat down in a real listening environment for 5 minutes A/Bing 16 vs 20 bit, 16 vs 24, etc. hears the difference immediately. There's no question. This is why you can buy ADAT 16 bit 'blackfaces' for $100, down from their original $4,000.

<comment><USER13> Yes, and anyone who has ever sat down infront of an LCD flatscreen watching their favorite movie on DVD/BD using gold-plated $200 HDMI cable instead of $4.99 Walmart HDMI cable see the extra sharpness immediately. This is why non- gold plated non-OFC HDMI cables are down to $4.99 a piece from their original $49.99 during introduction.

<comment><USER14> That's cute. Obviously you've never recorded a rock band while riding the pre to compensate for 16bit's terrible noise floor and horribly limited headroom. You've never had the joy of ruining a perfectly good take because of that wonderful sound it makes when the volume spikes into digital distortion despite compressing the wazoo out of the input source. Glorious sound, digital distortion. Run a dentist drill through an old Speak & Spell and you'd just about have it.

You've never rented an expensive tube EQ during a mix to cover up 16bit's grating harshness from 10k to 15k. Or tried like mad to make the bass drum sound like a freaking bass drum and not a pie pan slamming against the back of a plastic trash can. And yes, we had good mics and pres, all standard studio stuff. Decent, not brilliant, converters, but it was the 16bit that was the problem. Getting those 20bit XTs for the first time was like walking into the Promised Land.

Sure, there's lots of marketing ploys out there, lots of snake oil. Moving up from 16 bit was _not_ one of them.

<comment><USER15> Did you read the original article at all?

> Professionals use 24 bit samples in recording and production [11] for > headroom, noise floor, and convenience reasons.

...snip...

> Modern work flows may involve literally thousands of effects and operations. > The quantization noise and noise floor of a 16 bit sample may be > undetectable during playback, but multiplying that noise by a few thousand > times eventually becomes noticeable. 24 bits keeps the accumulated noise at > a very low level. Once the music is ready to distribute, there's no reason > to keep more than 16 bits.

The original article does say that yes, during recording and production, 24 bit audio gives you a lot more room to play with. That doesn't mean that you can hear the difference between 16 and 24 bits for the final recording; just that 24 bits give you more room to keep out of trouble during production.

<comment><USER14> Did you read the comment thread at all? I wasn't responding to the article, I was responding to a comment:

...snip...

>Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible

No, the difference is not audible at all.

...snip...

</comment></comment></comment></comment></comment></comment></comment><comment><USER4> For those of you who are interested in just how much of a golden ear you truly are: download Harmon's "How to Listen" software for Windows or Mac OS X <http://harmanhowtolisten.blogspot.com/> (scroll down).

Harmon requires its trained listeners to pass tests based on this software before participating in juries to evaluate Harmon products. It doesn't directly address the sample rate/bit depth issues discussed in the linked article, but it does address a lot of the issues brought up in the HN discussion, so you can have a chance to see how much those characteristics really matter.

You may be surprised.

</comment><comment><USER16> Even without debating the science and signal processing arguments raised...

 _In any test where a listener can tell two choices apart via any means apart from listening, the results will usually be what the listener expected in advance; this is called confirmation bias and it's similar to the placebo effect. It means people 'hear' differences because of subconscious cues and preferences that have nothing to do with the audio, like preferring a more expensive (or more attractive) amplifier over a cheaper option._

 _The human brain is designed to notice patterns and differences, even where none exist. This tendency can't just be turned off when a person is asked to make objective decisions; it's completely subconscious. Nor can a bias be defeated by mere skepticism. Controlled experimentation shows that awareness of confirmation bias actually increases rather than decreases the effect!_

Doesn't that completely negate his conclusion, that there is no point to distributing 24/192 music? If people want to pay for 24/192, and even he just admitted that they will legitimately enjoy it more, how can you conclude there is no point?

Life is short. I want to enjoy things. Whether or not my enjoyment can be quantified or scientifically defended, I really don't give a shit. But that's okay, if you don't want to sell me 24/192 music, Amazon will. Between this and DRM-free content, it's no wonder I buy all my music from Amazon these days.

<comment><USER17> > _If people want to pay for 24/192, and even he just admitted that they will > legitimately enjoy it more, how can you conclude there is no point?_

Sorry, no time to reply. I gotta run and write up my biz plan to distribute 32/384 audio.

<comment><USER18> SUCKER! I'm already working on 48/768 audio. It's amazing how clear the recordings are.

</comment></comment></comment><comment><USER19> This article is one of the most lucid and accurate that I have read on this topic.

However, one thing that's missing here (and in nearly all other similar pieces) is a full discussion of the prerequisites of the sampling theorem. For example, the signal must be bandwidth-limited (and no finite-time signal can be).

But this is a minor concern, as there are many elements in the analog domain of the recording and playback chains that serve as low-pass filters - <USER2> with the mics. So bandwidth-limiting is effectively achieved.

For a similar reason, the discussion of the "harmful" effect of high- frequencies to playback electronics and loudspeakers to be a bit overdone IMO. Peruse the excellent lab results of modern audio gear on Stereophile's web site. You'll find that bandwidths exceeding 30kHz are rare.

One last thing. When doing subjective "testing," keep in mind that what some folks are hearing may be limitations of their gear. For example, most DACs derive their clocks for higher sampling rates (88/96/176/192) by clock- multiplier circuits. IOW, 44kHz and 48kHz are the only ones clocked directly by a crystal. These multiplier circuits are often noisy, contributing to jitter. The audible effect of this jitter is hard to predict.

Bob

PS As an avid audiophile, I find the clash of subjectivists and objectivists on this normally-buttoned-down forum to be a bit of a trip.

</comment><comment><USER20> You always record stuff at 24-bit/192 kHz for many reasons usually involving minimizing analog artifacts and to give you a lot of information to work with. You use 32-bit float wavs to transport stuff around so you don't have to worry about normalizing levels and clipping. Lossless formats drastically improve the quality of transients by an enormous degree. But every single objection to this is either ignoring the points of the article, or talking about the benefits of recording at high fidelity, when this entire article is pointing out that once you have _finished a mix_, there is no reason to distribute things in 24-bit/192kHz. Most speakers can't even play about 20kHz anyway, which makes the entire point moot. I don't care if you have a bajillion kHz, the speakers can't play about 20 kHz, so your screwed.

<comment><USER21> You're getting two entirely different things mixed up.

192 kHz is the sample rate. 192,000 slices per second. It does not refer to the audible sound spectrum.

20 kHz in speakers refers to the cycles per second of the audible waveform. Normal human hearing rage is 20 hz - 20 kHz. For most people, it's less than that.

A speaker can certainly play back music sampled 192,000 times per second. Most of them can't play tones that are higher pitched than 20 kHz, which is fine because mostly only dogs can hear up there anyway.

<comment><USER20> I am not getting these things mixed up, because the sample rate is related to the maximum frequency that can be stored, and lo and behold, look at all these people claiming that those higher frequencies matter. 44.1 kHz sample rate can only encode tones up to about 22 kHz, whereas 192 can encode frequencies of up to 81 kHz, and those people up there are arguing that these higher frequencies are exactly why 192 kHz is superior. Now, if you want to say that sampling a tone at 44100 times per second somehow won't sound as good than 192000 times per second, I'm not saying that isn't possible, but I don't really take that claim seriously at all.

The fact is, simply distributing music in lossless format carries the vast majority of audible improvements. Arguing over whether or not its 24-bit or 16-bit or making a chunk of sound last 5.2 microseconds instead of 22.67 seems incredibly stupid to me, because you're better off simply improving the mix itself then fiddling over such microscopic differences. These things only become relevant if your mix and performance and recording equipment (or synths) are absurdly close to perfection. This becomes even LESS relevant in an age of indie-musicians.

</comment></comment></comment><comment><USER22> The sampling theorem is for static signals and perfect filters. Turns out, music isn't static. Once you have transients in the program, you need higher bandwidth or you will end up with phasing effects (time domain aliasing.) This is plain from the math!

Filters are also not perfect (but good oversampling filters are not the weakest link)

Further, even perfectly dithered 16 bit data can't go 20 dB below the quantization floor, unless you give up on frequency response on the high end. Again, this is plain math.

With a calibrated 105 dB low-distortion sound system, in a quiet room, I can hear imperfections from 16 bit, 44 kHz material, especially in soft flutes and triangle type percussion. Of course, D class amplifiers, and MP3 encoding, do worse things to the signal, so let's start there. But 20 bit, 96 kHz (or at least 64 kHz) are scientifically defensible, when analyzing the math and the physics involved. No snake oil needed!

</comment><comment><USER23> For an article containing a lot of "well, if you knew signal processing..." there are two fairly major oversights:

1) Any well-designed system is going to have headroom. Period. Just because 48kHz can capture the frequencies the human hear theoretically, it's always good to have a little wiggle room. This comes into play even more with interactive situations: humans are particularly sensitive to jitter. Having an "overkill" sample rate lets you seamlessly sync things easier without anyone noticing.

2) 192kHz comes with an additional benefit besides higher frequencies: it also means more granular timing for the start and stop of transients. More accurate reverb would be the obvious example. I don't know if the human ear can discern the difference between 0.03ms and 0.005ms but it's something I don't see mentioned often.

<comment><USER24> > I don't know if the human ear can discern the difference between 0.03ms and > 0.005ms but it's something I don't see mentioned often

That's the time it takes sound to travel 8mm. Do you think you could tell if an instrument was positioned differently by 8mm?

<comment><USER25> The ears distinguish directional audio in part from timing differences in what hits each ear.

<http://en.wikipedia.org/wiki/Sound_localization> cites [http://web.archive.org/web/20100410235208/http://www.cs.ucc....](http://web.archive.org/web/20100410235208/http://www.cs.ucc.ie/~ianp/CS2511/HAP.html) that suggests the brain is sensitive to timing differences between ears as low as 10 microseconds, or 0.01ms.

<comment><USER26> It's not the timing differences, it's the phase differences. The ear is exceptionally sensitive to phase differences between the ears below 1kHz. This information is captured exactly (to well beyond the naive precision of the sampling clock) for any frequency below Nyquist.

</comment></comment></comment></comment><comment><USER27> What I would love to have is: independent instrument/vocals tracks along with a default recommended "mix". The default mix would be used for normal playback and independent tracks would be great for custom mix / karaoke etc.

Is this too unrealistic to expect? Has something like this been tried before?

<comment><USER28> Trent Reznor / Nine Inch Nails has done it several times: <http://www.ninremixes.com/multitracks.php>

Plenty of other artists have as well, but this is the most high profile example I can think of. I agree it would be great if it happened more often.

<comment><USER29> I'm not a huge NIN fan, but Trent is truly awesome when it comes to digital music. You can add excellent mastering and dedicated surround mixes too..(rec: Social network soundtrack). Also a former oink'er.

The beatles multi-tracks are also available (although they were only recorded 4-track so not every instrument always has it's own track), and there has been a handful of artists who have released their samples of one song for remix competitions (Daft Punk, Royksopp, Booka Shade).

</comment></comment></comment><comment><USER29> I have to say that was probably the most comprehensive dealings with the issue of sample-rates I've ever come across. I'm not going to make the mistake others have of claiming falsehoods (all of which i've read so far have been debunked to my satisfaction by the HN users-- i'm impressed, guys).

As pointed out, mastering has _vastly_ greater effect on the audio quality (and is often pretty poor[1]), and is the reason vinyl records often can sound better than their digital counterpart, despite being an inferior technology[2]. The DAC used also has a massive effect on the sound once you get into decent quality equipment.

Like the author, i'd also love to see some expansion of mixed-for-surround music.

[1] a lot because of loudness wars, as pointed out in the post, but also just due to a lack of time/care/love(/demand?).

[2] [http://www.hydrogenaudio.org/forums/index.php?showtopic=6175...](http://www.hydrogenaudio.org/forums/index.php?showtopic=61758) This thread explores the bit-depth of vinyl records, beginning with a claim of a maximum 11-bit resolution-- limited by the width of a PVC molecule the record is made from.

</comment><comment><USER30> My hearing has declined over the years, to the point where audiophile gear is a complete waste of money. For example, I can no longer hear the difference between a cassette tape and an LP. I still listen to and enjoy music all day, but no longer worry at all about the sonic quality of it.

My advice to you younger guys is to keep the windows rolled up while driving. I have no other explanation why my left ear is much worse than my right.

</comment><comment><USER31> This is a really convincing article that makes me want to set up a double blind test for myself with my own equipment.

In my own tests I believed that I couldn't tell the difference between 16/44 and 24/96 on high quality loudspeakers, but I could with high quality headphones. The studies cited all seem to use loud speakers in testing.

Also worth noting, the article states that obtaining 24/96 source material sometimes means you get better mastered material, which still sounds better after down-sampling back to 16/44.

<comment><USER13> You weren't just believing things. The difference between 44khz and 96khz sample rate is very noticable even with mediocre audio equipment. It's an overstatement to refer to the situation as a "hi-fi case". 16/24 bits however makes no difference at all except on the size of the material.

</comment></comment><comment><USER32> I know a bit of sound engineering, waves and so.. I totally agree with the title and the first 60 lines of article, and I add my POV: 1\. Most of the people doesn't care, 2\. What apple did is just about marketing, 3\. Most of the people who says that care is pretending, 4\. Zeppelin still rock the shit in a poor quality mono mp3 recorded by a drunk guy in the audience of a concert in 73.

<comment><USER33> I do care, but I'm not the average user. Apple has always catered well for those in audio and video, up to professional levels. These are markets that retain Apple users, even when Steve Jobs was between Apples. It seems like Apple is only requesting masters to come in a higher resolution, not that consumers will generally end up with these. I think this is entirely fair since before you want to modify something (e.g. to remaster it for iTunes) you want to start off at a good quality high resolution.

That said, if Apple also allows high quality recordings to be sold, it will be useful. For example of their acapellas, instrumental tracks or samples, it would be convenient for others who want to want to remix it, and iTunes would be a platform for this trade.

Also for tracks DJs play. Most compression throw away a lot of the bass which people can't hear, but this is bass you can feel rumbling through your guts on a big sound system and is part of the experience.

For the rest, they were happy with low rate AAC files on the early iPods, they are happy with the sound coming from their crappy little iPod dock, for them it won't make a difference as long as it's a chart music track from a memorable and impressionable time of their life.

</comment></comment><comment><USER34> In normal listening conditions and for most people the difference between 16/44 and 24/192 is inaudible.

Given a 5 minute song, if I have the choice to download a 11MB file (320kpbs MP3) or a 330MB file (24/192) I would of course choose the 11MB file. The sound quality is perfectly acceptable and the file size much more convenient to manage (storage, backups, etc.).

In terms of the convenience of managing the file size and sound quality I think 320kbps MP3 is the best compromise.

Here's a file size comparision of a 5 minute stereo song:

MP3 128kbps > 5 MB

MP3 320kbps > 11 MB

Uncompressed 16/44 > 50 MB

Uncompressed 24/192 > 330 MB

When talking about sound quality there is a much more relevant issue: the amplitude compression (distortion) abuse used by mastering engineers and producers that totally destroys the dynamic and life of the sound. That is a real issue. When buying a song there should be two versions to choose from:

A) "Loud", dynamically destroyed / distorted version.

B) Normal, dynamic, non-distorted version.

Today only version A is available to buy.

<comment><USER35> But then for every 10 people like you there is 1 person who is willing to pay 20x as much so they can get a "higher fidelity" product.

For a producer and manufacturer the rational approach would be to cater to that craziness and extract as much money from it as possible. In other words if you are selling HDMI cables, spend $2/cable to make it, then sell most for $5 and then re-brand some and sell for $500. If only takes 1 out of 100 people to buying that to make the same profit. You know these people are obsessed and irrational so you cater to that. And that's basically how we end up with ridiculously overpriced Monster cables and recordings distributed to customers @ 192kHz.

<comment><USER34> Agreed, that market exists. My point is, why discuss the subtle difference between 16/44 vs 24/192 when there are far more audible and damaging practices going on in the music industry. For example, aggressive compression and brick limiting which adds distortion to achieve maximum loudness ('loudness wars').

</comment></comment></comment><comment><USER36> I mostly agree with the article in the context of distribution of a final mix. However, the article ignores one glaringly obvious reason to distribute in 24/192 format: to allow the listener to be a participant in the creative process, enabling better results for amateur musician listeners who want to sample or remix the audio or for DJs to get better results when altering the tempo for beat matching one track with another, etc. Of course, if you're going to do that, you might as well distribute in a multi-track format instead to maximize flexibility for the end user (Want to sing karaoke? Just turn off the lead vocal track for playback).

<comment><AUTHOR> Yea, and and if the bandwidth/storage is at all an issue 6x size bloat from 24/192 pays for 6 separated tracks. (Actually more, because multitrack is more losslessly compressible while 24/192 is less). If you're already providing multitrack then 24 bit audio would make sense... otherwise, meh.

</comment></comment><comment><USER37> There is no harm in releasing higher quality uncompressed or loss-less tracks. At the worst they will bring in some new customers, such as myself, that currently will not buy music online. Why would I pay $10 for an album as a highly compressed download when I can pay the same price for the CD and rip it to FLAC myself? I realize I am in the minority here, but as CDs phase out even more, there has to be some other way for consumers to obtain high quality versions of tracks.

Footnote, you don't have to have a >$10,000 setup to benefit from higher quality tracks (compared to the downloads that sometimes have 'questionable' quality). I have two systems, a full range stereo (front left and right) setup for nearfield listening at my desk thats +/- 1DB from 50hz-20khz. The other is a stereo setup in my media room; 2 way quarter wave transmission line, +/-3DB 40hz-20khz. The point is, there are a lot of people with less than $1200 in audio gear that still want lossless tracks made available. Who cares if the human ear can't discern _much_ of the extra information, we still want it.

</comment><comment><USER38> A few years ago I became really interested in recording music. I had been writing a little with a friend, using whatever crap equipment we could afford, the results weren't amazing but we were having fun and staying focussed on the music itself.

Then we starting recording other people. I became obsessed with gear, software and all the associated toys that go with any technical pursuit. I'm a programmer, so it's easy to understand how that happens but I totally lost sight of the music, spent way too much money and equipment that was nowhere near being required and generally lost the plot. I was tracking everything 24-bit/96kHz and bemoaning the loss of quality when I mixed down for CD.

Anyway, the TL;DR version of what followed was that we recorded quite a bit, lost interest in making our own music and then the whole adventure came to an end. Now my gear is leaving via eBay and I'm finding my way back to just playing guitar and trying to write good music.

24-bit/192kHz - pointless. Give me a small venue and a guy with an acoustic guitar any day.

</comment><comment><USER39> This is a good article, however the guy who has been pushing this for years and years now, is a man called Dan Lavry. In fact he wrote a very good, rigorous explanation a few years back,in very readable and well written form.

[http://www.lavryengineering.com/documents/Sampling_Theory.pd...](http://www.lavryengineering.com/documents/Sampling_Theory.pdf)

</comment><comment><USER40> Minor nitpick

> The FLAC file is also smaller than the WAV, and so a random corruption would > be less likely because there's less data that could be affected.

At the same time, if you flip a bit on a WAV file, you may hear a "pop" sound. On a FLAC file, the whole encoding block may be inaudible (or worse).

</comment><comment><USER41> The hearing of ears is a time-domain thing, not a frequency domain thing. It's the frequency response of all the frequency components added together. people might not be able to respond well to a single high frequency tone, but might respond well to a combination of tones.

<comment><USER26> No. It's both.

The basilar membrane is a loosely tuned resonator. The hair cells placed on it fire beginning on the positive zero crossing. So, to a first approximation, the ear is in fact a filterbank.

There is a time domain component in that the cochlear nucleus contains nerve cells that watch multiple hair cells at a time and correlate the firing in several different ways. Some attempt to discriminate pitch, some convolve and correlate in-phase firing energy, some look for tones to end, etc. This information is then forwarded on to the brain.

However, getting back to your point, no hair cells will fire if the basilar membrane doesn't move, and it's tuned to a frequency range.

</comment></comment><comment><USER42> I find mp3 and aac compression artifacts to be monstrously irritating. I have no idea how the majority of the world seemingly can ignore them.

Further, I can hear a difference between 44.1kHz and 96kHz. Whether you can hear that difference is up to you. (The word-length is a red herring - there's no new information contained in a 24-bit recording vs 16.)

IMO anything less than flac and you're missing something. Higher sampling frequencies do add to the sound, but in a way that is almost invisible to the untrained ear. Perhaps these should be distributed at a premium the way SACDs and similar "audiophile" formats were in the past?

</comment><comment><USER43> So, presuming we take this example:

<http://people.xiph.org/~xiphmont/demo/jaggy2.png>

The key to reproducing the original signal from the digital signal is a low- pass filter that rejects everything above the sampling rate, correct?

That is to say, what I am getting at is while the original signal can be reproduced, it requires properly tuned, and probably reasonably high performance, hardware to remove the higher frequency components of that square wave. Can you count on consumer grade hardware to do this well?

<comment><AUTHOR> Yes, thats basically it. They do this _exceptionally_ well in fact.

Typically the technique used inside DAC is to digitally upsample the signal (by duplicating samples, often to a few MHz— also allowing them to use a low bit-depth DAC) then it applies a very sharp "perfect" digital filter to cut it right to the proper passband (half the sampling rate). The analog output then contains only a tiny amount of ultrasonic aliasing which is so far out that it's easily rolled off by simple induction in the output.

This isn't just theory. Here is a wav file I made at a 1kHz sampling rate, where every other sample is -.25/.25: <http://people.xiph.org/~greg/1khz- sampled.wav> (so a 500Hz tone, the highest you can represent with 1kHz sampling).

Feeding that file to a boring resampler (I used SSRC, but anything should give roughly the same result— a least when not quite so ridiculously close to nyquist, most will attenuate near-nyquist data extensively) and get this: <http://people.xiph.org/~greg/1khz-sampled-to-48khz.wav>

Here are the two signals plotted against each other: <http://people.xiph.org/~greg/1khz-to-48khz.png>

As you can see— the 500Hz sinewave is reconstructed perfectly. (Of course, a 500Hz square wave would not be (you'd get a sinewave out) but this is because a 500Hz square wave contains energy far beyond the nyquist of 1kHz sampling).

Here is a spectrograph of the same signal <http://people.xiph.org/~greg/1khz- to-48khz-spec.png> showing that the tone is indeed pure (the faint background noise is the dither the resampler applies when requantizing its high precision intermediate format back to 16 bits).

</comment></comment><comment><USER44> I was under the impression that two inaudible high frequency tones could interfere with each other to create an audible interference pattern. (I think known as a "beat frequency").

If this is the case, then all of the arguments in the world about the maximum audible single frequency are irrelevant. Imagine music composed entirely of these beat frequencies and performed with a pair of oscillators between 25kHz and 35kHz. Without higher resolution encoding, it would be audible IRL but the recording would be silence.

<comment><USER45> If the beat frequency is audible, it will be on the recording. Obviously.

<comment><USER44> That would suppose that the recording device precisely matched the orientation of the listener, and the recording was not created digitally in (multi-track fashion for example). There would have to be air space in order for the interference pattern to set up in.

So you'd be right if your mics were head spaced and in the venue. But you'd still have secondary data, with the original lost.

<comment><USER46> > But you'd still have secondary data, with the original lost.

By that standard, the original is _always_ lost unless you have a completely holographic recording. 192kHz doesn't help with that problem at all.

</comment></comment></comment></comment><comment><USER47> TL;DR - long and detailed information about why if you got music in 24/192 format you couldn't tell the difference between it and 16/48 music.

I chuckled because this is so true, and yet tell that to the people who buy oxygen free copper 'monster' cables for their speakers, being careful to align the arrows with the direction of the music from the amplifier to the speaker. People, even otherwise reasonable people, will swear up and down they can hear the difference.

</comment><comment><USER41> A person can not hear a 22kHz tone doesn't mean he can not hear a sound that contains 22kHz components. For example, a square wave contains lots of high frequency harmonics, the more higher frequency harmonics it have, the "squarer" the square wave gets. An ideal square wave forms ideal "0" "1" states. A person's ear might not be able to hear a 22Khz sine wave tone, but he might be able to sense the steepness of "0" "1" state.

<comment><AUTHOR> First, if you can "hear the steepness" you really can hear higher frequencies, but I assume you meant "maybe you can hear higher frequencies but not higher frequency _tones_"...

People have suggested this. It's been tested in rigorous double blind tests— involving both real music signals as well as special test tones (Linked from the article). The tests were unable to show that people could hear the ultrasonics. Moreover, there isn't any physiological basis to expect people to be able to. You can't expect a stronger result than that.

Common 48KHz audio already goes a bit beyond what adults are known to be able to hear, so you've already got some headroom for "but what if a few people hear better than anyone the researchers have been able to find!".

</comment></comment><comment><USER48> I know this is slightly tangential but are hi-end DACs really worth it? I have always been amazed how much audiophile DACs cost ($300-1000). The reality is I listen to 320kbps music that was most likely recorded at 44100. DAC technology is not exactly new. So why the price?

Another tangent: To me it seems audio engineering should fix the "woofer". That is it seems subwoofers have terrible distortion.

<comment><USER49> A low-end dedicated DAC is likely to be a substantial upgrade over a built-in soundcard (I'm assuming we're talking PC sound here). A PC case is a pretty noisy place, electrically - I know one one work PC I had once you could actually here the mouse move, if you had heaphones on and cranked the volume with nothing playing - horizontal and vertical movement had different frequencies.

The move from a low end ($150-300) DAC to one much more expensive will be considerably less drastic, and likely won't matter until you've dropped at least $5k in to the rest of your system.

That said, you may already own a DAC without realising it...as long as you're taking the singal out _digitally_ (e.g. SP-DIF or digital coax) to an external receiver, you're already in a pretty decent place.

<comment><USER48> Oh yes I agree an off board DAC is better. I own the Fiio E7 which I highly recommend for laptops and only costs $80.00. In fact I run a 50 foot USB cable from my laptop to my DAC and the improvement is much better than running a 50 foot 3.5mm TRS.

But the high end ones that are 24 bit 192khz that cost $1k (Cambridge Soundworks DAC magic comes to mind) I have to seriously doubt I'm going to hear it. I really only hear the DAC difference (compared to my laptop and FIIO) when I use headphones.

</comment></comment></comment><comment><USER50> Has anyone had a look at their hi-fi amp recently? If probably probably doesn't handle much more than 80 kHz and your speakers probably dont respond to anything over 20 kHz. So yes, 192 kHz is pointless UNLESS you intend using it for studio quality editing/mixing - and I'm sure Steve Jobs would not have encouraged this!

</comment><comment><USER51> From Footnote 1: [...] If we were to use the full dynamic range of 24bit and a listener had the equipment to reproduce it all, there is a fair chance, depending on age and general health, that the listener would die instantly. The most fit would probably just go into coma for a few weeks and wake up totally deaf.

</comment><comment><USER52> The article AFAIK states little about distortions introduced in remixes & samples. I would expect certain high frequency samples, when mixed together to overlap in time, would introduce moire artifacts (beats).

<comment><AUTHOR> Not unless you pass them through a non-linear filter like a distortion effect.

(yes, the ear-brain system is non-linear too— but apparently it filters out the ultrasonics before they do anything measurable in this regard)

</comment></comment><comment><USER53> One of the strongest things that makes this article credible is that in it we have the author of Ogg Vorbis recommending that we stop using Ogg Vorbis (and all other lossy compression formats).

</comment><comment><USER54> I am told that a similar argument can be made between TV's that display at 120 Hz as opposed to 240 Hz. i.e there is no discernible difference!

</comment><comment><USER55> I just want floating point, then this silly loudness war would end (to some extent, since you can make the mix almost infinitely loud).

<comment><USER56> It would make no difference at all.

</comment></comment><comment><USER57> I think this only applies to headphones. People also 'hear' sound with there body (skin). Maybe you could call it experiencing sound. And then there are resonating sounds that cannot be heard but help to create other sounds. But maybe this won't apply to a recording because your will record the result and not the tones that make the result.

This is a great article but I'm still not convinced people cannot have a sensation of sound out of there hearing range.

<comment><USER57> Never mind. I read 192kbps instead of 192kHz. 24 bit might have some advantages but 192kHz not.

</comment></comment><comment><USER58> > Can you see the LED flash when you press a button? No? Not even the tiniest > amount?

I used to be able to see it when I was a kid (it looked very faintly red), but I just tried it and couldn't see it at all. That's actually a little bit disturbing.

<comment><USER26> I would guess that's because some earlier remotes used a higher frequency IR emitter that was in fact touching into the red.

These days, the various IR communication protocols have been standardized and virtually all use 920nm, 940nm or 980nm emitters, all of which will be invisible. I mentioned the Apple IR remote specifically because it's a remote most people reading TFA will have, and it's known to be a 980nm emitter.

</comment></comment><comment><USER59> Would someone explain should I use 44.1 or 48Khz?

</comment><comment><USER60> I think 192kHz is the sampling rate used by the A2D converter vice verca. It is not the actual frequency of the sound (data).

</comment><comment><USER13> There is no point with going over 16 bits, but there is definitely a point with going over 44.1khz, as it allows you to actually reproduce waveforms more accurately than 44.1khz. Try reproducing f.e. a sinewave accurately over 4-5khz with a sample rate of just 44.1khz - it cannot be done, and at this point we haven't even taken into account the issue of varying slew-rate characteristics of the thousands or so different DAC output stages in use in personal audio equipment.

44.1khz gives too much aliasing distortion, but 192khz is quite the overkill. Ideally, digital audio could sit on 16 bits of depth sampled at 96khz.

<comment><AUTHOR> No. This really is not the case. The article _specifically_ addresses this misconception.

The signal reproduced from your 44.1kHz sampled digital input is not a stair- step like some broken waveform editor might display: On output it goes through a matched reconstruction filter (which may, in fact, be digital and involve an oversampled DAC or it could be analog though those are harder to build without compromise). After the reconstruction filter the output is _EXACT_, assuming the input only contained energy below the nyquist (well, and was sufficiently far away from the reconstruction lowpass).

So even a 5khz sine wave is reproduced perfectly with 44.1kHz sampling.

<comment><USER61> @nullc: of course you're right, and the commenter you're replying to does not understand the Nyquist-Shannon sampling theorem. Which is a shame, because the article specifically addressed this point.

These discussions of audio standards always get sidetracked by people who don't understand or believe this result. (Have to admit, the result _is_ surprising).

I think there may be problems with the argument in TFA, which is based exclusively on standard linear systems theory.

Of course, the ear and some of its perceptual components may be significantly nonlinear, and thus not covered by the frequency response graphs of TFA.

These graphs assume linear systems, in which you put two frequencies in, and the same frequencies pop out in scaled form. Nonlinear systems can produce new frequencies in response, and this possibility is not discussed in TFA. Probably these effects are quite minor, but may be audible to some listeners on some equipment for some choices of source material.

<comment><AUTHOR> Indeed, but if there were non-linearies in the ear (there are many, of course) which allowed detection of ultrasonics (less likely, because the first stage of the ear is impressively linear) you'd expect them to show up in the actual listening tests.

The TFA does at least make this the-proof-is-in-the-pudding point somewhere in its depths. :)

</comment></comment></comment></comment><comment><USER62> Sometimes less is more. The debate goes on. Why not just let the music play? And by that I mean high resolution music. All you need is one person who can hear high frequencies, and all the technical mumble-jumble becomes hogwash.

People actually _believe_ the 20KHz argument that anything above is inaudible. That's hogwash. I know because I can hear (or sense) higher frequencies, and I do not have the absolute best ears I've ever "met."

For example, last week I attended a A/V equipment event with very high-end equipment. It was packed --- over 600 people for one evening. 6 rooms of equipment. I'm sure all six served the same fare according to the 20-20KHz argument of this piece, yet they all sounded quite (or even extremely) different.

The 20 KHz argument is a myth. For people who can't hear the difference, no problem. But please do refrain from ruining or hobbling music for the rest of us... who can hear a wider frequency range.

Yes, some people are color blind. Does that mean the rest of us shouldn't use color? I hope not.

Music is an important wholesome and potentially emotional part of human life. Please do not cap it with "false optimizations".

24-bit/192 KHz is not inferior to CD quality sound. If you don't believe me, try a Linn system sourced on a Klimax DS with some high bitrate Linn classical music (or the Beatles Masters USB release!). If you can't hear the difference compared to low bit-rate (including CD quality) material, I assure you someone can. The low bit-rate will sound flat, hollow, less lively, or/and more coarse. Any number of problems exhibit at inadequate bit levels.

Vinyl is analogue quality (no discrete digital distortion). CD quality is a large step down from vinyl. A/V is just trying to get vinyl like quality from digital. We don't need nay-sayers impeding progress. If you can't hear the difference, please let someone who can hear make the informed decisions.

Thanks.

<comment><USER63> It's not a myth, but a fact established in laboratory studies. Your anecdotal claims to hear frequencies that scientific evidence suggests you cannot hear doesn't overturn science. I'd be convinced if you correctly identified which speakers were reproducing 21 kHz frequencies in a double-blind test, though.

<comment><USER62> Isn't science verified through (wait for it...) experimentation? So how does my hearing not invalidate your science?

That's the problem with the theoretical science. When it's false, it's false. Come up with a new hypothesis; this one's false as it pertains to human hearing. There's information theory, and then there's auditory reality. Reality confounds the theory as applied to hearing. I don't know where the fault lies, and I don't really care.

But it's really annoying and frustrating having people nix progress out of idealistic theory, "laboratory" studies, and ignorance. The experiments (my experiences and numerous others) don't lie.

Double-blind is great, but I can already tell the differences between all six rooms of equipment from last week. One of the rooms was so extreme, I wanted to run out of the room due to discomfort (but I was polite and stayed all 30 minutes). In other words, double-blind was unnecessary. Someone whose ears I respect a great deal, loved that room. Even golden ears don't all hear the same. But I don't need double-blind to confirm trivial experience. The proof is already in the listening.

<comment><USER63> > So how does my hearing not invalidate your science?

Because it's not a blind study. In audio, claiming something sounds better than something else is low-strength evidence, because it doesn't: 1) distinguish psychological bias (which is very strong in this area) from actual audible results; or 2) distinguish which characteristics of speakers, if any, you may be hearing.

If you can consistently ABX two speakers that have similar characteristics except that one reproduces frequencies over 20 kHz while the other doesn't (with identical performance below 20 kHz), I'd be convinced. One possibility is to use the same speaker but insert a high-quality 20 kHz lowpass in the chain during part of the test; or use the same speaker but with 44 kHz versus 96 kHz source material. I've never seen a controlled, blind case where a human can tell the difference there.

<comment><USER62> The psychological component is a red herring. Even though I already have a system (bias), I don't care about the other systems. I went to the show for enjoyment, education, and appreciation. Some of the systems were unknown to me (no bias), and some were known and surprised me in some ways (again, some bias overridden). So bias can be important, but it's not relevant in this case. So bias doesn't invalidate my experience.

As for the double-blind and high frequencies, I believe I've already done the test. I have had my hearing tested several times. One of them, I recall the tester actually asked me to repeat some tests... it was funny. The testing was at very high frequencies. I believe she thought I was guessing the higher/lower frequencies... and getting lucky. So (I strongly suspect) she wanted to "prove" to herself what you want to prove --- that noone can hear above 20KHz. I disappointed her. I think she even threw in some placebo tests (no frequencies at all). It was funny. She never explained herself. I suspect she just thought I got lucky again.

How to really test this stuff? Get one of the audio designers to test... but they will laugh in the testers' faces. They do this stuff for a living... to build real products... for real live customers who can hear the differences. Dave Wilson was at the A/V show. Try listening to a pair of Wilson Audio speakers. I bet he can hear better than just about anyone... His speakers (when sourced and driven properly) are that good. But he wouldn't waste his time on such tests. He has customers to serve and a business to run.

I doubt lab experiments look to disprove their theories once and for all. That's a social prejudice built into the lab experiments. Fix that, and you'll end up with a better hypothesis.

</comment></comment></comment></comment></comment><comment><USER64> Unless you are a dj or producer and would like to sample or time stretch the tracks. That's why Beaport offers a wav download option, that many djs/producers prefer.

</comment><comment><USER65> Science be damned! Onwards with subjectivity!

• 24-bit audio is magical. When I recorded myself playing guitar in 24-bit and played it back through my amp, it sounded like I was still playing. 16-bit sounded like a CD.

• With MP3s, 192 kbps is a huge step up from 128 kbps. 192 doesn't exhibit any of the "swooshiness" heard in the upper range of 128 kbps MP3s for regular rock/pop/hiphop music.

<comment><USER66> I believe you are confusing 192kbit compression with 192 khz sampling rate. Not at all the same thing.

<comment><USER65> Whoops! Guess I'll read more carefully, then.

</comment></comment></comment><comment><USER67> One thing I don't see addressed is the experience of _feeling_ frequencies that can't directly be heard. There was a study done with a particular piece of classical music, with and without a particular inaudible component to it. The presence of the inaudible component drastically changed the listeners perception of the music. They described it as more dark or creepy (perhaps not the actual words used, but it matches the sentiment). The point is that there may be value in reproducing frequencies that we can't "hear", as inaudible notes can alter the experience of the music.

*not the study I was referring to but its along the same lines: [http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=5291...](http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=5291285)

<comment><USER68> The author completely ignores infrasonics and writes under the incorrect assumption that our only perception of wave pressure comes from our eardrums.

I've never been able to enjoy listening to my favorite classical music on headphones or even smaller speakers, and it's largely because of the effect you describe.

At this point I'm resigned to preserving my treasured (and cumbersome) vinyl collections. Maybe if Apple comes up with some snazzy marketing term (e.g. "Retina") for 24/192 or even 24/92, and starts distributing it on iTunes, things might start to change.

<comment><USER69> You don't need a higher sample rate to capture or play back infrasonic pressure waves, but most recordings are mastered to remove DC offset and rumble <20Hz, as reproducing those components requires specialized equipment, such as a rotary subwoofer.

<comment><USER68> Wouldn't the higher bit rate help though? I've long suspected my preference is largely due to the difference in mastering techniques more than any technical limitations.

</comment></comment></comment></comment><comment><USER70> I don't understand why anyone gets down on 24-bit consumer audio.

Specifically because CD-quality 16/44 audio has midrange distortion present during complex passages that is completely eliminated and non-present in 24/96 sources.

Listen to "Us and Them" off a 16/44 CD version of the Pink Floyd album Dark Side of the Moon. When it kicks into the chorus, it becomes totally distorted and everything in the midrange bleeds into each other. It's a mess.

Then, try listening to the 24/96 Immersion box set copy or a vinyl-sourced 24/96 rip and you'll find it's gone. When the song gets complex and loud, everything remains totally clear, each instrument stands on it's own, it doesn't become an awful distorted jumble.

You could argue that it's just the quality of the master that makes the difference; but if you take a copy of the original transcoded to 16/44 and compare it again with the 24/96 copy you can hear the same effect.

Why would anyone argue against high-resolution audio anyway? Sure, most everyone will probably just continue downloading 16/44 MP3s, but at least give us the option to have 24bit FLACs of the stuff we really like. Please and thank you.

<comment><USER71> You could argue it's the quality of the master, and the mastering process, and you'd be right. That's a no-brainer.

"but if you take a copy of the original transcoded to 16/44 and compare it again with the 24/96 copy you can hear the same effect." I could believe that, but do you mean to do the transcoding yourself? IN this case you become the engineer, and the tools you use and all that become vital as well.

Having heard stunningly awesome CD's of DSOTM on a homebuild heathkit amp and some old speakers and not believing my ears when I saw what the setup was, I'm skeptical... can't help it.

</comment></comment><comment><USER72> Huh, I think people truly advocating 192 as a distribution format will be few and far in between, a really good and cheaper sampling system can be put together at 96. Still, a lot of things in this article perplex me.

Human hearing is limited to 20k because frequencies higher than that are perceived as painful? Dont agree with that one.

24 bit doesn't offer any advantages to sound quality? Sheesh.

And the crux of the argument is intermodulation distortion increases when you try to represent more frequencies? Isn't that an argument for a faster power amp?

<comment><USER73> > Human hearing is limited to 20k because frequencies higher than that are > perceived as painful? Dont agree with that one.

You misread the article. It's because there is so little response that being able to hear it would blow your eardrums (and even then, it might _still_ be beyond your ability to hear it). There's no value in that.

> 24 bit doesn't offer any advantages to sound quality? Sheesh.

Not quite what TFA says. According to the article, 16 bits effectively covers the dynamic range of human hearing, so more than that is pointless _for music consumed_ by human beings (hence all the stuff about 24bit being a good idea for mastering & production). If you're storing integers in the 0~16384 range, going from 16 bit integers to 32 bit ones is not going to give you "better ints", it's just going to waste 2 bytes per int. Same thing here.

<comment><USER72> I can admit that I misread the article when it comes to hearing limits. I was reacting to my perception as an audio engineer that a lot of people dismiss the importance of that frequency range.

<comment><USER74> With good reason!</comment></comment></comment></comment>