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."
stop being so paranoid about what you think you can't hear, and enjoy the damn music.
I'm a composer who works in film/games. I can assure you this is exactly what I'd like people to do when they listen to my music. I spend 99% of my time trying to create good musical ideas, and I spend 1% of my time getting the mix down. I get criticized (rightly) for this quite a bit, but it is hard to care about someone sitting in a >$10,000 labyrinth of sound equipment when I'd rather write a catchy tune.
Then again, when I write sheet music I have to endure some of the most soul-crushingly awful midi sequencing in order to check my work, so perhaps I'm too tolerant to terrible sound quality. Still, I'd rather people listened to the music, not the sound of it.
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.
With a username like yours, I'm not surprised. :-)
> although the prescription isn't always as
> simple as "get a thicker rug" etc
A prescription is only as good as the likelihood that it will be heeded by the patient. A rug is an easy win; acoustic ceiling tiles and bass traps are a bit harder...
I might sound/be stupid for asking, but what's the actual physical response from something at 22 kHz+? I have a hard time picking up a pure sine > 17 kHz. I doubt I'd get any aural response from anything at 22 kHz, so what's the deal?
I agree. It should be about the performance, not the sonics. There are plenty of old Motown and even Beatles recordings with distorted vocals, bad edits, etc. Your brain passes right over them because of the emotional content of the music.
That's because they focussed on the most likely end-user experience:
>While Motown shortened song to fit into radio time, the company also produced records specifically with car radio audio quality in mind. Motown recording engineers set up car speakers in the studio so that they could simulate and perfect how a song would sound emanating from a car radio
- what's the point of engineering things to a set of conditions virtually none of your target audience possesses?
I worked out a long time ago, that I enjoy listening to _music_, not HiFi gear.
My advice to people who ask how to make their system sound better? Buy some music you enjoy more…
I can enjoy a wonderful performance of a great tune played through my laptop speakers - much more that I enjoy test tones or gear-demo-tracks through sound gear worth something north of a new car…
I'd disagree, at least among the people I know: they all have cheap HTIB systems and the single biggest, most cost-effective improvement you can make from there is to buy better speakers.
No, even in that case, the room can still overpower the speaker. A $200 HTIB system in a properly treated room will sound a lot better than $28,000 Wilson Watt Puppies in a bathroom for example.
You might be surprised how much of a difference EQ can make. As an experiment I once used 12 bands of parametric EQ to adjust the speakers in a cheap, old LCD monitor. Sure, you're not going to get any better bass response than before, but stereo imaging (nonexistent before EQ, perfect phantom localization after), spectral balance, clarity, distortion (due to not exciting resonances in the monitor), etc. were significantly improved. Most people could have added an EQ'd subwoofer to those tinny LCD speakers and been completely satisfied.
One of the many things EQ can't fix, of course, is room reflections, which can be helped by room treatments and speakers with a directivity better suited to the room.
I should point out that after room treatment, my next recommendation tends to be an amplifier with Audyssey MultEQ in-room calibration. I've never heard a listening environment that didn't sound unambiguously better with it enabled.
It is said that in most cases 192 kbit .mp3 is indistinguishable from >192, and blind tests support that. Granted, there are instruments like castanets which make it easier to hear the difference. In general though, I can't distinguish 128 from 192 and I listen to music a lot. Also it's unlikely that my hearing is already damaged because I try to keep volume low.
But I've noticed that where I put the speakers makes a huge difference. I can easily tell the difference from speakers on the floor versus speakers on my desk. Where I'm at the moment also matters a lot. If I lie on the floor, floor speakers don't sound as bad anymore.
In the end, I use headphones. Midrange Audio Technica ones, and I'm probably already overpaying a bit. But I bought them for build quality and comfort, and I wasn't disappointed. I can have wear them for hours (Not healthy I guess, but I'm used to wearing them even with no music being played). Headphones have the advantage that it suddenly stops to matter where your speakers are and where are you relative to them.
This effect isn't sooo surprising seeing as it even occurs with dumb mono guitar cab speakers and is very, very, VERY clearly audible there, even just moving your head a few cm in or out of the cones' axis.
> Stop being so paranoid about what you think you can't hear, and enjoy the damn music.
Yes. Do a couple of blind tests with your acoustic system first.
> It's true enough that a properly encoded Ogg file (or MP3, or AAC file) will be indistinguishable from the original at a moderate bitrate.
Disagree. This claim seems to be ungrounded compared with others.
I can believe limitations with bit depth and sampling rate (although I'll take a chance to test myself if I get near good enough acoustic system). However, I definitely could discern in a blind test whether music I listened to was stored using lossy format with reasonable bitrate. It's usually quite audible with rock music that involves cymbals.
There's a specific "bug" in the mp3 encoding scheme which means that you get a pre-echo effect on fast attack waveforms. It's inherent in the encoding, so it can't be eliminated (although the higher the bitrate, the less obvious it is IIRC). If you how know to listen out for it then you'll spot it immediately.
AAC / Ogg don't have that limitation & at high enough bitrate should be indistinguishable from the source in a blind listening test, as demonstrated in a number of Hydrogen Audio listening tests down the years, unless of course you're using crappy encoders at which point all bets are off...
(Really, LAME is very good indeed these days. I eventually decided that I was going to get with the program and just encode all my CDs (backed up to flac files) as mp3 for portable listening. It's good enough, and I've decided not to listen for the pre-echo artifacts so that I won't notice them :) )
A thousand times this. I never cease to be amazed at the number of people who will vocally argue the benefits of solid-silver wundercable, but who've never heard of mirror points or bass traps. $20,000 hifi systems in rooms with bare wooden floors and bare concrete walls. Subwoofers in untreated cubic rooms. People praising the transient response of their PMC MB2s in a room with chronic flutter echo. It's utterly dispiriting.
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.
> Digitally recording a triangle is the best example of why 48kHz is very limiting
The article's about distribution, not recording. I don't think anybody disputes the usefulness of higher sampling rates when recording.
> 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.
...and 48kHz audio contains those lower frequencies.
Stripping frequencies above 20kHz negates the effect on the lower frequencies since those lower frequencies are not "modified" by the higher ones. The human hear can actually hear the very high harmonics when they're combined with a lower fundamental frequency.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz. If it's played at 44.1kHz, the 30kHz frequency is gone and all you'll hear is 10kHz, not a "different sounding" 10kHz.
> The article's about distribution, not recording. I don't think anybody disputes the usefulness of higher sampling rates when recording.
Didn't read the article, so commenting out of context, however it needs to be said that in sample-based music genres the distributed music gets used as if it were a recording. Maybe then it could be argued that higher sampling/bit rates should be available, if only for those who are sampling.
> 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.
That may well be true. But those mixed-down harmonics that are heard "live" would then be captured by the 16/44 (or whatever) sampling. IOW, the recording captures what you heard. Those upper harmonics have no emergent properties. Their effect is captured.
I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.
Just because it is difficult to record a triangle does not necessarily mean it is impossible to accurately recreate the sound (to human ears) using 48kHz.
> I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.
Yes, you're right.
In fact, some of the section X references don't even mention hearing, they talk about "alpha-EEG rhytms" (in this case "listeners explicitly denied that the reproduced sound was affected by the ultra-tweeter") and "bone-conducted ultrasonic hearing" trough the "saccule" ("organ that responds to acceleration and gravity and may be responsible for transduction of sound after destruction of the cochlea").
--
In fact, most of the claims of the article are around the fact that there is energy over 20khz and how it can affect recording process.
This is a well known fact, and this is exactly why engineers filter out sub-sonic and super-sonic frequencies, especially today: stuff that you can't hear (or feel) will just suck your headroom and make you lose the loudness war.
The only "good sounding" triangles you'll hear are those buried in a mix. Alone, it always sounds weird and "muted".
EDIT: Listen to the triangle at the beginning of Rush's YYZ. It's an old recording, but it sounds significantly worse than the analog version. It's been digitally mastered some time ago so if it was mastered today, it would probably sound better, but still not great. I heard a rumor that Rush is remastering all their albums "for iTunes" at the moment, so hopefully we'll be able to compare soon!
Yes, but our ears only hear 20Hz-20KHz. So, according to Nyquist theory, you can recreate the entire signal that the human ear hears by recording those sonic artifacts that result from interference between supersonic harmonics.
So while it's true that the human ear can't hear well above ~18KHz, and the interference between high order harmonics are audible, it's also true that a properly recorded signal, sampled at 44.1KHz, oversampled, and filtered, can reproduce the exact signal the human ear is capable of hearing. At least according to theory.
The human ear is capable of detecting sound pressure as well as sound intensity, and while playback of the interference between harmonics can be reproduced faithfully in the sound intensity realm, the sound pressure levels will differ, and it is theorized that people may be able to tell the difference between the two. However, as far as I am aware, nobody has been able to demonstrate this reliably in practice.
What about sound outside 20-20k that affects us via mechanisms other than being directly sensed in the air by our ears? For instance, consider frequencies below 20 Hz that we can feel with our feet as vibrations in the floor, instead of hear with our ears? Or what about the possibility of sound above 20k causing a vibration in something other than our ears, which could have a subharmonic in 20-20k that gets conducted to our ears via bone?
I'd prefer recording technology to err on the side of capturing what we need to reproduce all of that, even if we aren't sure that we need it.
I'm an audio engineer, too, and I agree that this has been debated to death. And I agree that frequencies above the threshold of hearing are more important than standard dogma (based on Nyquist theory combined with Pure tone audiometry) allows. It helps explain how audio gear with a 100kHz bandwidth sounds clearer than gear with a 20kHz bandwidth even when they measure the same in the audible band.
Have you read the Audio Technology magazine interview with Rupert Neve?
Greg Simmons: Geoff Emerick, the famous British Producer ?
Rupert Neve: Yes, he started me off on this trail. A 48 input console had been delivered to George Martin's Air Studios, and Geoff Emerick was very unhappy about it. It was a new console, made not long after I had sold the Neve company in 1977. George Martin called me and said, "please come and make Geoff happy, while he's unhappy we can't do any work".
They'd had engineers from the company there, and so on. The danger is that if you are not sensitive to people like Geoff Emerick, and you don't respect them for what they have done, then you are not going to listen to them. Unfortunately, there was a breed of young engineers in the company ( I hasten to say this was after I sold it !) who couldn't understand what he was bitching about. So they went back to the company and just made a report saying the customer was mad and there wasn't really a problem. Leave it alone, forget it, the problem will go away. They were acting like used car salesmen. I was very angry with it. So I went and spent time there, at George Martin's request, and Geoff finally managed to show me what it was that he could hear, and then I began to hear it, too.
Now Geoff was The Golden Ears - and he still is - and he was perceiving something that I wasn't looking for. And it wasn't until I had spent some time with him, as it were, being lead by him through the sounds, that I began to pick up what he was listening to. And once I'd heard it, oh yes, then I knew what he was talking about. We measured it and found that in three out of the full 48 channels, the output transformers had not been correctly terminated and were producing a 3dB rise at 54kHz. And so people said, "oh no, he can't possible hear that". But when we corrected that problem, and it was only one capacitor that had to be added to each of those three channels, I mean, Geoff's face just lit up ! Here you have the happiness/ unhappiness mood thing the Japanese were talking about.
The article doesn't suggest only using 48kHz for recording and mixing. I don't think the author would disagree that recording triangles is difficult. He would argue that once you've decided what final audible frequencies you want to present to the listener, the best way to distribute them is at 16-bit 44.1/48kHz. It's a compelling case.
That's one thing I find concerning with the move to digital. With analog media, you can go back, re-record and get an improved result (provided the source is good) but District 9 (which was shot on Red One) will never have improved quality other than resampling because the source is set to a particular digital format with associated data quality.
I completely disagree with the article having heard the difference many times myself. You can't record at 192kHz and hope to keep the same quality by distributing the final mix in 44.1kHz. It just doesn't work that way.
I don't think I understand quite what you're saying and wondered if you could explain more. You and the article both say that humans can't hear above about 20kHz. If there are higher frequencies that create a harmonic at a lower frequency (e.g. a 33kHz harmonic that produces a sound at 16.5kHz) then surely that lower harmonic (16.5kHz in this case) will be recorded by the original recording equipment assuming it is recording at a frequency at least twice that of the highest audible frequency (let's say that this would be 48kHz, although there might be other DAC-related reasons to go higher).
Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics.
If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only our first frequency will be captured: 15kHz. It will be exactly the same as if you only recorded 15kHz alone. The harmonics don't modify the fundamental frequency, they just trick the human hear. But when they're gone, they have no effect whatsoever.
Hope this helps!
EDIT: the frequency numbers I used are actually somewhat of a bad example. Harmonics are never exactly double, triple the fundamental. Those would be mostly inaudible. But you get the idea.
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."
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.
The statement that frequencies above 20kHz don't matter rests upon the assumption that the ear is linear. If the ear is not linear (I don't know whether it is not not) then frequencies above 20kHz will matter, as the ear will be able to mix higher frequencies down to less than 20kHz. For example, if we have frequencies of 56kHz and 59kHz, the ear MIGHT be able to discern a difference frequency of 3kHz. No doubt this effect could be reproduced by signal with a sampling rate of 44.1KHz, but only if the analogue systems, before the sampling stage, reproduce any non-linearity in the human ear.
Incidentally, you can get speakers that create a localised beam of sound, that the person sitting next to you cannot hear. They work by transmitting frequencies above the audible range. These high frequencies can be beamformed by a relaitively small speaker array, so the sound is localised. They then rely on the non-linearity of the ear (or maybe the air around the ear?) to mix the ultrasonic frequencies down to audible frequencies. I guess there must be non-linearity in the human auditory system!
On the subject of 24-bits my understanding is that 16-bits is adequate, provided the levels (scaling) are set correctly in the recording. What 24-bits delivers is the ability to do a crappy job of the mixing, and still end up with the full dynamic range of the human ear. 24-bits is probably a temporary solution though, as manufacturers will engage in the usual Loudness War [1], and push the signal to the top of the dynamic range. Before long 24-bit audio will be equivalent to 16-bits (since the 8 least significant bits will be unused) and the next big thing will be 32-bit audio.
Having said all that, I'd guess that the speakers will be the limiting factor in most sound systems, not the recording format.
Another "ex audio engineer" here, you can believe or not at your leisure. Many hours spent in high-end recording and mastering environments.
I'm not sure what your background in audio is, but everything he says is correct. High end frequencies well past 15k and up (22.1k actually) are widely acknowledged to influence the lower frequencies and play a huge role in the perception of the quality of a recording. This is an old debate with pros and cons on both sides, but in general you'll find the "Golden Ears" mastering engineers (Stephen Marcussen, Bob Ludwig, etc.) come down on the side of higher sampling rates.
Now, if your original recording was mastered to 16/44.1, then a transfer by way of 24/192 will probably actually hurt the recording. But if you're mastering from an original analog or high-quality digital, in my experience there's no question, higher sampling rates deliver better experiences.
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.
>Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible.
This [1] (widely accepted in the scientific audio community) study's conclusions disagree with your assertion.
>44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad.
This is not the 1980s, hardware has progressed beyond that point. Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems. Also see [1]
>That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data.
I did not notice that in the article. It talks about IMD in the context of the analog chain and the transducers following the DAC, and it's possible that high frequencies can increase it.
> Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems.
True, but they do so using (long, high-quality) high-cut filters. And these filters are pretty sharp, as they have to close within, say, 18-22.1 kHz. You can design them as linear-phase FIR filters with oversampling and all the good stuff, but physics dictates that sharp filters introduce distortion. A sharp filter like that is audible.
You will still need a aliasing filter that cuts off between, say, 18 and 22.5 kHz to avoid aliasing noise. That is one sharp filter no matter how you look at it. You can use a high quality, long, linear-phase FIR filter, but you can't cheat physics: sharp filters necesserily introduce distortion, and such a sharp filter so close to the hearing threshold does not go unnoticed.
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.
why do you think "This difference gets lost in the THD already at the DAC "? Do you have numbers to back it up? What's the noise floor of DAC? What's the noise floor of an output stage? Do you have the number?
why do you think "This difference gets lost in the THD already at the DAC "? Do you have numbers to back it up? What's the noise floor of DAC? What's the noise floor of an output stage? Do you have the number?
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.
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.
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.
There is a perversion going on both ends here. And by perversion I mean a distortion of truth in a bid to make a profit. This is not the worst that can happen, but is just worth mentioning. You probably put more mildly, but I am bit more harsh. Some people are irrational and spend money of stuff that they don't need and another group of people are perpetuating the lies and the marketing in an effort to extract the maximum amount of money from the other group (In other words your basic market setup).
Audiophiles are quite a fascinating group. These are people that can be rather rational in some respects (they could be doing research in some lab somewhere) but when it comes to audio equipment they will shell $2000 for HDMI cables. The salesmen and manufacturers that make these things ("high end" HDMI cables, 192kHz recordings) know this very well and they aggregate around this target set of clients.
I think that is exactly what is happening here. At some point storage capacity is just good enough and one can distribute 48kHz, 16bit audio to everyone. But what do you do next? Everyone is getting that and it is not new and cool anymore. What to do? Well increase the frequency and sell everyone a newer, better, higher fidelity thing, even though objectively human years cannot really hear the difference. Subjectively though, there is a huge difference. If you ask someone who just spent $50 for a 192kHz record if they like it better than say a $20 48kHz one, I bet you 100% of people will confirm that 192kHz sounds better and will be ready to go and buy more.
> 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?
Ultimately, sure. The world is full of products and services which only add value in this weak sense.
If the same wine tastes better if it's priced higher, then it still tastes better. But I think it's only honest that the consumer be aware that the increased utility from being priced higher is due solely to the fact of it being priced higher. Beyond that, I don't care.
One thing we can all agree on is that music is much more enjoyable if you think you're listening to it through good equipment or from a good source. Ultimately it's only the `thinking' part that matters. So I would make two points:
1. One point he's making is that playing audio sampled at 192khz through regular equipement actively distorts the music in negative ways. So now if you know this now you should enjoy that music _less_.
2. If you're adept metacognition (maybe that's not the right word), you'll realize a) you can get most of the enjoyment by buying equipment that's `pretty decent', and then not worry about it too much. b) you're probably fooling yourself by spending so much time/money worrying about having the best equipment, so you're probably not getting the maximum utility from the experience anyway. Or maybe it's the experience of trying to get the best equipment it self that's enjoyable, not necessarily the increased audio fidelity.
Well, if we accept that argument, then just about any means of signalling "this sounds better!" will work. How about we choose something that doesn't waste bandwidth?
That's true. It's kind of the Monster Cable model. BTW, I'm not saying that marketing and whatnot should deceive less technical consumers and trick them into spending more money than they should (which is basically what Moster Cable does). But when you explain to technical people why something like 24/192 isn't better (other people in this thread have pointed out, this isn't totally accurate in the first place), and they understand what you're saying but still prefer it, by all means, let them buy it.
This is the same reasoning that somebody used when I was debating with her if insurances should reimburse homeopathic and other alternative treatments. Her reasoning was 'well if it works, it should be reimbursed, doesn't matter if it's from a placebo effect or not'; my position is that they shouldn't be reimbursed, but quite honestly, I don't really have a rational reason for it (at first I thought I had but it turned out I couldn't formulate it, which is the same as not having it).
So, while I have no option (for now) but to acknowledge your position, I still feel dirty for doing so.
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 - starting 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.
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.
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.
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.
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!
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.
2) increased sampling rate does not improve timing. This also has been researched in detail (because it sounds like it could possibly be true given that the ears can phase match to much greater granularity than the sample clock). It was found false in practice, and in retrospect, the sampling theorem explains why. The Griesinger link discusses this with illustrations, and provides a bibliography.
48kHz already has enough 'wiggle room'. How many people do you personally know that can hear a 24kHz sine tone?
> more granular timing for the start and stop of transients.
... it's something I don't see mentioned often.
Probably because it doesn't make sense. Human ears cannot hear frequencies about 24kHz and Nyquist tells us that 48kHz is enough to completely capture all the detail of a signal at that frequency and below.
> Having an "overkill" sample rate lets you seamlessly
> sync things easier without anyone noticing.
You can get the same theoretical benefit by oversampling on playback. And a lot of audio equipment does just that.
> 192kHz ... also means more granular timing for the
> start and stop of transients.
Not really, for two reasons -- unless you're talking about glitch music, transients are unlikely to ever be so sudden that the difference between 0.03ms and 0.005ms could possibly matter.
I'm pretty sure that #2 isn't true; signal processing folks will be able to phrase this better than I can, but I think that if you have enough information to capture the waveform at a given frequency, you also have enough information to precisely place it in time - phasing errors are more likely due to quantization error, which is about bit depth, not sample rate. No?
0.03ms is 33kHz - you can't, no matter how much you want to, make a granular timing that is faster than at least one cycle of the frequency you are using. 0.005ms is 200kHz BTW.
This isn't true. Sample a bandlimited impulse. The exact timing is encoded into the gibbs oscillations of the signal. So long as you have a high enough SNR you can have timing as precise as you want. (and because the ear doesn't work with ultrasonics— it is itself bandlimited— it uses the same phenomena for timing)
Humans are sensitive to jitter, but jitter isn't a major problem with modern digital electronics and reclocking strategies. This ArsT thread hashed out these issues a couple of months ago: http://arstechnica.com/civis/viewtopic.php?f=6&t=1164451...
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?
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).
There are two reasons I don't think this will happen:
1. People would use the tracks to create custom remixes which they would then distribute. What happens when a remix becomes more popular than the original track? Artists generally have to pay other artists to remix their songs (usually via royalties).
2. Creativity. When an artist creates something they want you to hear it the way it was intended. Allowing you to remix it however you like takes away a lot of the creative control from the artist.
Regarding remixing. Artists usually don't "pay" each other, but return the favor - if it's the right term to say. E.g. artist A remixes a song of artist B and artist B in turn does the same for artist A. Or if they are all on the same record label artist A does a remix for artist B and later B makes a collaboration with A. I've noticed this in electronica/edm music artists at least.
And another important remark: some artists are flattered when someone asks them to make a remix for their song. (Imagine you're an artist and your idol asks you to make a remix of his song.)
That was predicted (and suggested) by Glenn Gould some forty years ago. At the time, anything with higher fidelity than, say, a bad telephone connection, was analogue, but we were stepping into the world of quadraphonic sound (which died soon after in the analogue kingdom), but he was a big proponent of the listener as participant (hey, it was Toronto and McLuhan was still around) and was convinced that technology was the only limiting factor at the time. (To put thing in perspective, he was also very much anti-concert--he hated what he called the "non-take-two-ness" of live performance.) Let's just say that the idea was no more popular among artists then than it is now.
Many of the groups I listen to do do this -- this is certainly not that rare. Sometimes they go for a bit more money by releasing a separate CD with karaoke tracks, for example, but at least if you want it, it's available.
Of course, you can very easily get just the vocal track by subtracting the two. Sometimes the "non-vocal" track will still include backing vocals or the like in appropriate places, and just pull out the main vocal track.
For music where the vocal tracks aren't released separately, you can often pull them out nevertheless. The best is if you can get the audio in 5.1 -- vocals are almost always center-panned, which makes extracting them quite easy.
Some songs when released as a single have Acapella and Instrumental versions of them as well. There are also compilations with only acapellas and compilations with only instrumental versions of the songs.
And when you have them, just use something in the like of Ableton Live and that will be it. I think that that's what you mean right?
It will be a great idea to have tracks released as several `layers` so that the user can choose which of them to play and which not, for example the bass/beats layer, layer with melodies, layer with the percussions, layer with the vocals of course, but that sounds like semi-studio production.
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?).
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.
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.
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.
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.
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.
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.
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.
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').
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).
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.
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.
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.
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.
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.
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.
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?
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?
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
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).
Your question is somewhat amusing. A standard CD player uses 1-bit DAC (it's either on or off) at a yet-higher frequency to achieve better linearity. Filtering is quite easy in the analog world.
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.
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.
Alas, they don't— you can easily demonstrate this for yourself. Startup an audio editor and generate tones at 25k and 28k (make sure you can't hear them— otherwise you have severe distortion screwing up your test) then play both at once. You will not hear a 3kHz tone.
The tone you get from an acoustic beat is not a real tone— it's a perceptual quark that requires you to be able to hear the tones in the first place.
I tried this in Audacity, with the project set to 96kHz and two tones at 25kHz and 28kHz. I couldn't hear either of the tones individually, but I could hear a tone when played together. This is on Windows 7 with the sound card configured for 24-bit/48kHz. Am I running into resampling artifacts somewhere in the chain?
EDIT: it turns out Audacity won't generate a tone above 20kHz (the UI accepts the value, but when you reopen it the value has been rounded down), so both of my generated tones were actually 20kHz.
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.
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.
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!".
There's been countless A B X tests that showed conclusively that the vast majority of people cannot tell the difference between samples with and without the 22kHz+ frequencies.
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.
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.
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.
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!
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.
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).
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).
So you want your music clipped or limited by the DAC? Or do you envision amps and speakers capable of generating shockwaves powerful enough to level entire cities?
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.
> 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.
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.
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.
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.
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.
@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.
Couldn't agree more with you! 192kHz is overkill as a "final" format.
16 bits is very limiting for music with lots of dynamics (ie: classical). Very quiet sounds sound quite bad at 16 bits, but since most pop music has about 6-12db of dynamic range, it doesn't make much of a difference.
I always thought the sweet spot would be 96-24. But the truth is, the market wants smaller and portable digital files, not higher quality music. Anything MP3 encoded will sound significantly worse than a CD anyways.
16 bits is not "very limiting" for anything, unless you think your ears themselves are very limiting.
Many things are mastered poorly— recording engineers crushing the dynamics in order to get the loudest possible signals— mostly a problem for pop music, but nothing is immune.
It's been observed that the various 'higher-definition' recordings have less brutal mastering— no doubt owing to the different audience they are marketed to. But this isn't a property of 24-bit vs 16-bit distribution.
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.
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.
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.
• 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.
The article states that greater than 16 bit has value in recording, just not in playback. If you took your 24 bit recording and translated the best 16 bits of it than output it through your 24 bit DAC then they are saying you won't here a difference. I say output through your 24 bit DAC so you aren't simply hearing a better DAC.
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.
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.
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.
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.
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.
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?
"Human hearing is limited to 20k because frequencies higher than that are perceived as painful? Dont agree with that one."
Yeah, that's a silly one. I disagree with it, too. It's a good thing it appears nowhere in the fine article. Are you actually confused about the difference between frequency and amplitude? Or did you misread the article?
"24 bit doesn't offer any advantages to sound quality? Sheesh."
As brazzy rightly points out, "Sheesh" isn't a reasoned statement. It's an ejaculation. And, it turns out, the author talked about why sound engineers record with 24 bits; It has to do with pragmatic reasons about leaving room for the highest and lowest frequencies in the audio being recorded without clipping, as well as with the author's discussion of Nyquist considerations in the distributed product.
Your post is wrong in so many ways that would have been easily fixed by reading the linked article with even 8th-grade reading skills that the reasonable reader has to wonder if you're being deliberately obtuse. Are you?
> 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.
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.
"Don't agree with that one" and "Sheesh" are pretty weak counters to detailed, objective arguments based on extensive research and decades of test data.
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."
stop being so paranoid about what you think you can't hear, and enjoy the damn music.
I'm a composer who works in film/games. I can assure you this is exactly what I'd like people to do when they listen to my music. I spend 99% of my time trying to create good musical ideas, and I spend 1% of my time getting the mix down. I get criticized (rightly) for this quite a bit, but it is hard to care about someone sitting in a >$10,000 labyrinth of sound equipment when I'd rather write a catchy tune.
Then again, when I write sheet music I have to endure some of the most soul-crushingly awful midi sequencing in order to check my work, so perhaps I'm too tolerant to terrible sound quality. Still, I'd rather people listened to the music, not the sound of it.
What's wrong with caring about both sides of it?
A good sounding catchy tune is something work spending that little bit more time on.
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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.
With a username like yours, I'm not surprised. :-)
A prescription is only as good as the likelihood that it will be heeded by the patient. A rug is an easy win; acoustic ceiling tiles and bass traps are a bit harder...
I might sound/be stupid for asking, but what's the actual physical response from something at 22 kHz+? I have a hard time picking up a pure sine > 17 kHz. I doubt I'd get any aural response from anything at 22 kHz, so what's the deal?
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I don't really know much about this, but wouldn't the 22kHz sounds potentially create beats in the lower frequencies?
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I agree. It should be about the performance, not the sonics. There are plenty of old Motown and even Beatles recordings with distorted vocals, bad edits, etc. Your brain passes right over them because of the emotional content of the music.
That's because they focussed on the most likely end-user experience:
>While Motown shortened song to fit into radio time, the company also produced records specifically with car radio audio quality in mind. Motown recording engineers set up car speakers in the studio so that they could simulate and perfect how a song would sound emanating from a car radio
- what's the point of engineering things to a set of conditions virtually none of your target audience possesses?
http://web.wm.edu/amst/370/2005/sp3/machinery_marketing.htm
This.
I worked out a long time ago, that I enjoy listening to _music_, not HiFi gear.
My advice to people who ask how to make their system sound better? Buy some music you enjoy more…
I can enjoy a wonderful performance of a great tune played through my laptop speakers - much more that I enjoy test tones or gear-demo-tracks through sound gear worth something north of a new car…
(not that I haven't been "that guy" in my past…)
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I'd disagree, at least among the people I know: they all have cheap HTIB systems and the single biggest, most cost-effective improvement you can make from there is to buy better speakers.
Then you can worry about your room.
No, even in that case, the room can still overpower the speaker. A $200 HTIB system in a properly treated room will sound a lot better than $28,000 Wilson Watt Puppies in a bathroom for example.
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You might be surprised how much of a difference EQ can make. As an experiment I once used 12 bands of parametric EQ to adjust the speakers in a cheap, old LCD monitor. Sure, you're not going to get any better bass response than before, but stereo imaging (nonexistent before EQ, perfect phantom localization after), spectral balance, clarity, distortion (due to not exciting resonances in the monitor), etc. were significantly improved. Most people could have added an EQ'd subwoofer to those tinny LCD speakers and been completely satisfied.
One of the many things EQ can't fix, of course, is room reflections, which can be helped by room treatments and speakers with a directivity better suited to the room.
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I should point out that after room treatment, my next recommendation tends to be an amplifier with Audyssey MultEQ in-room calibration. I've never heard a listening environment that didn't sound unambiguously better with it enabled.
Did you read the article? Are you sure it wasn't just unambiguously louder? ;)
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Audio tweaker with hacker leanings but without carpentry skills? Start your engines: http://drc-fir.sourceforge.net/
We have a lot in common.
It is said that in most cases 192 kbit .mp3 is indistinguishable from >192, and blind tests support that. Granted, there are instruments like castanets which make it easier to hear the difference. In general though, I can't distinguish 128 from 192 and I listen to music a lot. Also it's unlikely that my hearing is already damaged because I try to keep volume low.
But I've noticed that where I put the speakers makes a huge difference. I can easily tell the difference from speakers on the floor versus speakers on my desk. Where I'm at the moment also matters a lot. If I lie on the floor, floor speakers don't sound as bad anymore.
In the end, I use headphones. Midrange Audio Technica ones, and I'm probably already overpaying a bit. But I bought them for build quality and comfort, and I wasn't disappointed. I can have wear them for hours (Not healthy I guess, but I'm used to wearing them even with no music being played). Headphones have the advantage that it suddenly stops to matter where your speakers are and where are you relative to them.
Is the 192 the bitrate or the frequency response? I thought 192 in MP3 was the bit rate, not the maximum frequency response...
This effect isn't sooo surprising seeing as it even occurs with dumb mono guitar cab speakers and is very, very, VERY clearly audible there, even just moving your head a few cm in or out of the cones' axis.
Amen. Someone got the point of the article.
> Stop being so paranoid about what you think you can't hear, and enjoy the damn music.
Yes. Do a couple of blind tests with your acoustic system first.
> It's true enough that a properly encoded Ogg file (or MP3, or AAC file) will be indistinguishable from the original at a moderate bitrate.
Disagree. This claim seems to be ungrounded compared with others.
I can believe limitations with bit depth and sampling rate (although I'll take a chance to test myself if I get near good enough acoustic system). However, I definitely could discern in a blind test whether music I listened to was stored using lossy format with reasonable bitrate. It's usually quite audible with rock music that involves cymbals.
There's a specific "bug" in the mp3 encoding scheme which means that you get a pre-echo effect on fast attack waveforms. It's inherent in the encoding, so it can't be eliminated (although the higher the bitrate, the less obvious it is IIRC). If you how know to listen out for it then you'll spot it immediately.
AAC / Ogg don't have that limitation & at high enough bitrate should be indistinguishable from the source in a blind listening test, as demonstrated in a number of Hydrogen Audio listening tests down the years, unless of course you're using crappy encoders at which point all bets are off...
(Really, LAME is very good indeed these days. I eventually decided that I was going to get with the program and just encode all my CDs (backed up to flac files) as mp3 for portable listening. It's good enough, and I've decided not to listen for the pre-echo artifacts so that I won't notice them :) )
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A thousand times this. I never cease to be amazed at the number of people who will vocally argue the benefits of solid-silver wundercable, but who've never heard of mirror points or bass traps. $20,000 hifi systems in rooms with bare wooden floors and bare concrete walls. Subwoofers in untreated cubic rooms. People praising the transient response of their PMC MB2s in a room with chronic flutter echo. It's utterly dispiriting.
I agree the damn room matters so much more. Also the distortion of most speakers is already the bottle neck in most people's systems.
Alan Parson's makes this point, too: http://boingboing.net/2012/02/10/alan-parsons-on-audiophiles...
Exactly! I could not have put it better myself!
" … when the most under-performing component of most people's sound equipment also has the lowest-hanging fruit:"
And I was _so_ sure the next sentence was going to be something like:
"No, do not have any suggestions that will make your sound equipment make Justin Bieber sound better…"
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
> Digitally recording a triangle is the best example of why 48kHz is very limiting
The article's about distribution, not recording. I don't think anybody disputes the usefulness of higher sampling rates when recording.
> 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.
...and 48kHz audio contains those lower frequencies.
Stripping frequencies above 20kHz negates the effect on the lower frequencies since those lower frequencies are not "modified" by the higher ones. The human hear can actually hear the very high harmonics when they're combined with a lower fundamental frequency.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz. If it's played at 44.1kHz, the 30kHz frequency is gone and all you'll hear is 10kHz, not a "different sounding" 10kHz.
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> The article's about distribution, not recording. I don't think anybody disputes the usefulness of higher sampling rates when recording.
Didn't read the article, so commenting out of context, however it needs to be said that in sample-based music genres the distributed music gets used as if it were a recording. Maybe then it could be argued that higher sampling/bit rates should be available, if only for those who are sampling.
> 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.
That may well be true. But those mixed-down harmonics that are heard "live" would then be captured by the 16/44 (or whatever) sampling. IOW, the recording captures what you heard. Those upper harmonics have no emergent properties. Their effect is captured.
Bob
I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.
Just because it is difficult to record a triangle does not necessarily mean it is impossible to accurately recreate the sound (to human ears) using 48kHz.
> I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.
Yes, you're right.
In fact, some of the section X references don't even mention hearing, they talk about "alpha-EEG rhytms" (in this case "listeners explicitly denied that the reproduced sound was affected by the ultra-tweeter") and "bone-conducted ultrasonic hearing" trough the "saccule" ("organ that responds to acceleration and gravity and may be responsible for transduction of sound after destruction of the cochlea").
--
In fact, most of the claims of the article are around the fact that there is energy over 20khz and how it can affect recording process.
This is a well known fact, and this is exactly why engineers filter out sub-sonic and super-sonic frequencies, especially today: stuff that you can't hear (or feel) will just suck your headroom and make you lose the loudness war.
The only "good sounding" triangles you'll hear are those buried in a mix. Alone, it always sounds weird and "muted".
EDIT: Listen to the triangle at the beginning of Rush's YYZ. It's an old recording, but it sounds significantly worse than the analog version. It's been digitally mastered some time ago so if it was mastered today, it would probably sound better, but still not great. I heard a rumor that Rush is remastering all their albums "for iTunes" at the moment, so hopefully we'll be able to compare soon!
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Yes, but our ears only hear 20Hz-20KHz. So, according to Nyquist theory, you can recreate the entire signal that the human ear hears by recording those sonic artifacts that result from interference between supersonic harmonics.
So while it's true that the human ear can't hear well above ~18KHz, and the interference between high order harmonics are audible, it's also true that a properly recorded signal, sampled at 44.1KHz, oversampled, and filtered, can reproduce the exact signal the human ear is capable of hearing. At least according to theory.
The human ear is capable of detecting sound pressure as well as sound intensity, and while playback of the interference between harmonics can be reproduced faithfully in the sound intensity realm, the sound pressure levels will differ, and it is theorized that people may be able to tell the difference between the two. However, as far as I am aware, nobody has been able to demonstrate this reliably in practice.
What about sound outside 20-20k that affects us via mechanisms other than being directly sensed in the air by our ears? For instance, consider frequencies below 20 Hz that we can feel with our feet as vibrations in the floor, instead of hear with our ears? Or what about the possibility of sound above 20k causing a vibration in something other than our ears, which could have a subharmonic in 20-20k that gets conducted to our ears via bone?
I'd prefer recording technology to err on the side of capturing what we need to reproduce all of that, even if we aren't sure that we need it.
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Nyquist is true for static signals. Music is not static. Brick wall at 20 kHz and get audible phasing artifacts! (Even if your filter is phase linear)
I'm an audio engineer, too, and I agree that this has been debated to death. And I agree that frequencies above the threshold of hearing are more important than standard dogma (based on Nyquist theory combined with Pure tone audiometry) allows. It helps explain how audio gear with a 100kHz bandwidth sounds clearer than gear with a 20kHz bandwidth even when they measure the same in the audible band.
Have you read the Audio Technology magazine interview with Rupert Neve?
Greg Simmons: Geoff Emerick, the famous British Producer ?
Rupert Neve: Yes, he started me off on this trail. A 48 input console had been delivered to George Martin's Air Studios, and Geoff Emerick was very unhappy about it. It was a new console, made not long after I had sold the Neve company in 1977. George Martin called me and said, "please come and make Geoff happy, while he's unhappy we can't do any work".
They'd had engineers from the company there, and so on. The danger is that if you are not sensitive to people like Geoff Emerick, and you don't respect them for what they have done, then you are not going to listen to them. Unfortunately, there was a breed of young engineers in the company ( I hasten to say this was after I sold it !) who couldn't understand what he was bitching about. So they went back to the company and just made a report saying the customer was mad and there wasn't really a problem. Leave it alone, forget it, the problem will go away. They were acting like used car salesmen. I was very angry with it. So I went and spent time there, at George Martin's request, and Geoff finally managed to show me what it was that he could hear, and then I began to hear it, too.
Now Geoff was The Golden Ears - and he still is - and he was perceiving something that I wasn't looking for. And it wasn't until I had spent some time with him, as it were, being lead by him through the sounds, that I began to pick up what he was listening to. And once I'd heard it, oh yes, then I knew what he was talking about. We measured it and found that in three out of the full 48 channels, the output transformers had not been correctly terminated and were producing a 3dB rise at 54kHz. And so people said, "oh no, he can't possible hear that". But when we corrected that problem, and it was only one capacitor that had to be added to each of those three channels, I mean, Geoff's face just lit up ! Here you have the happiness/ unhappiness mood thing the Japanese were talking about.
copy here: http://poonshead.com/Reading/Articles.aspx
The article doesn't suggest only using 48kHz for recording and mixing. I don't think the author would disagree that recording triangles is difficult. He would argue that once you've decided what final audible frequencies you want to present to the listener, the best way to distribute them is at 16-bit 44.1/48kHz. It's a compelling case.
What if you want to sample the song later?
That's one thing I find concerning with the move to digital. With analog media, you can go back, re-record and get an improved result (provided the source is good) but District 9 (which was shot on Red One) will never have improved quality other than resampling because the source is set to a particular digital format with associated data quality.
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I completely disagree with the article having heard the difference many times myself. You can't record at 192kHz and hope to keep the same quality by distributing the final mix in 44.1kHz. It just doesn't work that way.
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Hey cmer, thanks for posting
I don't think I understand quite what you're saying and wondered if you could explain more. You and the article both say that humans can't hear above about 20kHz. If there are higher frequencies that create a harmonic at a lower frequency (e.g. a 33kHz harmonic that produces a sound at 16.5kHz) then surely that lower harmonic (16.5kHz in this case) will be recorded by the original recording equipment assuming it is recording at a frequency at least twice that of the highest audible frequency (let's say that this would be 48kHz, although there might be other DAC-related reasons to go higher).
I'm possibly being very daft here!
Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics.
If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only our first frequency will be captured: 15kHz. It will be exactly the same as if you only recorded 15kHz alone. The harmonics don't modify the fundamental frequency, they just trick the human hear. But when they're gone, they have no effect whatsoever.
Hope this helps!
EDIT: the frequency numbers I used are actually somewhat of a bad example. Harmonics are never exactly double, triple the fundamental. Those would be mostly inaudible. But you get the idea.
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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....
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.
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Playing Devil's Advocate...
The statement that frequencies above 20kHz don't matter rests upon the assumption that the ear is linear. If the ear is not linear (I don't know whether it is not not) then frequencies above 20kHz will matter, as the ear will be able to mix higher frequencies down to less than 20kHz. For example, if we have frequencies of 56kHz and 59kHz, the ear MIGHT be able to discern a difference frequency of 3kHz. No doubt this effect could be reproduced by signal with a sampling rate of 44.1KHz, but only if the analogue systems, before the sampling stage, reproduce any non-linearity in the human ear.
Incidentally, you can get speakers that create a localised beam of sound, that the person sitting next to you cannot hear. They work by transmitting frequencies above the audible range. These high frequencies can be beamformed by a relaitively small speaker array, so the sound is localised. They then rely on the non-linearity of the ear (or maybe the air around the ear?) to mix the ultrasonic frequencies down to audible frequencies. I guess there must be non-linearity in the human auditory system!
On the subject of 24-bits my understanding is that 16-bits is adequate, provided the levels (scaling) are set correctly in the recording. What 24-bits delivers is the ability to do a crappy job of the mixing, and still end up with the full dynamic range of the human ear. 24-bits is probably a temporary solution though, as manufacturers will engage in the usual Loudness War [1], and push the signal to the top of the dynamic range. Before long 24-bit audio will be equivalent to 16-bits (since the 8 least significant bits will be unused) and the next big thing will be 32-bit audio.
Having said all that, I'd guess that the speakers will be the limiting factor in most sound systems, not the recording format.
[1] http://en.wikipedia.org/wiki/Loudness_war
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Another "ex audio engineer" here, you can believe or not at your leisure. Many hours spent in high-end recording and mastering environments.
I'm not sure what your background in audio is, but everything he says is correct. High end frequencies well past 15k and up (22.1k actually) are widely acknowledged to influence the lower frequencies and play a huge role in the perception of the quality of a recording. This is an old debate with pros and cons on both sides, but in general you'll find the "Golden Ears" mastering engineers (Stephen Marcussen, Bob Ludwig, etc.) come down on the side of higher sampling rates.
Now, if your original recording was mastered to 16/44.1, then a transfer by way of 24/192 will probably actually hurt the recording. But if you're mastering from an original analog or high-quality digital, in my experience there's no question, higher sampling rates deliver better experiences.
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That data doesn't back your point at all. That data concerns what frequencies are present, not what frequencies can be heard.
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.
>Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible.
This [1] (widely accepted in the scientific audio community) study's conclusions disagree with your assertion.
>44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters, which are audibly bad.
This is not the 1980s, hardware has progressed beyond that point. Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems. Also see [1]
>That bit about intermodulation distortion is complete bogus. He talks about problems when resampling high-fs audio data.
I did not notice that in the article. It talks about IMD in the context of the analog chain and the transducers following the DAC, and it's possible that high frequencies can increase it.
[1] http://www.aes.org/e-lib/browse.cfm?elib=14195
> Modern (i.e. anything from 1995 onwards) DACs do not suffer from aliasing problems.
True, but they do so using (long, high-quality) high-cut filters. And these filters are pretty sharp, as they have to close within, say, 18-22.1 kHz. You can design them as linear-phase FIR filters with oversampling and all the good stuff, but physics dictates that sharp filters introduce distortion. A sharp filter like that is audible.
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44100 is not a bad sampling rate, but it necessitates very sharp aliasing filters,
When you're talking about recording, sure, but in terms of storage and playback, we solved that problem 20 years ago with oversampling.
You will still need a aliasing filter that cuts off between, say, 18 and 22.5 kHz to avoid aliasing noise. That is one sharp filter no matter how you look at it. You can use a high quality, long, linear-phase FIR filter, but you can't cheat physics: sharp filters necesserily introduce distortion, and such a sharp filter so close to the hearing threshold does not go unnoticed.
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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.
why do you think "This difference gets lost in the THD already at the DAC "? Do you have numbers to back it up? What's the noise floor of DAC? What's the noise floor of an output stage? Do you have the number?
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why do you think "This difference gets lost in the THD already at the DAC "? Do you have numbers to back it up? What's the noise floor of DAC? What's the noise floor of an output stage? Do you have the number?
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.
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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.
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.
There is a perversion going on both ends here. And by perversion I mean a distortion of truth in a bid to make a profit. This is not the worst that can happen, but is just worth mentioning. You probably put more mildly, but I am bit more harsh. Some people are irrational and spend money of stuff that they don't need and another group of people are perpetuating the lies and the marketing in an effort to extract the maximum amount of money from the other group (In other words your basic market setup).
Audiophiles are quite a fascinating group. These are people that can be rather rational in some respects (they could be doing research in some lab somewhere) but when it comes to audio equipment they will shell $2000 for HDMI cables. The salesmen and manufacturers that make these things ("high end" HDMI cables, 192kHz recordings) know this very well and they aggregate around this target set of clients.
I think that is exactly what is happening here. At some point storage capacity is just good enough and one can distribute 48kHz, 16bit audio to everyone. But what do you do next? Everyone is getting that and it is not new and cool anymore. What to do? Well increase the frequency and sell everyone a newer, better, higher fidelity thing, even though objectively human years cannot really hear the difference. Subjectively though, there is a huge difference. If you ask someone who just spent $50 for a 192kHz record if they like it better than say a $20 48kHz one, I bet you 100% of people will confirm that 192kHz sounds better and will be ready to go and buy more.
> 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?
Ultimately, sure. The world is full of products and services which only add value in this weak sense.
If the same wine tastes better if it's priced higher, then it still tastes better. But I think it's only honest that the consumer be aware that the increased utility from being priced higher is due solely to the fact of it being priced higher. Beyond that, I don't care.
One thing we can all agree on is that music is much more enjoyable if you think you're listening to it through good equipment or from a good source. Ultimately it's only the `thinking' part that matters. So I would make two points:
1. One point he's making is that playing audio sampled at 192khz through regular equipement actively distorts the music in negative ways. So now if you know this now you should enjoy that music _less_.
2. If you're adept metacognition (maybe that's not the right word), you'll realize a) you can get most of the enjoyment by buying equipment that's `pretty decent', and then not worry about it too much. b) you're probably fooling yourself by spending so much time/money worrying about having the best equipment, so you're probably not getting the maximum utility from the experience anyway. Or maybe it's the experience of trying to get the best equipment it self that's enjoyable, not necessarily the increased audio fidelity.
> 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.
SUCKER! I'm already working on 48/768 audio. It's amazing how clear the recordings are.
Well, if we accept that argument, then just about any means of signalling "this sounds better!" will work. How about we choose something that doesn't waste bandwidth?
That's true. It's kind of the Monster Cable model. BTW, I'm not saying that marketing and whatnot should deceive less technical consumers and trick them into spending more money than they should (which is basically what Moster Cable does). But when you explain to technical people why something like 24/192 isn't better (other people in this thread have pointed out, this isn't totally accurate in the first place), and they understand what you're saying but still prefer it, by all means, let them buy it.
This is the same reasoning that somebody used when I was debating with her if insurances should reimburse homeopathic and other alternative treatments. Her reasoning was 'well if it works, it should be reimbursed, doesn't matter if it's from a placebo effect or not'; my position is that they shouldn't be reimbursed, but quite honestly, I don't really have a rational reason for it (at first I thought I had but it turned out I couldn't formulate it, which is the same as not having it).
So, while I have no option (for now) but to acknowledge your position, I still feel dirty for doing so.
If there's no point arguing against something that people will eat up regardless of evidence or fact,
why are you arguing against the conclusion of an article that has this many upvotes on HN?
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 - starting 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.
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.
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.
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.
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!
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.
1) 48kHz sampling does include headroom.
2) increased sampling rate does not improve timing. This also has been researched in detail (because it sounds like it could possibly be true given that the ears can phase match to much greater granularity than the sample clock). It was found false in practice, and in retrospect, the sampling theorem explains why. The Griesinger link discusses this with illustrations, and provides a bibliography.
To avoid the trouble of digging up the link: http://www.davidgriesinger.com/intermod.ppt
Slides 29-35 address this point.
> it's always good to have a little wiggle room
48kHz already has enough 'wiggle room'. How many people do you personally know that can hear a 24kHz sine tone?
> more granular timing for the start and stop of transients. ... it's something I don't see mentioned often.
Probably because it doesn't make sense. Human ears cannot hear frequencies about 24kHz and Nyquist tells us that 48kHz is enough to completely capture all the detail of a signal at that frequency and below.
You can get the same theoretical benefit by oversampling on playback. And a lot of audio equipment does just that.
Not really, for two reasons -- unless you're talking about glitch music, transients are unlikely to ever be so sudden that the difference between 0.03ms and 0.005ms could possibly matter.
I'm pretty sure that #2 isn't true; signal processing folks will be able to phrase this better than I can, but I think that if you have enough information to capture the waveform at a given frequency, you also have enough information to precisely place it in time - phasing errors are more likely due to quantization error, which is about bit depth, not sample rate. No?
[edited: I was wrong]
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> 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?
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.... that suggests the brain is sensitive to timing differences between ears as low as 10 microseconds, or 0.01ms.
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I get 10mm vs 1.7mm but it's roughly the same argument http://www.wolframalpha.com/input/?i=speed+of+sound+*+0.03ms... http://www.wolframalpha.com/input/?i=speed+of+sound+*+0.005m...
0.03ms is 33kHz - you can't, no matter how much you want to, make a granular timing that is faster than at least one cycle of the frequency you are using. 0.005ms is 200kHz BTW.
This isn't true. Sample a bandlimited impulse. The exact timing is encoded into the gibbs oscillations of the signal. So long as you have a high enough SNR you can have timing as precise as you want. (and because the ear doesn't work with ultrasonics— it is itself bandlimited— it uses the same phenomena for timing)
humans are particularly sensitive to jitter.
Humans are sensitive to jitter, but jitter isn't a major problem with modern digital electronics and reclocking strategies. This ArsT thread hashed out these issues a couple of months ago: http://arstechnica.com/civis/viewtopic.php?f=6&t=1164451...
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?
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.
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).
There are two reasons I don't think this will happen:
1. People would use the tracks to create custom remixes which they would then distribute. What happens when a remix becomes more popular than the original track? Artists generally have to pay other artists to remix their songs (usually via royalties).
2. Creativity. When an artist creates something they want you to hear it the way it was intended. Allowing you to remix it however you like takes away a lot of the creative control from the artist.
Regarding remixing. Artists usually don't "pay" each other, but return the favor - if it's the right term to say. E.g. artist A remixes a song of artist B and artist B in turn does the same for artist A. Or if they are all on the same record label artist A does a remix for artist B and later B makes a collaboration with A. I've noticed this in electronica/edm music artists at least.
And another important remark: some artists are flattered when someone asks them to make a remix for their song. (Imagine you're an artist and your idol asks you to make a remix of his song.)
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That was predicted (and suggested) by Glenn Gould some forty years ago. At the time, anything with higher fidelity than, say, a bad telephone connection, was analogue, but we were stepping into the world of quadraphonic sound (which died soon after in the analogue kingdom), but he was a big proponent of the listener as participant (hey, it was Toronto and McLuhan was still around) and was convinced that technology was the only limiting factor at the time. (To put thing in perspective, he was also very much anti-concert--he hated what he called the "non-take-two-ness" of live performance.) Let's just say that the idea was no more popular among artists then than it is now.
I'd love that too... damn, should have put it in the article...
Closest I've found is to take the .mogg files out of the Guitar Hero games and use those to make new mixes. :-)
Many of the groups I listen to do do this -- this is certainly not that rare. Sometimes they go for a bit more money by releasing a separate CD with karaoke tracks, for example, but at least if you want it, it's available.
Of course, you can very easily get just the vocal track by subtracting the two. Sometimes the "non-vocal" track will still include backing vocals or the like in appropriate places, and just pull out the main vocal track.
Some musicians have even released every single track of their work separately; see "Desperate Religion" on http://en.wikipedia.org/wiki/Trilogy_(ATB_album) , intentionally inviting remixes.
For music where the vocal tracks aren't released separately, you can often pull them out nevertheless. The best is if you can get the audio in 5.1 -- vocals are almost always center-panned, which makes extracting them quite easy.
Some songs when released as a single have Acapella and Instrumental versions of them as well. There are also compilations with only acapellas and compilations with only instrumental versions of the songs.
And when you have them, just use something in the like of Ableton Live and that will be it. I think that that's what you mean right?
It will be a great idea to have tracks released as several `layers` so that the user can choose which of them to play and which not, for example the bass/beats layer, layer with melodies, layer with the percussions, layer with the vocals of course, but that sounds like semi-studio production.
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... 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.
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.
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.
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.
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.
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.
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.
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.
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').
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).
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.
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.
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.
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...
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).
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.
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.
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?
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?
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).
Your question is somewhat amusing. A standard CD player uses 1-bit DAC (it's either on or off) at a yet-higher frequency to achieve better linearity. Filtering is quite easy in the analog world.
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.
If the beat frequency is audible, it will be on the recording. Obviously.
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.
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Alas, they don't— you can easily demonstrate this for yourself. Startup an audio editor and generate tones at 25k and 28k (make sure you can't hear them— otherwise you have severe distortion screwing up your test) then play both at once. You will not hear a 3kHz tone.
The tone you get from an acoustic beat is not a real tone— it's a perceptual quark that requires you to be able to hear the tones in the first place.
I tried this in Audacity, with the project set to 96kHz and two tones at 25kHz and 28kHz. I couldn't hear either of the tones individually, but I could hear a tone when played together. This is on Windows 7 with the sound card configured for 24-bit/48kHz. Am I running into resampling artifacts somewhere in the chain?
EDIT: it turns out Audacity won't generate a tone above 20kHz (the UI accepts the value, but when you reopen it the value has been rounded down), so both of my generated tones were actually 20kHz.
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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.
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.
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!".
Except that human ears don't directly perceive the waveform, only the frequency decomposition.
There's been countless A B X tests that showed conclusively that the vast majority of people cannot tell the difference between samples with and without the 22kHz+ frequencies.
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.
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.
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.
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!
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.
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).
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)
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).
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!
I just want floating point, then this silly loudness war would end (to some extent, since you can make the mix almost infinitely loud).
It would make no difference at all.
So you want your music clipped or limited by the DAC? Or do you envision amps and speakers capable of generating shockwaves powerful enough to level entire cities?
Hotblack Desiato, is that you?
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.
Never mind. I read 192kbps instead of 192kHz. 24 bit might have some advantages but 192kHz not.
> 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.
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.
Almost certainly because it was a different remote control when you were a kid. Some remote controls emit a lot closer to visible light than others.
Would someone explain should I use 44.1 or 48Khz?
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.
I think 192kHz is the sampling rate used by the A2D converter vice verca. It is not the actual frequency of the sound (data).
The article states the point to using this format; keeping master-quality originals.
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.
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.
@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.
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Couldn't agree more with you! 192kHz is overkill as a "final" format.
16 bits is very limiting for music with lots of dynamics (ie: classical). Very quiet sounds sound quite bad at 16 bits, but since most pop music has about 6-12db of dynamic range, it doesn't make much of a difference.
I always thought the sweet spot would be 96-24. But the truth is, the market wants smaller and portable digital files, not higher quality music. Anything MP3 encoded will sound significantly worse than a CD anyways.
16 bits is not "very limiting" for anything, unless you think your ears themselves are very limiting.
Many things are mastered poorly— recording engineers crushing the dynamics in order to get the loudest possible signals— mostly a problem for pop music, but nothing is immune.
It's been observed that the various 'higher-definition' recordings have less brutal mastering— no doubt owing to the different audience they are marketed to. But this isn't a property of 24-bit vs 16-bit distribution.
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.
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.
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.
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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.
I believe you are confusing 192kbit compression with 192 khz sampling rate. Not at all the same thing.
Whoops! Guess I'll read more carefully, then.
ahem 192 kHz. We're talking sampling frequency, not bitrate.
The article states that greater than 16 bit has value in recording, just not in playback. If you took your 24 bit recording and translated the best 16 bits of it than output it through your 24 bit DAC then they are saying you won't here a difference. I say output through your 24 bit DAC so you aren't simply hearing a better DAC.
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...
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.
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.
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Mmm loves me some inappropriate downvotes.
Probably because you couldn't come up with the actual study you mention.
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.
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.
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?
"Human hearing is limited to 20k because frequencies higher than that are perceived as painful? Dont agree with that one."
Yeah, that's a silly one. I disagree with it, too. It's a good thing it appears nowhere in the fine article. Are you actually confused about the difference between frequency and amplitude? Or did you misread the article?
"24 bit doesn't offer any advantages to sound quality? Sheesh."
As brazzy rightly points out, "Sheesh" isn't a reasoned statement. It's an ejaculation. And, it turns out, the author talked about why sound engineers record with 24 bits; It has to do with pragmatic reasons about leaving room for the highest and lowest frequencies in the audio being recorded without clipping, as well as with the author's discussion of Nyquist considerations in the distributed product.
Your post is wrong in so many ways that would have been easily fixed by reading the linked article with even 8th-grade reading skills that the reasonable reader has to wonder if you're being deliberately obtuse. Are you?
> 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.
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.
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"Don't agree with that one" and "Sheesh" are pretty weak counters to detailed, objective arguments based on extensive research and decades of test data.
Principle frequencies well above 20k as well as their sympathetic harmonics are pretty easily audible by me, try it.
24 bit is also extremely easy to hear. Arguably more important during the recording phase when headroom is valuable.
Its just as easy to qualify everything with "placebo effect", as it is to be dismissive
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