I even transcoded some 24/192 FLAC Pink Floyd I had lying around and made him do a double blind test to show him that he'd prefer the slightly louder song every time, even if the louder song was 192kbps vs the FLAC. He did. He still doesn't believe me.
He still thinks he can hear the difference between FLAC and MP3 to this day. He works as a sound engineer now.
I don't think any amount of reasoning will make some people change their minds. Some people buy $500 wooden knobs to make their volume pots sound better. (or was that a hoax? i can't tell anymore)
Some people buy small pyramids to elevate their cables off the floor, some people buy mats to put onto your CDs before putting the CD in a player (http://dagogo.com/millenniums-m-cd-mat-carbon-cd-damper-revi...), some people buy $1000+/meter digital interconnect cables (http://www.theabsolutesound.com/articles/transparent-referen...), some people buy $7200 power cords (http://www.theabsolutesound.com/articles/crystal-cable-absol...) and $350/m HDMI cables (http://www.theabsolutesound.com/articles/nordost-releases-fi...).
Self-styled audiophiles are, by and large, idiots with way too much money plagued by magical thinking. Developer bullshit has nothing on them.
An 80-minute, 700 MB CD-R fits 80 * 60 * 44100 * 2 * 2 / 2^20 ~= 807 MB of audio.
Why is that?
The 100MB difference is not just due to the audio TOC being of smaller size than the ISO9660 or UDF file system metadata. It's also because of differences in error correction. I don't have the spec on hand but I recall from when I was investigating this that CD-ROMs use more bits for error correction than audio CDs. That's why you can fit more audio data than "filesystem data" on a CD-R. Reading (ripping, digitally) an audio CD will likely result in different digital audio files every time, since the error correction is not that good, but good enough, for audio.
I read into this when I was wondering why my CD-DA extracted .wavs came out with a different checksum every time. Vibration is one of the factors that would make the same audio CD, read with the same CD player, produce different digital signals some of the time or even every time.
CD-ROMs however, which store digital data, need better correction - you definately don't want a bitflip in your .exe, while a minor amplitude diff — an uncorrected bitflip in the upper bits of a 16-bit PCM signal — is no biggie.
So… I'm not saying that the people using CD mats are informed (or have tested whether the mat makes a difference, or would even know how to go on about testing this, scientifically), but there's more to it than what I originally thought — which was "it's digital so it's never degraded". I wouldn't have known without checking the md5sum of my .wav, though.
This sub-bit jitter and interference can travel along with a digital file and sneak right past your ordinary bit-level error detection and correction, no matter how lossless you make it. That's because these errors aren't visible in the bits. They occur at a deeper and more subtle level, in between the bits.
Even if you prove mathematically that two files contain the exact same bits, you can't prove that the human ear won't hear any difference, can you?
Same file -> same playback.
If you hear the same sound file twice (or two identical files) and hear something different, you software is broken or you're imagining things.
Can you turn this into mere "bits"? Of course not!
That's why it is so important to protect against sub-bit quantization errors, and this can only be done with proper interconnects. Ordinary cables allow the bits to travel willy-nilly until they jam up against each other creating a brittle, edgy soundstage. Quality interconnects are tuned, aligned, and harmonically shielded to keep those precious bits - and the all-important spaces between them - in a smooth flow.
And then, we can hear all of the things that make us human.
That comment is just perfect.
(I put the scare quotes on because I haven't actually bothered to check if there is an audible difference. But it does confirm the GP's experience.)
I remember experimenting with writing a CD ripping program in the 90s, using Windows APIs, and I found, like you, that I got different data each time. But modern rippers such as EAC does this stuff much better and will for the most part give you bit-perfect rips.
That mat does nothing. And if you read that linked page, you will see that he claims it drastically improves audio quality (bass, etc.), which is pure nonsense.
That makes sense when you have an analog version of the audio picked up by an analog transducer (i.e. a vinyl record) but makes no sense with an isochronous stream of quantized samples.
I suppose a vibration could cause a small phase shift in when the sample physically appears under the LED, but but since the D->A conversion is clocked by a PLL it is irrelevant.
If you have extreme warping or shaking (e.g fling your discman onto the floor or stick your finger on the disk while it's spinning) then a sample might not appear at all, but that's something different than you are talking about.
I suppose it's theoretically possible that some extreme warping or vibration could cause a bit flip, but that's what the ECC is for.
"[…] The change in height between pits and lands results in a difference in the way the light is reflected. By measuring the intensity change with a photodiode […]"
I'm no signals expert. Are you saying that there is no quantization in that intensity change measurement?
Regardless of quantization, maybe you're right on vibration not being a major source of errors (I know little about electronics and PLLs).
But then, what are the error sources that made the engineers put an extra 276 bytes of Reed-Solomon error correction per 2352-byte sector on a mode-1 data CD-ROM (vs none on an audio CD, which has just has the frame ECC and nothing extra). See https://en.wikipedia.org/wiki/CD-ROM#CD-ROM_format .
Some statements just don't age well ...
You sure there isn't something in the wavs like a creation date field that would always cause the checksum to be different? That would make way more sense than "vibration"....
I agree on the "plagued by magical thinking" part, but not all these people are idiots. Some of them are quite intelligent, in fact. I think they just want to be "in the know", and are able to suspend their normal skepticism in order to belong.
One of the smartest and most productive programmers I ever met was taken in by this nonsense. He replaced all the metal bolts in his power supply with teflon because the metal bolts disrupt the magnetic field around the transformer and you can hear that, maaaan!
He did have a nice sounding system, for which he spent about $10k more than one that would have sounded the same.
Not to be picky, but it bugs me to see when people talk about two different things and don't understand each other.
Intelligence is not a linear value that could be compared like "person1 intelligence > person2 intelligence". With both of these terms there are always skipped implication of the area of intelligence.
By "idiots" he meant "small amount of/incorrect knowledge in the area of audio quality and human hearing", and by "intelligent" you mean "big amount of knowledge and efficiency in the area of writing computer code".
Now, if by "idiots" he means "ignorant people" then he's using the word incorrectly. But there's no actual indication that's what he meant. At some point you just have to assume people mean what they say.
And despite what people want to believe, the last fifty years of psychometrics research indicates there really is such a thing as "basic intelligence" (which they call "g"), and people with more of it do better on a wide range of intellectual tasks. So you really can say "person1 intelligence > person2 intelligence".
You can be very intelligent in one domain and be a complete idiot in every other domain. The result ends up being unless we're talking about that one domain, they're an idiot.
It does if he starts voicing 'iditotic' opinions and beliefs as to the construction of log cabins.
But I suspect the metal ones are better. More magnetic shielding
So, IF your audio system uses a linear power supply (and it should) AND it is badly filtered (it should have good filtering) you can hear the 60Hz/50Hz from the power network (assuming it's not creeping in your system through other means as well - most likely they are)
(Also, the reviews on this $15,000 speaker cable are amazing: http://www.amazon.com/AudioQuest-Terminated-Speaker-Cable-Di... )
Not just that. But the arrogance and fanboyism is rampant.
God forbid you ever consider buying a Bose or Beats product.
Subjectively, you might like them, but the faithfulness of audio reproduction is not a subjective matter. You can play a tone and measure how well that tone is actually played back.
You can then also objectively compare things that produce that playback quality at various price points and figure out if they're priced competitively.
There is plenty of fanboyism in high end audio, but that's not why they say Bose and Beats are shitty. It's because Bose and Beats ARE shitty.
*The Solo 2 Beats actually measure very well. They're even competitively priced... with other overpriced fashion statement headphones. They're still overpriced vs. headphones that are just meant to play music well.
I am finding it hard to believe that they are actually shitty, while I find it very easy to believe that they are way overpriced.
I have never listened to Beats headphones but I imagine they have a lot of base-boost (based on absolutely nothing), but that is not the same as shitty.
If you are going to reduce them to the basest level of what the purpose for a speaker or headphone is, to reproduce the input sound, then yes, they are shitty, because they are not good at that.
From a purely objective standpoint, you are going to have to judge them based on that. Why would you want the speakers or headphones to make a different sound than what the signal is?
If you want to move away from an objective measurement of what makes a headphone good or not to something that's purely subjective (i.e. 'I like how they sound'), it's impossible to answer that question.
The Solo2 are a pair of Beats headphones that actually measure really well - they're good at the base purpose of a transducer. But they're $250. You could buy a pair of Sony MDR-7506 that measure similarly (IIRC, a bit better, even) for $85.
Or Superlux HD668B which can be found for $30 last time I checked.
Or if you want IEMs, you're not going to beat Hifiman RE-400 for any IEM under $300.
Though I don't personally like the cold, base heavy sound of Beats, I don't really get how you could know this, because most people have no idea what a piece of music should sound like. They know how they think it should sound, they know how they like it to sound, but very few know how it should sound. The only real exception to this is music with "real" instruments like pianos who's sound is familiar to enough people that their reproduction can be reliably determined. Even then, however, unless you know the piece well, it's unlikely most of us are in a good position to make a judgement about the speaker's quality.
So what factors are you using to determine if the sound is reproduces correctly?
This is a pretty scientific matter - when I say "the purpose is to reproduce the input sound", we can tell exactly what is supposed to be reproduced, and we can tell exactly how capable the speaker is of reproducing it.
Some exceptions have to be made due to how having headphones on your head causes the sound to change, but again, these are pretty much known quantities - to get the equivalent of a flat response from a speaker, you will see change X in bass response, change Y in treble response, etc for headphones.
It's not a question of esoteric "The artist and recording engineer meant for this to be played on Kef blades powered by a Cary tube pre-amp feeding into a Mcintosh amp setup using a rail to rail ladder DAC", but a "We know how frequency response should look when measuring equipment and if it doesn't look like that then the sound you are getting out of it is different than the source material"
You are assuming the song was mixed by someone wearing headphones that perfectly reproduce the input sound.
Suppose the person who mixed a song was using beats headphones or other headphones that audiophiles consider inferior but that they know the majority of people use to listen to music. Wouldn't that then mean Beats headphones actually provide the listener with the actual, intended experience?
There's a few reasons for this. The most pragmatic is that doing so will produce the track that sounds the best on the widest variety of setups - EQed or not. There's also not any single headphone out there that is used so predominately that it would make sense to cater to it in specific. The closest might be apple earbuds, but people using those probably aren't too concerned about sound quality anyway, so it doesn't make sense to mix with those in mind either.
From a theoretical standpoint, you're not necessarily wrong, but it's just not how things currently work, and there's not really any reason why it ever would work that way in a professional studio.
I make no claim as to what the people making music exclusively in their bedroom are doing, though.
I have a set of Sennheiser HD 202 which don't have anywhere near the same leakage and cost £35. I haven't tried Beats so can't say much about audio quality, but in my experience high leakage usually means that the audio is poor too. It also means you will listen to music louder to compensate, which leads to more distortion.
> I am finding it hard to believe that they are actually shitty
In that case the marketing team have done a good job :-)
This makes the mistaken assumption that isolation and good sound are related, which -- as open headphones and speakers can attest -- is not true. The goal of a speaker or headphone is to reproduce music faithfully. Unless you are familiar with the music's origin or it has real instruments who's sounds you can easily identify, it's impossible for most people to tell if the music is reproduced "faithfully". So there are a couple of general rules that most "audiophiles" will consider when dealing with volume:
1. Music played at louder volumes generally sounds better than that at lower volumes. You can hear more of what you are intended to hear.
2. Music often goes up and down in volume, so you want to hear the broadest range of volume.
3. The best listening devices both allow high volumes without clipping and low volumes with clarity.
The point is, just because you can hear it, doesn't mean they are bad headphones.
It also doesn't mean they are good headphones or that the people aren't inconsiderate. It simply means that "sound leakage" isn't really a decent criteria unless it's something that important to you.
> Bose and beats* are by, every /objective/ measure, shitty products.
> God forbid you ever consider buying a Bose or Beats product.
If you need faithful audio reproduction, start with the room. There are reasons for buying a portable Bluetooth-enabled speaker, and also reasons one may consider specifically Bose SoundLink. Sound quality, in this sense, is not among them.
I've got excellent bookshelf speakers that were cheaper than the equivalent from Bose, but the reviews and tests showed them to be way better.
My (small!) speakers end up producing way too much bass for the room they're in, in fact, and I use Foobar 2000 with the "MathAudio Room EQ" plug-in to get a flatter speaker response from them. But their problem isn't that they can't produce bass notes.
There are three brands of headphones that pros use: Beyerdynamic (typically DT-100 or DT-770), Sennheiser (typically HD-5/650) and Sony (typically MDR-7506/9). Beyerdynamics have somewhat better isolation so they're more popular in music studios, Sonys are more comfortable when you have to wear them all day so they're more popular on film sets; I favor the 7506 and am on my 4th or 5th pair. Some people love Sennheisers but I personally don't care for the ergonomics.
They're not beautiful, lightweight, or fashionable, but they're a lot nicer to listen to - which is why one or other of them was almost certainly used at the recording stage. If it was good enough for the people who made the recording, it's good enough for you. Also, you'll save money compared to the 'quality' consumer brands.
But, your statements about "pro" headphones are accurate. They aren't the nicest looking, but they are really good, and I always recommend a good pro set of headphones over the marketed crap from Beats, Bose, Monster, etc. $250 will buy a lot of headphone quality from one of the pro audio manufacturers.
I can't stand the Sonys. They're specifically designed for tracking & editing - all that screech points up Bad Things Happening. But they fatigue my ears.
Laugh now, but I landed on the Koss KTXPRO1 ( which are $20 to $40 ) and have basically stopped looking :) Most comfortable thing I've ever used and they're actually pretty flat, except for a little bass bump and a smidgen of upper mid. I think I'm on my tenth pair. They're a bit too light weight - if you catch the cable on something they'll fly off your head.
And yeah - I bet the $500 vs $20 figures into my perception of things.
I can mix on 'em, at least to the rough stage.
And for heavy noise cancelling goals it was very good (like, being able to work with someone with a lawnmower or a drill next to you)
Granted, half of the noise isolation is passive, half active, still, very good
Your comments over those years don't seem bad at all - sometimes perhaps a little confrontational but not aggressively so. Perhaps HN could allow users above a certain karma threshold vote on [dead] posts, with those scores going towards a "repeal fund" - make decent comments over a certain period and get temporarily un-banned.
Sennheiser HD280 Pros have extremely good passive isolation, and will beat QC15s at a fraction of the price, in both isolation and audio quality.
So yes, Bose still loses when you look at the big picture. Bose is very good at marketing, they are not very good at making quality audio.
The sound is very clear, but not balanced. Neither an expert nor a musician though.
Erik Larson is pretty vocal about his use of LCD-2s for mastering. Which... Honestly, I'm not generally a fan of his work, so it's not necessarily a ringing endorsement.
Also kind of surprised at your lack of mention of AKGs - they're another very popular brand for studio work.
The MDR-7509 and its successors the 7910 and 7920 have a lower impedance than the 7506 (24 vs 63 ohms) so if you plug them into the same sound source the higher-numbered models will be a bit louder - and as we all know, 'louder = better' for most people. This plus the larger driver is somewhat helpful for DJs, who work in very loud environments, but that's a fast track to hearing damage.
Why I like the 7506 so much: on film sets I give them to eople to listen in and they say 'is it on? I don't hear anything.' Then I turn the volume down or make a small noise next to the boom microphone and jaws drop. Plugged into a quality microphone like a Schoeps, which has a very flat frequency response, it's like there's nothing there. I always have two pairs now because if one gets damaged I can't deal with other brands at all.
This has little to do with the article. If you've got 100 bucks to spend on a pair of headphones, it's only fair to point out that with certain products you're not getting the best sound out of your money.
Meh, they're okay, but there are better choices out there.
I've a pair of Sennheiser HD600 and it's one of those things that make you go "holy cow, all the hype is justified".
And no, I'm not one of those folks who think gold-plated cables make a difference. Right now I'm listening to MP3 Internet radio on a pair of cheap behind the neck street cans.
Work bought me a pair of 380 HD Pros, and I'm impressed on how much of an upgrade they are over the 280s - I can only imagine how good the other Sennheisers are.
The 280s are closed back. Great for isolation, for not letting ambient sound interfere with the music. It also changes the way the transducers work, a little bit.
The 600s are open back. Obviously there's no isolation, but the transducers work more freely. It's a bit easier to distinguish tiny sounds from a huge background.
I've both the 600 and the 280. Great phones both, in different ways.
I nearly went off on a tangent and bought an amp etc, but I'm happy with my much cheaper HD380's - great price/performance :) But those 600's are awesome.
 I've since learned it's called soundstage
 How would the source influence soundstage? Sounds irrational to me. Hey, one sounded better than the other and I don't know why.
I'm certainly not complaining.
Unless it's an insulator (like fiber optic cables), you are hooking both a 50 foot antenna and a digital transmission line to your box; if you want just the digital transmission line, you have to insulate the hell out of the antenna part of it.
I'll head back to my 40$ crappy mp3 player now.
Our company would go to the remote places of the earth to hunt down copper dragons (as in DnD) and harvest their veins to make audio cables.
The "natural", "organic" copper has a warmth to the audio signals flowing through it that artificially produces cables just can't provide (they have harsher undertones).
Then we'd also have silver and gold cables, harvested from, you guessed it, silver and gold dragons.
These would truly be "monster" cables.
As we melt the gold we mix in a few atoms of "rare earth" elements (rare == expensive == so good "they" don't want you to have access) which is then diluted by adding more melted gold until only the imprint of the rare earth atom remains.
The gold will the be hand drawn by virgins (in truth these will be strong, hairy, 50-yo virgins with dreadful hygiene and B.O. though strong enough to pull, but we need not add all that confusing stuff, we'll just say "virgins"). The wire will be lovingly laid into hand-made insulation made from organic pinniped leather.
I see a variety of future applications both in the home (connect your cable modem to your WiFi access point) and business (data centers). To quote Rony Abovitz, we'll soon be "the size of Apple".
Invest now, while you still can!
Let the 'idiots' spend their money driving an industry that is combining the creation of electronics with functional art. I'm not sad to see a $40,000 DAC. I don't have to buy it, and it's cool to know someone built something of silly 'value'.
For example, look at this thing: https://www.naimaudio.com/statement It's absolutely silly, and the cost is outrageous. I'm happy that they built it though. It was actually built as part of the acquisition that Focal made of Naim. It seems that they allowed the engineers at Naim to go nuts once the company was acquired.
I like seeing silly things that people build. It doesn't make me sad that someone spends thousands on ridiculous items that from an engineering perspective don't make a difference.
$40,000 DAC? someone really went ahead and fabbed their own silicon (~$1mil for low volume run)? or did they maybe picked up $100 (at most) part, put it in a shiny box and started looking for suckers?
Do you even realize what $40K gets you? we are talking military grade Agile^^key'hole/RohdeShwartz multi gigahertz arbitrary waveform generators here, not some pityfull audio stuff.
Here are some examples of multi thousand dollar scams:
Audiophoolery is on the same level as creationism, it only works on uneducated simple minds with no metacognition.
I don't even know what that is. But that's what I'm calling my next breakcore track.
In audio, you have people who love vinyl because they enjoy the distortion it makes, and that's perfectly fine; but you have others who somehow believe it sounds closer to the original, which is demonstrably ridiculous.
Trying to defend something through completely subjective argument is silly, I've a hard time discussing 'objectively better' technology with audiophole folks, but if someone said 'I like this more' I really can't hope to debunk that through any sort of mathematical characterization of performance.
Only if they shut up and never tell other people that should be listening at 24/192. However, the people being complained about here spout nonsense like that all of the time.
I think you find you have it backwards. I don't hear anyone here telling you that you should be listening at 24/192. Go look at all the comments and count them up. All I hear is people saying that you should be listening at 16/44, because it sounds exactly the same, or even sounds better, and if you think otherwise you're obviously an idiot, stupid, audiophool who spends $5000 on a power cable.
I sure know who I think should shut up. It's all those arm-chair-experts who don't even own any decent hifi gear. Why would they? It's all crap and my second hand ipod headphones beat it all hands down anyway. Right?
The other fun stuff is building your own Speaker kits, hook all this up with a Pi Music Box and you have yourself a kind of home made Sonos.
The best definition of audiophile I've heard is somebody who listens to equipment, rather than music.
Of course your favourite Pink Floyd sounds better on a decent stereo rather than a clock-radio, but if somebody is forever chasing the proper "colour" for their speakers, or swapping amps for the perfect tone... they might be an audiophile.
edit: s/you might be/they might be/
Audiophiles are an easy target. There are a lot that do stupid shit like buy $5000 power cables, expensive risers to lift cables off the ground, etc. A lot are pretentious, even if they're not insane or dumb.
But that's a rather inflammatory, and in my opinion, unfair position to take. I would probably be considered an audiophile - I have put quite a bit of money into audio equipment. But I love music. I listen to it basically constantly. It's one of my primary sources of entertainment - and I don't just mean 'I have music on when I do other shit.'
Each week I spend probably 20 hours doing nothing but relaxing with a bit to drink and some music on. Not reading, not surfing the net, not doing anything but closing my eyes and enjoying the sound. I'm listening to the music.
At times, yes, when I have been testing out new equipment before deciding if I want to buy it I go through and I do blind ABX tests with level matching. In this case, yes, I am listening to equipment. But this is a very minor portion of my total listening time.
I know you're probably not being totally serious with the post, but I do think it's a bit unfair towards those of us that love music, but also have invested time and money into getting a setup that sounds better for increased enjoyment.
192 KHz has nothing to do with 192 kbps.
I'm one of those audio engineers who mistakenly thinks I can tell between an MP3 and a FLAC. Yet somehow I understand the difference between sample rate and encoding bitrate, and you do not.
All of this is besides the point. I'd much prefer a 24/44 sample than a 16/96 or 16/192. Bit depth has a much larger impact on the sound than sample rate.
You'll frequently find 1-bit A/Ds and D/A at > 5Mhz on high fidelity systems. That 1-bit signal is converted to/from a higher bitrate, lower sampling rate signal without loss of fidelity. If you're interested in looking at alternate bitrate encodings you should just look at the Super Audio CD format https://en.wikipedia.org/wiki/Super_Audio_CD
On your second point, I agree, we live in a noisy world and hearing 144dB of dynamic range would require serious isolation.
What most people should be able to hear with 90Dbs of dynamic range, are the harmonics created by undersampling a high frequency signal. To quickly explain I'll use a 1-bit lower frequency scenario. Lets say we have a 2Khz sine wave and a 1-bit 5Khz sampling rate. The 2Khz signal is going to be represented by a different 2 samples every cycle. The result will be a signal that is no longer a sine wave and closer in frequency to 800Hz (wild approximation) than 2Khz. Low pass filters are used to to keep those harmonics from being too pervasive but they still sneak into the signal near the high frequency range. Transpose this example to our current audio standard and you might realize that in order to accurately represent the high end we need a little more than 16bit 48Khz.
In terms of attenuating the high frequencies before downsampling - have I misinterpreted Nyquist? I thought that there was no loss in fidelity, right up to half the sampling rate.
So you're right that you lose information as you go higher in frequency, but there is also commensurately less need for information to recreate it precisely because the filters remove the detail anyway (and if not the filters, the human ear).
Not if they can't hear the value in higher resolution audio. For many people the only difference in HD audio over 16-bit 44.1 Khz is that the files are bigger. If someone can't hear the difference, it's no surprise that they don't care to move to a new format.
The screen analogy isn't perfect as most people can still readily tell the difference between an HD image and a significantly lower resolution one. (Though yeah, we're getting closer to pixel densities surpassing people's ability to resolve pixels as well, provided they're not putting their nose to the screen. It won't be too long now.)
Don't argue from a position of ignorance. Make friends with a recording engineer and have them play you a 32bit mix followed by a 16 bit mixdown.
44.1kHz/100 = 441hz. But that's nonsense in the same way that saying that a signal at the Nyqvist frequency can be accurately encoded. @diroussle pointed out that you lose phase but there's another consideration, sync. If your signal at Nyquist is not in sync with the sampling frequency, then it's going to be represented as a signal offset, an out of phase line.
In any case, thanks for your patience, I'm glad to have cause to reconsider my position on this topic.
Take it from me, when you master 24 bit stereo tracks, and you don't dither, huge amounts of low level detail disappear. The detail in the quiets is there in 24 bit, and lost when its truncated to 16 bits. Add the dithering, and you get increased noise, but the detail comes back.
One could suggest that with dithering 16 bits can represent it. But that's with a whole bunch of noise added to the signal. You can argue that noise is not audible, but it is _just_, and when mastering you can audition the different dither spectrums to find which dither least impacts the music.
At that point the issue may become moot as other problems like standing waves, harmonic distortion, inaccurate speaker frequency response and so on creep in and affect music playback to a subjectively larger degree than '16-bit versus 24-bit does', IMO.
All that said, 24-bit is definitely the way to go since we might as well do it right even if only x percent of listeners will notice.
As an aside, thank you for being one of the conscientious 'good guys' in the studio. I collect music and wish I had a nickel for every sloppy recording I've heard.
But for so long, for those of us who want a higher quality (hearing it exactly as they would have heard in the studio during the production) there was nothing we could do. Willing to pay more for it, doesn't matter. You just can't get it. It's still that way.
What gripes me is the attitude of many, including this xiph article, that hires versions "make no sense", that "there is no point" and thus everyone should just be happy with what they've got and that anyone protesting is an "audiophool" or believes in magic fairies or something. We all get lumped in with those people buying $3000 IEC power cables. For many people it's all black and white, there is no room for grey. You either think that 128kbps mp3s sound identical to the analogue master tape, or you are a fool spending $20,000 on magical stickers to increase the light speed in your CD player.
All I want is to be able to buy the mix and hear it as the engineer heard it in the studio. That would be nice. I know it's not for everyone, but it doesn't make me crazy.
As food for thought, have a read of what Rupert Neve said about Geoff Emerick's hearing ability (being able to discern a 3dB rise at 54kHz) here: http://poonshead.com/Reading/Articles.aspx
"The danger here is that the more qualified you are, the more you 'know' that something can't be true, so you don't believe it. Or you 'know' a design can't be done, so you don't try it."
There's definitely nothing crazy about wanting to hear a recording with as much fidelity to the master as possible. Yeah, I do remember people saying that 128 kbps MP3 was "CD quality" in the early days of the format, and that was a laughable claim indeed. One would have to be pretty tin-eared to think 128 kbps was hi-fi, although I'd say there were valid use cases for it, at least back when portable music players had storage in the megabyte range instead of the gigabytes we have today.
So many of those audiophile tweaks are just outright scams, and a fool and his money are soon parted. I guess education is the only way to combat that.
As for Emerick's ability to hear anything at 54 kHz, much less discern a 3 dB difference there, well, I am really, really skeptical. I'm obviously not in a position to say it's impossible, but it strikes me as an outright superhuman ability that should have been tested scientifically.
I can only speak from my own experiences, and I record and mix in 24/96 but for reasons that don't really relate to music distribution. When doing further processing, some plugins sound better with their algorithms taking 96k instead of 44k. Every plugin has been written with compromises. And I find I can push hires audio further in the digital domain before unpleasant artefacts arise.
It's very much like image processing. If you take a picture with a cheap basic 1 mega pixel camera and then play with the curves and sharpness, at a certain point smooth graduated colour becomes "posterised". If you take the shot with a DSLR (with 12 bits of each primary colour) then you can push the image a lot further before the posterisation occurs.
I have found the same occurs for audio. I can manipulate the sound with less artefacts when its hires. The plugins sound more transparent and smoother. I tend not to go above 96kHz because this effect is achieved at 96, and 192 (to my ears) sounds no better and I'd just have bigger files and more CPU load from the plugins processing the extra data.
The bandwidth of 96kHz is just short of 50kHz, so if as an added benefit I satisfy the one in a million Geoff Emerick's, then all the better.
But then once the final mix is rendered and no more processing needs to be done, ie for distribution, then this hires advantage seems moot. Maybe there is still some advantage for people or devices that may post process the sound digitally in some way, like a digital equaliser in your playback device, or something like that. But then again, that device could always upsample before processing.
I tend to use 88kHz if the final destination is intended to be CD, and 96kHz otherwise (so there is less aliasing when sample rate converting to 44kHz).
The reason I harp on about the bit depth is because in my experience that is where we are falling short. If I take my hires sources and convert to 44 or 48 with a high quality SRC I hear no difference at all. But when I change to 16 bit the difference is enormous. There is always a degredation. And it's never a good thing. It seems silly to just be throwing away that bit depth because of a 1982 format that people aren't even listening on anymore.
Also on the topic of SRCs, this site has some interesting comparison. For the record I do my SRC conversion with iZotope RX 64 bit SRC. http://src.infinitewave.ca/
So in conclusion, I want 24 bit tracks. If they're given to me as 44, 96, 192... whatever. As long as they're 24 bit. Enough with the 16 bit! :D
One reviewer  notes: "The effect of the first few Shakti products was not as apparent as when the effect became compounded. Each built on the others' ability to eliminate EMI in the component on or under which it was placed. Music became more relaxed, with greater clarity. Space and ambience increased. The soundfield became considerably more open and defined. At a certain point, the effect became quite startling as another Stone or On-Line was added. Shazaam!"
PS: Sorry for bringing these up. They're quite the recurring joke in audiophile discussion.
I used ferrite beads to remove EMI in a pair of self-powered speakers. I was hearing the local college station broadcast at a very low level when nothing was playing. I added them to the speaker wires, along with a basic EMI power strip, and the interference was gone.
It could still be parody though, even though they actually sell this stuff... :P
For some reason people seem to latch on to the format thing, before being able to make judgements about the more important factors.
"I do think in the domestic environment, the people that have sufficient equipment don’t pay enough attention to room acoustics. The pro audio guy will prioritize room acoustics and do the necessary treatments to make the room sound right. The hi-fi world attaches less importance to room acoustics, and prioritizes equipment; they are looking more at brand names and reputation."
An actual live musician can still sound poor in an acoustically-terrible room.
Sometimes I think I'd like to buy some expensive measurement microphones and record at 192Khz for Science, eg to find out if there are tunes in cricket stridulations or whatnot. But then I get over it.
However, all I'll say is, it's very different to hear or feel a difference, than to prove it 100% without any doubt in the exacting conditions of an ABX test. You behave differently and aim your listening at different things in the special case of critical testing, than when normally listening.
You have to know and respect this to make good arguments against hardcore audiophiles. Only once you give credence to the possibility can you bring on the real science: that the true bandwidth of the ear and the nyquist theorem truly does mean that any signal within our range of hearing can be encoded perfectly in double the sampling frequency and some 65 thousand steps—assuming an ideal decoding, of course, which means, yes, you should respect the idea of DAC design.
The world is full of idiots who are easily parted with their money. But don't throw the baby out with the bathwater. Pursue good quality audio equipment, to a point, because damn, it is enjoyable.
so what you are saying is you know what sounds better if you read the label BEFORE listening to it? :)
There is almost always a difference, no one is claiming otherwise. Pointing which one is closer to the original (not "better", because "better" might mean louder/overdriven bass) is the real test, and EVERY SINGLE audiophool to date fails at this point - Randi foundation did run a $1M pot for someone to spot a difference between 'audiophile grade' power/speaker cables versus coat hangar at one point.
It's really not fair to compare FLAC vs MP3 to "hi-rez" FLAC vs regular FLAC
There's legitimately some instruments that do not compress well. The harpsichord is a particular example that you should be able to hear the difference on on any sort of decent equipment.
Bur hi-rez vs regular flac is something that I don't think can be really detected by humans. I've gone through and done the Philips golden ears challenge to completion, and have very high end equipment, blind ABX FLAC vs MP3 on a lot of songs I am familiar with, but have never once been able to successfully ABX between a 24/192 flac and a regular one.
You can just not use MP3 though. It's 2014! Use AAC!
Also, everyone is satisfied with AAC already, so there's no good reason to throw out your music collection or your HW accelerated decoding platform.
Are you sure? I thought that was just a problem particular to early encoders for the Vorbis codec, which were alleviated by altering the tuning parameters of the encoder.
I have not done any personal blind ABX tests on AAC or modern ogg vorbis, so I can't really speak to them.
I'm going to keep 'archival' quality stuff in FLAC anyway, just so I'm covered for any advances in compression tech or whatever, and I stream to my mobile stuff, so size concerns aren't a huge deal for me. My ABX testing has just been for the sake of the mp3 vs FLAC argument.
So, AAC and Vorbis might have very well solved the problem of compressing some of these instruments.
If it is better, then the person who thinks it is better benefits from it.
If it is no better, then the person who thinks it is no better doesn't benefit from it.
If it is better, then the person who thinks it is no better doesn't benefit from it.
If the objective is subjective benefit, then placebo is a benefit; assuming your bank account is large enough and you don't care to give your money to someone who really needs it.
Edit: An answer to this is the Carl Sagan quote at the end of the article:
"For me, it is far better to grasp the Universe as it really is than to persist in delusion, however satisfying and reassuring."
Of course, it isn't really possible to not 'persist in delusion'. One can try, but he won't know if by trying he is perpetuating a grander delusion.
He explores (and technically explains) how higher sampling rates can actually be much worse due to equipment.
If it is worse, then the person who thinks it is better benefits from it.
If it is worse, then the person who thinks it is worse loses value with it.
Only if everything else is equal. It's rare that's no downside to the benefit - for example, something costing more because it's "better".
I've personally done blind A/B testing in my (then) studio to discover the point at which I can't distinguish between MP3 and uncompressed audio. These days the encoders are really good, so it gets real hard at around 256kbps. I'm confident I could reliably pick out 192kbps though.
Yet, I only buy lossless music since I plan to keep my music library around for ages and this allows me to change to a different format in the future if needed. This is an aspect of lossless audio, which is often overlooked.
I still rip CDs to FLAC but only to transcode them to lossy formats for later listening. I do this in case I decide to switch lossy formats in the future (note: due to differences in psychoacoustical models, you should never transcode from one lossy format to another).
I don't know if I would tell the difference in your test, but where I have noticed it the cause might be bad MP3 encoding - MP3 encoding quality varies widely... The difference between good and bad encoding may be far greater than between good MP3 and FLAC.
It's the most overlooked part of the listening chain, and is in fact the most important part. In fact it always shocks me how many "audiophiles" will pump tens of thousands into audio equipment for their reflective, untreated, boxy listening space. A $1000 pair of speakers in a room with $5000 in room treatment will totally blow away $20,000 pair of speaker in a room with $0 in room treatment. Every time.
I humored him by teasing out which one was which without him noticing, and then saying the lossless one sounded better. Didn't want to hurt his feelings. And really, the sound system as a whole sounded awesome. I just couldn't tell the difference between the formats.
There is a reason the mp3 scene moved away from 192kbps, and it doesn't have anything to do with bandwidth availability. It's because 192kbps sounds terrible.
I can't on almost any modern music (which is the majority of what I listen to), but when I was going through the philips golden ears course, I did a fair amount of blind ABX on harpsichords, cymbals, and a few other instruments at v0/320kbps and didn't have much trouble identifying them.
Granted, at that point I had been going through something specifically intended to help train you for discerning differences in audio, but they were distinct enough I don't think I would have had any trouble beforehand, either.
On some stuff I couldn't immediately tell that some sounded better - just different. Though on some of the samples the FLAC was easily better to my ears.
(My criteria for a 'successful' ABX was accuracy of at least 8 out of 10 using the foobar ABX comparator plugin)
If we're to take Tidal and Spotify(at highest quality) as representative of those two (please correct me if I'm wrong, no expert) then the difference is night and day. Perhaps Spotify could use a higher quality mp3 encoding?
In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz) are recorded with lower amplitudes because our ears are more sensitive to them.
Sampling theory is correct and 16-bits can reproduce any waveform with ~100dB of range, however, in a complex waveform consisting of low, mid and high frequencies, the mid- and hi-range frequencies quite simply get shortchanged.
Imagine a recording of a bass sinusoid and a mid-range sinusoid of equal volume. It might use e.g. 10 bits to store the bass and only 6 to store the high frequencies. (2^10sin(200wt)+2^6sin(4000wt)). That means the resolution of the high frequencies is less than the lower frequencies. When the volume of those frequencies changes dynamically, the high frequencies' amplitudes are more quantized. That is quite simply why 16-bits are not enough.
This is similar to the problem with storing waveforms unprecompensated on vinyl. The precompensation makes up for the non-uniformity of the medium. It could be done with 16-bit digital as well. Or alternatively, larger sample sizes like 24 can be used.
I haven't A/B tested this. The A/B test in the article compares CD with SACD. SACD isn't PCM, so its artifacts are going to be totally different from 24-bit PCM.
So, the noise of that signal is -5dB SPL. 0dB SPL was set to be the lowest possible perceivable level of a single sound in an aechoic chamber. And that's not even considering other sounds in the recording, or ambient noise levels in a typical living room, etc.
In your example, moving to 24 bit would been a long way from having any effect (other than a 50% increase in file size). And if you use, say, an 8 bit signal as an example, then things are even less noisy. Note that the noise is the only consideration here: any fidelity loss is represented in that figure.
The audio engineers of yore who (among other things) decided that 16 bits was more than enough for final mixdown were much more competent than they get credit for (many were downright amazing at what they did, in fact). They thought of stuff like this.
If the results are no better than chance, then 24-bit doesn't matter, regardless of how sound the underlying argument is.
EDIT: The experiment would also be extremely difficult to design. For example, you'd need to run this test with music, not simple sounds. So the question is, which music? I think whatever is most popular at the time would be a good candidate, because if people are listening to music they hate, they won't care about the fine details of the audio. But that introduces an element of uncertainty and noise into the results which is hard to control for.
Some people might deliver accurate results with https://www.youtube.com/watch?v=2zNSgSzhBfM but not with https://www.youtube.com/watch?v=4Tr0otuiQuU whereas for others it's the opposite.
Or, it could be the exact opposite: Maybe you can only detect whether a sound is 24-bit when it's a simple tone, and not music.
Age is also a factor. My hearing is worse than a decade ago.
The headphones used by the test are another factor. If you feed 24-bit input to headphones, there's no guarantee that the speakers are performing with 24-bit resolution. In fact, this may be the source of most of the confusion in the debate. I'm not sure how you'd even check whether speakers are physically moving back and forth "at 24-bit resolution" rather than a 16-bit resolution.
A quick summary would be that most "popular" music has been mastered with the following goal: the song should be recognizable and listenable on a FM-radio with only a limited bandwidth midrange-speaker. One of the many things they do to achieve this is by eliminated almost all dynamic range through a process called "compression" (dynamic compression, not digital-compression).
They also limit the spectral range to not have "unheard" sounds cause distortion when played through limited bandwidth amplifiers and speakers.
This means that the kinds of musical pieces which could benefit from the increased dynamic range of 24-bit would be thoroughly excluded from the test.
And then you'd probably get the "expected" result, but only because you now test whether music mastered specifically not to have dynamic range benefits from having increased dynamic range. For which the answer is given.
Note: I'm not claiming 24-bit end-user audio has merits, of which I have little opinion. I'm just pointing out the flaw in the proposed experiment.
If you feed 24-bit input to headphones, there's no guarantee that the speakers are performing with 24-bit resolution.
Not sure if you're just imprecise in your language here or if you're genuinely confusing things. Speaker-elements, as found in both speakers and headphones are analogue. They operate according to the law of physics, and respond to changes in magnetic fields, for which there is practically no lower limit.
They have no digital resolution. A quick example: Take your 16-bit music, halve the volume and voila! You are now operating at "17-bit resolution". Halve it again. 18-bit resolution. Etc.
There's probably some minimum levels of accuracy, yes, but it just doesn't make sense to measure it in bits.
If you're aware of this and were just trying to adjust the language to the problem at hand, I'm sorry for being patronizing, but I just wanted to make sure we keep things factual here.
I disagree. I think all the factors you are concerned about can be eliminated with a large enough sample size, like in the thousands (or maybe 10s of thousands).
You allow each person to select the genre of music they like, and you play a few clips from a few songs of each bitrate. Then they guess which is 24-bit and which is 16-bit.
I'm not paying to set it up. But it could all be done online without too much grief. It would be good to track the other statistics (age, headphone brand, etc.) as well, and see if something falls out of that.
More realistically, the participant might choose music for which no 24-bit recording exists.
It's very important to control for every variable. It's actually not possible to gather info about what headphones the listener is wearing. Even if it was, it wouldn't be possible to know whether they're doing the experiment in a quiet room, or whether there's a traffic jam just outside their apartment window, or whether their dog is barking during the test. Stuff like that.
Crowdsourcing this is an incredibly cool idea, but it'd just be so easy to believe you've performed a reliable test even though some variable undermined it.
I forgot another variable: Whether the music was recorded at 16-bit resolution. Most musicians use 24-bit, but it's easy to imagine that some of their samples might've been quantized to 16-bit without them realizing it.
It's not, actually. Say you have 10,000 listeners and you randomly assign each one to 16-bit vs 24-bit listening. You have enough listeners that any differences between the groups are due to chance and will very close to even out. Now, if you find people are unable to distinguish between 16-bit and 24-bit you might want to try the test again with more control over the environment, but if you find a substantial difference in a large blind randomized test that's a real finding.
Well, obviously we'd need to have a limited set of music selections for which we have 24-bit recordings.
As you suggest, I expect the biggest impact on playback fidelity is going to be other factors like the noise in the system (likely a PC) and such.
But the flip side of it is that's also a good real world test. If the only time you can tell a difference is to be in an acoustically dead room with top end equipment, then the higher sample rate really isn't worth it.
Hey, that's a great point! Hadn't considered that.
Proving "most people can't tell the difference between 24-bit and 16-bit in real-world settings" is less compelling than proving "no one can ever tell the difference," but it's still very relevant.
So you'd have to decide in advance what difference is meaningful and choose your sample size to ensure you can detect it.
If I pay for music, I want to truly own it, including the possibility of someday making a mashup, a music video, a hip-hop beat, et cetera. A 24-bit source gives casual creatives the same quality material as the original masters, for a relatively paltry 1.5x increase in file size.
Chris Randall of Sister Machine Gun (at least used to?) use a low-frequency generator at live shows to produce a sound that the audience could feel but not hear in order to make the music more intense. I suspect that you'd gain some of that effect with a larger bit size.
...or I could be completely wrong. Whatever.
What you mention, though, points directly to what /will/ improve the quality of sound reproduction: speakers. It gets harder and harder to move that much air with precision as you get lower and lower in frequency. It's a definite technical limitation, but it's to do with very high-power amps and giant speakers, not the recording format.
We have (to the degree that humans can prove that they can perceive), perfect reproduction from digital recordings, perfect amplifiers for reasonable prices (at lower-than-concert-power-levels at least), but we haven't yet developed good enough speakers to cover the whole perceptible range of frequencies to anywhere near the same degree.
Audiophiles love to try and improve the whole chain, but really the only place it matters is at the very end.
What you feel is parts of your body resonating (because the low frequency sound is exciting modes of your body). This is unlikely to happen at high frequencies, partially because it would necessarily be much smaller parts of your body (see  for a diagram of typical body resonance frequencies) which we probably don't feel and because attenuation of sound greatly increases at higher frequencies (for example, see  for air), making it likely impractical to excite any such modes. My guess is that you might cause some tissue damage if you had significant ultrasonic excitation in your body (see  for something that may or may not be true...).
In fact, you should be very careful about to think about quantization in digital sound reproduction, because it can easily lead you astray. Think of bit depth as a measure of dynamic range. Do look at that digital music primer for geeks posted elsewhere in this thread, it makes for awesome reading.
24 bits is great for recording, but the end distribution medium only needs to be 16 bits, after you normalize the thing you tracked at 24 bits.
You can get libsndfile and FFTW and do the tests yourself.
While it would be possible to represent each of those individual sinusoids in 6 and 10 bits individually, the signal you describe has the high frequency signal "riding on " the low frequency signal. you need 16 bits to represent the amplitude of that signal at e.g. t = pi/4.
Sound attenuates proportional to f^2 in air. So, a 10-fold increase in frequency causes a 100-fold increase in attenuation. So, a 100KHz signal has 10,000 times the attenuation for a 1KHz signal over the same range.
In addition, attenuation due to water vapor is particularly bad above about 40KHz.
Much of that study cannot be replicated, and people have tried:
In addition, the effect went away when using headphones.
However, I can certainly believe that if you can pump enough energy to rattle things at ultrasonic frequencies you are going to get a result. Especially since ultrasonic frequencies rattle in water particularly effectively.
As an example, if I pump enough energy into an ultraviolet or infrared signal at your eye I will eventually get a detection result in your brain. However, pain and a burned retina are not what we think about when we consider a brain response.
I find the intermodulation argument a convincing one - it's hard to not have it affect the test, and if they haven't taken specific steps to avoid it then it would be easy to join a large amount of tests that have fallen prey to it.
I note, however, that Wikipedia doesn't mention any specifically brain-scan-based studies that counter it. If you know of any more I'd be very interested in hearing about it.
On one hand, higher frequencies above 20KHz can't be heard at all, so there's no point having them! You can't hear them!
Then on the other hand, higher frequencies above 20kHz affect the audible region of the sound (intermodulation distortion), so you can hear them, so make sure they aren't there!
What if the presence of the higher frequencies in a spectrum that shares a harmonic relation to the audible region causes intermodulation distortion that is pleasing and musical to the ear. What if the complete absence of this high frequency information, or alternatively a non-harmonic higher frequency signal (say some kind of switching or power supply noise) causes the audible region to be perceived in a less pleasant manner?
Certainly some people find certain kinds of distortion pleasant, but the people arguing for 192kHz claim increased fidelity, not pleasing distortion - when it is just the opposite for any stereo that introduces these artefacts.
High frequency limiting is not the only artifact that results from data compression.
I can reliably hear a pitch difference of ~0.2hz at this site: http://tonometric.com/adaptivepitch/
and that is after 15 years in a symphony orchestra having my ears blasted by the brass and percussion section (with a demonstrated hearing impairment from my time in the orchestra).
For example, he just released an album that was recorded in what is basically a phone booth. http://www.clashmusic.com/news/neil-young-makes-entire-album...
This is the same reason I'm convinced we're going to get 8k phone displays someday.
If the recording industry wants to sell me a "platinum" version of recordings, what I'd really like to have is different mastering of an album: at least one for noisy environments like the car, and one for higher-quality environments like my home theater. If you're familiar with "The Loudness Wars", this is a reaction to that. NiN tried to do this with their "audiophile" mix of Hesitation Marks (although a lot of people think they did not succeed, http://www.metal-fi.com/terrible-lie/ )
On the other hand, I don't need to buy any new equipment to support that, so the equipment guys aren't going to be happy. I don't know if there's any silver bullet for them--if there is a hypothetical advancement that would cause me to upgrade my system, I can't envision it.
clearly the display would blink "let there be light"* at startup.