I once spent ~2 hours explaining all of this (well maybe not all of it) to a friend of mine who was studying sound design at the time. He didn't believe me.
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)
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.
Uh, no. Bit-perfect ripping is trivial and routine, and tools like the AccurateRip DB (which has checksums for around three million different titles you can use to verify the checksums on your own rips) and the CUEtools database (which has recovery records you can use to correct bit errors on your own rips) prove it. I routinely get bit-accurate single-pass high-speed rips--no "paranoid" settings or re-reads--of discs dating back thirty years or more, and so do hundreds of thousands of other people. If you get different checksums on successive rips of the same CD, either the disc is damaged or the drive you're using is failing.
Oh sure, your rips may be perfect at the bit level, but how do you know that they're free of sub-bit quantization that isn't detectable by electronic circuits but can be heard by the human ear?
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?
Ah, well, the human ear is a much more finely tuned instrument than your decoders and players. Think of the feelings you get when you hear the ocean waves, the birds sing, a thunderclap!
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.
Interesting. I'll have to check those projects out. I have the same problem as the GP -- I have a script that rips CDs, taking multiple reads until it gets two bit-for-bit identical copies. And just about every time at least one track is silently "corrupted."
(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.)
CD-audio "Red Book" data does have error correction (Cross-Interleaved Reed-Solomon Code). Whether you get a bit-perfect audio rip depends on how much error correction and retrying you do.
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.
> 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.
Err, no.
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 .
There was a day when the clock going in to the D/A converter could be affected by the bitstream coming off the CD. Those days are long gone I'm sure. Everything is buffered in RAM, overclocked, and digitally processed before it hits the D/A.
"Oh, and virtually no PC on earth has that kind of I/O throughput; a Sun Enterprise server might, but a PC does not. Most don't come within a factor of five, assuming perfect realtime behavior."
It's generally the lack of synchronization and positioning information compared to data CDs that gets you. In particular, on many older drives you can't reliably start the rip at the same place each time, so even if all the corrections and fixups work perfectly and you get a bit-exact rip (which isn't hard) you still won't get the same file twice.
> I read into this when I was wondering why my CD-DA extracted .wavs came out with a different checksum every time.
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"....
FYI the difference is almost certainly due to the seeks not being sample-accurate, so your rips are bit identical for each sample, but you are starting in very slightly different places. Either that or you have a really broken CD-ROM drive (which is also possible).
>Self-styled audiophiles are, by and large, idiots with way too much money plagued by magical thinking.
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".
If you're going to be "picky", please be picky about something you understand. A "small amount of/incorrect knowledge" is ignorance, not a lack of intelligence. When you label someone an idiot you're not talking about his lack of knowledge. You're talking about his intelligence.
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".
What you are referring as "basic intelligence" is actually a combination of neuroplasticity and general knowledge. Neuroplasticity is a speed at which brain can learn new things, but even then you can't say "person1 neuroplasticity > person2 neuroplasticity". That is because brain is composed of many parts that can have different plasticity. Also neuroplasticity (i.e. "intelligence") is not static and can even change over time depending on what parts of the brain are most active.
James Randi has a speech (you can probably find it on Youtube) about how it is easier to fool smart people than average people. Smart people think they can't be fooled.
The most generous approach to audiophiles is to allow for a placebo effect. i.e. they get greater enjoyment from listening to what they believe is a perfect sound system, regardless of whether the gold-plated cables actually do anything.
> not all these people are idiots. Some of them are quite intelligent
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.
I would refer to that as knowledgeable and ignorant rather than intelligent and idiotic. It doesn't really make sense to say that Gary Kasparov is an idiot regarding the construction of log cabins. Rather, he is an intelligent person who is ignorant of 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)
Bose and beats* are by, every /objective/ measure, shitty products.
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.
Are Beats shitty or just expensive? How little would I have to pay to get same quality?
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.
So, we have to define, in your opinion, what would make a pair of headphones 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.
>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.
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"
> 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.
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?
So, headphone use in studios is not generally for creating the final mix. Monitor speakers are used nearly exclusively in professional studios as what you are mixing for. Headphones have multiple places in the production process where they are used, but they're not the final target.
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.
Interesting. I guess the main concept I'm exploring is the idea that if you don't control for the sound quality that the person mixing it (or more importantly, the person approving the mix) then it's hard to make any claims about how the sound was "meant to be heard".
Given whenever I stand near someone on a train with them I can hear a fair amount of their music (not anywhere near as bad as Apple earbuds though), I assume they can't be that great - that or the listener has very bad hearing.
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 :-)
>Given whenever I stand near someone on a train with them I can hear a fair amount of their music (not anywhere near as bad as Apple earbuds though), I assume they can't be that great - that or the listener has very bad hearing.
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.
Leakage is sometimes intended so it's not necessarily an indicator of quality. See the HD800's. You'll hear them in any open-plan office, for sure. There's no attempt to keep the music from leaking, their only priority is sound quality (which is, at this price, a matter of taste and preference).
Yeah, Dr. Dre is on record saying that he's not an audio engineer but he knows what makes hip-hop music sound good. So he never claimed they had "flat response" or anything.
In response to your and your parent’s blanket claims
> 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.
As far I've tried NC headphones, nothing comes even close to what Bose offer with QC25, any other brand I've tried cancel out less noise than the bose. Sound quality might not be the best, but the intented environment is the limiting factor anyway, and they do a great job at dealing with environment noise.
The biggest problem "audiophiles" don't seem to get is that accurate reproduction is not the end goal of music. Enjoyment is. Audiophiles have convinced themselves to find enjoyment from accurate reproduction, and that's OK. But the majority of the world does not see it that way.
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.
Whoosh ... I think my comment went over some people's heads. A "one note bass" isn't a good thing, technically. I did not say that Bose is great at making speakers with excellent bass.
Those are low-budget audiophile products, ie they are still marketed mainly to boost the ego of purchasers.
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.
There are a few more brands and models that pros use. The AKG K240 has been in use for decades, and I think they're quite charming. Audio Technica has made a lot of in-roads into mid-level pro studios in particular (i.e. not million dollar rooms, but still quite good studios that make good records regularly). I've also seen Shure SRH series headphones in professional contexts.
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 have to try the Beyer some time. Agreed on the Sennheiser ergonmics - and the price.
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.
That's nothing laugh-worthy. Audio equipment faces massive diminishing returns. If you're looking for midrange sound you might as well pay attention to the people recommending sub-$50 headphones as to the ones talking of $150. For instance, the $30 Panasonic HTF600s is better-sounding than the $160 ATH-M50x frequently recommended as an entry-level audiophile headset. As for studio-quality headphones, chinese Superlux/Takstar models are as good as the $200-300 range.
As a die hard fan of the M50x (and no qualifications whatsoever for judging headphones) I'll have to explore those Panasonics you mention. I haven't considered that company as a quality maker of audio gear since the portable CD player era. Did you do the test yourself, or are you going off of a review site? I typically rely on head-fi, but I'm always looking for a recommendation in this field.
From all the reviews I've seen, Bose noise-cancelling headphones are pretty much the best you can buy. Especially if you want earbuds (the QC20s). They're extremely expensive though. Do you (or anyone) have suggestions for alternatives?
For the price of Bose noise-cancelling headphones you can get headphones from the three brands the parent mentioned (throw in AKG for good measure) that sound better in 'lab conditions'. But if your discerning feature is 'noise-cancelling', i.e. headphones that sound excellent in noisy environments like trains, coffeeshops or open work environments I believe Bose is the king and will be as long as their patents are enforced.
I believe they're good as far as they go because Bose has the strongest patent portfolio in this area, but wearing any kind of noise canceling headphones immediately gives me the unpleasant sensation of having my eardrums sucked outwards. I'm not sure why; I think it's a side effect of the tiny latency inherent in the design. It's so unpleasant to me that I stopped paying attention to new products in that category so I'm a bad person to ask.
tomc1985: hope you see this - your account has been hellbanned for over two years (about 850 days, with one comment visible 270 days ago - not sure how that happened).
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.
Bose QC15s and QC20s are the best active noise-cancelling headphones out there... but the problem is, hey still have very mediocre sound, but they do the noise-cancelling part well. They are also massively overpriced.
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.
I use Klipsch x7i and they allow you to listen to audiobooks on a low volume setting in an underground train. Which, in Moscow, seems like a perfection.
The sound is very clear, but not balanced. Neither an expert nor a musician though.
There's a fair amount of pros that utilize Audeze and HiFiMan gear as well. Planar magnetics are popular.
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.
Almost every studio used to have nothing but AKG K240 phones for monitors in the room. I haven't worked professionally directly in the field for years (I work in live sound occasionally now, and do interact with recording engineers occasionally), but I still see them discussed regularly enough online to assume they are still common. I love the look of them, and always have. To me, they are the definition of "studio headphone". (They aren't what I use in my home studio, as there are better phones, if you're willing to spend more money, but they are a good headphone for a good price.)
These Yamaha headphones are excellent, http://www.musiciansfriend.com/pro-audio/yamaha-rh3c-profess.... Durable, they collapse and sound great. I have another set of open phones with foam surrounds, the foam is not stable and collects gunk if used in a backpack. The Yamaha phones very respectable replacements for the MDR unit and stay clean while taking up little space.
Slightly larger driver, slightly heavier, supposedly has a greatly-extended frequency range of 5Hz - 80 KHz vs 10hz - 20Khz in the 7506. Of course your typical D/A converter won't even render such low frequencies due to DC coupling, and if they were there you'd want to EQ them away pronto as they would eat all your headroom. While I continue to enjoy excellent high-frequency hearing even in my mid 40s (to my surprise), neither I nor anyone else needs a tweeter that goes to 80 KHz.
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.
thanks for the info; I'm considering upgrading my office headphones (I've got some random $20 over-the-ear pair right now). $80 is certainly reasonable, and I prefer transparent speakers in general.
>God forbid you ever consider buying a Bose or Beats product.
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.
> God forbid you ever consider buying a Bose or Beats product.
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.
I have a pair of Sennheister 280 HD Pros - they've lasted me about 7 years, an excellent set of headphones. I used it to help critique music for lots of artists on their production, and I know lots of artists who use it as a cheap pair of mixing headphones.
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.
A friend has had a set of these for ages. We found a difference between two source setups. A particular Sony DVD player sounded incredible - each note seemed perceptible in a 3D space [1]. A CD player he had, didn't. [2] We tried with Yamaha amp, without, different configs of widgets. That DVD player with nothing added was the best. He gave it to his sister and I haven't heard anything like that since.
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.
[1] I've since learned it's called soundstage
[2] How would the source influence soundstage? Sounds irrational to me. Hey, one sounded better than the other and I don't know why.
And there are jobs created for engineers who design that equipment - simultaneously recycling more money in the economy and contributing to the statistics of STEM job prospects! Everybody wins!
Sure, you need to take externalities into account. One can imagine an audiophile who would rather spend $100K on a sound system that sustains 3 jobs at a niche sound-system-design business, instead of angel-investing that money in a growth company that would eventually create 300 jobs. But what if the audiophile's daughter is then inspired to build another growth company when she grows up, because she was intrigued by how much her dad would geek out about the electronics in her sound system? Nothing is clear, it's incredibly hard to quantify probabilities about any of these things. As long as the audiophile isn't neglecting responsibilities or breaking windows to obtain his sound system... that is, as long as negative externalities are not a foregone conclusion... we should let him enjoy his passion.
Certainly, there are valid moral arguments to allow, even encourage, this state of affairs. I was just pointing out that the economical argument that was offered is a broken window fallacy. Refuting an argument in favor of X is not an argument against X.
Wouldn't most of the digital cable costs be due to analog interference in the machines they are connecting and them acting like antennas and not that the signal is corrupted on the way?
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.
On the other side of their spectrum, I went "near" an audiophile shop test booth and I was sucked by clarity and density of the sound in the air (this was a drum solo track). Some audiophileness is good.
I actually think you'll find many "audiophiles" enamored of the Sansa Clip Plus and Zip, which fall into that price range. Probably the most highly thought of MP3 players after the iPod classic 5.5.
Now that Android supports USB DACs, you can just get portable DAC+Amp combos that you can stick in your pocket with your phone. No need for a dedicated device anymore.
Far from me the idea to defend them, but $350/m is about 3 times higher than Monster's most bullshitty bullshit cable. Even their "2000HD HyperSpeed HDMI cable" only had a $115/m MSRP in 5ft, falling to $32/m in 35ft: http://www.monsterproducts.com/Monster_Video_ISF_2000HD_Hype...
My great business idea was to produce a line of "organic" cables.
Our company would go to the remote places of the earth to hunt down copper dragons (as in DnD) and harvest their veins to make audio cables.
The "natural", "organic" copper has a warmth to the audio signals flowing through it that artificially produces cables just can't provide (they have harsher undertones).
Then we'd also have silver and gold cables, harvested from, you guessed it, silver and gold dragons.
I plan to disrupt the audiophile business (and crush you in passing) with my homeopathic cables.
As we melt the gold we mix in a few atoms of "rare earth" elements (rare == expensive == so good "they" don't want you to have access) which is then diluted by adding more melted gold until only the imprint of the rare earth atom remains.
The gold will the be hand drawn by virgins (in truth these will be strong, hairy, 50-yo virgins with dreadful hygiene and B.O. though strong enough to pull, but we need not add all that confusing stuff, we'll just say "virgins"). The wire will be lovingly laid into hand-made insulation made from organic pinniped leather.
I see a variety of future applications both in the home (connect your cable modem to your WiFi access point) and business (data centers). To quote Rony Abovitz, we'll soon be "the size of Apple".
dragoncopper, have the website default to some non-existent northern european language/font with a translation button (british flag). Burled wood with reds and yellows. will buy.
When someone spends money on something in a silly way, do you consider them to be an idiot? Would you make the same statement about someone who spends 200$ on a bottle of scotch versus buying a 35$ bottle?
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.
You are mixing two things.
There is a difference between Intel charging you >$1K for a CPU that is 5% faster (in games) than overclocked $300 one versus scammer selling magical power cables (made by gypsy virgins in Romanian monastery on top of the highest mountain, in full moon) to some rich retards.
$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.
HP was known for building generally good test equipment, including arbitrary waveform generators. Op is humorously referring to the fact that the test and measurement division was spun off first as Agilent, then Keysight (seriously?), and probably something else by tomorrow (marketing is furiously brainstorming new meaningless names. They only need to merge with Danaheer and rename themselves Flukeronix for the circle of life to be complete.
It's one thing when you plop down a ton of cash for something where you embrace the "silliness" or whatever makes it special (e.g. buying exotic cars with monstruous engines to drive them into traffic). It's another, however, believing something is objectively "better" because it cost more; e.g. spending money on Monster Cables thinking you can hear a difference.
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.
I completely get that as I personally prefer to collect Old Stock vacuum tubes for my listening purposes (http://imgur.com/a/INXVX). I just think folks should be left to their own devices to enjoy an avocation as they please.
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.
>I just think folks should be left to their own devices to enjoy an avocation as they please.
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?
If you enjoy music (who doesnt) and are bending toward learning electronics I highly recommend projects like Twisted Pear kits and build yourself a DAC.
http://www.twistedpearaudio.com/landing.aspx
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.
http://www.woutervanwijk.nl/pimusicbox/
I would make the argument that if some of the silliness was stated for what it was, gullible rich people would spend their money on something equally frivolous that did more to drive innovation
That's rather a wide-ranging bit of character assisination. Some of us just enjoy listing to music on decent equipment (If you can buy it at your local big box store, it's probably not sufficiently "decent") properly setup (which doesn't mean expensively - just basic proper speaker placement and the like).
> Some of us just enjoy list[en]ing to music on decent equipment
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.
>The best definition of audiophile I've heard is somebody who listens to equipment, rather than music.
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.
You've completely misunderstood the article. It's discussing differences in the the PCM sampling rate, not the MP3 encoding bitrate. Or in otherwords, all these samples they compare are FLAC.
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.
Bit depth affects dynamic range, and that's it. The only thing a 24-bit sample can do better than a 16-bit sample is accurately reproduce the difference between very loud sounds and very quiet ones. That's all. For the vast, vast majority of music listeners, the difference will be insignificant, as they don't listen in an environment where a dynamic range of 144 dB can be used to anywhere near its full effect.
This is slightly incorrect. The combination of bit length and sampling rate determine both dynamic range and frequency fidelity. Although it's common to hear the two values used to represent these two separate physical measurements, it's just another case of explaining new tech (80s CDs) to old technical consumers (70s hi-fi types). You can measure a reduction in dynamic range by reducing either sampling rate or bit length.
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 your last example about the harmonics, I can see that being so in the most simple case, but surely that's a limitation of the playback system, not the storage medium. And isn't that addressed by the oversampling that is used almost universally in DACs now?
It's a limitation of the recording/storage medium and is addressed by oversampling and low pass filters in ADCs in acoustical recording. Most audio recording, processing and mixing is performed in 24 or 32 bit. Once the data is down sampled for distribution in 16bit 44.1Khz you run into the limitation again where you have fewer samples to represent higher frequencies. The only remedy is to attenuate those frequencies before downsampling. I'm unsure of the role oversampling plays in DACs so I can't speak to that.
My understanding is that oversampling is done in DACs so that digital filters can be applied without introducing the effects that you describe at the top end of the frequency range. Basically it interpolates to a higher sample rate so that there's more play in filter selection/application.
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.
I figured it had something to do with the application of digital filters. There are only two significant samples of a source at half the sampling rate, so it should be able to represent a frequency at exactly half the sampling rate but I can't see how it could accurately represent a frequency at just a few hz below 1/2 the sampling rate. Notice I'm using the word frequency. A sine wave with a frequency of 22,050hz encoded at 16 bit 44Khz is not going to look anything like a sine wave.
Right, 22,050hz looks exactly like a triangle wave on a computer screen. The thing is that a triangle wave is composed of a fundamental sine wave (22kHz), and a series of ascending odd harmonics above it. So after the filters nix everything above 22kHz it looks exactly like a sine wave on a scope.
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).
Sure 22,050hz and 11,025hz get smoothed out into perfect sine waves and the human ear can't hear 22Khz anyway. But the Nyquest frequency isn't some magical threshold that you cross and suddenly everything is perfectly preserved. It's a folding frequency that determines where aliasing is going to occur, or rather where it's not going to occur. A 44Khz sampling rate is based roughly on western tuning (440Hz A) and makes no attempt to accurately capture sounds and frequencies that are not tuned to western music. As you move away from the folding frequency, there are frequencies in the human audible range that can not be represented, so they're discarded or attenuated by anti-aliasing filters. As far as I'm concerned CD audio is outdated tech that most of the world just doesn't care enough to drop. It's ridiculous in a world of 5K retina screens that people can't see the value in higher resolution audio.
>It's ridiculous in a world of 5K retina screens that people can't see the value in higher resolution audio.
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.)
There's a difference between not being able to hear and not knowing what to listen for. Listen to the highs on a well tuned hi-fi system and you can hear the difference between CD and SACD. Listen for the sound of a singer taking a breath or the slow ring of a cymbal. If you've never heard these things in person to begin with then you're at a disadvantage trying to hear how badly they are represented in recording technology from the 1980s.
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.
Yes, and also remeber that as you approach the nyquest frequency, the ability to encode phase is lost. People always talk abou sampling capturing different frequencies but they forget to think about phase.
Which gets back to your original point about both sample rate and bit depth contributing to the dynamic range. I'm very curious: can you give me some search keywords that would get me to the math behind that? Also, I've never heard the claim that the 44.1kHz rate biases toward 440Hz tuning, is there somewhere I can read more about that?
pulse code modulation, pulse density modulation...
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.
I understand PCM, 1 bit DACs, et al. What I'd love to see are some equations relating the three quantities of bit rate, sampling frequency, and distortion. Turns out to be very hard to Google.
In any case, thanks for your patience, I'm glad to have cause to reconsider my position on this topic.
Yes. And 16 bit cannot by itself represent without distortion the full dynamic range of a lot of music. Most samples, most of the time, do not use a full 16 bits. This is why during CD mastering dithering is used.
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.
I certainly won't argue that 16-bit is just as good as 24-bit from an objective standpoint, as 24-bit is obviously superior, full stop. I'm just saying that for most listeners (everyone except those who listen at high levels in dedicated, treated listening rooms in very quiet environments) the difference will be inaudible almost all the time. Extremely low level detail doesn't really matter if it's lost in the >20 dB of natural noise in your room.
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.
Yes. I completely agree. From my perspective, even if one person in a hundred can hear a difference, then I'm going to pay attention. I don't want to boil everything down to meet the average. I think it's fine to release 16/44. I think well produced music sounds excellent in that format. It's one of the reasons the CD has done so well. It's just hifi enough to capture everything. And it's amazing to think this technology had its début in 1982!
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."
What's the argument in favor of using extremely high sampling rates though? Using 48 kHz instead of 44.1 seems reasonable (as in the Philips digital compact cassette that never really caught on), giving a little bit of headroom for wider frequency response, moving the filters a little higher or whatever, but I've seen D/A convertors that use 384 kHz, and I just can't fathom what the point is... It smacks of the "if some is good, more must be better" mentality.
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'm not sure there is a compelling reason to distribute final music pieces in 192kHz.
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
I think you're misreading what the OP said. He wasn't making the mistake of associating 192 kbps with 192 kHz. It was just a coincidence that he used 192 kbps MP3 transcodes. He could just as easily have used 224 kbps MP3s versus 24/192 FLACs in the test he described. The point was that his friend couldn't tell the difference between MP3s and FLACs.
One reviewer [2] 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!"
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 is baffling to me that people even talk about 24/192. There are such vast differences in audio quality related to speakers, loudness, amplifier, mastering, and EQ, before you even get to the source format.
For some reason people seem to latch on to the format thing, before being able to make judgements about the more important factors.
It doesn't end there. For most people, I strongly suspect moving around your furniture would affect your sound reproduction more than changing your equipment.
I have no source either, but I'd bet a very small sum of somebody else's dollars that "most people" in the US, at least, listen to most of their music in the car.
"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."
If it's any consolation, I'm a pro sound engineer and I entirely agree with you. I do like 24 bits (although I'd be satisfied with 20 bits) but I can't be bothered recording at anything higher than 96khz and even that is mainly wiggle room in case I want to do extreme pitch-shifting or suchlike. Most of the time I use 24 bits/48Khz.
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.
Preface: I consider myself an audiophile, but keep reading before you judge. I completely, wholeheartedly agree with the original article and the science behind it. There's no question that our hearing just isn't good enough to discern the minute differences between sound of sufficient quality (which is well defined in the Xiph article).
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.
>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.
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.
>He still thinks he can hear the difference between FLAC and MP3 to this day. He works as a sound engineer now.
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.
By '"hi-rez" FLAC vs regular FLAC"' the parent post means something like 24/192 vs 16/44.1, not the amount of compression. It has a higher resolution than the other example.
MP3 has specific algorithmic weaknesses above 16khz[0] which even the highest legal bitrate can't cover up… sometimes. It's actually easiest to hear it with cymbals, or the old LAME sample "fatboy".
You can just not use MP3 though. It's 2014! Use AAC!
The early encoder releases were tuned for voice chat, not music, so the best rate control mode is CBR and it hasn't seriously been tested otherwise. It is pretty amazing considering it's only so good by accident!
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.
> There's legitimately some instruments that do not compress well.
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'm basing it purely on modern (Within the past couple of years) encoded 320kbps/V0 MP3s.
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 no better, then the person who thinks it is better benefits from it.
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.
The DECT phone standard had to be marketed as "6.0" for the US market because non-technical people were trained to believe that 5.8GHz was better than 2.4GHz and a phone that ran at 1.9GHz would never sell in the US. This is the same thought process driving the audiophools' desire for bigger numbers regardless of reason. It doesn't help that the fundamentals of sampling theory aren't particularly intuitive.
The article is correct, but it's not true that nobody can't tell the difference between an MP3 and a FLAC.
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.
While I think that there are people who actually can tell the difference between lossless and lossy audio (with a decent bitrate), I'm not counting me among them.
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.
The article specifically mentions that lossless formats offer advantages over compressed formats. The argument is 24/192 encodes lossless files vs 16/48, which I feel the author soundly confirms.
In my personal testing a few years ago, I couldn't tell the difference between source and 192kbps MP3s either.
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).
> He still thinks he can hear the difference between FLAC and MP3 to this day
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.
Maybe it depends on the song as well? Classical music is supposed to have a much greater frequency spectrum - meaning, the effects of MP3 encoding becomes apparent when played on high fidelity equipment.
It has greater dynamic range (think volume, this is the domain of the bit rate properly called bit depth) than most compressed studio-produced music but the frequency range or spectrum (think pitch, which is limited on the high end by the sampling rate/2 via the Nyquist-Shannon theorem) is not different if not also clipped off by mixing.
mp3 sucks for bass and sub bass physical response. It's incredibly easy for a non audiophile (like my girlfriend back in the day) to tell the wooly, muffled bass of a quantized in my car vs the actual cd on the same system. Perceptual coding is not perfect.
I don't know about comparing digital to digital, but I know I can tell the difference between a 320kbs mp3 and a good quality LP record. Its in the treble. I guess any kind of digital just ruins that.
MP3 in particular has issues above 16KHz. This is solved in modern formats such as AAC and Ogg Vorbis. It has nothing to do with "any kind of digital".
I keep FLAC copies just so that I have lossless versions that I can convert into the lossy format du jour, but for actually listening, I mass-convert to lossy. I've listened to FLAC and properly encoded MP3 files (@192 and 160) on $50,000 audio equipment, and I can't tell the difference.
I'm curious, did you listen in a properly treated room? As in with bass traps, panel traps, diffusers, a cloud, first reflection points covered, etc? Because I can hear a difference on less that $10,000 of playback equipment, but in a fully treated listening room. And I know from the days when I used an untreated room, there would be no way I could tell in an untreated room, even with a million dollars of equipment.
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.
It was an audiophile who'd spent a small fortune decking out his "listening room". I imagine that he knew what he was doing.
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.
To put it simply and honestly, you are not paying enough attention. There is no human-discernible difference between FLAC and V0/320kbps. But implying that 192kbps and 160kbps are sufficiently perfect is somewhat ludicrous. You can still hear noticeable artifacts on cymbals at 192kbps. They will sound like an absolute mess of warbling, and all of the audio will be imbued with a slight tinge of white noise.
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.
>There is no human-discernible difference between FLAC and V0/320kbps
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)
> He still thinks he can hear the difference between FLAC and MP3
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?
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)