Can You Really Hear the Difference Between Lossless, Lossy Audio?

CWmike writes "Lossless audio formats that retain the sound quality of original recordings while also offering some compression for data storage are being championed by musicians like Neil Young and Dave Grohl, who say compressed formats like the MP3s being sold on iTunes rob listeners of the artist's intent. By Young's estimation, CDs can only offer about 15% of the data that was in a master sound track, and when you compress that CD into a lossy MP3 or AAC file format, you lose even more of the depth and quality of a recording. Audiophiles, who have long remained loyal to vinyl albums, are also adopting the lossless formats, some of the most popular of which are FLAC and AIFF, and in some cases can build up terabyte-sized album collections as the formats are still about five times the size of compressed audio files. Even so, digital music sites like HDtracks claim about three hundred thousand people visit each month to purchase hi-def music. And for music purists, some of whom are convinced there's a significant difference in sound quality, listening to lossy file formats in place of lossless is like settling for a Volkswagen instead of a Ferrari."

Comments

Depends on the bitrate

[Anonymous Coward]

Usually if the bitrate is above 256kb/s, i dont notice any difference.
Ofcourse it still effects some songs (especially the percussion parts).

Re:Depends on the bitrate

[jlfose]

It could be dependent on the gear that playback occurs on and the quality of the listener's ears. In watching Stan Lee's new show about "superhumans" it becomes clear that some people have, by training or genetics, better reflexes then the bulk of humanity. On my home gear I can't tell the difference above 160Kbs, but I'm more then willing to believe that some people can, either because they have much better gear to listen to, and/or they have superior hearing.

Re:Depends on the bitrate

[Joce640k]

You can actually practice listening to music, it's something you learn.

Sometimes the difference between two sets of speakers can be as little as one clarinet in the middle of an orchestral piece. On one set it sounds good, on another it doesn't (or it's hardly there at all).

It's not something you can pick out just by putting on a rap CD for ten seconds and turning the bass up to maximum in a store (which is how most "HiFi" systems are chosen these days and why the manufacturers produce so much garbage).

Re:Depends on the bitrate

[asliarun]

In my humble opinion, this old hoary debate will always remain a debate for several reasons. As you right mentioned, the reproduction environment in most cases is woeful at best. Most speakers are not even full-range to begin with, their cabinets resonate, their drivers cannot often keep up in complex multi-layered music, their passive crossovers do a half-assed job in distributing the sound to the various drivers, and so on. Then, the amps are weak so they start bottoming out and start clipping when the speaker impedance and phase dips sharply in certain frequency bands. Then the electronics, especially the capacitors and power supply cannot keep up. Then the cables are not fat enough or are not shielded enough so they load up the power amp even more. Then the pre-amp adds its own coloration to the already feeble signal coming from the source. Then the DAC does its own thing and further colors or degrades the source signal even more. Then the source adds its own share of noise and jitter to the audio signal that screws up not just the signal quality (bad enough) but even the timing of the music.

On top of it, the room comes into play. The room adds its own coloration and effect that is often a far bigger factor that the audio system itself - boosting certain frequencies while muddying and deadening others, and even adding echoes, reflections, etc.

Then there is the human being at the end of the chain. I personally can't even listen above 16KHz, and I have average ears. I suspect many people are like me too, at either end of our audible spectrum. On top of it, we humans hear music very differently - while our audio range may be fairly similar (20hz to 20khz by popular definition), our sensitivity to *variations* in tone and timing varies drastically - many often have off the charts sensitivity to even slightly off-key music (I do) or slightly off-beat music (I do not at all).

All in all, a decent headphone setup is far far more revealing than a decent audio setup. At a thousand dollars, you can probably assemble a decent headphone, but an audio system will sound atrocious, unless you are willing to spend a whole lot more effort and research in second hand discrete gear OR are willing to do serious DIY.

Anyway - I also wanted to say one thing - the thing that gets neglected the most in all this is actually the quality of the source recording - or what people call "mastering".

Most people who say something like "SACDs sound far better than redbook CD" or "vinyl sounds far better than CD" are most likely saying this because a whole lot more care went into recording the SACD or vinyl compared to the cheaper mass market CD or mp3.

If I look back at all the albums I have purchased or listened to (in whatever format), the one thing that stands out to me personally is that I have found less than 10% of them to be "recorded with care". And I'm not even being picky! Across the board, I can say that recording quality sucks when it comes to rock (which is what I listen to most often) - and I mean all kinds of rock.

If Neil Young's initiative (and even his Pono device) and Dave Grohl's initiatives are successful in improving the audio quality of music in general, I strongly suspect it will be because recording quality will be done with greater care, not because they decided to use a fancier digital format or use higher number of bits and samples to store their music. While everything becomes a factor by the time the music reaches your ears (heck, by the time it is processed by your brain, you even have to factor in psychoacoustics and gear bias and the "burn-in" syndrome) - the recording quality in general needs to improve (except for the jazz and classical pieces that audiophiles love to love, and are hence recorded with care), and this improvement will arguably make the biggest difference in audio quality.

A lengthy, thorough, and well-explained discussion

[EmagGeek]

There is a long discussion among very qualified individuals on this subject. You can read it here

Depends on the source

[Stentapp]

I am quite sure I prefer a lossy compressed version of a 24 bit, 96 kHz track than a lossless compressed version of a 16 bit, 44.1 kHz track.

Re:Depends on the source

[Hatta]

44.1hkz 16bit audio is completely transparent to the human ear. No one has ever been able to detect when a 16bit DAC ADC pair has been placed in a 24/96 audio path.

Your preference for 24/96 audio as a listener is entirely due to the placebo effect. There are good reasons to master audio in high res, but for listening 16 bit 44.1khz audio is as good as anything.

Re:Depends on the source

[QRDeNameland]

Your preference for 24/96 audio as a listener is entirely due to the placebo effect.

Well, in all fairness, listeners may actually hear perceptible differences between 24/96 and 16/44.1 audio sources due to different mastering, but of course that says nothing about whether they can actually tell the difference between the two bitrates when everything else is equal.

This article is a pretty good explanation of why 16/44.1 is as good as anyone needs for playback.

Re:Depends on the source

[femtobyte]

You sure can hear the difference if you stick a 44.1kHz DAQ in a 96kHz signal chain before filtering out ultrasonic high frequency components (if there are enough to make a difference). The advantage of 96kHz recording isn't that it can capture any more human-audible frequencies than 44kHz can, but that you have a lot more leeway to prevent aliasing of signals above the Nyquist limit down into the audible range (a 25kHz tone sampled at 44kHz results in a spurious, highly audible (25-44/2)=3kHz aliasing signal).

It's pretty much impossible to build analog frequency filters with a sharp cutoff (e.g. everything below 20kHz and below gets through, everything above 22kHz is -60dB attenuated), so recording at 44.1kHz sampling requires either being absolutely certain the original sound source has minimal high-frequency harmonics, or heavy analog filtering that cuts well into the audible high frequency range. With 96kHz sampling, it's much easier to build an analog filter that gradually rolls off high frequencies between 20kHz and 40kHz (...producing a >40kHz sound is tricky in the first place), preventing aliasing without the filter cutting into the audible range. Once digitized, it's trivial to make a *digital* filter with a perfect frequency cutoff to downsample the 96kHz to aliasing-free 44.1kHz.

Re:Depends on the source

[hairyfeet]

You are 100% correct, I have sat in a $100k studio with $5k reference monitors and heard my tracks played back at both 192k and at 44.1k and honestly? Couldn't tell the difference, i really couldn't. And while my midrange hearing may not be the greatest I'm picky as hell when it comes to low end and that is usually the first thing that goes when you compress but standard 44.1k? Couldn't tell the difference which if there was gonna be a difference i would have heard it on that system, it was top notch. I'm sure many here can bring citations showing double blind tests which i have no doubt show its all placebo, because if I can't hear it in a nice studio with the actual live instrument right beside it i doubt seriously anybody is gonna hear a difference with home gear, even high end home gear.

Re:Depends on the source

[chipschap]

44.1hkz 16bit audio is completely transparent to the human ear. No one has ever been able to detect when a 16bit DAC ADC pair has been placed in a 24/96 audio path.

Your preference for 24/96 audio as a listener is entirely due to the placebo effect. There are good reasons to master audio in high res, but for listening 16 bit 44.1khz audio is as good as anything.

As a former audio professional (specialized in location recording of choirs and orchestras) I must agree. But even my aging ears can hear the difference between 44.1 (or 48)kHz 16 bit uncompressed and a typical MP3. Side note: 24-bit has a few audible advantages for music with extremely wide dynamic range (from ppp to fff, say) where 16 bit will struggle a little at the very soft end.

Re:Depends on the source

[dgatwood]

Speaking as someone who frequently does recording, your comment suggests that no one has done that test with classical music in a properly controlled listening environment using quality gear while giving the test subject the ability to control the volume arbitrarily. When you crank up the volume, the noise floor difference in soft passages alone should make the difference between 16-bit and 24-bit signal paths a dead giveaway, even for someone with moderate to severe hearing loss. It isn't even subtle. Of course, if the person doesn't turn it down for the loud passages, he/she will likely suffer hearing damage, but perhaps that's why he/she has moderate to severe hearing loss in the first place. :-D

The 44.1 vs. 96 kHz difference is more subtle, requiring someone with top-notch hearing (very rare), headphones that can accurately reproduce frequencies above 20 kHz, and 96 kHz DAC hardware that does not have a bandpass filter starting at 16 kHz. If you fail to verify even one of those requirements, you would expect no one to be able to hear the difference, because there won't be any difference.

Re:Depends on the source

[QRDeNameland]

kinda like 640K?

Unless you want to argue that human hearing is improving similarly to Moore's law, then no.

Re:Depends on the source

[hedwards]

The point of the equipment is that you have quality in reserve as you go through the process of mastering the tracks. The more quality you have in reserve the more you're able to do before you start having to deal with artifacts and other nastiness. As with all such things, you have to think about the order in which you do things and the order in which you throw out data to get the best results.

The point of buying lossless music isn't so much that it's better for listening to, it's that you can compress it however you like later on without having to worry as much about the sound quality you get. Since you have more data to work with, you can get a better quality at a lower bitrate than if you were starting with an already compressed track.

Re:Depends on the source

[hairyfeet]

Well to be really REALLY fair I have noticed it also matter if the original music was recorded in analog or digital, as I've taken some tracks we've cut in a classic studio with the analog 8-track and its really fricking hard to get those to sound really..."right" for want of a better term as its really hard to describe, when it is compared to digital.

The closest I can get to describing it is this and sorry if you aren't a musician but they'll know of which I speak...you know how you have that great old tube amp for the guitar and it has that nice warm fat feel to it? Notice how the same amp when modeled digitally doesn't doesn't quite have the warmth? Its kinda...artificial sounding? That was the trouble we had, the tapes sounded nice and warm but trying to get that to switch over to digital was fucking hard, frankly it was easier to just cut the tracks again in a digital studio than it was to get the analog tapes to really convert well.

Sorry if I'm not describing it correctly but music is one of those things where my terminology often fails me, its so hard to describe feelings and emotions and music for me is an emotional expression so I end up having to try to describe how I felt as I listened or played and my vocabulary fails me, the analog was a little fuzzy but it was warm and lived in feeling while trying to convert that to digital something was lost in translation, no other way I know how to say it. the same tracks recorded natively in digital sounded great, analog sounded great, but putting the two together was just something we never could get to work.

Re:Depends on the source

[juniorkindergarten]

I will tell you now that the average person cannot hear to 20khz. Young children can. Anybody who has listened to loud music for any length of time have blown away the top couple of khz of their audio range.
If you have ever gone to a rock concert and been near the front or gone to most dance clubs and you will have sustained hearing damage. If you have ever left one of these venues with ringing ears, or been around loud machinery and noticed the same, then you have sustained hearing loss. Your hearing will recover mostly after the trauma and that will be indicated by the subsiding of the ringing of your ears.
If you want to find out how your good/bad hearing is, spend the money and see an audiologist. You will be surprised on to find out what your hearing is really like.

Re:Depends on the source

[femtobyte]

1) Digitally, yes you can. Take the DFT of the data; zero out all components above your frequency cutoff; reconstruct the signal as the sum of below-cutoff frequencies. Voila, a perfect sharp cutoff. The only subtlety is that you can only choose an exact cutoff corresponding to some integral number of cycles in your sampling window, so you can't cutoff at exactly sqrt(e*pi)kHz --- but you do have plenty of wave numbers from which to select a perfect cutoff (increasing with the size of your DFT window).

2) Untrue: a 44kHz *sampling rate* has a 44/2=22kHz Nyquist cutoff. Frequencies f>22kHz Nyquist limit "wrap around" to f-22kHz difference frequencies.

But yes, I agree, on the playback side there's no audible difference between a (sufficiently well made) 44.1kHz and 96kHz DAC.

Re:Depends on the source

[Entropius]

OT, as a choral performer:

Classical music has a stupid wide dynamic range, more than any other genre I know of, and (in particular) soprano sections have a nasty talent for pegging meters that were supposed to be set with plenty of headroom.

Re:Depends on the source

[nabsltd]

The closest I can get to describing it is this and sorry if you aren't a musician but they'll know of which I speak...you know how you have that great old tube amp for the guitar and it has that nice warm fat feel to it? Notice how the same amp when modeled digitally doesn't doesn't quite have the warmth?

The reason for this is that it's hard to capture distortion accurately.

That "warm sound" is a result of the inacurracies of the tube amp. You may like it better (and that's just fine), but it is does not accurately reproduce the original signal. For me, it's really no different than the current "loudness war" where re-mastered releases are much louder. Many of today's listeners like that sound beter, but it isn't accurate.

Re:Depends on the source

[djdanlib]

As a live sound engineer dealing with vocalists who do that regularly (sing at normal program levels and then BELT A PHRASE OUT)... let me say... ARGH.

I put a steep compressor on someone who's prone to doing that, and let me tell you, it makes my life much easier. I can't fix the clipping, but I can make sure they don't cause the audience to cover their ears.

One word: YES.

[Anonymous Coward]

Caveat: You have to have decent headphones (not Apple earbud BS), and/or good speakers, but that's about it. The difference is negligible once you hit ~320Kbps MP3, in my opinion, but anything under 256Kbps, regardless of lossy format, you can *clearly* hear cymbal hits turning to an underwater splooshy mess.

I can hear a slight difference

[jgtg32a]

I can't tell which one is better though.

No

[Hatta]

No you can't. Not with any reasonably modern encoder and bitrates above 256. Anyone who tells you otherwise is experiencing the placbo effect. BTW, you can't tell the difference between 16bit/44.1khz audio and 24/96 audio either. And vinyl might sound "better" than digital to you, but digital is objectively more accurate.

Audiophilia is saturated with woo. This is the same market that brought us $500 ethernet cables.

I usually can, but I rarely care.

[Clueless Moron]

I'm listening to a performance, not some audio benchmark. If a bit of loss bothers you, it must be some pretty damned uninspiring music you're listening to.

And if you're listening on some random mp3 player with bud headphones while walking around doing stuff, compression loss is the least of your worries.

44.1khz ought to be enough for anyone...

[scorp1us]

We recently discovered that human hearing beats the linear response assumptions used in lossy codecs. So yes, their criticisms are scientifically founded.

Re:44.1khz ought to be enough for anyone...

[Hatta]

Unless you have people that can ABX the difference, no their criticisms are not scientifically founded. An actual blind test beats any theoretical reasoning any day.

It doesn't matter

[Anonymous Coward]

The reason people use lossless compression for audio (i.e. FLAC or SHN) is not because they can tell the difference. Maybe you think you can, maybe you think you can't, but that's irrelevant anyway. The reason people choose lossless is that lossless is the only suitable solution for archiving. If you want to preserve your CD audio exactly as it appears on the CD, the only possible solution is lossless compression. If you choose lossy, you aren't making an archive or the original, but rather an approximation of the original.

That's all there is to it.

Re:It doesn't matter

[tuffy]

And you never have to re-rip physical discs. 128kb/s CBR MP3 used to be the standard. Then 192 VBR. Then AAC. And so on and so forth. So by keeping a lossless archive, one will always be able to transcode to the latest-and-greatest lossy codec without a lot of hassle.

Re:Audiophiles might.

[msauve]

Mine goes to fiveier.

Re:No

[rudy_wayne]

In medical tests, people are given a placebo and yet claim to feel better or feel the same effects as people who are given the real medication. These must be the same people who rail against mp3s.

Just because Neil young and Dave Grohl are famous musicians, it doesn't mean that they actually know what they are talking about. 40 years of exposure to loud music has probably damaged their hearing enough that they really don't know what they are hearing.

Saying that A sounds better than B is completely subjective and affected by many things. Not just how the music was encoded, but the quality of the DAC used for playback and the quality of the speakers/headphones used.

Re:the answer is obvious, isn't it?

[rudy_wayne]

It's like asking "can you win a race against a Toyoda?" where do you even start with that....?

Since Akio Toyoda is 30 years older than me, I'm pretty sure I could beat him in a race.

Re:Better question

[noh8rz10]

it doesn't matter how lossy or lossless the file is if you're listening with shitty white earbuds.

Re:Better question

[MarkGriz]

Or not using Monster Cable

Re:Better question

[Joce640k]

This is the real point: People are so used to listening to music with no dynamic range, on ear buds, in crappy acoustic environments that they wouldn't know where to start listening for a difference.

Re:Better question

[mwvdlee]

Look, you want your 0's and 1's to look like stupid Comic Sans 0's and 1's or like high quality, stylish Zapfino 0's and 1's?

Re:Better question

[t4ng*]

I think the real point is that there are known limits to human hearing and many audiophiles fantasize about their hearing being superhuman. It just ain't so. Dynamic range compression is one thing, but perceptual compression, sample rate, and bit depth are a different matter. No audiophile has ever heard the difference between FLAC and 320Kbps mp3 audio in an ABX test at a statistical rate that is better than guessing.

Any time this argument starts, I refer people to this well written article that lays out the limits of human hearing compared to the specifications of recording formats...