Can You Really Hear the Difference Between Lossless, Lossy Audio? 749
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."
Any good studies? (Score:5, Interesting)
Yes and No. (Score:2, Interesting)
People with normal, standard hearing cannot detect a difference, that' s the point of the compression.
If you are hearing a difference, it's because you have a hearing defect. If you can hear something that you don't hear after compression, it's because you're deaf to the sounds that's overlaying it (and killed it in the compression)
You were hearing the original differently in the first place, than anybody else with normal hearing.
Oblig (Score:4, Interesting)
Re:Depends on the source (Score:5, Interesting)
Depends on how good the sound engineers are. A lot can be gained by higher resolution and sample rate in the mastering stage, but by using a good low pass filter and dithering (and dithering is not really necessary, http://developers.slashdot.org/story/13/02/27/1547244/xiph-episode-2-digital-show-tell [slashdot.org] ) basically all audible information is captured in 44.1kHz / 16. Your speakers probably don't go much above 20 kHz anyway, so anything beyond 44.1kHz will only cause distortion (aliasing), see post by MetalliQaZ "Debunked" below.
Re:It doesn't matter (Score:4, Interesting)
Oh, for mod points.
While I can't (mostly) tell the difference between the original CD and a ~140Kbs VBR MP3, I _can_ tell the difference between a 140Kbs VBR MB3 made from the CD source, and a 140Kbs VBR MP3 made from a 256Kbs VBR MP3.
Lossless isn't for listening to, it's for archiving. And make sure you get the cuesheet, pregaps, etc. right when you're archiving too :)
Comment removed (Score:5, Interesting)
Re:Depends on the bitrate (Score:5, Interesting)
Re:Any good studies? (Score:5, Interesting)
I don't know if it's a good study, but I did exactly this test. Ten or fifteen years ago.
I took four musical selections (from the latest Rolling Stones album at the time, a solo piano performance, a classical orchestra, a female vocal), and encoded them at 128, 192, and 256 Kbps with the Fraunhofer codec of the day (remember that?). I re-expanded them to 44.1 KHz CD tracks, and put them on a burned audio CD (remember those?). Each selection on the CD had five versions - the first was always the original bit-for-bit copy from the source CD, then followed (in random order) the 128, 192, 256, and the original again.
I made ten copies, and handed them out to the audiophiles in the office to play on their home stereos, and gave them a test sheet - I asked them to identify for each selection which version was 128, 192, 256, or the original. Nobody came close to having a "golden ear" that could reliably tell the 128Kbps versions from the others, much less the higher bitrates. Overall, there was a slight ability to detect the 128 kbps versions - it got selected as the lowest quality one more times than random chance would suggest, but even it was still well below 50% (I don't remember the exact numbers any more).
And this was with ancient MP3 encoders.
Frankly, if you think you've got the golden ear, first of all I pity you - I'm sorry that you have to put up with all the crap you're going to hear. Second of all, I really recommend running the same test - prepare the tracks, have a friend randomly order them (but keep track), and then see if you can identify them. Don't simply say "Of course I can" - Actually do it and prove it.
And, if I can be an old man with a bit of advice for a minute: if you can't tell the difference, don't go out of your way to train yourself to tell the difference. It'll just be an annoyance to you for the rest of your life. Kinda like the person who taught me about the reel-change indicators on film at the movie theatres - I see it, and my whole body tenses up waiting for the change. I wish I had never known about it. I really appreciate the change to digital projection so I don't have to deal with those anymore. /frank
AIFF?, Flac!, Lossless in General. & Randomnes (Score:4, Interesting)
Also, it is definitely possible to tell lossless audio from lossy audio, even at higher bitrates. Around 2002 I had a friend who completely mocked my lossless ways, even though I'm not one of those gold-cable audiophile people -- just a normal guy who likes his music. I just had a decent pair of Klipsh speakers with a subwoofer. My friend was so certain that this was all in my head and I was so certain that it was not that we devised a simple test. He would show me two identical-looking files in iTunes, just showing the titles. One was a high-bitrate AAC and the other a FLAC file. I could click on them to play them as much as I wanted. I was then to decide which was lossless and which was lossy. We did this with 10 files. It was basically double-blind as he didn't know which was which either until he took the computer back to check my answer. He set up 10 files this way. All in all the test took just 5 or 10 minutes.
I got 9 of 10 right. It is hard to describe sounds, but the lossless music is "deeper," especially bass, guitar vibrations and high notes. This makes it obvious for many songs.
However, I expect not everyone has hearing like this. I suspect this because one day I heard this annoying buzzing sound and asked my girlfriend about it. She couldn't hear anything. So, I searched all over for what was causing it. It turned out it was a television that was on, but that was on a non-channel so it was completely black on the screen. However, the CRT television emitted a sound from being on in a silent room that I found annoying and my girlfriend couldn't even hear. My sister could also hear it when I tested her later. I also sometimes find the sounds fluorescent lights make annoying too.
Anyway, lossless is great and, yes, you can hear the difference if you have hearing which can hear the difference. It's sort of tautological, but it's the truth.
Comment removed (Score:5, Interesting)
Re:Depends on the source (Score:4, Interesting)
"There are good reasons to master audio in high res, but for listening 16 bit 44.1khz audio is as good as anything."
The reasons for having "extra" fidelity in master recordings is the same reason for having high-resolution photos in "raw" format: there is lots more wiggle room for editing while still maintaining good enough fidelity that the end user can't tell the difference.
For example: take a large (say 16M pixel) 8 x 10 photo, and reduce it to 4 x 5 at 600 dpi. Then take the same photo, edit it (for example, change some colors, remove a cloud from the sky, etc.) and reduce that to the same size and resolution. Even though the resulting photos are higher resolution (at arm's length) than the eye can perceive, they look different.
I knew this article was gonna be BS (Score:3, Interesting)
And nothing of value was lost in the remaining 85% of the *data* that is inaudible to the human ear.
"Young, in fact, created his own digital-to-analog conversion (DAC) service called Pono. Young has tweeted that the Pono cloud-based music service, along with Pono portable digital-to-analog players, will be available by summer."
There's your cash-in scheme lurking behind all the BS.
"Young's service would increase the quality, or sampling rate, of the music from 44,100 times per second in a CD (44.1KHz) to 192,000 times per second (192KHz), and will boost the bit depth from 16-bit to 24-bit."
I would like to repeatedly hit you over the head with http://people.xiph.org/~xiphmont/demo/neil-young.html [xiph.org]
"The sample rate of a digital file refers to the number of "snapshots" of audio that are offered up every second. Think of it like a high-definition movie, where the more frames per second you have, the higher the quality."
NO, do not think of it like that unless you're a charlatan. Refer to rebuttal on xiph.org.
"Millions of people in the world are audiophiles."
No doubt, Millions of people in the world are fools and they have money that could be yours.
"It's just common sense that the higher the resolution -- the more data that's in an audio file -- the better the sound quality, Chesky said."
Too bad this thing called SCIENCE has been trumping "common sense" for millenia now.
"The site also recommends high-resolution player software such as JRiver, Pure Music, or Decibel Audio Player. The software, which basically turns your desktop or laptop into a music server or a digital-to-analog converter,"
HILLARIOUS. I won't even begin to..
"The most popular music server among audiophiles, according to Bliss, is an Apple Mac Mini."
This is beautiful. I am not surprised in the least to see this audiophile-appleophile overlap.
Re:AIFF?, Flac!, Lossless in General. & Random (Score:5, Interesting)
AAC (like MP3) is a frequency-domain codec, and can therefore never provide transparent audio. It has nothing to do with "deeper". but instead is an inability to represent transients... non-tonal components like percussive sounds and other noise.
If you had performed the test with Musepack/MPC or even MPEG-1 Layer II at high bitrates, you would have failed the test.
http://en.wikipedia.org/wiki/MPEG-1#Quality [wikipedia.org]
Re:Depends on the source (Score:5, Interesting)
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 bitrate (Score:5, Interesting)
Re:Depends on the bitrate (Score:4, Interesting)
Re:Depends on the source (Score:5, Interesting)
The trick you're playing on yourself here is:
x = [1 0 0 0 0 0 0 0]; % x is only defined on 8 samples over the interval. There are an infinite number of continuous signals that could be sampled this way.
Following your procedure through to y:
octave:5] y = ifft(Y);
octave:6] y
y =
0.87500 0.12500 -0.12500 0.12500 -0.12500 0.12500 -0.12500 0.12500
so y is also defined at 8 sample points; as for x, there are an infinite number of curves that could fit these. One of these curves is the sum of frequencies indicated by Y. But what does fft(y,256); mean? From the Matlab documentation,
"Y = fft(X,n) returns the n-point DFT. fft(X) is equivalent to fft(X, n) where n is the size of X in the first nonsingleton dimension. If the length of X is less than n, X is padded with trailing zeros to length n."
So, now you have y defined in a larger window (y = 0.87500 0.12500 -0.12500 0.12500 -0.12500 0.12500 -0.12500 0.12500 0 0 0 0 0 .... 0). See my response above to another poster's question: when you enlarge the sampling window, you "create" a lot of possible "intermediate" frequencies that "don't exist" (i.e. are indistinguishable from sums of integral frequencies in the shorter window). By padding y with zeros to a larger window, you're looking at a *different* signal from the un-padded y alone; consequently, you need the "extra frequencies" that you ascribe to the "non-sharp-cutoff" to correctly describe the different "y+0,0,0,0,...,0" signal (which is distinct from y). But that doesn't mean the cutoff isn't perfect as defined on the original signal x->y. In fact, if you periodically *repeat* y (y->y,y,y...,y instead of y->y,0,0,0...) you'll see the "sharp cutoff" still applies since the periodic signal is still the sum of the original frequencies in y.
Audiophiles are, for the most part, gullible (Score:2, Interesting)
After all, we're talking about people who buy $1,000 Monster cables, even though in a blind test, they can't tell the difference between those and wire coat hangers. [consumerist.com]