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2nd Multi-Format 128kbps Public Listening Test
technology is sexy writes "Roberto Amorim has launched his latest public listening test evaluating the performance of different audio codecs at 128kbps, among them Apple's AAC implementation (used in iTunes), LAME, Ogg Vorbis fork auTuV, WMA, Musepack and even Sony's Atrac3 format, which is soon to be used in their own music store. Read more on Hydrogenaudio and check out the results of prior tests. As opposed to most evaluations of audio codecs, this is a scientific test adhering to ITU-R BS.1116-1 as much as possible while still allowing everybody to participate."
128kbps doesn't cut it. It's an absolute lossy, disgusting bitrate, no matter what it's in. They should test similar file sizes instead of by bitrate, to determine whether something is good or not- this gives a better impression of quality vs size, instead of a purely comparison based test.
Great, now all the ____ fanboys are going to forge results to make their codec look good. Talk about useless tests.
Not possible. All you will get is a bunch of WAV-files, you have no way to tell which file belong to which codec.
That said, I don't care which codec wins the test because Vorbis is still the only one free from patents and the margins are so incredibly small.
Vorbis will win for me even in the unlikely scenario that it comes out last.
My other account has a 3-digit UID.
How do you bas a listening test on the web? People with crappy speakers are going to say that all of them sound bad yet the people that have the better speakers are going to have the better responses. This should be something that is done in a controled environment so that the hardware that is playing back the audio is standard.
Yes... certainly this kind of listening test is important to access the capabilities of each codec.
But in the real world other factors may be more important to chose a coded, like for example general acceptance, freely available code and specs, and a large content base available.
You see: performance will increase allways in all codecs with time... so this kind of testing is only a minute factor amongst others.
You cannot proceed from the informal to formal by formal means
I wish there was a filter that scored any post with the words "You're new here, aren't you?" -5 stupid joke.
For example, conventional wisdom says that the human ear cannot detect sounds above roughly 20kHz, yet there is at least some anecdotal evidence that higher order harmonics shape what we hear.
If "normal" human auditory capacity was a completely decoded topic, there wouldn't be nearly as much a need for different approaches to music compression (it would be a much simpler problem with fewer possible solutions)
There are no karma whores, only moderation johns
Why does anyone still use 128kbps? I hate it when I download music (legal ;) and the only bitrate available for the song i want is 128. With 200GB+ hard disks being so affordable these days and everyone having high speed, I think everyone should encode their (mp3||ogg||aac) at 192 or 256.
Well I could be wrong, and forgive me if I've misinterpreted your post...but
Don't all of these compression algorithms rely on psychacoustic modeling to remove 'extraneous' information from the bitstream?
If that is correct, and the algorithms are implemented correctly, then really what we are looking for is the best perceived result.
Just because the output meets the algorithm input->output specs, justn't mean it's the best output as perceived by humans.
Maybe think of it as optimizing sort routines? Yep, bubble-sort or b-tree still output a sorted list, but the perceived value is that the b-tree is better because it performs it's function more quickly.
This isn't an exercise in getting the frequencies algorithmically correct - the end result has to be listenable.
Humans are analog devices...
They do that with small groups, but the point of making this study public is to get a larger sample size without having to plunk down serious cash to set up a "reliable test environment" for thousands of listeners. Also what kind of codec bias could you possibly be referring to?
BTW, I think the difference between MP3 and Vorbis at 128 kb/s is perfectly noticeable. MP3 sounds rather bad, vorbis sounds pretty good. And the point is precisely to tell which format sounds best, so you don't want to do 512 kb/s bitrate where all formats sound close to CD quality.
When you listen to compressed audio over inexpensive speakers / headphones, you can't hear the difference. With my Sony Studio Monitor headphones, I lost the difference at about 250k with mp3, so I started using 320K as that was the best at the time. Then I bought $2000 Martin Logan Mosaic Speakers, and the original CD was clearly better than even the 320K bitrate. So now I only do lossless compression. That's fine at home, but in any other environment, there's usually so much noise and distractions that even if you had excellent headphones or speakers, you wouldn't appreciate that little difference lossless brings over 256K or even 128K.
I'd read the thread when they were discussing which version of Apple's ACC codec to use for the test, and concluded based on a few samples that the new version was subpar.
If they'd included both versions of iTunes/QuickTime in this test, perhaps they could have helped shame Apple into fixing what they broke.
That is not lamer-proof.
One could just send in forms with the same ratings to manipulate the test arbitrary.
If you mod this up, your slashdot background will turn into a beautiful sunset!
So uh, why is this necessary, exactly?
...). therefore just comparing the decoded signal with the original won't do, because the "subjectively" heard difference is what matters.
hmm, the whole point of the "lossy" compression algorithms is to filter out information the human ear/brain is unable/unwilling to hear (psychoacoustics,
and adhering to a certain norm and "scientific method" when comparing those codecs can't be bad...
so what is it exactly that you find unneccesary??
The best replacement for r3mix.net in my opinion is HydrogenAudio . The forums are frequented by a lot of professionals, as well as developers of LAME, FLAC, Nero AAC, Musepack, Wavpack, and other codecs.
it's double-blind, so you don't know what you're testing. Good gear has practically no bearing on identifying compression artifacts - that you need good equipment to hear slight imperfections is a myth.
Jeremy
most encoders reach perceptual transparency at around 160 (MPC) to 190 (MP3) kbps. At these bitrates even the most trained listeners can't tell the damn difference on 99.999% of samples. Of course, audiophiles are the most succeptible group of people in the world to placebo, so they probably "think" they can hear it.
Jeremy
After a while, once you have weeded out bad ways, one is going to reach the following situation. Each algorithm will perform very well for a large set of music and poorly for some small set of music. Barring pathologies, The poor set will be assymtotically fixable by increacing the bit rate. By the way this is not just my opinion. Theres theorems that say this is true of any compression scheme when applied to all problems.
what does this mean? it means that the end user is never going to work at the truly low end of the bit rate specrrum because they want something that virtually always works. Plus they want a wee bit more just in case they have to transcode it. So if the recommended rate is 128 people will encode at 160.
So these comparisons need to be done not at the bitter edge where music flaws are easy to spot because NO ONE WILL ACTUALLY MAKE THAT THE OPERATING POINT THEY USE. That is to say everyone knows vorbis sounds so-so at 64KB while MP3 sound much worse. But no one wants So-So they want darn good. So they are going to recors their Mp3 at 160 and at 160 Ogg and Mp3 sound so close that the size of the test you'd have to do to pick up the difference is silly.
the proper way to do this is the following. Pick the gold standard format, say MP3 and its standard excellent operating point, say 160. now test all the others at lower bit rates than 160, and see which one has the lowest bit rate that scores as good as the Mp3 at 160.
comparing all methods at a constant bit rate, esepciall a low one, is stupid
Some drink at the fountain of knowledge. Others just gargle.
It is really refreshing to see someone so willing to demonstrate their wrongheaded ignorance. Saves us all a lot of trouble.
I've found most of the people on Hydrogenaudio to be incredibly pragmatic. Perfection isn't the only parameter of importance. If it were, they'd not be wasting time testing codecs at 128kbps, except to demonstrate their unsuitability compaired to losless formats. They'd not be wasting time letting phillistines with their waxy untrimmed ears particpate in listening tests with their $20 sony earbuds.
As for the vendors lauding useless gear, um, what vendors lauding useless gear?
But hell, why let any of that get in the place of a perfectly good piece of ranting rhetoric. Still, it would be better if you'd unloaded at a deserving target. There are certainly enough of them out there.