Audio Format Listening Tests Concluded
Pointing to the conclusions of this listening study, nullity writes: "The results are interesting, and show a high variation in the performance of the various codecs on different musical styles. Ogg seems to work well on dance music, WMA8 on chamber music, etc."
Ogg seems to work well on dance music, WMA8 on chamber music, etc.
Like requiem...
While mp3s encoded at lower bit rates seem to have a tingly sound sometimes, almost every wma file I've gotten or encoded ALWAYS sounds like someone is blowing windchimes in my music. I can't stand it.
In my opinion, 192kbps MP3 is the way to go, but then what do I know? I can't hear about 13khz.
Chris
These tests are all at 64 kbps and most people use much higher bitrates for real music. I'd like to see comparisons at 128k bits minimum, and preferably 160k or 192k, which is what most quality mp3's are at, for direct comparison.
Tests confirmed that attempting to encode "Aphex Twin" with any of these codecs caused the PC to tremble at a frequency that, when connected to a refracting laser stuck up Bill Gates' ass, had it spell out "we're all dead" on the nearest wall.
Invoicing, Time Tracking, Reporting
In some circles, it's believed that Ogg Vorbis is the future of lossy format music. You get higher quality in less harddisk space than with MP3, plus it's a far more open codec allowing more customisation with less legal risk.
Steven Woston
Lead Programmer, J-j-j-julius SoftwareWell....not quite. There's a different frequency distribution between electronic, pop acoustic and classical music.
Specifically, electronic music, which most dance stuff is, has a very flat frequency distribution. See this for yourself - load your favourite media player, siwtch on the graphic equaliser graph and watch how basically nothing happens except in the mid-range.
Now try again with an orchestral piece. There will be much more variation, though in most it will tend towards the top end.
Now try again with rock. Tends towards the bottom and top, with middle frequencies missing.
Keep going with any format you feel like mentioning...you'll get the same.
Actually, this is a striking example of how recording techniques can ruin sound as well. Take a look at the Apollo 440 album - Gettin' High on Your Own Supply. A good mixture of guitars and electronics, right? Well, look at the frequency graph again. See how virtually every guitar frequency variation has been cut out: this music was recorded digitally, mostly using samples by the looks of it. The normal variations you'd associate with having guitars play live are all filtered out, and the graph goes back to the flat digital sound again.
Cheers,
Ian
This isn't terribly surprising, as it was already known that the different formats have different frequency responses. More specifically, the way they compress the music dictates what frequencies are cut out. MP3s, for example, are notorious for removing high and low frequencies from music - not a big deal for casual listeners, but those with high-end stereo systems will definitely notice the lack of high overtones, and the "flat" low end bass response. WMA sort of sounds like certain frequencies are cut from the raw audio, leaving the rest to fill in as sort of an approximation of the original full sound - it sounds hollow and "chime-y". Ogg has its defining sound characteristics as well. Thus, it isn't surprising that different styles of music sound better encoded in different formats, as different styles take advantage of different frequencies. Rock music has high frequency cymbals and low frequency bass drums and guitars, as well as a very full mid-range, so a well-rounded encoding system works well. Classical is somewhat more compressed, as a result of the physical limitations in terms of sound reproduction of the instruments, so to the undiscerning ear, a format with especially good mids will suffice. The examples go on and on, but the point is that different tools are needed for different jobs - if nothing else, this study shows that having having a number of encoding tools on hand is actually a good thing. When you look in your tool box, you've got more than a couple Phillips-head screwdrivers - you should have enough tools to deal adequately with any job. The same applies to music.
Let's assume that anyone who likes Ogg and is seriously into music will compress their music with both Ogg variants and use the best variant for each file.
Therefore we should also consider taking the best of the two results and comparing it to mp3.
From a quick look at the results it appears that Ogg will still be edged out by mp3 when analysed in such a fashion, but it's much closer.
Also a test on several bit rates would be useful.
See http://etbe.coker.com.au/ for my blog.
I guess grip will have to use Genre info from CDDB to decide what to encode the the files as now. I wonder if you coudl set up something to optimize individual tracks. Like scan a wav and pick the best codec for the frequencies used in the audio.
Why not fork?
I'd be interested to know how these codecs perform when streaming things like news or talk radio or foreign language lessons. Clarity at a low bandwidth would support a lot of simultaneous listeners from a low-end server. Clarity at medium bandwidth could provide the extra sound quality needed for something like language learning/practice, again from relatively modest servers.
"Those who have never entered upon scientific pursuits know not a tithe of the poetry by which they are surrounded."
I noticed a number of confused posters here... The tested codecs were AAC/MP3PRO/OGG/WMA, not MP3. Had mp3 been tested, it would have lost every round as all of the tested codecs are vastly superior to plain MP3 at this bitrate.
It also should be noted that the only two samples that WMA beat OGG at (indeed the only ones that it didn't totally flop on) were two very simple samples that are demonstrations of two differnt weaknesses in the current revision of vorbis. Orignally the results page had some very interesting commentary from Monty on this, but it looks like it got pulled.
With the exception of those two samples, OGG clearly won. Even including those, it was only beat out by MP3PRO by a small margin. When you factor in that MP3PRO isn't available at anything but such low bitrates and that it's substantially more propritary then MP3, it seems like pretty much a no-contest.
Who cares what music was used?
I agree that each song should have been evaluated by every test subject, but I think it's completely valid to include Green Day or music for which the original recording was of mediocre quality; the purpose of the test was obviously not to see which codec had the best transient response, lowest weighted snr or what-have-you -- the purpose of the test was to determine which codec was subjectively preferred under realistic low bitrate application of the codecs.
People encode Green Day and the Mamas and Papas (frighteningly enough), so those artists are fair game.
How would the codecs compare in bitrate given a minimum quality requirement. Say eg at least 99.9% of the samples produced when decoding must match the original wave with 99.9% accuracy, at what bitrate can this requirement be met by the various codecs (which is smaller). And a nice graph of some sort for various musical styles.
Maybe the music the respective formats are best in represent the musical tastes of (most) of the authors, Microsoft with WMA chamber music, OGG vorbis getting good in techo type music (and even though i prefer Power metal (think manowar, western europe metal bands (not rammstien, ugh)), i know alot of nerds love techno.
Just a thought.
P.s. It doens't explain too clearly the diffirence between the two ogg vorbis formats, atleast for me, can anyone expand a little?
Microsoft IIS is to webserving as KFC is to healthy eating
There's been a couple of posts along these lines, citing cheaper hard disk costs and improved client side bandwidth.
The bottom line is that if a codec like Ogg Vorbis can deliver a similar experience at 64kbps to another codec at higher bitrates, that can be quite a saving in bandwidth at the server end. If you're running a streaming music service, that can be quite a saving.
OGG is not a sound format. Vorbis is the audio codec and OGG is the bitstream manager!
Tom
Someday, I'll have a real sig.
His team looked so enthusiastic mooting AAC audio in their MacWorld NY keynote presentation.
:p
They must be kinda broken-hearted
Ogg in peace,
Michel
Michel
Fedora Project Contribut
"Are you saying that the intended audience for these tests are people that are not interested in the quality of the music they're listening to?"
but you hit the nail on the head. And the answer is `Yes`.
I listen to loads of music, and I cant stand lossy compression systems. I can see the point, like Napster or whatever, but the idea of spending £400 or whatever on an iPod is laughable. Blank cds are 20p each, writers are £60 - why bother arsing around with wondering about 64Kbits vs 128 etc? Anyone listening to music at 64K is wasting their time. 128 still sounds like much of it was recorded underwater, especially the bass.
But obviously to some people its just about how much music they can hoard away.
that these codecs are lossy, and take advantage of the fact that the human ear is better at hearing certain things than others to pair out extraneous info and improve compression. IOW, it doesn't matter how technically different the new files are as long as they still sound the same to the human ear.
BlackGriffen
First of all, people should NOT confuse mp3 with mp3 pro.. mp3 pro has very little support, is only useful for lower bitrates, and is even more mangled than mp3 when it comes to patents and stuff. So even though it won by a small margin, it still doesn't stand a chance against vorbis in practice.
Besides, vorbis will improve at 64 kbps in the future, and probably beat mp3 pro one day. The vorbis-team has focused most on the 128-ish bitrates for now.
One more thing.. mp3 is not even included in this test.. but perhaps they should have, as I know from experience that mp3 at 64kbps sucks very much.. would it suck even more than wma8 and QT-AAC? Probably! But they should almost have included mp3 if only so some hesitant people could see how bad mp3s really are for streaming..
More tests will come in the future.. one at 128 kbps too I suppose.. and there I guarantee you that mp3 will be included and get its ass kicked by vorbis. Vorbis is a superior technology, don't get that confused. Its main competitor is aac, but aac is patented and not free, and currently not very tweaked.
Considering that different codecs do better at different music w/ different frequency spreads, who else thinks that the next generation of audio codecs will be multi-modal; in effect, be several codecs in one. Then have each codec specialize on certain types of music. Perhaps even have them run in an advanced mode where they do a frequency analysis of whole songs, rather than just using genre, to automatically select the best codec for the job. Perhaps even use different codecs for different sections of the song. That would definitely help songs like Bohemian Rhapsody and orchestas with movements, etc.
Would this be too time consuming to implement or what?
BlackGriffen
I'd like to know if there are and codecs that support constant quality but a variable bit rate?
A codec with a target bitrate of 64k but maintains quality by channing between say 1k(for silence) and 100k through the streem would be nice.
thank God the internet isn't a human right.
But is their some fundamental reason why nobody else insists on VBR?
Do you mind, your karma has just run over my dogma.
64 kbps can be useful for internet broadcasting, etc. Not the most important use now, agreed.
As for me, at 160 and 192 kbps ogg is better than mp3, and it does not take a sound expert to hear the difference: it is not negligible.
Computers make very fast, very accurate mistakes
I never thought I'd see a black metal band and one of my favorite melodic death/progressive metal bands BOTH listed as test material.
But when you think about it, it makes sense. Modern black metal with its "wall of sound" approach and all the progressive stuff with the wild tempo changes, switching from acoustic to electric guitars in the middle of the track etc. provides a whole lot of substance to really put a codec through its paces. And it's probably stressing entirely different frequency ranges than "boom boom" bassdrum heavy dance/hip hop stuff, providing a nice counterbalance.
Or maybe I'm just talking out of my ass, I'm not a sound engineer or anything.
Just a side note about the frequency distribution of different styles of music:
The reason why classical music generally compresses better is because the frequency distribution of the sound of natural instruments like for instance string instruments (including the human voice) is harmonic. This means that the sound spectrum consists mainly of a superposition of peaks at the base frequencies of the instruments played and their corresponding harmonics at higher frequencies.
If you were to make a two dimensional spectral analysis of a such sound recording with the time axis to the right, the frequency to the top and the amplitude as the color intensity of the point you would see a lot of wiggling lines at
regular distances. (BTW: this would make a great visualization plugin for xmms)
Since audio compression algorithms also make such a spectral analysis and after that discard some of the information below a threshold they can
reproduce a mainly harmonic spectrum easier than that found in pop or rock music, which is much more complex and more "noisy" because of the
use of distorting amplification and all kinds of
percussion.
Holger
Looking at the data, it looks the two samples where Ogg performed poorly ended up being encoded at a significantly smaller average bitrate than any of the other encoders.
The table at the end lists LiszBMinor with an average ogg bitrate of 45 and BachS1007 with an average bitrate of 47. Since the other codecs encoded those samples at a bitrate 64 or higher, this may explain the results.
The results may point to a flaw in Ogg's VBR login rather than in the lossy compression scheme.
OVERALL RANKINGS (12 SAMPLES)
mp3pro 49.00
oggq0 44.00
ogg64 40.00
wm8 24.00
aac 23.00
The AC above me speaks the truth. mp3PRO has no hope of gaining enough market share to become a worthy competitor. It's a very proprietary extention to MP3. OGG being open source and free (as in beer) has clear advantages for hardware vendors (where it really counts). Lets hope the codec is easy to embed into portable products.
I want my Portable OGG CD Player! I'll buy the first one that comes out. Could you imagine? Twice the capacity of normal players and it STILL sounds better (or same capacity truly indistinguishable from CD -- at only 128k). Right now I have to encode my mp3's at ~180-220kbit to get something acceptable. =/
Unfortunately I have no mod points at the moment.
Yours Sincerely, Michael.
Man did you guys even do the test or are you just looking at the results? I did the test with my klipsch speakers and I got nothing like what these crackheads got on this site. Makes me wonder what they were listening to. I was at my fiances house and used her speakers and the worse compression sounded the best. Then I cam home to my beautiful klipsch (best computer speakers I have heard) and the difference was amazing. In most of the tests not only could I see a huge difference but I could tell you which one was which. The only ones I had trouble with were the two oggs and on some files the mp3pro. The oggs tend to increase the highs a tiny bit making them very different than the other compressions that tend to remove highs. The mp3pro on some occasions wasas good or a little better than the oggs. But both oggs where in the top 3 every time I did the test. I call for a redo because I think either people were guessing, using bad speakers or had too much wax in their ears. I did my own test with the blind player with ogg and did the two styles of ogg versus mp3 at 96k and 128k. The 96k was horrible but at 128k you could hardly tell the difference from the 64k ogg files. The two oggs were almost identical in every test. The 128 was a little better in almost every test but at twice the size it was incredible. I also tried then to compress the ogg files as much as I could to 45k max bitrate. Man it still sounded good but on songs with lots of simbols you could see obvious compression. Bottom line if you use on a regular basis 128k mp3's switch to ogg at 64k without having them side by side you will _NEVER_ know the difference.
"We can no longer live as rats... we know too much." -Secret of NIMH
I haven't seen anyone else mention this yet. At the end, he gives a table of the bitrates for each song for each codec. The one with the greatest variation appears to be oggq0. I noticed that for the songs where that codec did well, the bitrate was much higher, and where it did poorly, it was much lower. I don't realy understand how the bitrate is chosen, but as I understand it, the encoder chooses it automatically somehow, right? I wonder how effective that really is.
The 60 files (12 songs * 5 formats) were all compressed at between 64 and 74 kbits/sec - except LiszBMinor and BachS1007 for OGG q0. They were actually stored at 45 and 49 kbits/sec respectively. No surprise the testers rated them low.
-
- - You can't take something off the Internet! That's like trying to take pee out of a swimming pool.
ABC/HR.. as in ABC/Hidden Reference... as in, there is a copy of the original track included as a hidden reference on every single trial.
The users are given 2 sliders per sample laid out on a panel. The samples are loaded in random order. On the sliders for each sample, one slider is for the original sample, and one is for the encoded. These are also randomized per sample. The user does not know which is which. If they happen to rate the original sample less than 5.0 (highest rating, meaning it should be transparent), then their results are disregarded entirely for that sample.
Does anyone else feel it would have been nice to see Red Book CD audio (16-bit 44.1KHz uncompressed) compared as a control? Seeing how "pure" audio compares to these compression standards could make the results seem more objective.
mp3pro is better than wma8 (99.9% confidence)
mp3pro is better than aac (99.9% confidence)
oggq0 is better than aac (99.3% confidence)
oggq0 is better than wma8 (99% confidence)
ogg64 is better than aac (97.2% confidence)
ogg64 is better than wma8 (96.1% confidence)
Is 99% good enough for you? Or perhaps you should just take the two at 99.9%?
Dammit, the lameness filter is kicking in. No, these are *not* junk characters - I'm trying to show the peon some useful statistical information, you worthless piece of software. I've already removed all the hyphens, what the hell more do you want me to do? Is a percentage sign 'junk'? Is a question mark? Is a space? What the *fuck* use is this, when it doesn't stop all the crapflooders in any way whatsoever - they just flood with random gay/incest/beastiality sex stories instead... I've been posting on this site for years, and for my sins haven't crap-flooded once - give me a LITTLE FUCKING LICENCE TO POST MATERIAL.
-- Help Digitise the Public Domain at DP.
Been using Ogg/Vorbis/Squish on Quicktime for a year. The Ogg/Vorbis/Squish codec got much better between 1.0rc2 and 1.0. At 128k it's already better than mp3 and the managed bitrate encoding is faster than the hard drive can read. The real value is of course, the ability to read these encoded files as long as there is UNIX. Mp3 is going to die and when it does there won't be any appliance makers interested in paying the $10,000 royalty to support mp3.
Please moderate the parent up - unlike the grandparent, this anonymous coward actually knows what he is talking about.
-- Help Digitise the Public Domain at DP.
Don't the codecs adjust their alogrithms according to their bit rate? Is it possible that some encoders don't proportionally sound better at higher bit rates than other encoders? I believe this test reflects sound quality for 64kbps and nothing else since different methods are used in encoding at this bit rate compared to at 128kbps and higher.
> These tests are all at 64 kbps and most people use much higher bitrates for real music.
Maybe the difference at 160kB is such small that any opinion is based on biased preferences. So if you do a test at 64kBit and then proof/estimate the "scaling" behaviour. You should be able to really tell about the quality of the codecs.
I use flac (lossless compression). What was the problem again ? ;)
Seriously, the reason I use flac, even though it takes up a shit load of space is that in the future, inevitably we will have more space to store everything. When we do, thousands will be cursing their crappy mp3s that they ripped at 128 to save space.
Of course, ahem, if you kept your original CDs to rip from then you can just re-rip them to flac or another lossless compression then, but still, why do it all twice ?
graspee
The header states: "Ogg seems to work well on dance music, WMA8 on chamber music, etc."
However, as far as I could tell the only chamber pieces here were LisztBMinor and BachS1007, and for both of theses pieces mp3pro beat out WMA8, in the case of Liszt by a rather large margin.
--- What
Looks like I'll have to get the white album, *again*!
--fetch daddy's blue fright wig, i must be handsome when i release my rage
Personally, I can hardly tell the difference between MP3 and CD and my old vinyl. When I play something on CD, then MP3, then CD and listen carefully for the 'crappy bits' I can hear them - but they don't bother me in the slightest when listening to them.
Are my ears just a bit shite? Are most of you guys able to tell the difference - or are the audiophiles just more vocal?
The participants could have just been scoring on "this is different to the unencoded track, therefore it must be worse".
Well, I don't see anything wrong with that. That is EXACTLY what the were supposed to judge. It is not a contest about what sounds better, it's a contest about which codec reproduces the source more faithfully (in the sense of Hi(gh)Fi(delity)). The more different from the source, the worse.
quake74I think, that for anyone who would actually be interested in which codec does best on which kind of music, it's a moot point, since by now they delete anything below 128kbps on sight
If you can't see the value in jet powered ants you should turn in your nerd card. - Dunbal (464142)
I do not know why Vorbis tried to compress those two so much more than the others. But until the
developers do something about that, I guess we'll want to specify the bitrate when encoding chamber music. Wait a minute, why would we want to listen to chamber music comprossed to this low-quality anyways?
For those of you that don't know, Opeth's Blackwater Park is one of the most earth-shattering CDs I've ever been privy to witness. They are my favorite band. Check out the last song, Blackwater Park. Wow.
You can get a taste of them in #mp3_metal or #mp3_death in dalnet. type @locator opeth blackwater park and you'll get plenty of results.
Caution - very harsh grunting vocals. May take some time to get used to, but their musicianship is absolutely brilliant.
Berto
People who argue about grammar and spelling are funny. Keep up the good work guys!
...For me to poop on!
Lesbian Nazi Hookers Abducted by UFOs and Forced Into Weight Loss Programs - -all next week on Town Talk.
Overall an excellent test done on a wide range of songs and codecs. Though, I would have liked to have seen how they all compare to the ATRAC codec used in minidisc recordings. Some say MD is dead thanks to MP3, however local stores in my area can't keep players and blanks in stock for long. Would have been nice to see how the two compare.
"I'm a leaf on the wind. Watch how I soar."
-Hoban Washburn
To thsoe of us who just want to listen to music on a PC, the newest greatest best algorithms are always good (mp3pro, oggs, wma8). But for many, the goal is to put that music on a MP3Player and listen to it anywhere. I'll summarize the support of these various codecs by MP3Players, as well as mention whether or not my MP3Player (RioVolt SP100) supports them.
MP3PRO -- little support on MP3Players. Not supported by RioVolt SP100.
Oggs -- little/no support on MP3players. Not supported by RioVolt SP100.
WMA8 -- little support on MP3players, though many support older WMA's. Not supported by RioVolt.
So, in summary, all of these new formats are completely useless to me on my MP3Player. The one option they present is if I want to encode something in two formats -- one for my computer, and another for the MP3Player.
Personally, I think more work should go into fractal endcoding, as most music has fractal patterns in it (especially Bach's music).
social sciences can never use experience to verify their statemen
The parent of all of this was certainly in err, but this still isn't a very good experiment. Look at the number of testers that were used. Most tests numbered in the 30s with respect to the number of subjects. While that number may be sufficient for small sample statistical tests, it is not a sufficient sample to test for such a normative value across the human population, such as judging music quality represents. Having achieved a small variance of opinion must not be determined to prove that the sample size was large enough to account for variance in opinion for the greater population, and while these tests are interesting, they are incomplete IMHO.
I believe all of the codecs have had claims of being equivalent to 128kb MP3 format audio at 64kb. The issue, though, is that they didn't include a baseline with 128kb MP3 compression in the test. Sure, you know how the various formats compare with each other at 64kb, but you can't really tell if the claims that they can perform with 128kb MP3s is accurate or not, assuming that anyone will even take the word of this to begin with.
-PainKilleR-[CE]
I am Karma Man, hear me Whore.
An honest double-blind listening test is extremely difficult to arrange, and there is no evidence whatsoever on such on the site.
This is how the test was conducted.
The test required access to a Windows machine (probably Win95 and up, didn't try with Win3.1) with a sound card. Users were required to download the ABC/HR "practice" Zip file, which includes the ABC/HR program, the Ogg Vorbis 1.0 command-line encoder and decoder, a LAME command-line encoder/decoder (I forget which version), a FLAC command-line decoder program, and a .flac sample file (the instrumental introduction to The Eagles' "New Kid in Town").
After unzipping this, the user had to run a batch file (encdec_foobar.bat) which un-FLACced the sample file, then encoded it with Ogg Vorbis and LAME, then decoded both of the resulting files back to .wav.
Then the user executed the ABC/HR program, which is a Win32 GUI application. After loading the sample into the application (pull-down menu and file selector dialog), the interface became a row of double-slider pairs. Below each slider was a "Play" button. Below each slider pair was a "Play Ref" button. Below that was a "Stop" button. There was a pair of sliders for each decoded sample -- so for the practice run, there were two pairs of sliders: one for file #1, and one for file #2. The user did not know which file was Ogg Vorbis, and which was LAME MP3.
The user then listened to the Reference file by clicking any of the "Play Ref" buttons. After hearing the Reference, the user could then click any of the normal "Play" buttons. The first task was to determine, for each pair of sliders, which one was the original and which one was the encoded file. Having determined that, the user used the slider (which went from 1.0 to 5.0 in increments of 0.1) to "score" the sample on the subjective quality of the result. There were also text labels on the slider: 4.0 was "perceptible but not annoying", 3.0 was "slightly annoying", 2.0 was "annoying" and 1.0 was "very annoying".
Finally, there was an ABX button, which launched a different window. In the ABX window, the user could select "Original", "Sample 1", or "Sample 2" for the "A" and "B" samples. Normal ABX testing proceeded from that point. (If you don't know what ABX is, go to pcabx.com.) I found that the ABX window sometimes helped me to focus on a specific sample so that I could find its flaws; armed with that knowledge, I was able to make a determination of which of the two sliders, right or left, was the encoded version.
Once a slider was pulled down from the default 5.0 position, another button became active above that slider. Clicking on it opened a new window with a text box, into which comments could be typed. When the user was finished with the test, the slider positions, the comments, and the ABX results (if any) were written to a plain text file (DOS CR/LF format), which was to be mailed to the test administrator. (Though, of course, you weren't supposed to mail the practice results.)
Now, that was just the practice session, which was a prerequisite for participation in the actual test. For the actual test, the process was similar, but differed in a few details.
The actual test samples included copyrighted, patented codecs for which there are no freely distributable decoders. Therefore, the WMA, AAC and MP3Pro samples were distributed as FLAC files, and decoded by the batch file. Since MP3 did not participate in the listening test, the LAME encoder was not used during the actual test. The Vorbis encoder, of course, was used twice: first with -q 0, and then with -b 64 --managed.
With 5 encodings per audio sample in the actual test, there were 5 pairs of active sliders instead of only 2 pairs. But otherwise, the actual test was exactly like the practice session.
(Personal note: I did 10 of the 12 samples, skipping the two classical ones. Out of 50 encoded versions of the 10 samples, there was only one case where I couldn't tell right from left -- "The Source", encoded with MP3Pro.)
Any self-respecting test should have known that the Yo-Yo Ma rendition of the Bach Cello Suites is utter drivel.
If they want to test with any authority, they should have worked with the Pablo Cassals version.
I believe he means -paying attention to it-. As in "my wife was talking to me, but as usual I wasn't paying attention to a word she said"
I have to agree. Not showing the rating of the control in the charts makes the chart pointless. It would be interesting to see how the user ratings for the control line up against the codecs.
It's possible that the data from the control showed that the listener preference had little to do with how the rated the codec reproduced the original.
* As is generally the case, my opinions do not reflect those of my employer.
I know that the thread is about compression formats, but hey - go to a bar/club with "LIVE" music, pay $10 at the door, have a drink and a good time.
Hopefully, the guys playing are getting a percent of the door, and they'll be happy to see you in the audience. Feel that bass!
Here's what I do: Bitty Browser & Andromeda
Check out Hydrogen Audio
Its pretty much the best audio discussion you can find on the 'net.
Yes - my 4/5 may have no connection to a 4/5 given to a different sample, or to a 4/5 given by a different person. In fact, he should have done the ranking to the individual results as well, which would further reduce the 'power'.
Unless we can have a set scale of non-transparency which we can guarantee is being adhered to (which is only the case in certain special circumstances), all we can keep is the ordinal information.
-- Help Digitise the Public Domain at DP.
By definition, the original was rated 5.0, or perfect. If the listener failed to rate the original 5.0 on any codec for a particular music sample, then all of the ratings for that listener on that sample were discarded.
This is a rather drastic way to screen, and certainly I might not have done this if there was less data and if more people had rated the original less than perfect. However, given the large amount of people who participated and the level of experience they had, I had that luxury.
A couple of things I personally would have like to have included but didn't for the sake of getting more reliable statistics: a 128 kbit/s mp3 anchor, and something like a 7 kHz lowpassed anchor. Just to kind of keep the ratings in perspective.
Oh, and to answer the criticism that the test doesn't represent the general population, that is quite true. The people who participated were not randomly selected off the street, but rather volunteered their services. I agree with the person who replied to this criticism, though, that I'm much more interested in the opinions of these motivated volunteers, who are much more likely to care about audio quality, than those of the average joe.
ff123
Because the story is more than 30 minutes old, so all the moderators have got bored and moved on. :)
-- Help Digitise the Public Domain at DP.
Apollo 440 is actually a prety damn good test bed for this stuff, especially (in my opinion) the Rapid Racer theme. Vocals, guitars, high frequency synth and ultra-low "theta-bass". Of course, you could have just used one of those Dynamic Frequancy test CDs, but what fun would that have been?
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Personally, I find the differences in bit rates and codecs obvious only when heard thru a good pair of speakers at a high volume so that distortion exposes the missing bits. As an audio engineer, I would always bet that even a professional could not tell a 256kbps MP3 from a MiniDisc playback at reasonable, personal (not loudspeakers or PA's) gain and EQ levels, and if they can differentiate it from CD playback it is only due to the compression that makes them sound like MiniDisc.
Anyway, they had listening booths set up which allowed for blind comparisons of mp3 and wma encoded audio (classical, rock, jazz ...). I tried it and as I stepped out of the booth an excited MS flak asked me what I thought of WMA audio. I told him that it sounded like a cow in a milk carton, watched him try to suppress his consternation and I walked away.
From time to time I have again tried listening to WMA encoded audio and it continues to sound to me like the cow isn't out of the box. Actually, one of the things that drives me nuts about both MS and Real audio is a high frequency phasing/envelope effect that I hear. To be fair to MS, I really don't think that true quality has ever been their objective; only sufficient quality so that Joe Sixpack will accept it, so that they can move on to the next stage of their total Internet/PC audio domination plan.
Does anyone know which encoder Creative's Playcenter 2/3 uses? I was curious as to whether it was their own proprietary encoder, or if they used a third-party one.
Running any car over a cliff will destroy it, regardless of safety systems. This is not a valid test of a car safety system.
I'm the stranger...posting to
This is an interesting and relatively well done test (although it appears that the listeners knew which format they were listening to, so it wasn't truly double-blind, and a anti-MS and pro-Ogg bias can't be ruled out).
However, some discussions seem to be focusing on this saying AAC is bad or WMA is bad, when really it refers to the particular implementations in codecs of those formats.
For example, the Apple MPEG-4 AAC-LC encoder was used for AAC. This is a Low Complexity version of the format. Also, the Apple encoder has a strange limitation where it automatically converts 44.1 stereo to 32 stereo at that data rate. This isn't required by the AAC format. Other AAC encoders yield MUCH better results, and beat MP3 Pro in double-blind testing. I haven't seen any double-blind comparisons between AAC and Ogg.
Also, the WMA8 encoder is due to be replaced by the backwards-compatible WMA9 in early September. Of course, there may well be improved versions of the other encoders by then as well.
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The statistical technique used to evaluate differences in each individual sample was a parametric method: Tukey's Honestly Significant Difference, using each listener as a "block." That is, the fact that different people use different parts of the rating scale is taken into account. The Tukey's method also takes into account the fact that multiple samples are being rated, not just two.
The statistical technique used to rank the codecs overall was a non-parametric method: A Friedman omnibus test to see if there was a difference at all anywhere in the experiment, followed by a non-parametric Fishers Least Significant Difference (also "blocked"). A non-parametric method means that ranking (first, second, third, etc.) was used instead of rating points (4.7, 3.5, 2.6, etc.).
The ranking method was used for the overall evaluation because ratings for one sample don't necessarily mix and match with ratings for another sample.
ff123
Nowadays I rip at 192 as standard, but still there's noticable artifacts on certain songs.
What we need is a better analysis of what exactly the weak points and strong points of these compression algorithms are. Perhaps some kind of a test involving compression noise with many different patterns and such. A test could be designed using noise and sine waves in different ways that would test all aspects of the compression. Then do an FFT on the result and compare to the original. This would yield results that are much more useful to the developers, rather than just saying which is best.
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All of whose base are belong to the what-now?
I'm sorry if this has been said before, but I couldn't find it:
Why didn't they also try playing the original, uncompressed music, to see how high it scored...?
RMN
~~~
Indeed, I was just replying in the style you'd initiated "I think he's full of s**t."
I use VBR, end up with files less than half the size, and it sounds good to me.
If you ever wonder why people don't like you
*cough* *splutter* And yes, I have a proud following of freaks too, but you'd expect that from someone who has done lots of trolling.
Indeed, I was just replying in the style you'd initiated "I think he's full of s**t."
Well, where we differ then is that he's my friend. I tell my friends when I think they're full of shit, and they don't get offended and I don't try to offend them. It goes from, "you're full of shit" to, "look, I'll show you" to, "damn, you're right" or "told ya so", or whatever, but it's a gateway to understanding. But when random people do much the same thing, it's, well, it's just bad etiquette. That's my opinion.
It's because you are passing your opinion on the quality of codecs/bitrates when I would claim that it is blatantly obvious that the lower bitrates sound crap.
If it were blatantly obvious, you wouldn't have people encoding at 128kbps and claiming they can't tell the difference. I can tell the difference btw 128 and 192, but I can't tell the difference btw 192 and 256. Or maybe I just haven't tried hard enough.
I'll spend some time tonight and see if I can figure out what all the fuss is about.
And yes, I have a proud following of freaks too
I saw that. Very amusing. I was actually referring to how you currently post at +0. People don't like to be talked down to, they like to be educated, and they like to be told what a big dick they have, not what a big dick they are. I'm not trying to be a dick here. Can't we all just get along? Or, barring that, agree to disagree?
Synergy is your friend
mmm the warm bass... only type of music i ever noticed a vinyl difference was for music with loud or enhanced bass range... it just seems "warmer" in vinyl.
;-)(
I post, therefore I am.
CAn'T CompreHend SARcaSm?
to quote:
Actually, this is a striking example of how recording techniques can ruin sound as well. Take a look at the Apollo 440 album - Gettin' High on Your Own Supply. A good mixture of guitars and electronics, right? Well, look at the frequency graph again. See how virtually every guitar frequency variation has been cut out: this music was recorded digitally, mostly using samples by the looks of it. The normal variations you'd associate with having guitars play live are all filtered out, and the graph goes back to the flat digital sound again.
Actually, this has little to do with digital vs. analog recording. The phenomenon you are refering to here is the overuse of compression (not data compression but audio compression) in dance music. Most recording engineers, especially in classical or jazz music, try to maximize the dynamic range of instruments, to mimic a live listening situation. Dance music is often hypercompressed, where the loud sounds are made quieter, then the overall mix amplified, to create a mix with a much higher overall loudness, but with the same peak levels. This kind of record will stand out in a mix as "punchier", getting a dance floor more worked up. It does however, severly blunt the natural sound of live instruments which comes from their dynamic range. But for a bunch of kids in a dark club high on pills dancing at 4am, this is highly desireable.If you were to make a two dimensional spectral analysis of a such sound recording with the time axis to the right, the frequency to the top and the amplitude as the color intensity of the point you would see a lot of wiggling lines at regular distances.
In the field of acoustics, that type of plot is generally called a "Spectrogram". See recent stories about Aphex Twin on Slashdot and/or Kuro5hin to learn more.
(BTW: this would make a great visualization plugin for xmms)
Winamp (AOL's proprietary freeware media player for Windows, coming soon for Linux) already has one: Nullsoft Tiny Fullscreen.
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In my view the whole test is bogus because we don't have figures for the original CD track.
By definition, the original track is a 5.00, because the scale measures perceptual similarity between two tracks on a 1.0 to 5.0 scale, and the testers were provided with a .flac (lossless compressed wav) version of the original CD to test against.
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Thanks for the info man, i understand how it works now.
=)
The answer would be once ripped, leave it ripped- but that isn't always feasable
Once ripped, leave it ripped, and stored on music CD-R. (Use music CD-R instead of data CD-R to make sure that the recording artist and songwriter get paid.)
few CD players will (yet) play compressed music.
Let me clarify my perception of your "few". Most newer CD players that also play DVDs (DVD video or DVD audio) will also decode play RCA's MPEG audio layer 3 ("MP3") format. In addition, a growing number of portable CD players can handle MP3 audio. But currently, the majority of CD players play only Red Book linear PCM audio.
Microsoft's media player will play a wma file WITH its RIAA trojan even if the wma has been misnamed to MP3.
Not if I've associated .mp3 to Winamp, where I've turned off the WMA plugin.
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>Basically, after the first listening, you have
>already instinctively 'decided' how the sample
>sounds.
If this were true, it would never have been possible to make conclusions with 95% certainty, and there would have been no correlation between the results of the listeners.
Both are false, and hence, your unsubtantiated claim.
>How many times were the tests repeated?
One test per listener per sample.
>Was there a control group?
There was a control group in the form of 'trusted' listeners also participating in the tests. The results were similar to the general ones.
>Also, there is no information on the sound
>system used and no measurements on its effect on
>the test.
If you had bothered to actually look, you would see that each tester used the system he generally uses to listen to music on his computer, whatever that was.
>My suspicion is, if the test were repeated, the
>result would be completely different.
There is nothing that supports this claim and past tests contradict it.
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GCP
the MP3Pro (not plain MP3) was also at 64kb/s.
Well duh. Read my post again.
Supposedly the MP3 version [of the Aphex Twin "Face" clip] obliterates [the image hidden in the spectrogram] because of lost frequencies.
I tried it with LAME at 64 kbps mono, and the face was still recognizable.
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It would not matter if ogg was capable of reproducing music at any bitrate, no body will care with a name like that.
It would not matter if mp3 was capable of reproducing music at any bitrate, no body will care with a name like that. Heck, the middle syllable sounds like something you do in the bathroom.
The name of a product can make or break its success in the consumer world
Coca-Cola sells, even though the first half of its name is the first half of "cocaine". Trust me, any name can be promoted by the entertainment media.
As it is now, [the name of Ogg technology is] based on two bits of extremely nerdy trivia
And "Motion Picture Experts Group Audio Layer 3" isn't?
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That's like calling streamed movies HTTP's or something...
No. It's like calling them ASFs or AVIs when the underlying codecs are DivX and MP3.
Will I retire or break 10K?
No integer only codec. This means no portable player support as they generally don't have a FPU.
Nerd: Derogatory term typically directed at anybody with a lower Slashdot ID than you.
Just to clear up any confision, I don't *own* a recording/sound engineer, I *am* one. :-)
...such as widely heard Hip-Hop music? I'm really surprised it didn't make to this comparision. Even more: I was really curious if I'm lame to store all my songs in mp3 format.
Don't forget that car audio plays only mp3, by the way.
Plain old sigh.