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."
I know you can do frequency analysis on the output of these various codecs. Just compare that to the average human auditory capacity and you can get an objective measurement of the merits of these various compression methods.
Because "human auditory capacity" is not fully understood. Sure we can give standard frequency response graph, but most of these codecs take advantage of psycho-accoustic hearing models -- where certain frequencies mask other frequencies in our perception. Since this is a developing field, objective listening tests could really help determine what's working and what's not.
Because "human auditory capacity" is not fully understood. Sure we can give standard frequency response graph, but most of these codecs take advantage of psycho-accoustic hearing models -- where certain frequencies mask other frequencies in our perception. Since this is a developing field, objective listening tests could really help determine what's working and what's not.
From my understanding of MP3 compression and others, the compression protocols take advantage of this frequency masking, so if humans can't hear it, it removes it. It also obviously takes into account frequency ranges of hearing.
As a side note, I think it might be neat to be able to compress 30-50% better based on your personal hearing characteristics, but it'd stink if you got old and had to not only wear a hearing aid, but also start collecting MP3's all over again.
Also what kind of codec bias could you possibly be referring to?
Apparently he doesn't realize that this is a double-blind test - meaning neither the listener nor the tester knows what codec is being presented at any given time.
I'm taking the test now (well, not right now, taking a break) and it's about as scientific as I think you could make a public test taken in the home. Yes, the samples get compressed and then put in easily accessible folders with proper file name extensions, but you never know what you're actually listening to when you're running the testing program. All you have is a source file for comparison, then two buttons marked "1" and "2", one of which is the source again, the other a randomized codec. You never know which of the two buttons is the uncompressed source and you also never know which codec you're hearing. The results are also encrypted, so it's not as if you can just go into the results files and look at what codecs you favor.
I suppose someone who's truly got the Ear of the Gods could listen to the samples outside of the testing program, pick various identifiable traits out of each, then listen for those traits in the testing program and vote up or down whatever codecs he or she chose, but that would be exceedingly difficult and more than a little time-consuming. I can't see how it would be worth it, especially as no single test result is going to skew the overall results to any significant degree.
This is the first time I've ever taken a test like this and I am honestly pretty shocked at how good all of these codecs sound. I am having a really hard time even deciding which is the compressed track most of the time, and I consider myself something of an audiophile. I'm even listening in a fairly controlled environment with a good pair of headphones, at a volume loud enough to hear any background noise clearly but below any clipping whatsoever. I will be surprised if any codec really does significantly better than the others consistently when we see the final test results.
That is a great idea in theory, however there is much debate on how psychoacoustics work, i.e. what information really "needs" to be there in music in order to be perceived.
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)
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.
by Anonymous Coward
on Thursday May 13 2004, @05:11PM (#9144958)
The purpose of a "perceptual" encoder such as MP3 is to remove the frequencies one cannot perceive. The frequency graph therefore need not be the same as the original and yet the encoded version may not be distiguishable from the original.
Also, a frequency plot tells us nothing about the phase or frequency distribution at certain times in the signal. I can make a sine sweep that would match exactly the spectrum of a pop song, but obviously would sound nothing like it.
There are ways of objectively measuring the performance of perceptual encoders, but frequency analysis isn't really one of them.
Frequency analysis only gets you part way there. For those who didn't look around at the articles (I'm not refering to you, of course; just some hypothetical/. reader), there are time domain audio effects that are not visible on FFT plots. An example of this is pre-echo. With pre-echo you get a n echo of an upcoming sound (like a drum beat) before the actual sound happens. This can happen when linear-phase FIR filters are used, but is also an artifact of some frequency domain encoder/decoder systems. The FFT is only part of the story.
The different formats don't simply limit the frequencies stored. A given compression format will change the sound in different ways depending on what input soundfile is. Some codecs perform well with some types of sounds, but poorly with others (for example, the compression your cell phone uses is good at speech but lousy at music).
Also, all frequencies aren't of equal importance to a our ears. Our hearing is best in the middle range (near where the important elements of speech are), and taper off above
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.
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.
Uh, if the sample is the same length, and the but rate is the same, won't the file size be the same as well? A 10 second sample at 128 Kb Per Second should be 1280Kb regardless of the format, no?
And, just FYI, MOST people, something like 95% of listeners cannot tell the difference between 128kbps sample and the original. I generally can't, even with decent headphones on.
I think that all you compression elitist snobs work for HD manufacturers, trying to get me to buy a 250GB drive to store the same amount of music as my 60GB will hold!
a given audio stream, at a given bitrate, for a given length of time, always has the same filesize. What else do you think bitrate measures?
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.
That kind of reasoning will be your downfall my friend. Should I have my webpages 100k a piece with 30k images all over the place, just because the majority of people have broadband? Should I insert obscene amounts of worthless features into my application, requiring it to have a 2000+ rated processor and buttload of ram, just because this is what the average home user has?
When you go above 128kbps, most formats become indistiguishable from the uncompressed sample. I mean hell, most people CAN NOT hear the
q6 is imperceptible from redbook in 19/20 samples for me, LAME --alt preset extreme (200-220 kbps VBR for most samples) is better at about 29/30 samples. This was from a double blind computer generated arangement using the same equipment and one listener with good hearing.
Not even if it's about average quality speakers? Not even if it's about some rather cheap speakers?
I can't say I hear much of a difference with modern codecs, and I own some average speakers. Maybe 128 kbps mp3 can sound bad (although that depends a lot on the kind of music), but that's an aging codec anyway. I think encoded files in the 192 - 256 kbps range is the best, and 128 kbps ogg's often acceptable, especially with the DFX plugin (or similar) for Winamp to compensate for shortcomings in compressed formats.
I'd definitely not call 128 kbps in modern codecs "disgusting". In ogg's I've found it to be roughly as 160-192 kbps mp3's and that's perfectfly fine for my ears.
Different codecs and implementations of those codecs may be optimized for different bitrates, so its important to test codecs at various target bitrates.
And how do you know what you are asserting? Have you done properly controlled listening tests with 128kbps encoding using a variety of codecs?
The fact is that for a lot of people, knowing the best codec at 128kbps is worth knowing because:
1) They are using portable devices where they are space constrained 2) They are using portable devices that may not have the perfect fidelity of a high-end sound system, but can go anywhere with them. 3) They are using their portable device in a somewhat noisy environment that overshadows any sound quality issues caused by a lower bitrate.
I must be deaf, I just did the test on a the kraftwerk sample file, and it took me a lot of relistening to finally pick out 3 out of 6 encoded files (although the first one - whatever it was - was fairly easy). The other 3 sounded exactly like the reference sample to me. This is using Sennheiser HD500 headphones and an Audigy ZX2 sound card.
Try doing the test, you might be surprised, or conversely if you're not surprised, you might contribute valuable information to the project.
While the sine wave's frequency is known exactly (within the resolution of your sampling frequency) the amplitude is not- you always have loss due to quantization noise. You may be thinking of the fact that the fourier transform will have only one harmonic and thus the quantization noise doesn't come into play.
Consider the signal to quantization noise rate (SQNR):
SQNR (dB) = 20log(Vsignal/Vquantization_noise)
With linear quantization, your quantization is evenly spaced and the noi
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.
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.
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.
Vorbis does variable bit rate and you set the quality you want. That way you don't waste lots of bits where they are not needed. My 4MB ogg file sounds as good or better than my little brother's 6MB mp3. The difference is more songs on my 256MB compact flash card. Yes, it's easy to play that music on my Zaurus, which cost about as much or less than DR
That was my first reaction, who uses 128. What I want is a blind test with experts and thousand dollar audio systems to find at what point the experts are no longer able to tell the difference between the compressed and uncompressed audio.
I use `lame --preset standard`, which ends up being VBR in with a max of 110-290, hovering mostly around 190-210 range. It's one of the reasons I don't use OGG, it doesn't have any preset's so I'm supposed to just decide on a good level myself. I'd rather use something th
by Anonymous Coward
on Thursday May 13 2004, @05:22PM (#9145079)
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.
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.
There is no proof that Vorbis is patent-free. "We didn't consult any implementation documents for patented algorithms" is not the same thing as "none of our algorithms are covered by a patent."
Of course, proving the patent-freeness of Vorbis requires searching every single patent with a fine-toothed comb, further indicating how messed-up the whole patent system is at this point.
I just have to wonder how many companies are waiting to pounce on the first major commercial user of Vorbis with a patent suit
Not possible. All you will get is a bunch of WAV-files, you have no way to tell which file belong to which codec.
Check the contents of the sampleXX.zip files; you actually get an mp3, an.ogg vorbis, an mp4 and 3 flacs. If you want to be biased either for or against mp3/oggvorbis/quicktime itunes AAC, you can.
A.wav file at 128kbps is going to sound absolutely awful. At 8 bits per sample (which sounds pretty bad no matter what), 128kbps gives you a sample rate of only 16khz, so any frequencies above 8khz will be lost. If you up the sample quality to 16 bit (CD quality), the sample rate goes down to 8khz (4khz frequencies).
And this is for monaural sound. If you want stereo, cut the sampling rate in half -- this might cut it for voice, but it won't work for anything else.
r3mix.net died because people actually did objective analysis of his recommended LAME settings and found they were crap. IIRC, the main guy behind it wasn't very accepting of criticism. Plus, he was a message board spammer [arstechnica.com].
The best replacement for r3mix.net in my opinion is HydrogenAudio [hydrogenaudio.org] . The forums are frequented by a lot of professionals, as well as developers of LAME, FLAC, Nero AAC, Musepack, Wavpack, and other codecs.
The r3mix tuning (--r3mix), while a small step forward, was inherently flawed because of his insistance on tuning based on pictures instead of acual listening tests. As a result, the --dm-presets were invented and improved by Dibrom (the HydrogenAudio founder) along with a multitude of testers. eventually those were included in LAME as the --alt-presets (and in the latest version they just replace the normal --presets). In short, Hydrogen Audio is THE place to go for this stuff now.
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.
So the whole goal is to find the system that compresses music the best in the smallest number of bits.
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
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
Listening test? (Score:5, Funny)
Ogg! (Score:4, Funny)
Now that that's out of the way, let the insightful comments begin.
Re:Ogg! (Score:5, Informative)
Parent
Re:Ogg! (Score:4, Funny)
Nope, it just doesn't have the same ring to it.
Plus, vorbisty just doesn't work.
Parent
Re:Ogg! (Score:4, Funny)
Now that has a ring to it!
Parent
Re:Ogg! (Score:4, Funny)
Vorb-Vorb Vorbbity Vorb Vorb.
Bissy Bis... ba bis bis bis.
Vorbbity Vorbbity va va vorb. bissity bis.
Parent
How about: (Score:3, Informative)
Of course, if that turns out to be inferior to any of the other formats, it would prove that something's wrong with the tests.
Re:How about: (Score:5, Funny)
Aflac? What does a silly duck have to do with sound compression?
Parent
Re:How about: (Score:3, Funny)
Re:Ogg! (Score:5, Insightful)
Parent
Re:Ogg! (Score:5, Funny)
I, for one, would welcome our new filter overlords.
Parent
Objective audio analysis (Score:2, Informative)
So uh, why is this necessary, exactly?
Re:Objective audio analysis (Score:5, Interesting)
Parent
Re:Objective audio analysis (Score:5, Interesting)
Because "human auditory capacity" is not fully understood. Sure we can give standard frequency response graph, but most of these codecs take advantage of psycho-accoustic hearing models -- where certain frequencies mask other frequencies in our perception. Since this is a developing field, objective listening tests could really help determine what's working and what's not.
From my understanding of MP3 compression and others, the compression protocols take advantage of this frequency masking, so if humans can't hear it, it removes it. It also obviously takes into account frequency ranges of hearing. As a side note, I think it might be neat to be able to compress 30-50% better based on your personal hearing characteristics, but it'd stink if you got old and had to not only wear a hearing aid, but also start collecting MP3's all over again.
Parent
Re:Objective audio analysis (Score:4, Interesting)
Apparently he doesn't realize that this is a double-blind test - meaning neither the listener nor the tester knows what codec is being presented at any given time.
I'm taking the test now (well, not right now, taking a break) and it's about as scientific as I think you could make a public test taken in the home. Yes, the samples get compressed and then put in easily accessible folders with proper file name extensions, but you never know what you're actually listening to when you're running the testing program. All you have is a source file for comparison, then two buttons marked "1" and "2", one of which is the source again, the other a randomized codec. You never know which of the two buttons is the uncompressed source and you also never know which codec you're hearing. The results are also encrypted, so it's not as if you can just go into the results files and look at what codecs you favor.
I suppose someone who's truly got the Ear of the Gods could listen to the samples outside of the testing program, pick various identifiable traits out of each, then listen for those traits in the testing program and vote up or down whatever codecs he or she chose, but that would be exceedingly difficult and more than a little time-consuming. I can't see how it would be worth it, especially as no single test result is going to skew the overall results to any significant degree.
This is the first time I've ever taken a test like this and I am honestly pretty shocked at how good all of these codecs sound. I am having a really hard time even deciding which is the compressed track most of the time, and I consider myself something of an audiophile. I'm even listening in a fairly controlled environment with a good pair of headphones, at a volume loud enough to hear any background noise clearly but below any clipping whatsoever. I will be surprised if any codec really does significantly better than the others consistently when we see the final test results.
Parent
Re:Objective audio analysis (Score:5, Insightful)
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)
Parent
Re:Objective audio analysis (Score:4, Insightful)
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...
Parent
Re:Objective audio analysis (Score:5, Informative)
Also, a frequency plot tells us nothing about the phase or frequency distribution at certain times in the signal. I can make a sine sweep that would match exactly the spectrum of a pop song, but obviously would sound nothing like it.
There are ways of objectively measuring the performance of perceptual encoders, but frequency analysis isn't really one of them.
Parent
Re:Objective audio analysis (Score:4, Informative)
Parent
Re:Objective audio analysis (Score:3, Informative)
Also, all frequencies aren't of equal importance to a our ears. Our hearing is best in the middle range (near where the important elements of speech are), and taper off above
No matter *what* (Score:2, Insightful)
Re:No matter *what* (Score:4, Insightful)
Uh, if the sample is the same length, and the but rate is the same, won't the file size be the same as well? A 10 second sample at 128 Kb Per Second should be 1280Kb regardless of the format, no?
And, just FYI, MOST people, something like 95% of listeners cannot tell the difference between 128kbps sample and the original. I generally can't, even with decent headphones on.
I think that all you compression elitist snobs work for HD manufacturers, trying to get me to buy a 250GB drive to store the same amount of music as my 60GB will hold!
Parent
Uh, file size *is* bitrate... (Score:5, Insightful)
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.
Parent
Re:Uh, file size *is* bitrate... (Score:3, Insightful)
When you go above 128kbps, most formats become indistiguishable from the uncompressed sample. I mean hell, most people CAN NOT hear the
Re:Uh, file size *is* bitrate... (Score:3, Interesting)
Re:No matter *what* (Score:5, Insightful)
Not even if it's about average quality speakers?
Not even if it's about some rather cheap speakers?
I can't say I hear much of a difference with modern codecs, and I own some average speakers. Maybe 128 kbps mp3 can sound bad (although that depends a lot on the kind of music), but that's an aging codec anyway. I think encoded files in the 192 - 256 kbps range is the best, and 128 kbps ogg's often acceptable, especially with the DFX plugin (or similar) for Winamp to compensate for shortcomings in compressed formats.
I'd definitely not call 128 kbps in modern codecs "disgusting". In ogg's I've found it to be roughly as 160-192 kbps mp3's and that's perfectfly fine for my ears.
Parent
Re:No matter *what* (Score:4, Informative)
Parent
Re:No matter *what* (Score:5, Insightful)
The fact is that for a lot of people, knowing the best codec at 128kbps is worth knowing because:
1) They are using portable devices where they are space constrained
2) They are using portable devices that may not have the perfect fidelity of a high-end sound system, but can go anywhere with them.
3) They are using their portable device in a somewhat noisy environment that overshadows any sound quality issues caused by a lower bitrate.
Parent
Re:No matter *what* (Score:4, Informative)
Try doing the test, you might be surprised, or conversely if you're not surprised, you might contribute valuable information to the project.
Parent
Re:No matter *what* (Score:3, Informative)
While the sine wave's frequency is known exactly (within the resolution of your sampling frequency) the amplitude is not- you always have loss due to quantization noise. You may be thinking of the fact that the fourier transform will have only one harmonic and thus the quantization noise doesn't come into play.
Consider the signal to quantization noise rate (SQNR):
SQNR (dB) = 20log(Vsignal/Vquantization_noise)
With linear quantization, your quantization is evenly spaced and the noi
Speakers (Score:3, Insightful)
Performance is only one more factor (Score:5, Insightful)
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.
What's the point of 128kbps? (Score:4, Insightful)
Re:What's the point of 128kbps? (Score:3, Interesting)
Same deal for MPEG-2 encoders, they all look great at 7 Mbit+/sec but the real test is 3-4 Mbit/sec.
VBR? (Score:3, Informative)
Vorbis does variable bit rate and you set the quality you want. That way you don't waste lots of bits where they are not needed. My 4MB ogg file sounds as good or better than my little brother's 6MB mp3. The difference is more songs on my 256MB compact flash card. Yes, it's easy to play that music on my Zaurus, which cost about as much or less than DR
Re:What's the point of 128kbps? (Score:3, Informative)
I use `lame --preset standard`, which ends up being VBR in with a max of 110-290, hovering mostly around 190-210 range. It's one of the reasons I don't use OGG, it doesn't have any preset's so I'm supposed to just decide on a good level myself. I'd rather use something th
Sound quality is in the speakers (Score:5, Insightful)
Too bad they didn't challenge Apple (Score:3, Insightful)
If they'd included both versions of iTunes/QuickTime in this test, perhaps they could have helped shame Apple into fixing what they broke.
Re:Honesty of responders (Score:5, Insightful)
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.
Parent
Re:Honesty of responders (Score:3, Interesting)
Of course, proving the patent-freeness of Vorbis requires searching every single patent with a fine-toothed comb, further indicating how messed-up the whole patent system is at this point.
I just have to wonder how many companies are waiting to pounce on the first major commercial user of Vorbis with a patent suit
Re:Honesty of responders (Score:4, Interesting)
Check the contents of the sampleXX.zip files; you actually get an mp3, an
Parent
Re:Okay... (Score:5, Informative)
The sad thing is that somebody went to the trouble of putting together a perfectly reasonable, logical post just to throw in a porn link. *sigh*
Parent
Re:The best 128kbps audio format (Score:3, Informative)
And this is for monaural sound. If you want stereo, cut the sampling rate in half -- this might cut it for voice, but it won't work for anything else.
you have no clue ... (Score:3, Informative)
The bit rate of
Re:you have no clue ... (Score:3, Funny)
>All you have to do is limit the length of the song to
good lord. There's TWO of them.
Re:What ever happened to r3mix.net? Any replacemen (Score:5, Insightful)
The best replacement for r3mix.net in my opinion is HydrogenAudio [hydrogenaudio.org] . The forums are frequented by a lot of professionals, as well as developers of LAME, FLAC, Nero AAC, Musepack, Wavpack, and other codecs.
Parent
Re:What ever happened to r3mix.net? Any replacemen (Score:5, Informative)
Parent
Re:A nice idea (Score:3, Insightful)
Premise of test is somwhat flawed (Score:4, Insightful)
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
Parent
Re:There is no satisfying audiophiles (Score:3, Insightful)
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