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AAC vs. OGG vs. MP3
asv108 writes "Yesterday, Apple unveiled their new music service claiming that the AAC format "combines sound quality that rivals CD." Here is a little comparison of lossy music codecs, comparing an Apple ripped AAC file with the commonly used MP3 codec and the increasingly popular OGG codec. Spectrum analysis was used to see which format did the best job of maintaining the shape of the original waveform." Wish they had WMAs in there too. And for the spoilage, it looks like OGG comes out on top.
Some decent quality properly blinded listening tests would be more interesting than a graph though.
When VHS established dominance of the video market, there were high barriers to change - your player and media were committed to that format.
There are far less barriers to change in the ripped audio format, although there will still be some inertia, but there is nothing* to stop ogg vorbis becoming the dominant format.
Where's my ogg pod then?
* apart from the silly name.
Humorous signatures are over-rated.
I agree that Ogg is a better format, better quality sound for similar bitrates to MP3, but until the portable devices I use, in-car CD/MP3 players, etc. accept the Ogg format as readily as they do MP3, then I (like most people) are stuck with the MP3 format. At least nowdays storage is cheap, so I whack everything to MP3 at a high bitrate.
Please read my Canon EOS tech blog at http://www.everyothershot.com
Nevertheless, I encode into flac now, as 1) it sounds much better than vbr mp3 or ogg, and 2) at 20-30MB per song, it really discourages people from downloading songs from me when I tell them how big they are.
how are you encoding your mp3s?
try lame with --alt-preset extreme
can you tell the difference then?
My portable HD music jukebox, and my car stereo, and tons of other devices out there ONLY play MP3s.
... ew. There's got to be a better way.
But any new music I buy through Apple is AAC encoded, in an m4p "protected" file.
So here's a purely technical question: What's the shortest path to convert these shiny new "protected" ACC files into plain MP3s so that I can take the music that I've just paid for and listen to it on my Archos MP3 Jukebox? I've already successfully gone from AACs to audio CD, and then re-ripped and re-encoded the album as MP3 but
And yes, I know Apple and Big Music and the RIAA and Homeland Security don't want me to be able to do this (easily, or maybe at all) but at this point I'd like to sidestep the politics and focus on a technological solution that works for me- a legit, paying user.
So: what's the closest we can get to "acc2mp3", or better yet "m4p2mp3"?
-Mark
How do you know it's ADAT rather than analog 1"? Probably is ADAT in alot of cases, but nevertheless, I recall Jobs' statement (for whatever it's worth) in which he claimed: (paraphrased) "Sometimes they sound better than CD's themselves"
I hate Grammar Nazi's
Beta-Max!
... hell my grandma even knows what it is.... that means Ogg is screwed!
Ogg = Too little, too late, overmatched and unknown to the masses. Also, too geeky. No hardware support to speak of. Walk down a street anywhere in the world and ask them what Ogg is, then ask them what MP3 is..... I guarantee you 1000 more people will know what a MP3 is compared to Ogg. It may be smaller, but in the age of 200 Gb harddrives for $200 size is no longer an issue.
MP3 = Widely known, was first on the scene, its everywhere, tons of hardware on the market, good quality, reasonable size
AAC = Already has an installed user base, sounds just as good as Ogg or MP3, plays nicely with the best known\most widely sold MP3 player on the market. Promising, but probably the lesser of the three unless this thing takes off.
You may not like what I have to say, but it is the truth.... and you all know it!
My personal experience with Ogg is that it takes forever to rip a CD using the format. I personally don't know why this is (perhaps just a problem with the software I was using?) but if it's going to take 20 minutes to rip three tracks on a 48x CD-ROM drive connected to a 1.8 (don't laugh, it's fast enough for piracy!) gig processor, then I might as well just rip to mp3 at 192 kbps. Storage is cheap as hell nowadays, and most people (myself included) don't need 40 gigs on their hard drive but somehow ended up with it.
HI, MY NAME IS ISAAC.
Ogg is a container. Vorbis is the audio you speak of.
.m4a with their AAC files. Because it's MPEG-4 Audio.
Oh, and as far as 3 letter extensions go, Apple use
I you have a really good system (probably anything over 3k nowadays) then it is not worth it to use any lossless compression.
In my system we can hear the difference between mp3 320 and wav files. That said, the difference is small and you have to be listening critically... so
it comes down to cost. If compression is 10% worse, and you spent 5k on a system, then using compression costs you $500 of system quality. $500 at $.90 per gb for a hdd can give me plenty of capacity.
Also, with WAV I know I won't have to re-rip my music when the next new compression algorythm comes out.
Of course for a portable with anything but highquality headsets it is unlikely you could tell the difference between a good compression and lossless...
How can this parent be +5 insightful? It is wrong and uninformative.
I worked with MPEG4 (AAC) and OGG a lot (for my phd. thesis) and spectral analysis IS very important. Although it is correct that it doesn't show precisely what information is left out because of what our hearing system doesn't register. However, these hearing curves and integration times are already known (although not the same for evry human) and most post-MP3 encoders do this rather correct. Most profit nowadays is in clever signal processing. The spectrum of a decoded signal shows almost all artifacts very well and is therefore something which helps a lot in showing artifacts in a coding scheme.
Of course listening test must also be done. They show that modern encoders make choices (not all our ears are the same, and so isn't all the music) which very often pays of in a certain test.
Theoratically AAC and OGG are rather similar, but AAC has a few nice extra's like the Temporal Noise Shaper. However in practice OGG seems good enough (unless MP3) and is free, while AAC is not that much better and unfree, so my choice is obvious.
I will wait for the OGG hack of the IPod, now it had a better processor.
MP3 this, OGG that, AAC somewhere in the middle... Sorry, I don't use any of the above. I encode all of my music into Musepack. At high bitrates, it's the best lossy audio codec, period. For more information on Musepack, see Case's Musepack Page</a>, or <a href="http://www.hydrogenaudio.org/index.php?act=S T&f=11&t=1927&">List of Recommended Musepack Settings</a>.
Musepack encoders and decoders are available for both Windows and Linux, with Winamp plugins available. The only real downside to Musepack is there is currently no hardware support. But having tried each of the codecs mentioned in this article as well as Musepack at the Quality 8 setting, Musepack is music to my ears each time.
"We are all in the gutter, but some of us are looking at the stars." - Oscar Wilde
And as all people who have taken advanced math knows: Sound can be described with equal precision in the time-domain and in the frequency-domain.
It's called a Fourier-transform.
And in the frequency-domain you still got phase, in case you wondered. It's covered by the use of imaginary-numbers.
So analysing the signal in the frequency-domain should uncover the same errors as an analysis in the time-domain, if it's extensive enough, that is.
I don't bother going into the theorys behind this, but google for Fourier-transforms and wise up :)
Not Buzzword 2.0 compliant. Please speak english.
Understandably, most of the discussion here is about the pros & cons of various compression formats. But the first thing that jumped out at me when I clicked on the apple.com link was:
"Preview any song for free, when you find a song you want, buy it for just 99... It's what music lovers have been waiting for: a music store with Apple's legendary ease of use, offering a hassle-free way to preview, buy and download music online quickly and easily."
FINALLY, a business model for downloading music that makes sense! (Now if only I could afford to switch to Apple products.)
While I'm not sure, I would say yes.
I noticed last night that the protected AAC files played both in the Finder's preview pane and in Quicktime 6.2 itself. I assume the actual AAC-Protected decoding is done in quicktime, and no modifications were made to the finder to allow it to explicitly play AAC-Protected files. This implies that any program that can use quicktime can also play protected AAC files.
I'd be suprised if may of the other mp3 players on the mac didn't already support playing via Quicktime, and by extension, playing AAC-Protected
And as you surely know: FFT stands for Fast Fourier Transform and is one special implementation which is not a complete Fourier Transform. I'll not go into boring details here, as you seem informed enough :)
As I know little specific about MDCT I will not go out on a troll raid either, but it's still a Cosine-based transform. Hence a Fourier-derivate work.
So my point still stands. Given a proper transform (I never mentioned FFT), you will keep phase information even in the frequency-domain, and a frequency-domain analysis will not be inferiour to a time-domain based analysis.
Not Buzzword 2.0 compliant. Please speak english.
Two points:
1) Apple does not explicitly mention how their Music Store songs are encoded (neither what the source is nor what encoder they are using)---they very well could be using a higher quality AAC encoder than what ships with QuickTime, which has reviewed poorly. There exist, it should be noted, other professional level encoders that have reviewed much better.
2) That being said, Apple released QuickTime 6.2 at the same time as iTunes 4 yesterday, and one of the headlining new features is an enhanced AAC encoder. It is entirely possible that Apple has addressed problems with their encoder, and perhaps the new version would stack up better in blind listening tests.
Of course, it would have been nice if Apple could step out of the Reality Distortion Field for ten seconds, and do the "Right Thing". They had to have known that AAC--because of current, community-reviewed blind listening tests--would be a controversial choice. Why they didn't undertake/commission prior subjective testing and why they haven't bravely taken their encoder to the "street" and up against OGG and MP3Pro, I don't know...if they had, we wouldn't be arguing about how crappy their encoder was, we'd be arguing subjective listening differences. Now, this potentially great new service will suffer from a 3 to 6 month "shake out" in the more discriminating audiophile community (the people who recognize that CD is better than cassette, and can hear that 128 CBR MP3 is NOT CD quality) because of the technical merits of the quality of the encoder. No new service needs such hesitancy to overcome, much less one from Apple. I predict that the stigma of the quality demon is going to be a major adoption speed bump for this service among the group most important to its widespread adoption--the audiophiles.
Once again Apple (read Steve Jobs) makes the mistaken assumption that just because they SAY their stuff is better, everybody should just accept that--it is a clear misread of their (new) market demographic, which is proving to be growing more and more into a Slashdot crowd. If they keep ignoring the fact that their fastest growing fanbase is a fairly technical, information hungry group, they will certainly lose them as fast as they gained them...if there is one thing I have learned in my years of being a Slashdotter is that we are a fiercely loyal, but not easily fooled community, and we certainly don't suffer fools gladly.
Scott
"Hokey religions and ancient weapons are no match for a good blaster at your side, kid."
At low bitrates, AAC is very weak, at 128kbps it was the worst of all:
Study
I was one of the 3000 participants, btw. And my ranking which I gave (blind, I did not know which sample was which) confirms pretty much the results, at 64kbps, AAC was unbearable, while ogg was not distinguishable (by me anyway) to the original.
The only test where AAC didn't fail miserably was the "expert test" with only 8 listeners.
OGG has beaten all other codecs consitently at all bitrates.
Let's see. Given the task of creating a codec de novo and the financial and political means to have access to the original source material rather than a version sent through a horribly non-linear sampling mechanism out of your control and beyond your specification, which would you choose?
I'm sure most Slashdot readers will be familiar with the Nyquist limit and understand the complete inability to represent information above the limit, but how many are familiar with the degradations that occur near the Nyquist limit when you have non-infinite signal lengths? This is why oversampling is so important. In general, if you have a signal at frequency f that you want to accurately capture, you should be sampling (by rule of thumb) at 5f or greater. If you sample at lower frequencies, the distortions in phase and amplitude are difficult to predict and statistically analyze as they tend to have uniform rather than Gaussian distributions.
So again, I re-pose the rhetorical question: given the task of creating a new codec rather than rewriting an old one, wouldn't you want to use the least-filtered signal possible as a source, especially when the extant filtering is non-linear, and be able to select by design which parts to encode and which parts to ignore? I sure would.
Put my fist through my alarm clock with its ding-dong death inside my ear. - The Blackjacks.
I bought about 10 songs from Apple's music service yesterday, and they all sound great. When I got home, I ripped Would? from Alice in Chains's Dirt and compared it to the 182kbps VBR MP3 I already had. The AAC sounded about the same as the MP3. It didn't sound worse, and I was running this through my iMac G4's audio system and then a pair of Polk bookshelf speakers I have on my desk (and a Pioneer receiver/amp). I'll stick with AAC, and I'll stick with the iTunes Music Store. For my money, it's a good deal.
Sure he's flamebait, but he's right. When I decided to rip all of my CDs and store them on my computer, I tried various formats. MP3, MP3pro, WMA, and yes OGG. In all honesty I could not hear the difference between any of them whether I played them via headphones or through my Sony STR-DE475.
Thus the choice was easy because only one factor remained: ubiquitousness.
Will it work with any portable player I buy, or will my hardware choices be limited?
Will I be able to share them with friends without having to explain how to play them?
Will it work with programs such as Nero without decoding the files to a different format first?
One format fit that criterion and it was MP3. Sure it's proprietary. But so is my car. I'm not going to stop using something that works merely because its proprietary. Computers are tools, not a religion!
If someone says he and his monkey have nothing to hide, they almost certainly do.
If you love free as in sunshine software, and pride yourself on using open protocols your allowed to : STOP COPYING MUSIC. If you want free music, accept your ripping people off, and do the whole job
It just seems to me that with all the self-praising of opensource slashdot does, it's shooting itself in the foot - haven't you seen any of the rocky films ; it's always the underdog who wins ; free software can only improve while people will admit it needs improving, and thats not going to happen with all the brown-noses on slashdot.
It's absolutely accurate, there's no question about it.
Here's a simple proof. Imagine you have a lossless compressor with a guaranteed compression ratio. Now, consider the set of all possible files with length N or shorter. (For the record, there are 2^(N+1)-1 files in this set.)
Now, using the hypothetical compressor, compress every file in this set. Because of the guaranteed compression ratio, all the output files are shorter than length N. But there are only 2^N-1 files that are that short (and 2^N-1 is smaller than 2^(N+1)-1), so it's impossible for the compressor to have a different output for each of its input files. There just aren't that many unique files that are that short. What this means is that somewhere along the line, you had two different files (call them A and B) and when you compressed them, they translated into the exact same thing as each other! Now, how is the decompressor going to take that same compressed data and produce file A in some cases but file B in others? Unless it's clairvoyant, it just can't know when to produce A and when to produce B. And that means you do not have lossless compression, because you'd lost the distinction between file A and B.
In my workflow, I want to keep a big bunch of high data rate files on the home server (about 140 GB of 320 Kbps MP3 files), and then recompress to more portable formats to carry around on the PowerBook or whatever. This used to work fine. I'd use the Import feature of iTunes, and would convert from the 320 Kbps master file to ~150 Kbps VBR MP3 files for the road. While the lower data rates wouldn't work on my home Paradigm speakers, they were fine for listening to on airplanes.
However, this doesn't seem to work in iTunes 4. I see the Import option, but all the MP3 files in my current library are grayed out. Is this operator error, or does this not work anymore? If not, what is the Import function for?
Obviously I'd like to switch to 128 Kbps AAC-LC for my mobile music. But heck, I'd live with being able to make my old MP3 files!
-Ben
My video compression blog
Are you sure that the problem isn't in the mastering engineers, not the CD format? Almost all pop music is dynamically compressed within an inch of its life to make it sould louder on cheap equipment. I am told that this is much less of a problem with classical music, but classical music also tends to have a much higher crest factor than pop, and is therefore more sensitive to compression as well.
The noise floor and dynamic range of a CD with a high quality DAC should be better than almost anybody's ears, if correctly mastered. DVD-Audio should be even better than CD, with multi-channel to boot, and also gives recording engineers a lot of headroom in the ultrasonic to avoid aliasing while using low order filters that are in principle somewhat gentler on the sound. SACD on the other hand is a travesty, superbly wasteful of bandwidth, while having less resolution and more noise in the highest octave of the audio range and much, much more noise in the ultrasonic, which is inaudiable, but can have negative effects on the audible spectrum because of effects in the tweeter.
Please note: the post said, "To do a true test [. . .]" It did not say, to tell the difference.
I think there was a typo there. I reckon he meant to say that you might not be able to hear the same difference on PC speakers. As the fidelity is less, that makes perfect sense.
My original post way up the chain was mainly because I've heard so many people compare an mp3 on their PC speakers/headphones through an on-board soundcard to a CD played on their HiFi. That's just bad science.
If you care that much about music, then why not just listen to CD's or pure WAV form? Why mess with lossy compression at all?
Because when it's done properly, the "lossy" issue is not a problem, as you will have already decided what your minimum requirements are. I use the r3mix mp3 encoder preset (site seems to be down, very odd), and I get great results through my AWE64 soundcard hooked up to a separates system.
The open-source cd -> mp3 ripper/encoder CDex has an encoder option to use this quality preset. Ideal.
The problem in your test is that if you know which file you're listening at, you're just not fair in your comparison and by listening several times, your brain just makes you hear stuff that is just not there.
A test was made where people would listen to two WAV file, one supposedely was an MP3 (that was expanded to a WAV). 25% of the people could hear a difference between the two WAV files where they were actually the same...
Write boring code, not shiny code!
The Russians have won. They have made the world a cesspool of distrust, greed, fear and hate.
Then, you have to do a blind test with all of them. You also need to use a variety of source material, because different genres of music compress better under some encoders.
I don't disagree with you, but I just wanted to throw in my own 2 cents worth of informal experimentation:
I recently discovered the sourceforge cdex ripping software, so I finally had a chance to rip all my music to the superior sounding ogg format instead of mp3. Before doing so, my wife and I ran a couple double blind tests with one another to see where the best encoding was.
The only pair of speakers I had to test this was a pair of old Yamaha YST-M7's. These are Yamaha branded $20 single driver computer speakers that came with some computer I bought a while ago. They are pretty bad speakers. For the test, I selected a reasonable genre swath of music:
Dixie Chicks "There's your Trouble"
Oingo Boingo "On the Outside"
Samuel Barber "Adagio for Strings"
W. A. Mozart "Queen of the Night's Vengeance Aria"
REM "Nightswimming"
Each piece was selected because of particular aspects of song such as use of strings, use of horns, or use of voice. Each song was tried in a variety of encodings in both ogg and mp3, constant and variable bit rate, with the original CD wav file thrown in amongst the samples. The mp3 encoder was Lame v 1.27 engine 3.92 Alpha 1 MMX, the ogg encoder was Ogg Vorbis DLL Encoder v 1.09 enging 1.05.
The results strongly disagreed with conventional wisdom. In every case, across genres, on these low end speakers, 320Kbps mp3's were the only ones that fooled our ears. Low bit rate ogg and mp3 recordings were different, but we didn't take time to notice which was better... they were both unquestionably inferior to the source material. Ogg's 350Kbps encoding was good, but inferior to the smaller 320Kbps mp3 files of the same work.
Reading some of the posts on this article, I am rather shocked how many people find sound reproduction to be anywhere between "very good" and "excellent" on mid end equipment listening to 192Kbps encoded audio.
After running this experiment, I ripped about 30 of my CDs to 320Kbps mp3's and noticed another benefit to CD quality rips: I could listen to the music longer without my ears feeling fatigued. I had always thought that it was pumping sound directly into my head from my headphones that caused my ears to become tired of the music. For whatever reason, it takes much longer now. Perhaps 3 or 4 hours compared to 1 to 1 1/2 before.
Education is a better safeguard of liberty than a standing army.
Edward Everett (1794 - 1865)
Taking the bass guitar as an example, depending on the mic, the pickups, the amp, and the cabinet, you're going to have a lot of different possibilities on the sound. I've recorded bass with a "woof" sound and with a "Seinfeld" sound, and everything in between. Regardless, if the HF driver was on the opposite side of the room in any of those cases, I'd be able to tell where the LF unit was.
From that LF unit, the waves (depending of course on the frequency) radiate in basically a circular pattern. Yes, I know there are lobes and a slight reduction at the rear portion of the cone and all that jazz. However, the location of the reproducing LF cabinet is easily located using psychoacoustic principles illustrated by the kunstkopf and the various implementations and understandings of the Haas effect.
The project would have probably been the predecessor. We went to a lot of cathedrals around Germany and one in Holland to record some really funky sounds using various prototypes of the kustkopf.
As for a reverberant chamber, I used the illustration of materials I did to make it more clear for the less knowledgeable people on the forum who wouldn't have a difficult time understanding the concept using something they can easily demostrate has a low absorbtion and transmission factor and a high reflection factor. As you're probably aware, sheetrock will reflect enough at 12k to produce the phenomenon I described (although not as well as glass *GRIN*).
These problems aren't specific to satellite systems, but all current sound reproduction systems. When you take the drivers and remove them from a single source point, you begin to introduce major timing issues which the average Joe can perceive. Look at the Tannoy web site and the Meyer web site about dual-concentric technologies. When you move the drivers away from each other, you introduce timing differences. I've illustrated this to friends and strangers in the local Circuit City or Best Buy. It's not hard to hear when you stop listening to the marketing hype from Bose. (BTW, "böse" in German means "evil"...just another reason to stay away from that company. heheheh)
If you were standing on a forward or downward firing sinlge driver cabinet, you would have basically an equal radiation all around you. However, you'd still be able to tell the cabinet was below you through means other than the fact your feet are vibrating. The studies leading up to and since the naming of the Haas effect will be able to explain to you what I mean.
If you take those same LF cabinets, put them 100 meters away on a giant turntable with you standing in the middle, you'll be able to locate it as it moves around you. You will be able to do the same at 10m and even 1m. So, although the sound coming from the cabinet is for sake of argument, omnidirectional, the source point can still be located. Your brain is still able to determine the location of it in the field around you.
The Haas effect I believe is the ruling factor here. I'd read up a bit on the Meyer site as John Meyer (along with his brilliant staff) has done some amazing studies in their anechoic chamber and in real life situations (Speech Intelligibility Papers) using systems like SIM II where you could acually measure the effect I'm trying to illustrate here.
Plant a tree in a developing country.