I have read everywhere that it is better than MP3, but will i be able to notice the difference?
I encode all my files to VBR -V0 and i don't know anything about AAC, what would be the equivalent encoding settings for AAC?
Can you people really notice that theyre is a reall difference in quality? I want to test it by myself (encoding always it max quality off course) but first i want to ask here.
Can I tell the difference between ~256kbps MP3 and ~256kbps AAC? No. Can you? Maybe. If you wish to know, you must (properly) test with your ears on your equipment.
Im going to make my first ABX test soon, but i wanted to know a general opinion about AAC behing better than MP3.
I read everywhere that "AAC is better than MP3" so that is why i ask this. I know this applies at low bitrates but at high bitrates too?
Nero AAC at q.65. behing the same as LAME MP3 V0, in terms of filez size... does it means that AAC will sound better because of the highter bitrate it produces at the same file size? (at least technically)
AAC is, technicaly, a bit more advanced codec than mp3.
As for the sound, in the testing of low bitrate samples, it came out as slightly better than mp3 - I think it was something around 96 kbit test. At high bit rates (160 kbit and omore), it is near impossible to differentiate the two, except on problem killer samples.
So, if you are using over 200 kbit encodings, it would have no advantage whatsoever. But, that depends on your hearing - do ABX tests, see what is the lowest bitrate where you can't tell the difference between original CD audio and mp3 (or aac) compressed audio. For me it's rather low, between 128 and 160 kbit - so I'm using around 192 kbit encoding, to be on safe side.
If you do hear a difference in a certain piece of music at these high bitrates, it might be worth posting ABX logs and short clips demonstrating the problem, because it likely means there's room for improvement in whatever encoder you used, rather than a shortcoming of one of the formats in general.
I did a quick test with foobar and im surprised i can not even hear any difference between q.35 AAC and a FLAC file! And .35 means about 100kbps, with MP3 im sure i would have heard a difference... i guess.
Anyway im pretty sure it deppends on the kind of music, i was just listening pop.
What is the best music genere to test music? Maybe clasical music?
I will test more when my new senheisser hearphones arrive, i read somewhere that with hearphones you ear more "sound bugs".
Anyway im pretty sure it deppends on the kind of music, i was just listening pop.
What is the best music genere to test music? Maybe clasical music?
Densely produced metal and electronica.
An example of a problem sample (specific for Nero AAC): mmasq.wav (http://www.hydrogenaudio.org/forums/index.php?showtopic=83246)
And BTW, according to the July 2011 AAC listening test (http://listening-tests.hydrogenaudio.org/igorc/aac-96-a/results.html), Apple AAC and FhG AAC perform better than Nero.
This is yet another "discussion" initiated solely to promote AAC not so subtly...
This type of posts usually start of sounding clueless but quickly reach definitive "conclusion" that 128kbp AAC (aka iTunes) is just like CD.
Never do they mention LAME, never do they mention VBR V2. No ABX logs, no technical details whatsoever, just pure admiration for AAC and Apple.
This is unscientific CRAP. Please try to ignore this.
This is yet another "discussion" initiated solely to promote AAC not so subtly...
Where did you get that idea? I think Bahamut2 would be justified if s/he were offended by that accusation.
This type of posts usually start of sounding clueless but quickly reach definitive "conclusion" that 128kbp AAC (aka iTunes) is just like CD.
Where has this been said? No one said it here, and I bet it has not been said anywhere else and you are just conjuring it up for the purpose of your trolltastic anti-AAC agenda.
Since you don’t seem to know, the iTunes store no longer sells any 128 kbps AAC files; everything has been 256 kbps for a couple of years. And ABX tests are generally not taken as “definitive” proof of anything (especially not positive assertions), never mind that 128 kbps AAC files sound just like CD audio; the best that could be said about that is that some might, to some people, which might equally apply to MP3—and again, no one has said that here
anyway.
Never do they mention LAME, never do they mention VBR V2. No ABX logs, no technical details whatsoever, just pure admiration for AAC and Apple.
LAME is mentioned here. VBR settings are mentioned here. ABX tests are mentioned here, and tests of AAC vs. MP3 are plentiful if you would care to look rather than ignoring them for the purposes of your rant (not that any of us use them to claim that MP3 is rubbish and that everyone should switch to AAC, as you appear to be implying). As for Apple, it is not the only purveyor of AAC, in case anyone is in any doubt, and the Apple-branded encoder is mentioned here because it is one of the main encoders and not because of any inherent bias or superiority (it is noted as being effectively equivalent to that of FhG in the latest test).
Again, where are these mythical posts that you appear to be referring to? I see no AACist crusade here. The only person who appears to be on a crusade is you, and I don’t know why you have such a personal animosity towards AAC.
This is unscientific CRAP. Please try to ignore this.
Your posts about AAC seem to be of similar quality. I advise others to ignore them. I advise you to stop making unsubstantiated assertions.
I think for most people to be able to tell the difference between AAC and MP3 you'd have to compare low bitrate samples, in the range of 96-128kbps.
I encode all my files to VBR -V0 and i don't know anything about AAC, what would be the equivalent encoding settings for AAC?
Just in case you have not heard this before:
For the mp3 files you already have, you should keep them the way they are. Transcoding from a lossy format is an almost-universal no-no.
I used to use LAME v1 for all of the files I would put on my iPod. When they released the iTunes Plus preset a while back, I started to test that out and found that I couldn't tell the two apart and that I was getting better battery life on my iPod when using AAC files.
LAME v1 is in the 200 to 250 kbs VBR range IIRC and iTunes Plus is in the 256 kbs VBR. (That reminds me that they need to update the bitrate listing in iTunes when looking at a files info. It says 256 on all files no matter how simple or complex the track is.)
I haven't tested at lower settings as I have enough space on my iPod for the larger file sizes.
CK
What is the best music genere to test music? Maybe clasical music?
Densely produced metal and electronica.
As a matter of fact classical, of which I'm a big fan, seems to be not particularly troublesome for modern encoders. I found the most revealing instrument to be harpsichord, especially on solo quieter passages. I successfully ABXed it at LAME -v2 (some very specific killer tracks, of course), nevertheless failed at 256Kbps on both codecs.
Hi everyone,
This is my first post in this forum, and I "accidentally" found this thread on Google. However, I find your discussion quite interesting, that's why I'd like to join in.
First, I have a small recording studio for mostly private recordings. I do not use expensive equipment, but still I think that the setup is pretty decent (Yamaha HS80M, DYNAUDIO BM5A MKII, AKG K 601 and AKG K 701). So much for my equipment - I'd like to place emphasis on the reason why there are Lossy Audio Compression Codecs at all. Well, nomen est nomen, they make your audio tracks smaller. I know a lot of people with decent or expensive equipment who compress their audio with a Lossy Compression Codec using a very high to highest quality setting (e.g. LAME MP3 -V0 or Nero AAC -q > 0.8). In my honest opinion, this doesn't make any sense.
Let me explain: Every Lossy Compression Codec makes use psychoacoustic models to omit parts of a sample, which are irrelevant to the human ear (I do not explain this any further, as most of you should know that). The downside is: There are so called "problem samples" (as already mentioned in another post above) - and I'm talking about samples and not necessarely entire tracks - which cause the compression algorithm to fail. Failing in compressing a problem sample is, most importantly, less a matter of the bitrate or the quality you are using, but more a matter of the algorithm itself. So, for example, if a LAME MP3 -V2 compression fails to return a clean sample of such a "problem sample", it is most likely that it will also fail at -V0. Same applies to other Lossy Compression Codecs.
The question is: Does your audio track contain such problem samples? I can't know, but in a well produced and mastered mainstream audio track, problem samples occur very rarely. I encourage you to give LAME MP -V3 or even -V4 (or equivalent "medium" settings with other codecs) a try at ABX (but please, do it correctly). The reason behind those kinds of compression is, as I've already mentioned, to significantly reduce the size of your audio tracks. If you encoded a track with a lot of problem samples at -V0 (or equivalent in other codecs), you'd be busted and could have used a lossless compression instead, since the difference in size wouldn't be that big anymore.
Back to the topic itself: The Nero AAC or simply AAC codecs are very powerful on lower bitrates compared to MP3. Any 96 kbps MP3 almost always sound horrible, while AAC-files with a similar size does very well. If you've got plenty of space (and battery on portable devices), it doesn't make any significant differences whether you use LAME or any kind of AAC, as long as the settings are decent.
Have a nice day!
Any 96 kbps MP3 almost always sound horrible
I feel more info would be needed to support that claim. In my experience, as long as the input is sufficiently lowpassed, 96 kbps MP3 as produced by FhG or LAME sounds great. No high end, of course, but rarely (if ever?) any artifacts. So...careful with generalizations
I found the most revealing instrument to be harpsichord
That appears to be a rather common observation.
Any 96 kbps MP3 almost always sound horrible
Sources and test results for this claim?
I apologise for that claim. Since English is not my native language, I poorly expressed that sentence. What I wanted to say is: If a 96 kbps MP3 sounded badly, a similar encoded AAC-sample might do a lot better. I will correct the original sentence above - it is wrong. And no, I do not have any proof.
Supplement
Great, I can't edit former replies.
Any 96 kbps MP3 almost always sound horrible
Sources and test results for this claim?
First of all he should give an operative definition of "horrible", one to test against.
I found the most revealing instrument to be harpsichord
That appears to be a rather common observation.
I think it's because of the rather "metallic" sound and sharp transients of this instrument, which is difficult to setup and record, in the first place.
Let me tell you an anecdote: I was listening to the Trevor Pinnock recoding of Bach Partitas (a GREAT and well recorded performance, which I own since its first release on LP) at 256K AAC, on headphones and quite loud, looking for artifacts. On a quiet passage I heard a very feeble, strange ringing sound, quite high in pitch, which surely didn't belong to the original harpsichord notes: here we go! I thought, and went for the lossless version, to possibly arrange an ABX set. But... what a surprise: the "alien" sound was there too! Turned it up a little more and... guess what? There was actually a bird, singing in the far distant background!
If someone has this recording and wants to try, they are more evident in the Sarabande from the fourth Partita: once you know he is there you can't miss him!
I was listening to the Trevor Pinnock recoding of Bach Partitas (a GREAT and well recorded performance, which I own since its first release on LP) at 256K AAC, on headphones and quite loud, looking for artifacts. On a quiet passage I heard a very feeble, strange ringing sound, quite high in pitch, which surely didn't belong to the original harpsichord notes: here we go! I thought, and went for the lossless version, to possibly arrange an ABX set. But... what a surprise: the "alien" sound was there too! Turned it up a little more and... guess what? There was actually a bird, singing in the far distant background!
Haha. I've heard similar at low volumes too, only that they become much clearer when I turn
off.
LAME v1 is in the 200 to 250 kbs VBR range IIRC and iTunes Plus is in the 256 kbs VBR. (That reminds me that they need to update the bitrate listing in iTunes when looking at a files info. It says 256 on all files no matter how simple or complex the track is.)
iTunes 'VBR' used to be constrained VBR at least in the older versions, even though the QuickTime engine is capable of encoding true VBR. That is probably why all your files are reading 256kbps. I have not used the iTunes frontend in a while so I'm not sure if this is still the case. XLD for OS X uses the Quicktime encoder and can be set to produce true VBR, I'm not sure whether there are programs for Windows capable of the same thing using Quicktime.
I have heard people speculate Apple chose CVBR for iTunes because they were worried about users being confused about songs resulting in different bit rates even though the quality settings were the same, but I don't know how accurate that is. iTunes (on OS X at least) also used to have issues with calculating bit rates on true VBR AAC files but this may have been fixed since.
Edit: Just did a quick test with the latest iTunes using 256kbps VBR, the resulting file reads as 256kbps in iTunes and 269kbps in OS X Finder. XLD using constrained VBR produced a file that reads as 269kbps in both Finder and iTunes. I suspect iTunes is deliberately displaying the bit rate as 256kbps when the files are produced through iTunes conversion.
Edit 2: Someone posted this (http://tmkk.pv.land.to/qtaacenc) in another thread for Windows.
Anyway, if one uses XLD or any other QuickTime frontend, to produce AAC from his own lossless material, it's better to target for quality (range 0 ~ 127) instead of bitrate. Target quality of 110 roughly corresponds to 255Kbps VBR, but the real average bitrate of a track, which is the value both Finder and iTunes show, actually depends on source complexity.
Quick question.
Wikipedia says:
The MPEG-2 audio tests showed that AAC meets the requirements referred to as "transparent" for the ITU at 128 kbit/s for stereo, and 320 kbit/s for 5.1 audio.
Who and when were these tests conducted by/at? How did MP3 score at 128kbps?
The modern lossy bitrate is far too bloated. Comparisons for mp3 Vs AAC are difficult as the average bitrate is now 250 .. 300k . Several years back the motivation of AAC and others was to provide near transparency @ 128k and better handling for problem samples where mp3 192k wasn't cutting it. Other encoders used VBR to implement these improvements. An early example was the MPC encoder which gave great quality @ 160..200kbit
Things where really interesting back then on the lossy side of things - 2002 ~ 2006. The other thing is mp3 was always competitive at 192k if you don't count rare problem samples. Even then it could probably sound very acceptable and satisfy 90 something % of people . This is probably true even for the old CBR 192 encodings.
Where its at today you could have just stuck it out with 256 CBR mp3 ten yrs ago , ignored any audio lossy development since and still be competitive .
The modern lossy bitrate is far too bloated. Comparisons for mp3 Vs AAC are difficult as the average bitrate is now 250 .. 300k . S
But I think video games and other entertainment media are still using sub-standard 128-160kbps mp3 or oggs.
Yes. What I am trying to say is that at 250..300k you can use a lame encoder from the dark ages and still be fine.
Quick question.
Wikipedia says:
The MPEG-2 audio tests showed that AAC meets the requirements referred to as "transparent" for the ITU at 128 kbit/s for stereo, and 320 kbit/s for 5.1 audio.
Who and when were these tests conducted by/at? How did MP3 score at 128kbps?
http://mpeg.chiariglione.org/quality_tests.php (http://mpeg.chiariglione.org/quality_tests.php)
http://mpeg.chiariglione.org/working_docum...AAC_results.zip (http://mpeg.chiariglione.org/working_documents/mpeg-02/audio/AAC_results.zip)
http://mpeg.chiariglione.org/working_docum...ts_overview.zip (http://mpeg.chiariglione.org/working_documents/mpeg-02/audio/NBC_results_overview.zip)
In my opinion it's true. High quality AAC encoder(s) provide(s) high quality at 128 kbps for statistically average lsitener. Some people won't hear the difference already at lower bitrate than that, other will need to rise bitrate.
I read everywhere that "AAC is better than MP3" so that is why i ask this. I know this applies at low bitrates but at high bitrates too?
I'm sorry, but I can't help it, a car analogy popped into my mind (and such, as everybody knows, are the benchmark in ~computing...):
I read everywhere that "AAC is better than MP3" seems a bit like "higher octane fuel is better" ...yeah, sure, it kinda is, in how it allows for "more on the edge" things technically, potentially higher-performing engine designs.
But virtually any engine is tuned & works with particular, perfectly sufficient octane levels - so pouring into the tank a "better" (than specified for the engine) octane fuel won't really bring any relevant improvement in its operation (at least in a non-malfunctioning engine),
despite what many people think & say.(I guess hearing system is a rough equivalent of the engine here... hm, maybe this analogy didn't turn out entirely horrible after all - and now that I think about it, there's plenty of audiophile-like snake oil salesmen in automotive field, too)
Now, on to make my post less silly...
I did a quick test with foobar and im surprised i can not even hear any difference between q.35 AAC and a FLAC file! And .35 means about 100kbps, with MP3 im sure i would have heard a difference... i guess.
[...]
I will test more when my new senheisser hearphones arrive, i read somewhere that with hearphones you ear more "sound bugs".
Such surprises tend to happen quite often, when people do a proper ABX test ...and I think you really should also do an ABX of LAME encodes at similarly low bitrates (~2 lower than your usual encodes); the results might very well surprise you similarly.
(BTW, ~training yourself to notice particular kinds of "sound bugs" also helps in hearing them - but IMHO one should really reconsider the utility of essentially
trying to notice them, if the idea is supposedly
listening to music)
[...] do ABX tests, see what is the lowest bitrate where you can't tell the difference between original CD audio and mp3 (or aac) compressed audio. For me it's rather low, between 128 and 160 kbit - so I'm using around 192 kbit encoding, to be on safe side.
Is that really "rather low"? Per the MPEG tests linked nearby (for just one example), it would seem close to typical, at worst.
the iTunes store no longer sells any 128 kbps AAC files; everything has been 256 kbps for a couple of years
I don't think that's quite as universal? Quickly searching for some confirmation brings up http://support.apple.com/kb/ht1711 (http://support.apple.com/kb/ht1711) "Why is iTunes Plus format available for certain music, but not all?" (plus I believe it's a bit country-specific, also in how some areas still get DRM'd music)
I have heard people speculate Apple chose CVBR for iTunes because they were worried about users being confused about songs resulting in different bit rates even though the quality settings were the same, but I don't know how accurate that is. iTunes (on OS X at least) also used to have issues with calculating bit rates on true VBR AAC files but this may have been fixed since.
Edit: Just did a quick test with the latest iTunes using 256kbps VBR, the resulting file reads as 256kbps in iTunes and 269kbps in OS X Finder. XLD using constrained VBR produced a file that reads as 269kbps in both Finder and iTunes. I suspect iTunes is deliberately displaying the bit rate as 256kbps when the files are produced through iTunes conversion.
It would certainly seem more elegant if iTunes and iPods simply displayed, from the start, some "quality preset expressed as nominal bitrate" tag - while the actual bitrate (essentially hidden from the user, normally) were allowed to float in a true VBR fashion.
But maybe they weren't so confident in the fully VBR mode, at the beginning? (I suppose CBR, maybe also ABR, is somewhat more straightforward to implement properly at the beginning / any codec dev here would like to shine a light on it?)
The modern lossy bitrate is far too bloated.
[...]
The other thing is mp3 was always competitive at 192k if you don't count rare problem samples. Even then it could probably sound very acceptable and satisfy 90 something % of people . This is probably true even for the old CBR 192 encodings.
Where its at today you could have just stuck it out with 256 CBR mp3 ten yrs ago , ignored any audio lossy development since and still be competitive .
I'd go further: since for many listeners, in many parts of the world, p2p (certainly with quite "random" quality - heck, how many are transcodes from other lossy files?), Youtube (etc., similarly random) videos, or low-bitrate streams are a major medium of music...
And people seem to be generally happy with those. Likewise with FM radio, or non-plus DAB in areas using "too low" MP
2 (two!) bitrates (or at least, research suggests they're fine (http://www.digitalspy.co.uk/media/news/a45367/ofcom-reveals-dab-sound-quality-opinions.html) - in fact, the complaints seem to be at least partly 'audiophile' in nature)
Overall, IMHO, the main utility of AAC (particularly as HE-AAC v2; generally, any of the "more advanced" codecs) lies in being able to cram more music, in perfectly acceptable quality, into portable players and mobile phones (or the just mentioned radio streams - but then, many stations don't seem to bother...) - after all, as HA tests show, we're getting where 96 kbps is already quite decent, and it (and lower) will only improve.
OTOH, it looks like space constraints might become moot, even on the cheapest of devices, in relatively near future... (well, maybe even then it will be still "which format and bitrate gives best battery life on this particular player?" - energy storage doesn't improve nearly so fast; likewise bandwidths, for large part of human population)
iTunes Store is 256 AAC with no DRM everywhere no more 128 with DRM.
Does anyone know what does apple mean by the 0-127 numbers for their TVBR? Other than just "higher the better". Like what do those numbers literally correlate to in the technical sense?
Why do you think that it has some hidden meaning?
[...] do ABX tests, see what is the lowest bitrate where you can't tell the difference between original CD audio and mp3 (or aac) compressed audio. For me it's rather low, between 128 and 160 kbit - so I'm using around 192 kbit encoding, to be on safe side.
Is that really "rather low"? Per the MPEG tests linked nearby (for just one example), it would seem close to typical, at worst.
For me it is - considering that most people choose quite higher bitrates for encoding their music.
the answer is yes.
I moved years ago and I haven't regretted it.
Also, the AAC codec is under active development by apple, while LAME is pretty much glacial in its development.
Also, the AAC codec is under active development by apple, while LAME is pretty much glacial in its development.
Latest LAME release: v3.99 (October 2011)
Not sure why you think that. Lame is quite active:
http://lame.cvs.sourceforge.net/viewvc/lam...ml/history.html (http://lame.cvs.sourceforge.net/viewvc/lame/lame/doc/html/history.html)
most of it is for stupid stuff like tagging files or minor bug fixes. Not many quality improvements.
So if what you really meant is that Lame is active, but that the Apple encoder is more active, I'm curious how you know this. Got a link to the Apple encoder changelog?
3.99 fills bitrate gap between the V0 and CBR 320 kbps . And it's useful.
But last time me and halb27 have discussion whether there were any real quality improvement while keeping the same bitrate since LAME 3.97. http://www.hydrogenaudio.org/forums/index....mp;#entry779727 (http://www.hydrogenaudio.org/forums/index.php?showtopic=92413&st=50&p=779727&#entry779727)
Shortly, no improvement since 3.97.
Anyway, if one uses XLD or any other QuickTime frontend, to produce AAC from his own lossless material, it's better to target for quality (range 0 ~ 127) instead of bitrate. Target quality of 110 roughly corresponds to 255Kbps VBR, but the real average bitrate of a track, which is the value both Finder and iTunes show, actually depends on source complexity.
While I agree that theoretically, a properly designed "true VBR" mode is better than ABR - the recent AAC listening tests here showed that for Apple's encoder, the "Constrained VBR" mode (which I suppose is like ABR) actually scored better.
I'm no expert at statistics or reading these listening test graphs, but it seemed that the Constrained VBR setting did score marginally better. (And both modes scored better than Nero AAC). My guess is that Apple has not tuned their true VBR mode as much as they have the constrained VBR.
the recent AAC listening tests here showed that for Apple's encoder, the "Constrained VBR" mode (which I suppose is like ABR) actually scored better.
Sorry, that is
not what the test showed. The average scores alone don't tell the whole story. Please read the text above (and below) the result plots on the test page (http://listening-tests.hydrogenaudio.org/igorc/aac-96-a/results.html).
I'm no expert at statistics or reading these listening test graphs, but it seemed that the Constrained VBR setting did score marginally better. (And both modes scored better than Nero AAC). My guess is that Apple has not tuned their true VBR mode as much as they have the constrained VBR.
Again, I disagree. True VBR averaged 6-7 kbps lower in bit-rate than constrained VBR while
not performing significantly worse in terms of quality. So it seems very well tuned. (That said, on the items tested, TVBR is tied to the Fraunhofer encoder whereas CVBR is significantly better than it).
Chris
I found the most revealing instrument to be harpsichord
That appears to be a rather common observation.
This is due to the fact, that the harpsichord produces tones with very high harmonics, that are very difficult to encode properly by means of compression. Compressors encode wave harmonics. A sine wave has very few of them (starting from 1), and square waves have the most. Actually, since you get ringing at the vertical parts of a square wave, which actually is a type of singularity, the number of harmonics in a square wave is infinite (and you can only get so close to it).
The harpsichord produces waveforms with mostly triangular waves, which are one order of complexity above sine waves.
Theoretically, all digital signals are the most difficult to encode, along with random seeds, like white noise with no cut-off limitation.
most of it is for stupid stuff like tagging files or minor bug fixes. Not many quality improvements.
Quicktime 7.7 was security fixes only. Quicktime 7.6.6 has been around since March 2010, and that release was to address H264 and iMovie issues. The last version that affected AAC encoding was 7.6.0 I believe (Jan 2009)
most of it is for stupid stuff like tagging files or minor bug fixes. Not many quality improvements.
Quicktime 7.7 was security fixes only. Quicktime 7.6.6 has been around since March 2010, and that release was to address H264 and iMovie issues. The last version that affected AAC encoding was 7.6.0 I believe (Jan 2009)
Actually, they often update AAC encoder without any announce.
As far as I know, the most recent update was on QuickTime 7.7.1, CoreAudioToolbox 7.9.7.8.
most of it is for stupid stuff like tagging files or minor bug fixes. Not many quality improvements.
Quicktime 7.7 was security fixes only. Quicktime 7.6.6 has been around since March 2010, and that release was to address H264 and iMovie issues. The last version that affected AAC encoding was 7.6.0 I believe (Jan 2009)
Actually, they often update AAC encoder without any announce.
As far as I know, the most recent update was on QuickTime 7.7.1, CoreAudioToolbox 7.9.7.8.
I very much doubt 7.7.1 affected the AAC encoder given it wasn't even released on OS X (correct me if I'm wrong, it wasn't released for 10.6 at least) and was released on Windows to address a specific buffer overflow exploit that was fixed with a system update on OS X. The latest version for OS X 10.6 is 7.6.6. They do occasionally make undocumented changes to QuickTime but given they don't even bother to mention them in the version announcement I think these are usually very minor. I've been using QT AAC for years and the last major change I noticed was from 7.5.5 to 7.6.0 (significantly increased bit rates at -q 127) which was documented in the 7.6.0 release notes as 'AAC encoder improvements' or something like that. But the QT release notes are very vague at the best of times. In any case I don't think there's any reason to suggest that QT is any more actively developed than LAME (and the LAME changelog is much more explicit).
I very much doubt 7.7.1 affected the AAC encoder given it wasn't even released on OS X (correct me if I'm wrong, it wasn't released for 10.6 at least) and was released on Windows to address a specific buffer overflow exploit that was fixed with a system update on OS X.
Look at
http://www.hydrogenaudio.org/forums/index....showtopic=91484 (http://www.hydrogenaudio.org/forums/index.php?showtopic=91484)
and
http://www.hydrogenaudio.org/forums/index....showtopic=90678 (http://www.hydrogenaudio.org/forums/index.php?showtopic=90678)
They have been improving the AAC encoder.
Even if you didn't know the fact above, at least you can simply compare the resulting AAC bitstream, and you will find QT 7.7.1 result is not bit-identical with the older ones.
As a practical matter, if you notice artifacts that you think are correctable (bug or poor encoding), report them to apple tech support (google apple bug report). However, it is a minor hassle setting up an account to do so (btw, with microsoft, most teams use connect.microsoft.com).
The guys who work in quicktime were pretty good at replying to bug reports and they're at least "aware" of this site. In April 2011, they told me that an aac quality bug I reported had been fixed in a subsequent release. ("We believe this issue has been addressed in Mac OS X Lion Developer Preview Build 11A419.")
Personally, i'd say stick with what you know and trust. I used to be occasionally tempted to re-encode to Vorbis or, only possibly, AAC. What stopped me is... why bother? MP3 is, as Wikipedia states "the de facto standard" for digital music. Thanks to the LAME devs, MP3 continues to be the most popular format after all these years. Another thing to consider is that in a few years, all the patents will have expired on MP3. There's every chance that at that time, even your toaster will be "MP3 compatible"!
The only reason i would use AAC is if you live in the Apple ecosystem. Lame mp3's still have issues with gapless.
I presume you mean the problem when streaming from iTunes. How is it that Lame is at fault?
It's not an actual fault of Lame, the issue is to do with streaming over 'Home Sharing". Fact is Apple won't or can't fix it so i would avoid Lame Mp3 if one was to be using Apple devices. In terms of sound quality AAC & Lame MP3 are going to be pretty much identical outside of ABX testing.
The modern lossy bitrate is far too bloated. Comparisons for mp3 Vs AAC are difficult as the average bitrate is now 250 .. 300k . Several years back the motivation of AAC and others was to provide near transparency @ 128k and better handling for problem samples where mp3 192k wasn't cutting it. Other encoders used VBR to implement these improvements. An early example was the MPC encoder which gave great quality @ 160..200kbit
Things where really interesting back then on the lossy side of things - 2002 ~ 2006. The other thing is mp3 was always competitive at 192k if you don't count rare problem samples. Even then it could probably sound very acceptable and satisfy 90 something % of people . This is probably true even for the old CBR 192 encodings.
Where its at today you could have just stuck it out with 256 CBR mp3 ten yrs ago , ignored any audio lossy development since and still be competitive .
To me QT AAC (-V 91 -q 2) sounds more compressed than 3.99.5 LAME (-V 2, -M j -q 0); my ABX between wav and qt aac below : -
foo_abx 1.3.4 report
foobar2000 v1.2.6
2013/06/14 20:47:35
File A: C:\Users\Apro\Desktop\3TalkAboutLove.wav
File B: C:\Users\Apro\Desktop\3TalkAboutLove.mp4
20:47:35 : Test started.
20:48:09 : 01/01 50.0%
20:48:53 : 02/02 25.0%
20:49:25 : 03/03 12.5%
20:50:06 : 03/04 31.3%
20:50:43 : 04/05 18.8%
20:51:16 : 05/06 10.9%
20:51:33 : 06/07 6.3%
20:53:03 : 06/08 14.5%
20:53:55 : 07/09 9.0%
20:55:33 : 08/10 5.5%
20:55:58 : 09/11 3.3%
20:56:12 : 10/12 1.9%
20:56:29 : 11/13 1.1%
21:05:47 : Test finished.
----------
Total: 11/13 (1.1%)
To me QT AAC (-V 91 -q 2) sounds more compressed than 3.99.5 LAME (-V 2, -M j -q 0)
You should remove the "-M j" and "-q 0" from the command line, as the defaults in LAME are tuned for best performance. Besides, the -q setting is irrelevant for VBR encoding. So, just use "V2".
That being said, either something's wrong when decoding AAC on your equipment and that's the difference you're hearing, or you have pretty privileged hearing. At that quality the result should be transparent in most situations. Perhaps you could post the sample you used so the rest of us can test as well. Make sure it's <= 30 seconds long.
It doesn't matter if -mj and -q0 are there. They do absolutely nothing when using VBR.
It doesn't matter if -mj and -q0 are there. They do absolutely nothing when using VBR.
So why include them?
Edit: I know I said "should remove" when in fact there's no problem in leaving them there. So, point taken.
It should be specified that “-M” is not a valid switch. So, not only would the actual switch be pointless to include (or yes, technically, remove) here, but in fact, the incorrect version used by aprofromindia would never do anything because it’s not a recognised option.