Hi everyone.
I've been following these forums for quite a while, but today I decided to register and post my experience.
After reading on forums I decided that AAC 128 VBR would give me transparent results, so I re-ripped everything.
But today I ABX'ed Rammstein - Sonne and got 20/20. I'm so mad, I need to re-rip everything again
And I'm only using Sennheiser HD439, cheap headphones, plugged into my cheap Acer laptop. This is insane. The cymbals are what give it away every time, and there's this "swoosh" sound in general, especially in the chorus.
foo_abx 1.3.4 report
foobar2000 v1.1.18
2012/11/26 15:44:29
File A: C:\Users\Carsi\Downloads\Rammstein FLAC\Mutter\Sonne.flac
File B: C:\Users\Carsi\Music\iTunes\iTunes Media\Music\Rammstein\Mutter\Sonne 1.m4a
15:44:29 : Test started.
15:46:29 : 01/01 50.0%
15:46:49 : 02/02 25.0%
15:47:13 : 03/03 12.5%
15:48:08 : 04/04 6.3%
15:48:40 : 05/05 3.1%
15:49:30 : 06/06 1.6%
15:50:35 : 07/07 0.8%
15:52:02 : 08/08 0.4%
15:52:21 : 09/09 0.2%
15:52:50 : 10/10 0.1%
15:53:13 : 11/11 0.0%
15:53:41 : 12/12 0.0%
15:53:53 : 13/13 0.0%
15:54:15 : 14/14 0.0%
15:54:41 : 15/15 0.0%
15:55:24 : 16/16 0.0%
15:55:39 : 17/17 0.0%
15:56:16 : 18/18 0.0%
15:56:36 : 19/19 0.0%
15:56:52 : 20/20 0.0%
15:57:11 : Test finished.
----------
Total: 20/20 (0.0%)
Please be more specific? How did you encode to AAC? Which encoder/software?
Thanks.
Please be more specific? How did you encode to AAC? Which encoder/software?
Thanks.
FLAC to ALAC in DBPoweramp and then to AAC 128 VBR in iTunes 10.7
If you still have the FLACs or ALACs, you don't need to re-rip, just mass transcode. It shouldn't take more than a few hours.
Perfect thank you. Do you have time to test more and would like to know where is your transparent spot? I can encode the file to different bitrates for you if you'd like me to. I'd really like to know where most people don't hear any difference between the original and the transcoded with AAC.
But today I ABX'ed Rammstein - Sonne and got 20/20. I'm so mad, I need to re-rip everything again
And I'm only using Sennheiser HD439, cheap headphones, plugged into my cheap Acer laptop. This is insane. The cymbals are what give it away every time, and there's this "swoosh" sound in general, especially in the chorus.
1) If you kept the lossless, you wouldn't have to rerip.
2) Why did you do your whole collection before testing the result?
3) Things that cheap headphones don't do well are different than things that encoders don't do well. $20 can get you phones with 20+ khz response and be plenty fine. AFAIK more money gets you: better bass, smoother freq response, better isolation from external noise, comfort. AFAIK, none of that is relevant to the usual lossy artifacts.
Note that you may be able to hear artifacts on cheap headphones, due to their deficiencies, that you will not hear with more expensive ones (same for speakers and other hardware).
Why don't you just convert the one problem song/album to a higher bitrate/quality? IMHO there is no point to be obsessed about having all files at the same quality/bitrate. The main focus of lossy encoding is
transparency, and you use any means to get there.
Note that you may be able to hear artifacts on cheap headphones, due to their deficiencies, that you will not hear with more expensive ones (same for speakers and other hardware).
That's still a deficiency of the codec (settings), which is (probably) adjustable by using higher settings. I don't think it's an good argument that "better" gear will mask certain artifacts. If the artifacts are there, it is not transparent with the currently used hardware, so you need to use different encoding settings to compensate for that. In the end this shows just how much lossy encoding settings are a subjective and equipment dependent choice.
I would re-rip/transcode the albums with VBR mode 5 of Winamp's AAC (~192 kbps) if you want to go safe, can't really imagine that you could still ABX it (I can't at least).
I don't have the FLACs anymore, yes I'm stupid I know. Anyway, I tried a few other songs, and it wasn't as bad as "Sonne". Guess I'll just wait till iTunes Match will be released in my country then I can upgrade them all to 256 VBR. I know that may be overkill, but at least I won't notice the artifacts as in 128 VBR.
What confuses me though is that if you look at articles and listening tests on the internet in the last 2 years, almost everyone come to the conclusion that 128 AAC VBR is 99% CD quality. That certainly isn't true for me, even with no audiophile equipment....
I may have been too optimistic about 128 AAC VBR.
I'll do some more tests and post logs if I get 20/20 again like in the Rammstein song.
I don't have the FLACs anymore, yes I'm stupid I know. Anyway, I tried a few other songs, and it wasn't as bad as "Sonne". Guess I'll just wait till iTunes Match will be released in my country then I can upgrade them all to 256 VBR. I know that may be overkill, but at least I won't notice the artifacts as in 128 VBR.
Correct me if I'm wrong, but if you're going to go up to 256 VBR in AAC, you might as well consider other lossy options at the same time. For example -V0 LAME MP3s are around that bitrate, and transparent (or close to it).
What confuses me though is that if you look at articles and listening tests on the internet in the last 2 years, almost everyone come to the conclusion that 128 AAC VBR is 99% CD quality. That certainly isn't true for me, even with no audiophile equipment....
In fact: no one in his right mind can assure you a lossy codec is 100% transparent for everyone and every source material on earth, whatever the bitrate, and I've never seen such a claim made on HA.
There are also chances that you could spot a specific killer sample to which you are highly sensible even at the highest bitrate allowed by that codec: have you made, for example, different tests with that same track at higher bitrates?
I would re-rip/transcode the albums with VBR mode 5 of Winamp's AAC (~192 kbps) if you want to go safe, can't really imagine that you could still ABX it (I can't at least).
You could even start with VBR 4 and check how close to transparency Sonne sounds to you. Winamp's VBR-4 bitrate will probably end up near 150 kbps on that song since it's a difficult one.
Chris
By the way: how old are you? Maybe instead of re-ripping, you could just let time pass and nature make his job...
Seriously, just for the record you could keep that sample you ABXed and test from time to time in the years to come. As far as I know, there are still no long term studies about perception of compression artifacts vs aging, are there?
FLAC to ALAC in DBPoweramp and then to AAC 128 VBR in iTunes 10.7
What software did you rip with (not that this will cause the artifacts). I dont understand why you converted from CD, to FLAC, to ALAC, to AAC, and then remove the originals. Seems illogical to me.
Hm. If I am allowed to post an ad I have no economic benefit from: Those who need big batch ripping jobs done can maybe head over to http://forum.dbpoweramp.com/showthread.php...eply-discounted (http://forum.dbpoweramp.com/showthread.php?27368-Kodak-KDK-1000-03-and-KDK-1000-04-robots-steeply-discounted) .
Myself I used a Sony XL1B 200-disc changer. A couple are available on eBay now.
But today I ABX'ed Rammstein - Sonne and got 20/20. I'm so mad, I need to re-rip everything again
It sounds like you found a *particularly* bad problem sample here. Methinks a lot of HA regulars might want to listen to it (hopefully leading to improved encoders in the future). Could you upload a 15-30 second clip of the track to the Uploads forum?
If you absolutely must have perfect transparency on every single sample you will ever encode, and you will be angry at finding the slightest ABXable difference, you have two options: you can test every single song you encode to find the bitrate at which it becomes transparent for you, or you can go with lossless. Nothing else will suffice.
Imagine if everybody had access to a database containing ABX trial results from every living human listener for every sample of music ever recorded up to the present day. Would you really insist on choosing a bitrate so high it'd never been ABX'd? But even if that was so high it was no significant savings over lossless, somebody might record a piece of music tomorrow that would be ABXable at that bitrate!
I don't think that's a rational position to take. Instead, you have to realize that with any lossy encoder you face a tradeoff between bitrate and the frequency and severity of artifacts.
The only "transparency guidelines" that are useful have nothing to do with claims that the encoder is absolutely always transparent at that bitrate for every conceivable sample, listener, and test setup. Rather, a useful guideline is a bitrate at which a supermajority of normal content is no longer ABXable for a supermajority of users and at which the remaining exceptions are unlikely to have any serious adverse affects on one's listening experience.
Different people have different priorities and thus different optimal points on the bitrate vs quality curve. But for listening purposes, unless you place practically zero value on space, it's unlikely that your optimum is above 160kbps for iTunes AAC.
Given that you got rid of your originals, your disappointment may be because you were been expecting a lossy format to serve as an archival format and not just a listening format. Even if you find a setting at which all your music is transparent to you, using a lossy format for audio archival is a somewhat bad idea given the availability of cheap multi-terabyte hard drives. (1TB= ~3000 hours of stereo FLACs.) Someday there will be superior formats, and devices most likely won't play AAC forever. Transcoding from one lossy format to another is quite likely to introduce unnecessary artifacts.
stuff
+1 vote for adding that to the lossy section of the wiki.
Someday there will be superior formats, and devices most likely won't play AAC forever.
Both wrong.
At the high bitrates you mentioned, i.e. >= 160 kbps stereo, (xHE-)AAC and Opus are so close to the theoretical maximum in compression you can get with reasonable encoding and decoding speed that any improved codec won't sound significantly better, except on some very few signals. I promise.
Do you know any modern optical disc drives which don't play audio CDs any more? So why should devices stop supporting AAC in our lifetime? IIRC, a few years from now the patents expire, so AAC decoders can be used free-of-charge then. At least that should be true for AAC-LC, which is all you need for >= 160 kbps.
Carsi, did we overload you with answers? Have you tried Winamp's AAC encoder? I'd be curious how it does on e.g. Sonne. You can rip to FLAC or WAV with any software if the free Winamp version doesn't rip to AAC (don't remember if it does) and then use the "Send to -> Format Converter" feature.
Chris
The future is probably moving towards more generally re-programmable devices like smartphones, smart tvs, media centers, DAPs with apps, and so on so in a sense I think "hardware" support probably isn't very important in the long term. Dumb hardware devices that can't run user supplied media apps will certainly exist, but probably not anywhere near the extent that they used to.
Both wrong.
Ridiculous.
Your statements about "theoretical maximum at that bitrate" are meaningless baloney. If you had actually read my post rather than just rushing in, you'd know I'm quite aware AAC is close to transparency at >160kbps. But Opus is already a superior format which can achieve the same quality at slightly lower bitrates. It's only a small incremental improvement over AAC for stereo music (unless you need low delay), but future work will result in future improvements, at some point between 12 and 20 years from now AAC will likely look as outdated as MP2 does today.
That MP2 is reasonably close to transparent at 256kbps and up doesn't mean that it's anything but an ancient outdated curiosity given the availability of vastly better codecs that achieve the same quality at substantially lower bitrates.
I don't see mainstream portable music players with built-in support for MP2 audio anymore, and neither iOS nor Android support MP2 by default. People aren't going to leave support for long-since outdated codecs in forever. This isn't a question of patents, it's a question of the continuing effort to maintain compatibility and prevent bitrot with code that no longer sees frequent use. I'm not suggesting that this will happen to AAC any time in the near future. But if you honestly don't think it will happen in your lifetime, either you're fooling yourself or you're reaching the end of the actuarial tables.
C.R.Helmrich, did iTunes upgrade to 256kbps just because people were whining? Not sarcasm, I'd just like to understand how crazy marketing is.
But Opus is already a superior format which can achieve the same quality at slightly lower bitrates. It's only a small incremental improvement over AAC for stereo music (unless you need low delay), but future work will result in future improvements
Still it need to be verificated. As for last personal test from Kamedo2 Opus is already better than AAC at 75 kbps and on par at 100 kbps. My findings are the same. Of course Opus has a lot of room for improvement.
But here I tend to agree with Chris. We probably will stick with MP3, (xHE)AAC, Vorbis and Opus for a long time.
There were a lot of effort for a new image lossy format. JPEG reigns the world.
There were a lot of effort for a new image lossy format. JPEG reigns the world.
And now we have PNG to complement FLAC nicely
As to the format discussion, which is slightly off-topic here:
- any lossless format can be replaced, at least, provided it isn't DRM'ed to the extent that it cannot be copied in the digital domain. It isn't unlikely that the fruity company might succeed with a new format to carry their miraculous hi-rez streams which will magically resolve the complex problems of 44.1/16 audio (except the ones along the real axis, which would have been addressed if I were dictator :-P )
- an MP3 is an MP3 – it may be encapsulated in a container, but it is unlikely that anyone will be able to improve over the MPEG stream until storage and bandwidth makes it uninteresting to anyone but the “because it is possible!” geeks. Which is, I guess, already.
C.R.Helmrich, did iTunes upgrade to 256kbps just because people were whining? Not sarcasm, I'd just like to understand how crazy marketing is.
I don't work for Apple, but I know they did listening tests just like Carsi did and found they needed to go to ~256 kbps stereo to be transparent even on the most difficult material they could find. Which was a great move in my opinion. It's plain obvious that 128 kbps AAC isn't transparent on several items. Neither is Opus, nor WMA, nor Vorbis, nor any future codec. The more relevant question is how close to transparency can you get at that bitrate on non-transparent items.
at some point between 12 and 20 years from now AAC will likely look as outdated as MP2 does today. etc.
Talking about meaningless baloney here... MP2 was designed in the mid 80s. Doesn't even use an MDCT yet. MP3 was designed around 1990 and clearly beat it. AAC was designed around the mid 90s and beat MP3, but less clearly. xHE-AAC and Opus at high bitrates are marginally better than AAC (
if at all, thanks for helping out, Igor) and designed almost 15 years after AAC. Do you really think codec improvements are linear? In what way can a new codec improve? Quality? Most unlikely, marginally at most. Speed/efficiency? Most unlikely, marginally at most. For the record, I'm working on the next generation of perceptual coding 50 hours a week. So don't tell me what I write is meaningless baloney.
Chris
But today I ABX'ed Rammstein - Sonne and got 20/20. I'm so mad, I need to re-rip everything again
It sounds like you found a *particularly* bad problem sample here. Methinks a lot of HA regulars might want to listen to it (hopefully leading to improved encoders in the future). Could you upload a 15-30 second clip of the track to the Uploads forum?
Sure, I could do that, but isn't that against the forum rules? After all it's copyrighted material... If not, then I'll upload straight away, I'd love to hear what people on this forum think about this sample.
But today I ABX'ed Rammstein - Sonne and got 20/20. I'm so mad, I need to re-rip everything again
It sounds like you found a *particularly* bad problem sample here. Methinks a lot of HA regulars might want to listen to it (hopefully leading to improved encoders in the future). Could you upload a 15-30 second clip of the track to the Uploads forum?
Sure, I could do that, but isn't that against the forum rules? After all it's copyrighted material... If not, then I'll upload straight away, I'd love to hear what people on this forum think about this sample.
Thirty seconds or less is generally accepted as falling under the "fair use" category, and so is allowed under forum rules.
Ok so I uploaded the sample in the uploads section:
http://www.hydrogenaudio.org/forums/index....showtopic=98111 (http://www.hydrogenaudio.org/forums/index.php?showtopic=98111)
Let me hear what you think.
For the record, I'm working on the next generation of perceptual coding 50 hours a week. So don't tell me what I write is meaningless baloney.
I'm quite completely aware of the historical facts, of your job, your ideological commitments, and your self-important bull. Trying to pull authority and telling people off just because "I'm with FhG, I'm soooo important" doesn't win you any points.
When you say "(xHE-)AAC and Opus are so close to the theoretical maximum in compression" that single statement is demonstrably meaningless baloney, and that's the only thing you said that I singled out as being simply baloney. No theory gives anything near objective convincing support to the claim that no future codec can achieve a markedly superior bitrate-quality curve on normal audio. Adding the qualifier "with reasonable encoding and decoding speed" doesn't help, since new algorithms, along with Moore's law, quickly bring many "unreasonable" methods and optimization problems into the realm of realistic possibility.
Again, back when MP2 was being put together, people made claims that perceptual entropy limits meant that nothing could ever do much better than MP2. Also, the kind of calculations done by a modern AAC encoder would have seemed well outside the limits of reasonable encoding speed in the late 80s.
No, I don't think codec improvements are linear. Nothing I said suggests that. Of course there's diminishing returns to effort. Nor am I saying that Opus or any other codec currently under development is going to be what obsoletes AAC. Anything which could possibly deliver that kind of an improvement will be substantially different from existing codecs and would be at best in extremely early experimental stages at this point, like Monty's chirp tracking Ghost work. (Heaven only knows whether that particular effort will lead, a decade or so down the road, to both a format which allows substantial bitrate/quality improvements and algorithms that can do the required encoder-side work with the required precision and speed, or whether it will prove to be a dead end. But it does seem to exemplify the kind of conceptual departure and exploration that may be required.)
The banter between jensend and C.R. has me curious - I wonder what a theoretical "best possible" lossy encoding algorithm would look like, when transparency and space efficiency are the only two factors of consideration (i.e. 100% transparency for all listeners for all samples in the smallest possible package is the only goal; encode and decode times, compatibility with current hardware/software, and all other factors are irrelevant, as it is assumed that future tech would catch up). It almost makes me want to dust off my programming skills.
And yes, I know "100% transparency for all listeners" is probably never achievable in a lossy codec. But this is just theoretical.
I'm quite completely aware of the historical facts, of your job, your ideological commitments, and your self-important bull. Trying to pull authority and telling people off just because "I'm with FhG, I'm soooo important" doesn't win you any points.
While philosophically speaking, arguments from authority are fallacious, Mr. Helmrich is simply backing up his assertions with his area of expertise. If you would kindly grace us with a reason we should trust your opinion over that of a legitimate professional involved in reserarch, that would be much appreciated.
As a moderator, I'd also appreciate it if the tone of this discussion became less abusive.
While we don't know where the next improvements in lossy audio are going to come from, the fact is that despite many different implementations, we seem to be unable to really push the boundaries much further than where they are today. Even the people pushing the boundaries are trying to tell you that. This doesn't preclude the possibility of a huge breakthrough, but I think such a huge breakthrough is unlikely any time soon.
And yes, I know "100% transparency for all listeners" is probably never achievable in a lossy codec.
I know it can't compete on bitrate, but has lossyWAV been reported to be non-transparent lately? At quality setting "standard" and above, that is.
If you would kindly grace us with a reason we should trust your opinion over that of a legitimate professional involved in reserarch, that would be much appreciated.
The reason is that this "legitimate professional" has made extraordinarily sweeping universal claims without any evidence to back them up.
In my first post in this thread I made two remarks that should be entirely uncontroversial: someday there will be superior formats, and devices most likely won't play AAC forever. His arrogant dismissive "Both wrong." means that he's claiming both that
there never will be any format superior to AAC and that
all future devices will play AAC until the heat death of the universe. There is absolutely no good reason to believe anyone who says those kinds of things, no matter how many hours per week they're being paid to do exactly what they are saying is impossible.
We can bicker about our predictions about the
magnitude of improvement future encoders will show
a given number of years from now, or about
how widespread backward compatibility with AAC will be
a given number of years from now. I continue to think the claim that it'll be longer than 20 years before there's either a) any substantial improvements over today's AAC encoders or b) any devices which are no longer compatible with AAC is entirely incongruous with both history and theory, and as far as his "authority" goes, I'll remind you of Clarke's first law (http://en.wikipedia.org/wiki/Clarke%27s_three_laws). But bickering about particular predictions is beside the point.
All that's required for my argument that using AAC for archival is non-optimal is that
someday someone might want to re-encode their audio with another lossy encoder because of either an improvement in the bitrate-quality curve or because they own a new device which no longer supports AAC. The claim that these are impossibilities is unreasonable, no matter who it's coming from.
I hate to interrupt the argument by going back on topic, but is that wav sample the encoded output, or the original?
Either way, can you upload a lossless copy of the encoded output or the original, whichever we don't already have..
I'm at work and don't have any tools here to really test with, but I'd like an original and a copy that suffers the problem he's describing to play with..
And yes, I know "100% transparency for all listeners" is probably never achievable in a lossy codec. But this is just theoretical.
I think the real problem here is that as today we rely on psychoacoustic models to develop lossy codecs and on listening tests as the only way to roughly measure output errors and feedback the developers. Those models are external, although good, approximations of the real listening process, so the codec development process has an internal cause of uncertainty which prevents from asserting full transparency.
Of course medical science is not engineering as physicians don't have access to the source code...
I wonder what a theoretical "best possible" lossy encoding algorithm would look like, when transparency and space efficiency are the only two factors of consideration (i.e. 100% transparency for all listeners for all samples in the smallest possible package is the only goal
The goal would likely be way more ambitious. For a “best possible” it should be impossible to simultaneously improve both sound quality and bitrate – not only at the threshold of transparency (if that is well-defined!), but for any (reasonable) bitrate below.
But part of the argument seems to be the codec/format, and that goes beyond the single encoding algorithm. Even when an encoder can be (significantly) “improved” in the above sense, it does not necessarily mean that you have to “improve” the format in order to carry the “improved” lossy signal. Like, current LAME is better than last-century encoders – meaning, it was not necessary to introduce AAC in order to get improvements of that order of magnitude. I would consider a format to close to optimal on the above parameters, if for any “best possible” benchmark it can carry an equivalent quality without requiring much extra bitrate.
I hate to interrupt the argument by going back on topic
Oh ... THAT was beyond the pale!
His arrogant dismissive "Both wrong."
I agree, "wrong" was inappropriate. Sorry about that, jensend. Please re-interpret it as "highly unlikely".
Regarding your statement that there's no theory to support my claim "theoretical maximum in compression": there's Shannon's rate-distortion theory (http://en.wikipedia.org/wiki/Rate%E2%80%93distortion_theory), where the maximum allowed distortion D is given by the theory of threshold of audibility, which in turn can be derived from the psychoacoustic theories of hearing threshold (http://en.wikipedia.org/wiki/Threshold_of_audibility) and of auditory masking (http://en.wikipedia.org/wiki/Auditory_masking) (both temporally and spectrally). In my experience, encodings that clearly aren't transparent clearly violate - i.e. introduce D which is above - the threshold of audibility at some point in time, and I also think the psychoacoustic theories have become quite solid. Your opinion may differ, of course.
yourlord, sorry, yes, back to topic. To me it looks like Carsi uploaded the original. Spectrum goes to 22 kHz and looks "original".
Chris
Let's get back to topic I uploaded the original. Anyone tested the sample yet?
Not yet, but you can save us some time by uploading the .m4a iTunes encode on which you heard the artifact(s).
Chris
Let's get back to topic I uploaded the original. Anyone tested the sample yet?
I've tested it with the recent master of Opus and the Winamp AAC encoder (both at 128 kbps), couldn't find any obvious flaws. I'm not too good at ABXing in general, though...
I tested it with Vorbis down to 96kbps yesterday on some ear buds and I couldn't tell them apart using a poor man's blind test..
I'll encode the original as aac this weekend and see if I can ABX on a decent set of headphones.
@OP: exactly which parameter you passed to AAC encoder? VBR quality target? CVBR bitrate?
Anyway, I encoded at 128 CVBR (130kpbs average) and this was my first, promising attempt to ABX:
ABX Test Completed: 2012-11-30 17:54:49 +0100
Number of tests performed: 10
Number of correct answers: 8
Percentage correct: 80%
File 1 = /Users/nessuno/extempora/Sonne_chorus.m4a
File 2 = /Users/nessuno/extempora/Sonne_chorus.wav
File placement was static.
n [A] [X] [B] Choice Score
1 [1] [2] [2] B 1/1
2 [1] [1] [2] A 2/2
3 [1] [1] [2] A 3/3
4 [1] [1] [2] B 3/4
5 [1] [2] [2] B 4/5
6 [1] [2] [2] B 5/6
7 [1] [1] [2] A 6/7
8 [1] [1] [2] A 7/8
9 [1] [2] [2] B 8/9
10 [1] [2] [2] A 8/10
--------------------------------------------------------------
But then made some other trials and my percentage worsened, about 70% or 60% so I was clearly guessing even the first time.
I haven't installed iTunes, but I use qaac. Encoded with foobar2000:
qaac 1.46, CoreAudioToolbox 7.9.7.9, AAC-LC Encoder, CVBR 128kbps, Quality 96
and real bitrate was 130 kbps.
This one was rather easy to ABX, at second time listening to the complete sample I spotted a problem with high-hat/cymbals (or something like that). Besides distortion with these instrument sounds I noticed a different stereo balance, but that could be caused by different level of distortion on left and right channel.
foo_abx 1.3.4 report
foobar2000 v1.1.18
2012/11/30 18:24:32
File A: C:\Users\Alexxander\Desktop\Sonne_chorus.wav
File B: C:\Users\Alexxander\Desktop\Sonne_chorus.m4a
18:24:32 : Test started.
18:25:37 : 01/01 50.0%
18:25:53 : 02/02 25.0%
18:26:04 : 03/03 12.5%
18:26:09 : 04/04 6.3%
18:26:24 : 05/05 3.1%
18:26:59 : 06/06 1.6%
18:27:16 : 07/07 0.8%
18:27:30 : 08/08 0.4%
18:27:51 : 09/09 0.2%
18:28:04 : 10/10 0.1%
18:28:08 : Test finished.
----------
Total: 10/10 (0.1%)
i can not abx sample converted with
afconvert -v -f "m4af" -s 3 in.aif out.m4a
(thats default quality, fully VBRish, 126 kbps average)
(possibly due to high intolerance to specific song/band.)
The iTunes 128kbps (High Quality) qaac setting is -a128 -q1 (ABR 128kbps Quality 1) as you can read here: https://github.com/nu774/qaac/wiki/Encoder-configuration (https://github.com/nu774/qaac/wiki/Encoder-configuration)
qaac's Quality 2 (qtaacenc's --highest) is enabled by default if variable left empty.
a. obviously i dont need qaac on mac, i dont use itunes, so irrelevant for me to care what defaults there
b. there was a slightly hidden suggestion to abx against true vbr (if anyone cares)
smok3, I was talking to the people who test, didn't really care about your post.
Listening to that sample, I find it interesting that without the hint about the cymbals I probably wouldn't have noticed the difference even at considerably lower bitrates.
I agree, "wrong" was inappropriate. Sorry about that, jensend. Please re-interpret it as "highly unlikely".
Regarding your statement that there's no theory to support my claim "theoretical maximum in compression": there's Shannon's rate-distortion theory... I also think the psychoacoustic theories have become quite solid. Your opinion may differ, of course.
Well, if you'd said "highly unlikely
to happen in the next 20 years" I might disagree but I wouldn't have felt the need to argue the point. And I'm sorry that I got overly worked up about this.
I agree that
distortion models now seem to be reasonably accurate. And that does give you a lower bound on what compression you can get
for your source model. But to say that this is the absolute lower bound for real-world music is to make an
extraordinarily strong and
entirely unjustified claim about your source model. In other words, it's baloney.
Are you really willing to claim that your source model contains
all the prior information available when we know that a signal is, say, the PCM from someone's CD collection rather than a stream of entirely random bits? Are you also claiming that you have objectively convincing evidence of this? You might as well be saying "the asymptotic optimality of LZ for Markov sources means that the GZIP'd size of the Library of Congress is so close to its Kolmogorov complexity that no compression algorithm will do significantly better." In both cases our source models are nice and useful but certainly wrong.
Hate to drag out the OT aspect of the discussion, but I saw a prime opportunity to drudge up ol' Jan Sloot (http://en.wikipedia.org/wiki/Jan_Sloot)'s story. According to some discussion on doom9 and elsewhere, reports indicate the quality was poor but the movies recognizable. It was postulated that the system was somewhat similar to MIDI -- huge database of specially selected "generic" content referenced within some sort of compressed text file ("as little as 8kb!"). Most folk (myself included) still think the trick was all smoke and mirrors, but it just goes to show there are plenty of tricks out there due to the inherent subjectivity of quality. Transparency might not be so easy to extend from this because of that pesky "indistinguishable" bit, but the point (shaky as it is) remains -- there may be breakthroughs yet, be they algorithmic, infrastructural, or downright tricks. SBR (and to lesser degree, band folding), PMS, and so forth remain consigned to the category of "tricks" merely because they weren't good enough data models to achieve transparency ("super" SBR for much higher frequencies, carefully used IS, and some other tools and cases notwithstanding). They remain useful for other things (increasing the perceived fidelity of non-transparent music!) but not for achieving transparency at a lower bitrate. It is conceivable that clever new strategies will be devised (and are currently being devised) that will.
But I'm certainly given to agree with C.R.Helmrich that the emergence of breakthrough technologies is becoming increasingly unlikely/difficult given our current understanding of lossy complexity models (RD as a function of ATH and the PA models used to approach these bounds). The seemingly asymptotic pattern of lossy codec improvement over time certainly seems to reinforce this understanding. Most work nowadays is in defining and quantifying this perceptual distortion and addressing it through code -- within and beyond standard limits. It's just that within the standard limits you have to work with a more limited format toolset to make improvements. The benefit of new formats in this area is the ability to leverage our evolving understanding of these models, masks, quantization metrics, etc in terms of addressing shortcomings in our existing models and adding new tricks to the toolset. Unless a viable MDCT alternative emerges, a codec needs to deal with the spread of quant noise over the transform window (detecting these occurrences and upping the bitrate locally is a crude fix for the hated "pre-echo problem." I think Opus can divide a problem block near the boundary, controlling the transient masking window). So we'll probably continue to need tricks to get around the problems introduced by our other tricks. MP3 was beleaguered, among other things, by the trade-offs taken to ensure backwards compatibility with MP1/2.
It's an open problem that there is no 100% accurate psychoaccoustic model in existence. Maybe there will never be, since humans with different ear and brain physiologies are more susceptible to different forms of distortion. This is one reason we rely on larger listening tests.
Are you really willing to claim that your source model contains all the prior information available when we know that a signal is, say, the PCM from someone's CD collection rather than a stream of entirely random bits? Are you also claiming that you have objectively convincing evidence of this? You might as well be saying "the asymptotic optimality of LZ for Markov sources means that the GZIP'd size of the Library of Congress is so close to its Kolmogorov complexity that no compression algorithm will do significantly better." In both cases our source models are nice and useful but certainly wrong.
The need for audio formats to be vaguely streamable, to require relatively limited amounts of memory for decode, and to decode using some reasonable level of power is a much more relevant limit on compression. You can of course come up with very clever formats that squeeze all kinds of redundancy out of bitstreams, but realistically people aren't going to be interested in using them.
Are you really willing to claim that your source model contains all the prior information available when we know that a signal is, say, the PCM from someone's CD collection rather than a stream of entirely random bits? Are you also claiming that you have objectively convincing evidence of this? You might as well be saying "the asymptotic optimality of LZ for Markov sources means that the GZIP'd size of the Library of Congress is so close to its Kolmogorov complexity that no compression algorithm will do significantly better." In both cases our source models are nice and useful but certainly wrong.
The need for audio formats to be vaguely streamable, to require relatively limited amounts of memory for decode, and to decode using some reasonable level of power is a much more relevant limit on compression. You can of course come up with very clever formats that squeeze all kinds of redundancy out of bitstreams, but realistically people aren't going to be interested in using them.
That's an excellent point. Vaguely streamable or
extremely so in low-delay Opus' case. Some (CM-based?) wide-window redundancy models are certainly off-limits for this reason. Most songs are at their core extremely formulaic -- it is often through a distinctive spread of beat, repeating instruments, and somewhat "context-predictable" aspects that a song is born. Methods to take advantage of these longer-term redundancies would most likely kill streaming potential (solid archive anyone?) or lack speed and/or fidelity at today's computing power requirements.