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80 kbps personal listening test (summer 2005), AAC MP3 Ogg Vorbis WMA
Razor70
post Jul 11 2005, 02:45
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QUOTE (HotshotGG @ Jul 10 2005, 08:35 PM)
QUOTE
So can I be the noob and stupid one here and ask..what does this tell us? Why do all the tests always run towards the lower bitrates and not the higher bitrates? Okay you can flame me now lol.


It's more difficult to actually hear any substantial differences between codecs. Guruboolez is really the only one around here who has golden ears biggrin.gif. Personally I couldn't really tell the difference beyond -q 5 and up with Vorbis, but that's just me I am sure some folks have found some problem case samples. wink.gif
*



Ok another stupid question on my part then (sorry I know this is all noobage stuff here and that's what I am), is using lower bitrates a better thing then? Or is it just used to save space? I get so lost on quality issues that I don't know what to use for a format. I have a Ipod and would like to get the best quality for use on it and don't want to go loseless..so this is were I get confused on formats. I know the thing to do is abx myself for what I think sounds the best..but isn't there a consenses on one format over another that would be the best format? Right now I am using 128AAC but the only reason is because I see it as a good go between for space and quality. But I really do want the best quality I can get. So any help would be appreciated guys.
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sld
post Jul 11 2005, 02:46
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Wow... as a satisfied user of Vorbis for my flash player, what more can I say?

3 thumbs up!
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guruboolez
post Jul 11 2005, 02:48
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QUOTE (HotshotGG @ Jul 11 2005, 02:35 AM)
Guruboolez is really the only one around here who has golden ears.
I won't say that. I'm just trained to catch artefacts and distortions (at least some of them). I'm rather an artefact hunter than a blessed audiophile.
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guruboolez
post Jul 11 2005, 02:52
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QUOTE (Razor70 @ Jul 11 2005, 02:45 AM)
Right now I am using 128AAC but the only reason is because I see it as a good go between for space and quality.  But I really do want the best quality I can get.  So any help would be appreciated guys.


Right now, you have two possibilities:

- keeping your current setting. If you're happy with 128 kbps encodings, you won't get any audible benefit from higher bitrate.

- if you really want the absolutely best quality with AAC, just go with Nero AAC and set the bitrate to CBR 448 kbps. It's totally insane, but you'll obtain what you've asked for: "the best quality I can get".

This post has been edited by guruboolez: Jul 11 2005, 02:54
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Enig123
post Jul 11 2005, 03:02
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Guru, you always bring us such brilliant articles. Very impressive and convincing.
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HotshotGG
post Jul 11 2005, 04:56
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QUOTE
Vorbis apparently embed some encoding tools (point stereo?) which are remarkably suited for this bitrate (but which are maybe interfering too much at higher bitrate: see this test and this test).
Quality is not perfect of course; usual vorbis problems are here: noise boost, coarseness, fatness. Distortion


I wonder how much noise normalization play's a large role in part due to low-bitrate encoding? I think a lot of the noise is characteristic in Vorbis has a lot to do with the the noise-floor is encoded via VQ approach, which is more pleasent sounding at least to me. I have been browsing through trying to figure out with Aoyumi had adjusted for educational purposes and I think I understand what he did at least for the B2 tunings that were merged into 1.1. Hmm.... thank you for the results though Guru. wink.gif

QUOTE
Right now I am using 128AAC but the only reason is because I see it as a good go between for space and quality. But I really do want the best quality I can get. So any help would be appreciated guys.


I was going to say the exact same thing that Guru said, but seeing that he answered your question first I would just stick with what you have now biggrin.gif

This post has been edited by HotshotGG: Jul 11 2005, 04:59


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kl33per
post Jul 11 2005, 06:17
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Wow,

Thanks for putting the effort in Guru.


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spoon
post Jul 11 2005, 09:16
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QUOTE (guruboolez @ Jul 10 2005, 08:25 PM)
QUOTE (spoon @ Jul 10 2005, 09:08 PM)
RE Bitrate problems: if your samples are 10 seconds could you create one large sample using that 10 seconds looping over and over 10 times, encode and then calc the bitrate and divide by 10?
*

I don't understand. Could you explain?
*



The problem is:

|---short audio data---| + container padding is not giving the true bitrate (without fudging the stream), so duplicate your short audio data x10:

|---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + |---short audio data---| + container padding

and calc the bitrate as divide 10.


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guruboolez
post Jul 11 2005, 10:47
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> Spoon: I undernstand better the purpose. Good idea, but fastidious if I have to apply it to so many samples.
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Aoyumi
post Jul 11 2005, 12:17
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Guruboolez, I appreciate the large-scale test of you. smile.gif

QUOTE
Vorbis apparently embed some encoding tools (point stereo?) which are remarkably suited for this bitrate (but which are maybe interfering too much at higher bitrate: see this test and this test).

Control is simultaneously difficult although point stereo is powerful. However, probably, in dealings by the low bit rate, it will be indispensable. Although there was a case to which improvement which is aoTuV beta3 expanded artifact of point stereo, it has improved in beta4 (a part of channel coupling was changed).

QUOTE
I wonder how much noise normalization play's a large role in part due to low-bitrate encoding? I think a lot of the noise is characteristic in Vorbis has a lot to do with the the noise-floor is encoded via VQ approach, which is more pleasent sounding at least to me. I have been browsing through trying to figure out with Aoyumi had adjusted for educational purposes and I think I understand what he did at least for the B2 tunings that were merged into 1.1. Hmm.... thank you for the results though Guru.

Although noise normalization can control ringing(and metallic warbling), there are side effects. However, it is needed especially for the low bit rate (especially q-1/-2).
I think that the feature of Vorbis is in Floor(1) encoding, Vector Quantization, and Channel Coupling. These are involved closely.
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guruboolez
post Jul 11 2005, 15:12
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Aoyumi> congrats! I hope that your work will soonly be merged in the official branch.

I starting to think about the 96 kbps test. I'd like this time to make a pool dedicated to MP3 at this bitrate. The old idea about Fraunhofer superiority at bitrate < 128 kbps is still alive, and I'd like to evaluate its validity. I didn't find any test comparing modern release of lame and modern implementation of Fhg.

My problem is: what software should I use? I have some possibilities:
- the new ACM encoder bundled with WMP10
- Nero Burning Rom
- iTunes
- Adobe Audition
- or maybe something else?


Does someone have an idea about the possible best FhG implementation?
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dev0
post Jul 11 2005, 16:10
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QUOTE (guruboolez @ Jul 11 2005, 03:12 PM)
My problem is: what software should I use? I have some possibilities:
- the new ACM encoder bundled with WMP10
- Nero Burning Rom
- iTunes
- Adobe Audition
- or maybe something else?

Does someone have an idea about the possible best FhG implementation?
*

iTunes does not use FhG, it's only identified as such by Encspot.

I'd vote for Adobe Audition, because it looks like the most configurable (using the 'Best - Current' encoder) FhG encoder.


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rjamorim
post Jul 11 2005, 16:21
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QUOTE (dev0 @ Jul 11 2005, 12:10 PM)
I'd vote for Adobe Audition, because it looks like the most configurable (using the 'Best - Current' encoder) FhG encoder.
*


I agree, but the ACM in WMP10 is more recent. I suggest a quick (only a handful of samples) listening test to select one of these.


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rutra80
post Jul 11 2005, 16:44
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Can we have listening-tests like this announced on the front-page news when they are finished, just like it was with Roberto's tests?

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Garf
post Jul 11 2005, 16:48
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QUOTE (rutra80 @ Jul 11 2005, 05:44 PM)
Can we have listening-tests like this announced on the front-page news when they are finished, just like it was with Roberto's tests?
*


IMHO roberto's tests had much more validity since they span a larger number of testers. The results of this test are entirely relying on guruboolez personal preferences, which may or may not be representative of the average person (and I suspect they are not).
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guruboolez
post Jul 11 2005, 16:54
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Something annoys me with Audition: it's a bit expensive for the basic user. Other problem: there are various settings. Like Nero AAC, testing Audition's encoder is a lot of work. But it could be worth.

I think I'll limit the MP3 pool to four contenders (three would be ideal).


+ The encoder embedded in WMP10 will probably be tested (it's an interesting one, which could be used without any expense on Windows, which works very fast, and which could -thanks to nyaochi- benefits from features such as gapless or direct reencoding with foobar2000).

+ LAME of course

+ Audition (maybe directly the "slow" encoder?)


Last one could be iTunes. I suppose that CBR would be better at this bitrate. Does someone experienced something else with it?
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rjamorim
post Jul 11 2005, 17:18
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QUOTE (guruboolez @ Jul 11 2005, 12:54 PM)
+ Audition (maybe directly the "slow" encoder?)
*


There's no such thing as slowenc in Audition. The last versions of slowenc were MP3enc 3.1 and AudioActive 2.04j.

All three encoders in Audition are different versions of fastenc.


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guruboolez
post Jul 11 2005, 17:48
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I have three choices (they're translated in french - I'll translate them in english):

- Current (best quality)
- Legacy - high quality (slow)
- Legacy - average quality (fast)

I thought that "Legacy Slow" corresponds to the old slow encoder [indeed, the Slow encoder isn't slow, and obviously can't correspond to "slowenc". Thanks for the precision Roberto].
But what really annoys me with Audition is the defaulted settings. Lowpass at 96kbps set to 11480. I don't know the exact lowpass set by LAME at the same bitrate, but even at 80 kbps LAME lowpassed to a more confortable value (~13000). To be honest, I really believe that Audition will end last of the pool with such lowpass (except of course if another contender really sucks).
Changing the lowpass would be more pertinent, but it's a game I don't want to play with. My purpose is to evaluate the quality of current encoders, and not to tune them... If lowpass was set to 11,5 KHz by default, there's probably a reason. Any suggestion?

This post has been edited by guruboolez: Jul 11 2005, 17:50
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Pio2001
post Jul 11 2005, 17:50
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QUOTE (Garf @ Jul 11 2005, 05:48 PM)
IMHO roberto's tests had much more validity since they span a larger number of testers.
*


On the other hand, they relied on much fewer samples. 20 ones, with many users only listening to half of them.
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guruboolez
post Jul 11 2005, 17:59
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QUOTE (Pio2001 @ Jul 11 2005, 05:50 PM)
QUOTE (Garf @ Jul 11 2005, 05:48 PM)
IMHO roberto's tests had much more validity since they span a larger number of testers.
*


On the other hand, they relied on much fewer samples. 20 one, with many users only listening to half of them.
*


12 for the 5 first tests; 18 for the 2 last ones.
Both ways have their Achilles' heel: limited by the personal subjectivity of the only tester, or limited by the number of samples tested.
And in both cases, the conducer did his best:

- I can't multiply my subjectivity
- Roberto can't force people to test 50 samples

However, I must add that all samples are online (I gave the link for my 150 classical samples, and the 35 others should be somewhere on Rarewares), and I'd like to see other people testing them.
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rjamorim
post Jul 11 2005, 18:34
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QUOTE (guruboolez @ Jul 11 2005, 01:59 PM)
and the 35 others should be somewhere on Rarewares
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http://www.rjamorim.com/test/samples/


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Zurman
post Jul 11 2005, 20:44
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Simply a m a z i n g Guru, as usual blink.gif

My understanding of the results : mp3@128 is the best choice for mobile devices, no need to bother with other codecs tongue.gif
(especially wma, really disappointing blink.gif )
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rjamorim
post Jul 11 2005, 20:55
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QUOTE (Zurman @ Jul 11 2005, 04:44 PM)
mp3@128 is the best choice for mobile devices
*


Dude, that's the high anchor.


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guruboolez
post Jul 11 2005, 20:57
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QUOTE (Zurman @ Jul 11 2005, 08:44 PM)
Simply a m a z i n g  Guru, as usual blink.gif

My understanding of the results : mp3@128 is the best choice for mobile devices, no need to bother with other codecs

If you want 128 kbps encodings for your player, vorbis and AAC are probably better than MP3.
And if you want LAME 128 kbps quality, you can probably reach it at lower bitrate (90...120 kbps) with other formats, and therefore increase the musical content of your player.

MP3 128 performed the test as anchor, not as competitor. It's here as reference.
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a_aa
post Jul 11 2005, 21:45
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First: Thanks for a very interesting article, I admire your work!

QUOTE (guruboolez @ Jul 11 2005, 09:57 PM)
[If you want 128 kbps encodings for your player, vorbis and AAC are probably better than MP3.
*


I do understand you are mentioning vorbis here, but are there really any large-scale testing to support a claim that any implementation of aac performs better than LAME at 128 kbps? Robertos multiformat test showed iTunes and LAME to be practically tied at this bitrate (both beaten by vorbis and MPC).

Problem is, everybody tells me that aac theoretically is much better than mp3, but I havent seen much reliable testing of aac implementations to substantiate this...

Got any good links?
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