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Topic: Multiformat@128kbps listening test - FINISHED (Read 182418 times) previous topic - next topic
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Multiformat@128kbps listening test - FINISHED

Reply #175
Quote
My point of view is that when it comes to bitrate, the only thing that counts is the long time average.

Yep. Of course. And here the problem with mpc lies.
So, to summarize, my IMHO, formed by last test and this thread:
1. MPC is a VBR encoder (or at least vbr is the mode it performs much better).
2. Test on many different albums showed (see 128Kbit test discussion thread) that 4.15 setting produces and average bitrate of ~130Kbit and that is ok.
3. The idea the samples for the test are selected is to make encoders job harder.
4. So, average mpc bitrate rises for test samples from 130 to 142Kbit.
5. To maintain avg mpc bitrate about projected (as a result - 136Kbit) additional easy sample(s) (one was selected ocassionally) was chosen.
6. MPC failed on this samples.
So, the question - do mpc have a high score because of it's quality or because of samples selected ?
Correct me, if I wrong somewhere...
BTW: ff123 idea to use equal number of overbitrated and underbitrated samples can correct a situation. May be. That's why I wrote, that error could be in test setup, not in mpc...

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Reply #176
The Debussy sample is a great sample for Frank Klemm when he does future MPC tuning.  If indeed the bitrate was too low for MPC to give good quality, then that is an issue that needs to be fixed.

From the Vorbis side, I think guruboolez tested the 1.0 encoder on one sample (may have been creaking sample or brahms) at q 0 that produced a low bitrate of 40 kbps or something.  It sounded pretty bad.  Monty took note of this and made some tweaks to produce 1.0.1 which now gives a more realistic bitrate (somewhere close to nominal 64 kbps) and sounds much better now.

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Reply #177
Quickly contribution some more humor to this thread. Someone posted the results on some Windows tech forum, someone else found it funny to reply with this:

Quote
I have audible problems with LAME so I use Blade Enc and all's fine now.

http://www.neowin.net/forum/index.php?showtopic=171304

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Reply #178
QuantumKnot> I've sent last years two samples: Stockhausen - Stimmung (vocal music) and Liszt - Harmonies Poétiques et Religieuses (piano). Both at very low volume (original, I didn't changed it).

With 1.01 encoder, the problem corrected (or close to be so).


I've played yesterday with mpc 1.14 -q4.15. I've encoded 10 CD of piano music. Average bitrate was < 100 kbps (for a complete work of Erik Satie (5h30, digital recording), bitrate was ~90 kbps. In other words, the average bitrate of the Debussy isn't an accident... or is a very usual one!

Low volume is just a part of the quality problem. I've encoded a piece of contempory music, very quiet too, and without background noise (lossless encoding reached 20%): bitrate was inferior to 80 kbps, and quality was much better, far from the disaster of Debussy.wav. This mean that MPC could achieve good reproduction even with very low bitrate...
Problem for mpc is maybe low volume + background noise? To be verified...
Wavpack Hybrid: one encoder for all scenarios
WavPack -c4.5hx6 (44100Hz & 48000Hz) ≈ 390 kbps + correction file
WavPack -c4hx6 (96000Hz) ≈ 768 kbps + correction file
WavPack -h (SACD & DSD) ≈ 2400 kbps at 2.8224 MHz

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Reply #179
Heh... just as the Slashdot hammering on my site starts to subside, I get Slashdotted again.

In Japan!!! 

I wonder if the SNR there is as big as in slashdot.org...



Edit: BTW, I recommend you don't use Babelfish

Quote
Using part the some overseas, when the domestic technician builds up in the country, when it is called "domestic production" is many, because is.
So, including to the part, when it makes from one, it becomes "the purity domestic".


Your head will explode.

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Reply #180
Roberto,

Can you clarify the sample sizes again? What your N means, is that the total people that listened to all songs or the number of people per song? And what is the sample size for the final ANOVA?

Btw, can you make the actual dataset available for other people to analyze?

As a final comment, I think this is too low a sample size.

Jaester

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Reply #181
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Can you clarify the sample sizes again? What your N means, is that the total people that listened to all songs or the number of people per song? And what is the sample size for the final ANOVA?

What do you mean by sample sizes?

N is the amount of results I received for that sample minus discarded results.

Quote
Btw, can you make the actual dataset available for other people to analyze?


It's already there.
http://www.rjamorim.com/test/multiformat12...s/comments.html

Quote
As a final comment, I think this is too low a sample size.


?

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Reply #182
Hi !

I just switched on my iRiver H120 in shuffle mode. I'm currently hearing a Madonna Song, I encoded many years ago with a Fraunhofer encoder in 128 kbps. It sounds very cool, probably due to the usage of intensity stereo. I guess LAME could benefit of intensity stereo usage in the 128-ish bitrate area. I really wonder what ranking it would have got...

I'm currently too lazy to search all postings but in case anyone knows: why has LAME been chosen for mp3 encoding in this test ? It lacks intensity stereo support.

I guess the aoTuV beta 2 encoder would sound lousy in the 128-ish area if it would not make use of intensity stereo (or point-stereo, whatever you want to call it)

bye,
SebastianG

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Reply #183
Quote
As a final comment, I think this is too low a sample size.

Typically, a sample size of 30 is recommended to be representative of a population.

However, we already know going into these types of open tests that the group of listeners who'll respond are not going to be representative of the general population.  So if you accept the proposition that this group of listeners represents the smaller population of "listeners who care (LWC)," then all you need are enough samples to produce a significant result.

You can get significant results this way from just one person, as long as he represents the LWC (e.g. Dibrom tuning lame).  However, more listeners are desirable, of course, to average out individual biases (for example, my limited high-frequency ability predisposes me to the sound of *gasp* wma9 standard).  As was shown in the Vorbis and mp3 listening pre-tests, even a handful of listeners listening to multiple samples can produce reliable and accurate result.

ff123

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Reply #184
Quote
I just switched on my iRiver H120 in shuffle mode. I'm currently hearing a Madonna Song, I encoded many years ago with a Fraunhofer encoder in 128 kbps. It sounds very cool, probably due to the usage of intensity stereo. I guess LAME could benefit of intensity stereo usage in the 128-ish bitrate area. I really wonder what ranking it would have got...

I'm currently too lazy to search all postings but in case anyone knows: why has LAME been chosen for mp3 encoding in this test ? It lacks intensity stereo support.

I guess the aoTuV beta 2 encoder would sound lousy in the 128-ish area if it would not make use of intensity stereo (or point-stereo, whatever you want to call it)

FhG does not use intensity stereo at 128 kbit/s.  IS is a low bitrate technique, in the same vein as spectral band replication, and isn't meant to produce near transparent encodings.

However, that isn't to say that old FhG encodings can't sound competitive.  Roberto's last mp3 test (designed to find the best mp3 encoder at 128 kbit/s, and which lame won) did not include the super slow FhG encoder, which many people with very good high frequency hearing might like best as their mp3 encoder at 128 kbit/s.

ff123

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Reply #185
Quote
why has LAME been chosen for mp3 encoding in this test ? It lacks intensity stereo support.

IMHO because:
1. It is most widely used mp3 encoder.
2. It, I think, is on pair (or even better, may be) than old mp3enc31.
I 'm sure Gabriel could answer this question more correctly.
BTW, do you know that mp3enc31 cost 199$ ? And you can not purchase it now.
I think, most people used it illegally 
Oh, and I think lame USES joint-stereo for 128Kbit bitrate.
And IS (intensity stereo) corrupts stereo image, thus it is not recommended for such a *high* bitrate as 128Kbit.
EDIT: ff123 was faster 
EDIT2:
Quote
Roberto's last mp3 test (designed to find the best mp3 encoder at 128 kbit/s, and which lame won)

Forgot to mention it as 3. Sorry...

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Reply #186
Lame is the most widely used MP3 encoder?  Perhaps around here.  I would have to say most people are using variations on royalty paying MP3 encoders, probably Fraunhafer, that are included in various all-in-one music solutions.  Because of its so so legal status none of these programs can incorporate Lame, even if many popular applications work with it.  What about Music Match, it comes on nearly every PC?

Frankly, I am amazed at all the tiny details that seem so fascinating around here.

IMO, Roberto's test is a blockbuster.  Look at the politics.  An unofficial build of the open source ogg vorbis encoder blows everything away.  Two proprietary solutions, one from the hated MS and the other from Sony, a mega copyright holder, make a weak showing.  The hightly compatible and easy to find Lame MP3 shows it has a bunch of life left in it.  That is headline news in digital audio compression if there ever was any.

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Reply #187
Quote
Lame is the most widely used MP3 encoder? Perhaps around here.

Heh. You are probably right.
Personally, I (nor my friends) 've never use any *box* solutions(except nero, which comes with all my writers), so I simply did not count them. My fault 
Musicmatch, ITunes and so on have a huge auditory, really...

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Reply #188
Quote
FhG does not use intensity stereo at 128 kbit/s.  IS is a low bitrate technique, in the same vein as spectral band replication, and isn't meant to produce near transparent encodings.

However, that isn't to say that old FhG encodings can't sound competitive.  Roberto's last mp3 test (designed to find the best mp3 encoder at 128 kbit/s, and which lame won) did not include the super slow FhG encoder, which many people with very good high frequency hearing might like best as their mp3 encoder at 128 kbit/s.

ff123

I've used an old l3enc which DOES make use of IS, even at 192 kbps.

As for "near transparency": Current Vorbis encoders make use of IS at up to -q5.99. They just don't call it Intensity stereo. Monty seems to have a very different (official) point of view regarding this. He talks about diffuse and point images in the specification. Well, it's basically the same as intensity stereo.

(Maybe seeing/interpreting things from a different angle helps avoiding patent issues, I don't know...)

Anyway, I'm surprised that LAME peforms so well WITHOUT Intensity Stereo in the 128-ish bitrate area - Same for FAAC.  (no IS AFAIK)
I guess Vorbis will have strong competitorw when LAME and FAAC start making use of IS for that kind of bitrates (perhaps PNS for FAAC, too).

time will tell.

bye,
Sebi

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Reply #189
Quote
BTW, do you know that mp3enc31 cost 199$ ? And you can not purchase it now.
I think, most people used it illegally 


Believe it or not. I registered l3enc back in 1997 together with the WinPlay3 software for Win3.11

Quote
Oh, and I think lame USES joint-stereo for 128Kbit bitrate.
And IS (intensity stereo) corrupts stereo image, thus it is not recommended for such a *high* bitrate as 128Kbit.


Yeah, I wasn't talking about joint stereo coding. LAME does M/S coding as one possible Joint-Stereo coding technology but not IS.

How you define stereo image ?
It is a widely believed fact that we are unable to perceive phase differences of high frequencies, so IS is an appropriate tool, even for near transparency encodings.

bye,
Sebi

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Reply #190
Quote
Roberto's last mp3 test (designed to find the best mp3 encoder at 128 kbit/s, and which lame won) did not include the super slow FhG encoder

It did. Audioactive (I.E, slowenc with some tunings done in AudioActive)

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Reply #191
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IMO, Roberto's test is a blockbuster.

Thank-you, but that's the reason conducing listening tests is nearly impossible now

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Reply #192
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I've used an old l3enc which DOES make use of IS, even at 192 kbps.

Buoy...

From l3enc documentation (taken from good ol' ReallyRareWares):

*****For l3enc 2.0*****
Quote
For bitrates <= 96 kbps, the default is intensity stereo (-mod 1). For
bitrates >= 112 kbps, the default is ms-stereo (-mod 0). For
more details about encoding modes, please refer to section 1.11 'Encoding
Recommendations'


Quote
For coding of stereo files with bitrates <=96 kbps, the use of intensity
stereo is highly recommended. This is also the default configuration of
the encoder. Note, however, that the use of intensity stereo will destroy
information which is needed for sound processing schemes like
Dolby Surround. For bitrates >= 112 kbps, intensity stereo is not used by
default.


*****For l3enc 2.72*****
Quote
For the coding of stereo files with bitrates <=96 kbit/s, the encoder
will use the intensity stereo technique.
Note, however, that the use of intensity stereo may demage information
which is needed for sound processing schemes like Dolby Surround.
For bitrates >= 112 kbit/s, intensity stereo is not used.



What means that, if you got IS at 192kbps, you were messing where you shouldn't

Quote
As for "near transparency": Current Vorbis encoders make use of IS at up to -q5.99. They just don't call it Intensity stereo. Monty seems to have a very different (official) point of view regarding this. He talks about diffuse and point images in the specification. Well, it's basically the same as intensity stereo.


I always understood Vorbis' implementation as a variation on M/S stereo, not IS.

After all, it's very well known that IS completely ruins the stereo image. There were some pre tests for my listening tests that came to that conclusion (look for a post by tigre, IIRC)

Quote
Anyway, I'm surprised that LAME peforms so well WITHOUT Intensity Stereo in the 128-ish bitrate area - Same for FAAC.  (no IS AFAIK)


Same for MPC and iTunes.

Actually, IS was once available in MPC, and IIRC Andree removed it because it had no place in a codec targeted at high quality

Quote
I guess Vorbis will have strong competitorw when LAME and FAAC start making use of IS for that kind of bitrates (perhaps PNS for FAAC, too).


I keep my point that using IS at bitrates above 96kbps is a very bad idea, except on very specific cases.

Regards;

Roberto.

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Reply #193
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Because of its so so legal status none of these programs can incorporate Lame, even if many popular applications work with it.

afaik, you are wrong, you are required t pay a license for the right to implement/use an MP3 encoder, so after that, you can use LAME if you want LEGALLY.

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Reply #194
Quote
Quote
Roberto's last mp3 test (designed to find the best mp3 encoder at 128 kbit/s, and which lame won) did not include the super slow FhG encoder

It did. Audioactive (I.E, slowenc with some tunings done in AudioActive)

Audioactive is a different beast from the very slow codec, which is best represented by mp3enc31, or by using fastencc.exe in -hq mode (this version of the very slow codec has a higher lowpass than mp3enc31).

Audioactive/Opticom/"radium" can be grouped together, but not in the same family as mp3enc31/fastencc.exe -hq.

mp3enc31 is recognizable by low frequency glitches.  Ironically, bAdDuDeX (an mp3 connoiseur from long ago), who could hear a 16 kHz lowpass in applaud.wav, loved mp3enc31 despite the glitching and despite its relatively low 14.5 kHz lowpass because it was free from high-frequency ringing.

ff123

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Reply #195
Quote
Frankly, I am amazed at all the tiny details that seem so fascinating around here.

IMO, Roberto's test is a blockbuster.


I agree completely with both of these statements.  They both, in their own way, provide interesting reading.  Roberto's tests "always" provide informative, useful and constructive information, while the former simply serves to amuse and utter the occasional "WTF is that about?"

Later.
"Did you just say he contacts you through a bird? Did I just hear you say that?" Sonny Valerio (Cliff Gorman). Ghost Dog: The Way of the Samurai.

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Reply #196
Quote
Heh... just as the Slashdot hammering on my site starts to subside, I get Slashdotted again.

In Japan!!!  

I wonder if the SNR there is as big as in slashdot.org...

It seems to be even worse than the one in .org. I found some guys say that Sony rules and this test sucks, or they should adopt Japanese songs for samples. A guy who has his doubts about the test didn't even know that has been a double blind test

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Reply #197
Quote
It is a widely believed fact that we are unable to perceive phase differences of high frequencies, so IS is an appropriate tool, even for near transparency encodings.

The problem with MP3 IS is that it´s not possible to restrict IS usage to certain frequencies - you can only switch stereo modes on a block level, not on a frequency one.

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Reply #198
Quote
Quote
I've used an old l3enc which DOES make use of IS, even at 192 kbps.

Buoy...

Ahhh!

I don't know what was going wrong....
I just trimmed a frame out of the middle and checked for myself:

Header: FF FB 90 64 = 1111 1111 | 1111 1011 | 1001 0000 | 0110 0100
=>
MPEG 1, Layer 3, 128 kbps, 44100 Hz, Joint-Stereo
mode_extension = 10 => M/S coding: yes  and  IS coding: no.

I apologize for that. Might have been fuzzy memory or something.
Also, I wasn't that experienced back in 1997.

But I remember that i fed l3enc with out-of-phase high frequency sines that got cancelled ... 

edit:

As for Vorbis: Trust me.  It's intensity stereo for q<6 (called "point-stereo")
I stick to "IS is very powerful if done right."
Vorbis Stereo Stuff
At q<6 and at a certain frequency Vorbis encoders switch from lossless coupling to point-stereo. In point-stereo the angle value will always br zero. Therefore, the (unscaled) MDCT samples will be the same for both channels after inverse square polar mapping. Intensity is controlled by the floor curves.

In Vorbis, decorrelation and intensity stereo is achieved by square-polar-mapping and channel-interleaved vector quantization.

bye,
Sebastian

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Reply #199
Quote
Quote
It is a widely believed fact that we are unable to perceive phase differences of high frequencies, so IS is an appropriate tool, even for near transparency encodings.

The problem with MP3 IS is that it´s not possible to restrict IS usage to certain frequencies - you can only switch stereo modes on a block level, not on a frequency one.

well, in mp3 you can signal the use of IS stereo without using IS at all**.
if IS is used, you will have to use it for the whole frequency range beginning from the last scalefactor band down to some arbitrary but fixed scale factor band. you can use L/R or M/S coding for the lower bands.

---
**) I'm actually not sure if the sfb21--where no scalefactor band exists--wouldn't have to be IS coded with 0 degree direction in this case