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Topic: Multiformat@128kbps listening test - FINISHED (Read 181387 times) previous topic - next topic
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Multiformat@128kbps listening test - FINISHED

Reply #200
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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.


This is a quote from the mp3 specification:
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Intensity Stereo
This mode switch (found in the header: mode_extension) allows switching from 'normal stereo' to intensity stereo. The lower bound of the scalefactor bands decoded in intensity stereo is derived from the "zero_part" of the right channel. Above this bound decoding of intensity stereo is applied using the scalefactors of the right channel as intensity stereo positions. An intensity stereo position of 7 in one scalefactor band indicates that this scalefactor band is NOT decoded as intensity stereo.


I guess this means the encoder can choose some kind of split frequency. Below this frequency L/R or M/S coding is applied and above IS coding is used.

Agree ?

bye,
Sebastian

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Reply #201
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I guess this means the encoder can choose some kind of split frequency. Below this frequency L/R or M/S coding is applied and above IS coding is used.

Agree ?

yes.

M/S coding is some special case of doing some main axis transformation of the stereo plane and transmitting the rotation angle, the sum and the difference signal. for mid/side coding the rotation angle is fixed and is not transmitted.

IS coding is some simplification where you leaf out the difference signal.

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Reply #202
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ff123: 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.

BTW, it was officially recommended to change it's default lowpass to -bw 15995 (number can be wrong).
And it's ability not to produce ringing and very small pre-echo always was a point all free developers gonna to achieve, as I remember. 
Interesting, may be now, when mp3 seems to be already mature standart, we could find somebody from Fhg and ask how they avoid ringing in their encoder ?
If it is not already known, of course...

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Reply #203
I've had a strong feeling that I already saw this discussion about mp3enc.
I've find it at the end: Test old Fhg encoder or not

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Reply #204
I can also confirm that the current Vorbis encoders use a mix of lossless stereo (full mag, full ang preserved) and point stereo (zero ang) below q 6.  Point stereo kicks in for components above a certain frequency which is dependent on the quality.  For lower quality values, more point stereo is used, hence the recognisable 'stereo collapse'.  It does not appear to be the optimal way of doing things but considering the quality we get from current Vorbis, it doesn't do a bad job either.  Monty has plans of implementing a better stereo model.

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Reply #205
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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.

Lame project is only providing a technology implementation. It is up to the the company wanting to use it to acquire a patents license regarding the mp3 patents.
Several companies choosed this solution and are using Lame in their products.

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Reply #206
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Anyway, I'm surprised that LAME peforms so well WITHOUT Intensity Stereo in the 128-ish bitrate area - Same for FAAC.  (no IS AFAIK)

There is absolutely no need to use IS @128 kb/s  for MP3 or AAC.

And I also disagree with the claims that IS could bring good quality - there are lots of cases with stereo configuration impossible to code properly with IS, because IS saves only ILD information (level difference)  and not ITD (time difference) and inter-channel cross corellation. 

Equalized and mixed "left" (IS) channel could completely distort the phase information, and you end up with something which is quite different from the original when the coloration of the sound comes into the question.

Applaud is one of the examples that is impossible to code properly with IS.

Smart psychoacoustic would be able to disable IS for such frames, but @128 kb/s there woud be no need for lossy bit savings,  same goes for PNS (in AAC) more or less- we did a lot of tests with PNS @128 kb/s and in most cases is pretty much useless, or degrades the quality.

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Reply #207
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I guess this means the encoder can choose some kind of split frequency. Below this frequency L/R or M/S coding is applied and above IS coding is used.

Agree ?

yes.

M/S coding is some special case of doing some main axis transformation of the stereo plane and transmitting the rotation angle, the sum and the difference signal. for mid/side coding the rotation angle is fixed and is not transmitted.

IS coding is some simplification where you leaf out the difference signal.

Thanks alot for the explanations. I want to apologize for making obviously wrong statements about MP3 IS.

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Reply #208
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And I also disagree with the claims that IS could bring good quality - there are lots of cases with stereo configuration impossible to code properly with IS, because IS saves only ILD information (level difference)  and not ITD (time difference) and inter-channel cross corellation. 
[...]
Smart psychoacoustic would be able to disable IS for such frames, but @128 kb/s there woud be no need for lossy bit savings,  same goes for PNS (in AAC) more or less- we did a lot of tests with PNS @128 kb/s and in most cases is pretty much useless, or degrades the quality.

Thanks for your reply.

Let's compare ogg to aac. Monty said once, lossless coupling would be like wasting space for 128kbps and 160kbps modes. You guys keep telling me IS is inappropriate for that bitrates. Sure, this mapping is irreversible and only preserves the channel's energy levels - not all their phase relations. But AFAIK phase relations are not that important to us at above 10 kHz because the wavelength is already very short. So if an advanced encoder would make use of this psychoacoustic effect properly by using IS this could save some space and allows to use smaller scalefactors to improve the SNR.

AFAIK IS can be switched on/off for each scalefactor band (AAC). Another cool thing: IS can be done in-phase and out-of-phase. How about the following sheme for scalefactor bands above 10 kHz:

- treat MDCT samples for a scalefactor band as multidimensional vector
- compute the cosine of the angle between L and R by cos_a := \frac{<L,R>}{||L|| ||R||}
- use in-phase IS for cos_a > 0.5
- use out-of-phase IS for cos_a < -0.5

These thresholds (in this case 0.5) could be chosen depending on the quality-preset and frequency area.

Well, I don't know, if Intensity Stereo is or is not appropriate for 128 kbps. But I do know that Vorbis makes use of it and "won" the listening test.

edit: corrected cos_a correlation formula

bye,
Sebastian

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Reply #209
Suprisely! How it is possible Lame mp3 better than Itunes aac?
are previous tests wrong?

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Reply #210
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Suprisely! How it is possible Lame mp3 better than Itunes aac?
are previous tests wrong?

the encoder-version + settings used for lame mp3 during this test for sure weren't the same as in previous large-scale tests.

Lame is not = Lame. In recent versions, lame seems to have made good progress in improving at mid-bitrates. You can see this in the lame 3.90.3 vs. lame 3.96 thread.

- Lyx
I am arrogant and I can afford it because I deliver.

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Reply #211
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Suprisely! How it is possible Lame mp3 better than Itunes aac?
are previous tests wrong?

the encoder-version + settings used for lame mp3 during this test for sure weren't the same as in previous large-scale tests.

It's also worth mentioning that Lame is not better in the test. It's officially tied, with a tendency to be a little worse.

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Reply #212
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ok, so it looks like the vbr contenders did very well and itunes's cbr held its on. how safe would it be to assume that using vbr with AAC (for instance the most recent FAAC with FB2K) would be a contender?


okay, it has been proven that iTunes AAC is better then FAAC for instance.
http://www.rjamorim.com/test/aac128test/results.html

but how sure could one say how good iTunes AAC would be if it had VBR implemented?

and, one silly question:

how sure is it how good/bad one encoder would perform at higher bitrates,e.g. 160kbps?
i mean is it right to say that vorbis for instance can reach transparent level at a lower bitrate then MPC or AAC?

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Reply #213
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okay, it has been proven that iTunes AAC is better then FAAC for instance.
http://www.rjamorim.com/test/aac128test/results.html


This is an rather old test, here are newer test results: http://www.rjamorim.com/test/aac128v2/results.html but iTunes is still the winner.

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how sure is it how good/bad one encoder would perform at higher bitrates,e.g. 160kbps?
i mean is it right to say that vorbis for instance can reach transparent level at a lower bitrate then MPC or AAC?


You can´t extrapolate the results! WMA is generally better than MP3 at 64kb/s, but at 128kb/s MP3 is better.

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Reply #214
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You can´t extrapolate the results! WMA is generally better than MP3 at 64kb/s, but at 128kb/s MP3 is better.


okay, thank you, but
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i mean is it right to say that vorbis for instance can reach transparent level at a lower bitrate then MPC or AAC?


i assume we would need a listening test at this high bitrate, right?

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Reply #215
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i assume we would need a listening test at this high bitrate, right?

Indeed

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Reply #216
even if i accounted harashin's private listening test i wouldn't be able to do so?
see, we know auTuV is very good at 128kbps and at ~200kbps (at least for harashin).
now we should be able to say how Vorbis would perform at around 160kbps shouldn't we?

and what about my other question:
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but how sure could one say how good iTunes AAC would be if it had VBR implemented?

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Reply #217
My post will not be very informative 
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but how sure could one say how good iTunes AAC would be if it had VBR implemented?

No one knows.
As example - Fhg mp3 encoders on low bitrates tends to be better with CBR, than VBR (search forum, if you wish to have more info) but no one would tell (I hope  ) VBR is loosely implemented there...
VBR have it's own problems, as discussed in this thread...
EDIT: grammar

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Reply #218
oh gosh!

trying is superior to studying - isn't it?
(well, I tried to translate a german saying)

okay.now for me it IS like this:

aoTuv is superior to iTunes AAC even at 160kbps and above and second personal truth is iTunes AAC would be better with VBR implemented presumed that it is decent implemented. harrrharrr!

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Reply #219
I wonder how ATRAC3plus performs...it's a pity that it wasn't included in the test
--alt-presets are there for a reason! These other switches DO NOT work better than it, trust me on this.
LAME + Joint Stereo doesn't destroy 'Stereo'

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Reply #220
It's currently not possible to oppose atrac3+ to other encoders at 128 kbps. For a simple reason: there's no 130 kbps mode with current and public atrac3+ encoder. Only low bitrate (48 & 64 kbps) and high bitrate (256 kbps) setting.

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Reply #221
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It's currently not possible to oppose atrac3+ to other encoders at 128 kbps. For a simple reason: there's no 130 kbps mode with current and public atrac3+ encoder. Only low bitrate (48 & 64 kbps) and high bitrate (256 kbps) setting.

interesting...so I wonder what bitrate is used in Sony's music store...I read they use ATRAC3plus...don't tell me they use 64kbps 
--alt-presets are there for a reason! These other switches DO NOT work better than it, trust me on this.
LAME + Joint Stereo doesn't destroy 'Stereo'

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Reply #222
Hello, I'm back from holydays...

I just wanted to point out an odd thing that happened to me during this test : contrary to the common way of things, I could only ABX the 7th sample (gone) with speakers, and not with headphones ! (Dynaudio Gemini speakers vs Sennheiser HD600 headphones).


A picture of me ABXing "gone" : listeningtest.jpg 
To avoid any background noise, the picture is video-projected on a screen in front of me, and the computer is in the next room. 5 meters mouse, keyboard, SPDIF, and DVI cables.

Mhhh, actually, I must admit that my speaker setting is often refered by my father as "the biggest headphones I've even seen".

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Reply #223
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A picture of me ABXing "gone" : listeningtest.jpg 
To avoid any background noise, the picture is video-projected on a screen in front of me, and the computer is in the next room. 5 meters mouse, keyboard, SPDIF, and DVI cables.

Mhhh, actually, I must admit that my speaker setting is often refered by my father as "the biggest headphones I've even seen".

   
I am arrogant and I can afford it because I deliver.

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Reply #224
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A picture of me ABXing "gone" : listeningtest.jpg 

Nice settings