Skip to main content
Topic: best mp3 encoder with something better than a command line interface? (Read 4174 times) previous topic - next topic
0 Members and 1 Guest are viewing this topic.

Re: best mp3 encoder with something better than a command line interface?

Reply #25
I think I was misunderstood... Excuse my french, but I am not a native english speaker. The improvement of the .mp3 encoded with the modified psy extended settings must be compared with the .mp3 encoded with the default settings (filters ON, no other psy settings only the default,...).

At the bitrates you are using it won't be possible to have an improvement generally, so that won't work.  Instead, you start with a sample where you can reliably detect a difference and then make adjustments until you cannot.  Then you compare the encode you make to the original and try to show that it is now transparent.

My goal is to tweak the lame settings, psy included, to obtain a mp3 file as close as possible with the original WAV, so that they can not feel any difference in ABX tests.

At the bitrates you are using it is very unlikely you'll find many files with a difference that is detectable.  Do you actually have one? 

Re: best mp3 encoder with something better than a command line interface?

Reply #26
If you upgrade your gear in the future all you need to do is redo the EQ settings to suit and not have to rerip and reencode your music again in to further "tweaked" mp3s.
You can't save EQ settings as metadata?
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: best mp3 encoder with something better than a command line interface?

Reply #27
At the bitrates you are using it won't be possible to have an improvement generally, so that won't work.  Instead, you start with a sample where you can reliably detect a difference and then make adjustments until you cannot.  Then you compare the encode you make to the original and try to show that it is now transparent.
I hope to obtain good results with all filters OFF and with the psy fine tuning (interchannel masking ratio and sticking around with the psy masking values). The default settings output something crappy and the differences are obvious for any real Hi-FI audio system. The increment of the psy masking values is +/-0.25 dB, added/subtracted to the default psy values. The sfb21 is treble dependent and, along with the low pass filter, cuts the high frequency sounds brutally. Working in progress.

At the bitrates you are using it is very unlikely you'll find many files with a difference that is detectable.  Do you actually have one? 
With the settings I used in lame, I saw in spectrograms some kind of denoise, but the music sounds apparently identical, from the subwoofer bumps to cymbals. I tested >30 samples. Now I have to prove there are improvements based on the psy fine tuning and concludent ABX tests.

 

Re: best mp3 encoder with something better than a command line interface?

Reply #28
I hope to obtain good results with all filters OFF and with the psy fine tuning (interchannel masking ratio and sticking around with the psy masking values). The default settings output something crappy and the differences are obvious for any real Hi-FI audio system.

Default 320kbps audio, of almost any format, is generally transparent.  If you don't already realize this, you should stop what you are doing and try it out. 

With the settings I used in lame, I saw in spectrograms some kind of denoise, but the music sounds apparently identical, from the subwoofer bumps to cymbals.

Spectrograms are meaningless.  Do not even waste your time with them.

Now I have to prove there are improvements based on the psy fine tuning and concludent ABX tests.

Maybe this is a language barrier, but logically to improve something you must first find a problem.  If you don't find any problems, then how are you expecting to improve anything?  You are not making a lot of sense. 

Re: best mp3 encoder with something better than a command line interface?

Reply #29
With the settings I used in lame, I saw in spectrograms some kind of denoise, but the music sounds apparently identical, from the subwoofer bumps to cymbals. I tested >30 samples. Now I have to prove there are improvements based on the psy fine tuning and concludent ABX tests.
Proof based on psy fine tuning?  No.

Concludent [sic] ABX tests?  Yes.  Where are they?

So far you've done nothing to justify your hijacking this topic.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: best mp3 encoder with something better than a command line interface?

Reply #30
If you upgrade your gear in the future all you need to do is redo the EQ settings to suit and not have to rerip and reencode your music again in to further "tweaked" mp3s.
You can't save EQ settings as metadata?
The EQ would be set up for his current setup, with different speakers\headphones or even device the EQ might need to be changed to suit. still far less painful then manipulating the sound of the mp3s.
Who are you and how did you get in here ?
I'm a locksmith, I'm a locksmith.

Re: best mp3 encoder with something better than a command line interface?

Reply #31
I hope to obtain good results with all filters OFF and with the psy fine tuning (interchannel masking ratio and sticking around with the psy masking values). The default settings output something crappy and the differences are obvious for any real Hi-FI audio system. The increment of the psy masking values is +/-0.25 dB, added/subtracted to the default psy values. The sfb21 is treble dependent and, along with the low pass filter, cuts the high frequency sounds brutally. Working in progress.

High frequency content (>16kHz) is difficult to encode, this is why the low-pass filter removes them. As you say yourself, you cannot hear any difference. Only the spectrogram shows the filter, but having nice spectrograms should not be the aim. (If it is, forget mp3 altogether and just save a png of the original spectrogram :)

When you disable the low pass filter, LAME starts wasting bits on encoding the high frequency content (which you cannot hear), and then it has less bits to encode the low and mid frequency information, so it will sound worse.

Thank you for sending me the samples. I cannot ABX any difference between the wav and your mp3, but that is also the case with the LAME default settings. And the default settings preserve the original 44.1kHz sampling rate. Resampling to 48kHz will not improve quality.

Re: best mp3 encoder with something better than a command line interface?

Reply #32

If spectograms do not mean anything, I wonder why there are still exist.

At a glance, it can be seen some denoise in both encoded samples, especially at high frequencies which anyway bring only annoying HF hiss in ears.

I lowered the maskers of bass, alto and treble to not lose any particular sound from the original sample, which may be covered by stronger maskers.

The original sample (44,100 Hz 16bit WAV) was resampled to 48,000 Hz 24bit and the encoded samples to (48,000 Hz 24bit MP3) and look in spectograms almost identical and sounds almost identical, even on the whatever Hi FI audio system.

I put sfb21=4 to cut any noise >20,000 Hz and to save bits for a better quality encoding at lower frequencies. My goal is to make good quality mp3's for my android. The 4.8X WAV/mp3 dimension ratio it's pretty OK.

The best settings I tested so far:

lame.exe -mj --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.25 --ns-sfb21 +3.75 --short --verbose -q0 -b320 --cbr --resample 48 --highpass 0.001 --lowpass -1 --clipdetect aaa.wav encoded1.mp3 --bitwidth 24 --interch 0.0002 --scale 1.5

@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
Now, the samples, the spectograms and the ABX test.

-original.wav

-encoded1.mp3 (this sample was encoded with the above mentioned psy settings)
lame.exe -mj --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.25 --ns-sfb21 +3.75 --short --verbose -q0 -b320 --cbr --resample 48 --highpass 0.001 --lowpass -1 --clipdetect aaa.wav original.wav --bitwidth 24 --interch 0.0002
[--ns-bass -1.0 --ns-alto -0.5 --ns-treble -0.275 --ns-sfb21 -4 --interch 0.0002]

-encoded2.mp3 (this sample was encoded with the default psy settings)
lame.exe -mj --short --verbose -q0 -b320 --cbr --resample 48 --highpass 0.001 --lowpass -1 --clipdetect original.wav encoded2.mp3 --bitwidth 24
[--ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.025 --ns-sfb21 -0.5 --interch 0.0000]
@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@

Conclusion:

I did not find significant differences in spectograms or audition, and ABX tests demonstrate that there is uncertainty in both situations (original.wav vs encoded1.mp3 / original.wav vs encoded2.mp3).

The psy settings --ns-bass, --ns-alto, and --ns-treble appears to have very little impact. Maybe --ns-sfb21 and --interch may be a game changer, I dunno.











Re: best mp3 encoder with something better than a command line interface?

Reply #33
Spectrograms are useful for looking at analog equipment, but for a perceptual encoder they can't tell you anything. The "denoise" you see is just your imagination. If you were making changes based on that, then your changes are random. That is a bad way to do things.

If you couldn't ABX the original file then what you are doing is pointless; you can't improve on something that doesn't have any audible flaws. You need to find something that can be improved first.

If you want to learn about audio codecs, I suggest starting at lower bitrates, maybe 100 kbps, and then finding files where you can hear a problem. Then you can try tuning. However, this is going to be immensely more difficult then you're assuming, and will require you to have a much deeper understanding of how perceptual encoders actually work.

Re: best mp3 encoder with something better than a command line interface?

Reply #34
I feel like I should bin all this "eyes are for listening" nonsense.  It is in no way advancing the original discussion, or even worth having in the first place.

It cannot be helped if BrilliantBob is unwilling to accept the fact that he is misguided.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: best mp3 encoder with something better than a command line interface?

Reply #35
I feel like I should bin all this "eyes are for listening" nonsense.  It is in no way advancing the original discussion, or even worth having in the first place.
It cannot be helped if BrilliantBob is unwilling to accept the fact that he is misguided.

LAME v3.100 64bit. ATH: using type: 5 (what is it?)

I experimented many types of settings in LAME to obtain a quality .mp3, as transparent as possible related to the original .wav. I disabled Y using VBR. Now the encoding speed is double. I used this batch file for faster bulk encoding (a possible solution too for dnewhous who started this topic)
Code: [Select]
@echo off
@setlocal
color 1E
echo  ÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜÜ
echo  Ý                                          ÚÄÄÄ¿Þ
echo  Ý         LAME v3.100 64bit unleashed      ³ û ³Þ
echo  Ý                                          ÀÄÄÄÙÞ
echo. ßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßß
echo.

:: this batch is very fast for bulk encoding. drag-and-drop the .wav in the batch window or write it with quotes ("FileName.wav")
:: the encoded .mp3 is as transparent as possible related to the .wav source
:: double encoding speed compared with 320 CBR
:: highpass filter disabled. polyphase lowpass filter disabled.
:: Y disabled * VBR 0 enforced to 320 kbps 48,000 Hz 24bit
:: ATH: using type: 5 (what is it?)
:: interchannel masking ratio: 0.0002
:: using LR stereo
::      for access to dev settings (--help dev) put "#define _ALLOW_INTERNAL_OPTIONS 1" in parse.c and compile
:: default lame psychoacustic tuning:                        --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.025 --ns-sfb21 0.5
:: adjusted psy masking: --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 (psychoacoustic disabled?)


echo ---------------------------
Set /P _infile=drag/enter source:
echo ---------------------------
set _outfile=%_infile:~0,-5%"

lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --resample 48 --bitwidth 24^
 --lowpass -1 --highpass 0.001 --clipdetect %_infile% %_outfile%.mp3 --scale 1
ECHO 

:reload
echo ---------------
Set /P _scale=new scale:
echo ---------------
lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --resample 48 --bitwidth 24^
 --lowpass -1 --highpass 0.001 --clipdetect %_infile% %_outfile%.mp3 --scale %_scale%
ECHO 

echo ----------------------------
     Set /P _abort=exit? (y, *):
echo ----------------------------
If /i "%_abort%"=="Y" goto terminate
goto reload
:terminate
pause

LAME returned these messages and I saw ATH: using type: 5
Code: [Select]
Warning: highpass filter disabled.  highpass frequency too small
LAME 3.100 64bits (http://lame.sf.net)
Resampling:  input 44.1 kHz  output 48 kHz
polyphase lowpass filter disabled
Encoding E:\music\09-metallica-of_wolf_and_man.wav
      to E:\music\09-metallica-of_wolf_and_man.mp3
Encoding as 48 kHz stereo MPEG-1 Layer III VBR(q=0)

misc:

        scaling: 1
        ch0 (left) scaling: 1
        ch1 (right) scaling: 1
        huffman search: best (outside loop)
        experimental Y=0
        ...

stream format:

        MPEG-1 Layer 3
        2 channel - stereo
        padding: all
        variable bitrate - VBR mtrh (default)
        using LAME Tag
        ...

psychoacoustic:

        using short blocks: channel coupled
        subblock gain: 1
        adjust masking: -6.8 dB
        adjust masking short: -6.8 dB
        quantization comparison: 9
         ^ comparison short blocks: 9
        noise shaping: 1
         ^ amplification: 2
         ^ stopping: 1
        ATH: using
         ^ type: 5
         ^ shape: 1 (only for type 4)
         ^ level adjustement: -7.1 dB
         ^ adjust type: 3
         ^ adjust sensitivity power: 1.000000
        experimental psy tunings by Naoki Shibata
           adjust masking bass=-8.5 dB, alto=-8.25 dB, treble=-8.025 dB, sfb21=-
15.5 dB
        using temporal masking effect: no
        interchannel masking ratio: 0.0002
        ...

    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
 10695/10695 (100%)|    0:43/    0:43|    0:44/    0:44|   5.8555x|    0:00
256 [   20] %
320 [10675] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
-------------------------------------------------------------------------------
   kbps        LR  %     long switch short %
  319.9      100.0        93.7   3.6   2.6
Writing LAME Tag...done
ReplayGain: -5.9dB
WARNING: clipping occurs at the current gain. Set your decoder to decrease
         the  gain  by  at least 0.7dB or encode again using  --scale 0.92
         or less (the value under --scale is approximate).

---------------
new scale:

Never heard before about ATH type 5. I don't post spectrograms to not make people angry, but now, the encoded .mp3 looks like the original .wav. And sounds identical.

Re: best mp3 encoder with something better than a command line interface?

Reply #36
but now, the encoded .mp3 looks like the original .wav. And sounds identical.

Please Follow the TOS especially #8

Atm anything you have said ant be considering of any value much above a drunken mans tale.
You have completely lacked the ability to prove any of your statements  even though you have been asked for several times.

The only thing you have proven so far is a complete lack of understand on any of the tools you have used
- Resampling
- ABX testing
- Spectrograms

Not understading the usage of the tools and not listening to any advice on how to do things correctly is not helping your case.

Even if we regard the clear lack of understanding of tools and lack of evidence. the entire synopsis of your case is outright wrong.
You start by some made up belief that something is wrong and just goes out to say it fixed now. Not even analyzing if your initial issue is present or not.

If you want your case to be taken any kind of serious you need to seriously understand how the basis of proving by evidence works.
All advice I would be able to give you has already been mentioned by others. Try listening to them.


Quote of TOS #8
TOS 8. All members that put forth a statement concerning subjective sound quality, must -- to the best of their ability -- provide objective support for their claims.  Acceptable means of support are double blind listening tests (ABX or ABC/HR) demonstrating that the member can discern a difference perceptually, together with a test sample to allow others to reproduce their findings.  Graphs, non-blind listening tests, waveform difference comparisons, and so on, are TOS not acceptable means of providing support.

For that alone you post should be deleted as the inane rambling it appears to be.
You have been giving plenty of leeway to try to adhere to this, but have rejected any and all of them.
Sven Bent - Denmark

Re: best mp3 encoder with something better than a command line interface?

Reply #37
Please Follow the TOS especially #8
 
 

I asked what is "ATH: using type: 5"

Samples and ABX tests for my settings will follow, working in progress, but I want to know first about this ATH type. Can you explain me about ATH type 5 please?


Re: best mp3 encoder with something better than a command line interface?

Reply #38
Quoting the mod telling you to stop violating ToS and replying with the same nonsense you're in trouble for is not only disrespectful to the community, it is very foolish.

Re: best mp3 encoder with something better than a command line interface?

Reply #39
best mp3 encoder with something better than a command line interface

Well, it's a matter of taste, for me there isn't *anything* better than command line - which is more ergonomic, more accurate, quicker to use...

Re: best mp3 encoder with something better than a command line interface?

Reply #40
Samples and ABX tests for my settings will follow, working in progress, but I want to know first about this ATH type. Can you explain me about ATH type 5 please?


That's a pretty meaningless ABX test. You don't need to prove that you can't distinguish a encoded file from the source.

What you should be doing: Post an ABX where you successfully distinguish the (in your opinion) inferior Mp3 with default encoder settings and same bitrate from lossless source. Then you should do ABC/HR tests to show that you can distinguish the two Mp3s from each other and that you consistently rate the one with your snakeoil commandline settings as higher fidelity.

Re: best mp3 encoder with something better than a command line interface?

Reply #41
That's a pretty meaningless ABX test. You don't need to prove that you can't distinguish a encoded file from the source.

What you should be doing: Post an ABX where you successfully distinguish the (in your opinion) inferior Mp3 with default encoder settings and same bitrate from lossless source. Then you should do ABC/HR tests to show that you can distinguish the two Mp3s from each other and that you consistently rate the one with your snakeoil commandline settings as higher fidelity.
I apologize if someone felt embarrassed by my posts. It was not on purpose. I'm new to this forum and I do not know the procedures very well. Others have posted spectrograms and have not been admonished for it. They are only indicative, OK, they do not have scientific value. But as you can see from the posts of others, nobody knows all. I thought the forums were for questions, discussion and exchange of ideas, I do not think they are canonical churches with rigid rules like "forget it and stick with the default settings."

I'm trying to find a combination of settings in LAME to get a .mp3 as transparent as possible, regardless of size. For storage on android. Maybe it's silly what I do, I do not know, but it's worth trying. I'll complete the analysis with the evidence suggested by Deathcrow, which I thank, and let you draw the conclusions. Thanks for understanding and patience.

Re: best mp3 encoder with something better than a command line interface?

Reply #42
I thought the forums were for questions, discussion and exchange of ideas, I do not think they are canonical churches with rigid rules like "forget it and stick with the default settings."

It's not forbidden to ask questions. People are just pointing out to you that it is extremely unlikely that you found some secret 'magic' settings for lame that are better than the default settings. LAME is a very mature codec, that has been worked on for many many years by people much more experienced than you in psychoaccustics.

It comes across very arrogant and exposes the Dunning-Kruger-Effect if you think you can improve general applicability by tweaking some command line settings.

Re: best mp3 encoder with something better than a command line interface?

Reply #43
R3mix? dibrom?

You haven't the faintest clue how this forum started.

We aren't angry, we simply know that you have no idea what you're doing, yet you insist that you do. Please, stop wasting our time. I would say your endeavors are useless, however, they will prove useful if you come away from this learning why what you're doing is the wrong way round.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: best mp3 encoder with something better than a command line interface?

Reply #44
It's not forbidden to ask questions. People are just pointing out to you that it is extremely unlikely that you found some secret 'magic' settings for lame that are better than the default settings.
You haven't the faintest clue how this forum started.
Certain extended LAME settings outsmart the default settings and preserves the original quality.

The original.wav sample was extracted from the "Alan Parsons Project - I Robot (1977) - 10 - Genesis Ch.1. V.32.wav" as you can hear in any original CD.

original.wav  = the sample extracted from the original CD file
MyEnc.mp3      = the sample encoded with my extended settings of LAME.
DefaultEnc.mp3 = the sample encoded with the default LAME settings.
Code: [Select]
1. The default settings
================================
lame original.wav DefaultEnc.mp3
================================
2. My settings
=======================================================================================================================================================================
lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --lowpass -1 --highpass 0.001 original.wav MyEnc.mp3
=======================================================================================================================================================================
C:\>lame original.wav DefaultEnc.mp3
LAME 3.100 64bits (http://lame.sf.net)
Using polyphase lowpass filter, transition band: 16538 Hz - 17071 Hz
Encoding original.wav to DefaultEnc.mp3
Encoding as 44.1 kHz j-stereo MPEG-1 Layer III (11x) 128 kbps qval=3
    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
  1169/1169  (100%)|    0:02/    0:02|    0:02/    0:02|   10.469x|    0:00
-------------------------------------------------------------------------------
   kbps        LR    MS  %     long switch short %
  128.0        1.0  99.0        96.6   1.8   1.6
Writing LAME Tag...done
ReplayGain: -8.0dB
............................................................................................
C:\>lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --
verbose -V0 -b256 -B320 -F --lowpass -1 --highpass 0.001 original.wav MyEnc.mp3
Warning: highpass filter disabled.  highpass frequency too small
LAME 3.100 64bits (http://lame.sf.net)
polyphase lowpass filter disabled
Encoding original.wav to MyEnc.mp3
Encoding as 44.1 kHz stereo MPEG-1 Layer III VBR(q=0)

misc:

        scaling: 1
        ch0 (left) scaling: 1
        ch1 (right) scaling: 1
        huffman search: best (outside loop)
        experimental Y=0
        ...

stream format:

        MPEG-1 Layer 3
        2 channel - stereo
        padding: all
        variable bitrate - VBR mtrh (default)
        using LAME Tag
        ...

psychoacoustic:

        using short blocks: channel coupled
        subblock gain: 1
        adjust masking: -6.8 dB
        adjust masking short: -6.8 dB
        quantization comparison: 9
         ^ comparison short blocks: 9
        noise shaping: 1
         ^ amplification: 2
         ^ stopping: 1
        ATH: using
         ^ type: 5
         ^ shape: 1 (only for type 4)
         ^ level adjustement: -7.1 dB
         ^ adjust type: 3
         ^ adjust sensitivity power: 1.000000
        experimental psy tunings by Naoki Shibata
           adjust masking bass=-8.5 dB, alto=-8.25 dB, treble=-8.025 dB, sfb21=-15.5 dB
        using temporal masking effect: no
        interchannel masking ratio: 0.0002
        ...

    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
  1169/1169  (100%)|    0:03/    0:03|    0:03/    0:03|   7.6477x|    0:00
256 [   0]
320 [1169] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
-------------------------------------------------------------------------------
   kbps        LR  %     long switch short %
  320.0      100.0        95.5   2.3   2.2
Writing LAME Tag...done
ReplayGain: -8.5dB

C:\>

The Results:

MyEnc.mp3 is almost identical with the original.wav and is clearly different than DefaultEnc.mp3, as you see in the ABX tests.

The BASS area in the DefaultEnc.mp3 is boosted (sounds more loud) probably due to the [ATH type 4] and/or the psychoacoustic default settings.

MyEnc.mp3 was encoded with my experimental LAME settings (including the dev settings ON), with [ATH type 5] and the psychoacoustic settings lowered to minimum. And sounds almost identical with the original.wav (no BASS boost). According with the LAME developers from 2001, the ATH type 5 is based on collected data. The other LAME ATH curves are simply based on experimental data. https://sourceforge.net/p/lame/mailman/message/4770856/

The HF 16,000 Hz-20,000 Hz area wiped out by the LAME default settings is irrelevant for people who can't hear anything in this range. Maybe the youngers may feel any differences. Not my case.

It's up to you to judge if my LAME settings are snakeoil or not. Feel free to comment.
         

    

Re: best mp3 encoder with something better than a command line interface?

Reply #45
MyEnc.mp3 is almost identical with the original.wav and is clearly different than DefaultEnc.mp3, as you see in the ABX tests.

Two problems here:

1)  You are comparing a 128k mp3 to a 320k mp3.  Most people will assume that the higher bitrate file will be at least as good as the lower bitrate file, so this is not a good comparison.

2)  In spite of the comparison presumably favoring the higher bitrate file, you failed to show that it is better.

So we have a comparison that is not useful and then a failure to get a result.   Why even post those results?

Re: best mp3 encoder with something better than a command line interface?

Reply #46
Two problems here:
1)  You are comparing a 128k mp3 to a 320k mp3.  Most people will assume that the higher bitrate file will be at least as good as the lower bitrate file, so this is not a good comparison.
2)  In spite of the comparison presumably favoring the higher bitrate file, you failed to show that it is better.
So we have a comparison that is not useful and then a failure to get a result.   Why even post those results?
Oops, I will test again by re-encoding the default.mp3 with the "--preset insane" included to see if the <bass boost> is still there.

So, <lame --preset insane original.wav DefaultEnc.mp3>
against
<lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --lowpass -1 --highpass 0.001 original.wav MyEnc.mp3>
is it OK?
What other tests to make?


Re: best mp3 encoder with something better than a command line interface?

Reply #48
https://hydrogenaud.io/index.php/topic,116217.msg963803.html#msg963803
lame --preset insane --verbose original.wav DefaultEnc.mp3
vs
lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --lowpass -1 --highpass 0.001 original.wav MyEnc.mp3
Code: [Select]
C:\>lame --preset insane --verbose original.wav DefaultEnc.mp3
LAME 3.100 64bits (http://lame.sf.net)
Using polyphase lowpass filter, transition band: 20094 Hz - 20627 Hz
Encoding original.wav to DefaultEnc.mp3
Encoding as 44.1 kHz j-stereo MPEG-1 Layer III (4.4x) 320 kbps qval=3

misc:

        scaling: 1
        ch0 (left) scaling: 1
        ch1 (right) scaling: 1
        huffman search: best (outside loop)
        experimental Y=0
        ...

stream format:

        MPEG-1 Layer 3
        2 channel - joint stereo
        padding: off
        constant bitrate - CBR
        using LAME Tag
        ...

psychoacoustic:

        using short blocks: channel coupled
        subblock gain: 1
        adjust masking: -10 dB
        adjust masking short: -11 dB
        quantization comparison: 9
         ^ comparison short blocks: 9
        noise shaping: 1
         ^ amplification: 1
         ^ stopping: 1
        ATH: using
         ^ type: 4
         ^ shape: 0 (only for type 4)
         ^ level adjustement: -12 dB
         ^ adjust type: 3
         ^ adjust sensitivity power: 1.000000
        experimental psy tunings by Naoki Shibata
           adjust masking bass=-0.5 dB, alto=-0.25 dB, treble=-0.025 dB, sfb21=0.5 dB
        using temporal masking effect: yes
        interchannel masking ratio: 0
        ...

    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
  1169/1169  (100%)|    0:03/    0:03|    0:03/    0:03|   9.1483x|    0:00
-------------------------------------------------------------------------------
   kbps        LR    MS  %     long switch short %
  320.0       88.0  12.0        95.9   2.1   1.9
Writing LAME Tag...done
ReplayGain: -8.5dB

...

C:\>lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --
verbose -V0 -b256 -B320 -F --lowpass -1 --highpass 0.001 original.wav MyEnc.mp3
Warning: highpass filter disabled.  highpass frequency too small
LAME 3.100 64bits (http://lame.sf.net)
polyphase lowpass filter disabled
Encoding original.wav to MyEnc.mp3
Encoding as 44.1 kHz stereo MPEG-1 Layer III VBR(q=0)

misc:

        scaling: 1
        ch0 (left) scaling: 1
        ch1 (right) scaling: 1
        huffman search: best (outside loop)
        experimental Y=0
        ...

stream format:

        MPEG-1 Layer 3
        2 channel - stereo
        padding: all
        variable bitrate - VBR mtrh (default)
        using LAME Tag
        ...

psychoacoustic:

        using short blocks: channel coupled
        subblock gain: 1
        adjust masking: -6.8 dB
        adjust masking short: -6.8 dB
        quantization comparison: 9
         ^ comparison short blocks: 9
        noise shaping: 1
         ^ amplification: 2
         ^ stopping: 1
        ATH: using
         ^ type: 5
         ^ shape: 1 (only for type 4)
         ^ level adjustement: -7.1 dB
         ^ adjust type: 3
         ^ adjust sensitivity power: 1.000000
        experimental psy tunings by Naoki Shibata
           adjust masking bass=-8.5 dB, alto=-8.25 dB, treble=-8.025 dB, sfb21=-15.5 dB
        using temporal masking effect: no
        interchannel masking ratio: 0.0002
        ...

    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
  1169/1169  (100%)|    0:04/    0:04|    0:04/    0:04|   7.4718x|    0:00
256 [   0]
320 [1169] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
-------------------------------------------------------------------------------
   kbps        LR  %     long switch short %
  320.0      100.0        95.5   2.3   2.2
Writing LAME Tag...done
ReplayGain: -8.5dB

C:\>
I put all the result files in archive. I can't process all results files with this abc/hr java thing. The ABX results: 18/18 positive.

You can process files again by restoring the archived ABC/HR session, config and results. There is my ABX test result.

My conclusion is my settings are better than the default settings, even at 320 kbps encodings. MyEnc.mp3 sounds like original.wav(flac) and sounds different than DefaultEnc.mp3. In this particular case, the HF distortions of the MyEnc.mp3 and original.wav(flac) sound identical, but the HF distortions of DefaultEnc.mp3 are different.

The differences are not very significant or annoying, I think a normal ear with a normal audio system can't hear the differences.

Re: best mp3 encoder with something better than a command line interface?

Reply #49
Quote
ABX Results:
E:\abchr-java-0.5b\abchr-java\MyEnc.mp3 vs E:\abchr-java-0.5b\abchr-java\DefaultEnc.mp3

Do I understand correctly that you did the test wrong for the third time?

 
SimplePortal 1.0.0 RC1 © 2008-2018