Hi all,

I'd like to generate a very bad sounding MP3 file, in order to expose as many obvious artifacts as possible.

Also, it would be nice to have a bitrate that's as high as possible (256 or 320kbps).

I've already played with the following options a tiny bit (using LAME 3.92):

LAME.EXE Test.wav
-ms => full stereo, for a nice bitrate waste
-q 9 => worst possible noise shaping
--athonly => disable psychoacoustic model altogether - except ATH
--noshort => optimize for maximal transient smearing
--nores => disable bit reservoir to decrease quality on difficult to encode parts B)
-k => disable all filters, for nice ringing and aliasing artifacts
--strictly-enforce-ISO => should cripple the output file quite a bit.

The sound now has several clear artifacts @ 128kbps, but that's not obvious enough... How would you suggest to decrease the quality even more, while increasing the bitrate ? It should be bad-sounding, even for someone who hasn't heard the original.

I thought about a few "improvements":
=> use short blocks all the way, in order to kill coding efficiency, and to lose most advantages of transform coding
=> re-enable the psymodel - and break it using cmdline-options only.

Transcoding from a 20kbps WMA is not an option... I'd like it to be a pure printine MP3 encode, from the original CD and with an unmodified mp3 encoder.

By the way, it's just for a fun experiment, to see how far we can go.
I love LAME, and no harm is intended !!! BTW, if you think the crime could be commited with a better tool [blade/xing_old/etc] please let me know.

What would you suggest?
Thanks

Try velvet.wav (from mp3dev.org) with Blade @128 It will fit your purpose


Or ask MS - what MP3 encoding setup they use when comparing MP3 to their coding solution

Ivan,

Wow ! what a fast response..

Thank you for the tip, I've just tried it - it sounds nasty but not quite enough..

After a quick comparison, on this sample Blade is (for me) a tiny bit higher quality than with "tweaked" LAME as above :-(

Actually I'd need to make the encoder sound awful on pretty much anything.. and since there doesn't seem to be room for tweaking Blade..

As for MS, that's a clever advice but I usually don't deal with them

Plugger is an unbeatable codec. Even JuraXing is better.
Try Plugger :

http://ec2000.xperiment.net/
Direct link

Thanks for the link Guruboolez !
I must admit, that Plugger codec gives quite amazing treble @ 128kbps with Velvet.wav

However after trying it @320kbps I'd say it doesn't sound shockingly bad (for the average person of course).. well, except with the fatboy sample.

So far, the worst 128k setup (for me) is:
LAME 3.92 -ms -q 9 --athonly --noath --noshort --notemp --nores -k --strictly-enforce-ISO

This is impressive, but not quite sufficient for an increase of bitrate yet...
Maybe someday, I'll try to get back to the origins (2.x?) of LAME, gentlemen

How about -p for a nice waste of bits for CRC?

I love this thread .

Hmm.. try resampling to 48000 as well Why not waste extra bits/sample

Also, is there an option to use dual-channel independent bit allocation (dual mono)? that way the allocation will be fixed to 64 kbps per channel, without a chance to donate bits or to exploit intra-channel redundancy.

Also,

-m i intensity stereo - maybe without psychoacoustic IS can make even more horrible artifacts than dual mono, worth trying

Thank you Ivan !!
Sampling rate bloating.. What a nice way to decrease storage efficiency by almost 10% :-)

I have included the useless CRC calculation from JensRex also.

The new command line is:
lame.exe --resample 48 -ms -q 9 -p --athonly --noath --noshort --notemp --nores -k --strictly-enforce-ISO

Isn't there a way to disable the %@&"~/&* huffman coding ? I mean, with this thing were disabled or at least seriously broken, we could pack more artifacts per kilobyte..

By the way Ivan, I can't seem to enable Intensity stereo with -mi... that's too bad - this option could really do wonders.

not without entirely breaking compatibility with mp3 decoders


kaps = kilo-artifacts per second?

I think this line should be added to list of recommended lame settings.

Oh... you mean "those". You're right, we must keep the huffman codin... but wait !! They could also be used to INCREASE the file size couldn't they?

Too bad they'd bring that benefit losslessly though.. that would have allowed us to go 320kbps while still decreasing quality

Well, you don't have nothing for less than nothing I guess.

KAPS ? Excellent ! The KAPS and NSR (see below) are now an official measure of sound crapness


Agreed. After all, I've just developed the command-line with the highest NSR(*) ® ™ !!

Also, the advantage with these settings, is that transients are no harder to handle than noise... since both produce artifacted noise at the output.

(*) Noise/Signal ratio.

If MP3 can handle custom huffman codebooks, you could make catastrophically poor ones that decrease coding performance by around 800% I would imagine (if the zero-run codes can be changed). A custom tool would be needed of course, or if LAME already creates codebooks internally, the frequency sort could be inverted so the longest codes go the the most common symbols.

-h

oOOOoOOh... 800% better NSR for the same bitrate... this sounds so beautiful to me !
How come no genius had implemented this wonderful idea yet ?!??

Just one thing, we should be careful when choosing the mp3 side-information, because 32-bit addressing might be an issue while huffman coding them...

Especially the CRC.. this will be the most difficult part to losslessly encode !

Also, remember that the requirement is, at least to be able to code silence at 320kbps... of course, we could switch to freeformat if a difficult-to-encode passage (ie. one or more non-zero samples) is to be encoded.

If the inherent limitations of the MP3 format (ie. bitrate ceiling too low for encoding silence ) indeed proves to be an issue, well... I guess I'll develop my own format with highly efficient 2^256-bit huffman codes.. any takers?

I think a really good way of increasing the bitrate while maintaining poorest possible quality (constant ka/s) would be, in addition to the already presented excellent command lines, by activating a dubious VBR mode. Something like -V 7 or -V 8 should do the trick

The problem is that it possibly might sound better than CBR, although I seriously doubt it. Just remember to use a decent bit-eating sample, and you're all set

The modified Huffman tables idea is pretty good, but I consider it cheating. It has to sound like crap, fair and square

Edit:

You also could normalize the sample to 110% or so in order to get some serious clipping...

Edit2:

Nah, artificially introducing clipping would also be cheating. I also thought of using the experimental switches, why not all of them at once? -X -Y -Z should really wreck some havoc.

Also a really simple way to discover ultra-tuned settings is to look around discussion boards for anything that has the magic words "mY LaMeLiNe r0xX, s0uNDs beTTer tHan aps!!!!!1111". The Newbie settings are usually extremely tuned for artifacts.

Instead of using -m s for middle stereo, you can try to use -m d for dual stereo. 2 channels with no coupling / channel distribution might help out a bit. Also, try setting an extremely high low pass filter, such as --lowpass 24. This is too much for 128 to handle. Have fun

PS - If you want to cheat, just set either --lowpass 3 OR --highpass .5 (thats right - .5 - its ALL you need B) )
--highpass .5 will give you the EXACT effect you want

Does --scale allow you to drive the files into clipping? That would just have to sound like shit.

Too bad there is no --allshort switch. Or if there was an --ath raise function, you could jack the noise floor up until it sounded as bad as you want.

Damn right! I have a couple of mp3s encoded with Plugger at 128kbps Stereo and they sound simply wonderful!

Try decreasing ATH with negative number of dB - that should make sound even more beutiful

Also, dual mono stereo coding mode should cripple file a little bit further

I think that you should try Shine...

Damn my stomach hurts now

People.. I didn't expect such strong support ! From the bottom of my heart I'm thanking you (don't know if my ears agree though).


Well, thanks for the suggestion. I'll try to badly enable VBR this way, and hope there'll be some bitrate increase @ constant ka/s.
For the actual sample - even though fatboy.wav is good to measure ka/s - it will be a Dream Theater song. It still contains some nice attacks.

Edit: Wow ! Even -V9 uses 320kbps almost all the time, because of the crippled psychoacoustics I guess. So, we should find a setting where psychoacoustics is enabled, but fails.


Yes, I agree. To get those artifacts, we've got to earn'em fairly.


Well, that's what WMA actually does when encoding.. so I think maybe, we might want to consider this, to get as many advantages of WMA as possible. Does someone know what the WMA scaling factor actually is ?


Oh ! I didn't know there was something better than -ms ! What a sweet hidden setting you found here :-P
This shall be added to the cmdline immediately.
About the lowpass, I've already done the worst possible thing: disabling every filter. You should listen to the real sweet twinkling !
Also, I've tried --lowpass .5 already, and am still considering it.. but maybe that's cheating already.. Hell, isn't full bandwidth, and max ka/s what we want?


Yes.. currently trying this.


Actually --allshort does exist. Before enabling dual mono it didn't sound worse than --noshort to me, but now @ 128kbps it might just decrease the quality if we're lucky :-)
Yeah, I've even tried negative values to increase the absolute threshold of hearing but it didn't seem to change anything.. Maybe a LAME developer could give us directions here?


Shine? haven't heard of it.. (which might be a good sign). Jumping on Google right away !

Updated cmd line is:
lame.exe --resample 48 -md -q 9 -p --athonly --noath --noshort --notemp --nores -k --strictly-enforce-ISO

Please be aware that this lame commandline still sounds better than Plugger on the few samples I tested. Further tweaking is required. Perhaps this level of quality can only be reached via code level modifications.

That's pretty funny.

You should add -V9 --cbr

LOL ! Thank you for the information... but code level modifications are out of the question.. it *should* really be possible to get to the needed level of performance just by commandline tweaking.

Here's a new commandline that uses VBR technology; the ka/s unfortunately lowers of fatboy.wav (because of the 320kbps frames) but it is much more interesting on real-life material:

lame.exe -V9 -q9 --resample 48 --interch 1 -md -p --allshort --notemp --nores -k --strictly-enforce-ISO

Is it possible to tweak the --interch parameter further?

After lots of intensive research, I found this near-optimal VBR cmd line that performs brillantly on both difficult samples and regular music !

Here it is:

lame.exe -V2 -q9 --resample 48 --interch 1 -md -p --allshort --notemp --nores -k --strictly-enforce-ISO
--nspsytune --ns-bass 850 --ns-alto 1000 --ns-treble -2500

I think, we have truly unique quality at attractive bitrates here.

You'll see that a significant amount of care is taken with treble. Sharp high frequencies and very smooth alto should be the word here.

What do you think about this ? :-)

Edit: At this level of tweaking, I think we could now focus on elaborating the ultimate command line - that is using --noshort instead of --allshort. This would bring truly exceptional tonal purity !!

Plugger has been surpassed! Lame now fails miserably on bagpipe music (we need tonal purity here) but it only allocates ca 160kbp/s for that particular sample, so there is still room for improvement. All others were close to 320. "Dumonde - Memory" (trance) surprisingly still sounds comparably good though. It only "chirps out" on each and every bass drum hit. One more thing to try might be the --voice switch.

Try the QDesign MP3 encoder.. might be worth trying, too

I tried QDesign, but it's way better than plugger

Messing with the --ns bass/treble/alto looks a little like cheating. Well, with ath enabled, you may want to play with the --ath-lower switch.

Here's an even better setting.. the output will be a highly-optimized, uniquely-sounding, pseudo-VBR 256kbps file:

lame.exe -V0 -q9 -b256 -B256 --resample 48 --interch 1 -md -p --noshort --notemp --nores -k --strictly-enforce-ISO
--nspsytune --athlower -56 --ns-bass 2 --ns-alto 12 --ns-treble 9

You can see that I've finally solved the transient response issues by disabling the short blocks.

Also, the bitrate allocation is now much more efficient (thanks Gecko !)

Yeah, Plugger is definetely blown out of the water here :-)

The fatboy sample sounds very pure now ! Also, your trance music should encode a bit better now.

Oh shit - that plugger must be very very bad then

I'll try this "LAME" line if I have time.

Note - Intel ships "demo mp3 encoder source code" with IPP 3.0 library - and also it produces pretty bad sound (and it is very fast ;-) also worth trying (requires IPP download )

Hmm... maybe there is a way to tell LAME to use the worst huffman codebook for each scalefactor band and to disable section merging? Gabriel?

Is this completely sain thread?

Imo this is more waste of time than useful...
In worst case some newbie with bad english skills could just read "Here's an even better setting..." and actually start using those.

I think I'm gonna ruin the fun and move this to off-topic eventually..

<mode newbie : ON>

I've just tested this line, and quality is fantastic !! Near CD quality !!! Curves are perfect ! Thanks to hydrogenaudio for giving us the best of audiocoding

<mode newbie : OFF>

[sarcasm]

I think .. the tonal purity is fantastic.. I mean.. FFT graphs are perfect! almost like a original... way better than MPC.. and the bandwidth.. full 24 kHz.. wow

[end of sarcasm]

hehehe

Hey, I've finally reached the ultimate switch!!!. B)
It's a real bit-catastrophe!!!!

I'm currently using mitiok's compile
lame.exe -V0 --preset radio -k -q9 -b256 -B256 --resample 48 --interch 1 -md -p --noshort --notemp --nores -k --strictly-enforce-ISO
--nspsytune --athlower -56 --ns-bass 2 --ns-alto 12 --ns-treble 9

This will ensure PERFECT CD quality, test it by yourself! Enjoy the noise!!! 100% NSR and TAPS [TeraArtifacts per second] guaranteed

Hmm, I tried using these settings with Lame 3.90.2 and Razorlame but the percent completed kept going up and no file was produced!

Have you tested mitiok's compile? Try that, it should work. You'll notice what "transparent" sound is then

WARNING!!!!! It seems that using a front-end (I use Speek') ruins the job and doesn't sound as bad as it should (¿?).
Open a DOS window and set the settings manually. If possible, use short names for the files.

Ok, I used Mitiok's 3.93 compile and sYeLtH's command line, but the result is better than a Plugger 128kbps one I have!

JohnV, yes, I think some brain is functioning a bit around here.. technical sound, purity without transient to kill tonal purity. WMA was beginning only. Here is audiophile truly underwater sound improve. Neurons like simple sound, no superior detail, simplicity is king. We us me do not like no short bloK.. Here is now optimiSzed setting at hand is for experiences as uniqueness.


Yes Ivan.. Tonal purity really iSt like purple sky because all transients are removed.. in fact isT more pure than the original CD ! And no need to equalize when playing the songs back, because all frequencies are melted up together ! This is one big advantage !

Oh, and temporal masking is abolished here.. this was so death ! Pure frequencies, in sweet bubblebath ® ™ sound... what a delightful experience isn't it?

sYeLtH, I must admit, your command line update iSt DeAth ! WMA was iLLnESs only.

Please note, however, that your change just adds "--preset radio". The rest (-k and -q9) were already present before.

I think this addition should be labeled as for hardcore audiophiles only. Regular people won't need this level of quality IMHO.

Of course, I really admire the efficiency of your setting.. especially at these bitrates !

For Shiki:
Don't use any front-end because it will encode faster and better, even if it just sets the settings, it does something that ruins the job.
You MUST open a DOS window, go to the lame directory and then set those parameters EXACTLY as they appear.

As for filenames, USE A SHORT FILENAME on the wav file, so it should appear something like this:

lame.exe -V0 --preset radio -k -q9 -b256 -B256 --resample 48 --interch 1 -md -p --noshort --notemp --nores -k --strictly-enforce-ISO
--nspsytune --athlower -56 --ns-bass 2 --ns-alto 12 --ns-treble 9 D:\musi-k\temp\01.wav

The encoded file will be, then, 01.wav.mp3 .

Enjoy the 200dB NSR and TeraArtifact/s Rate!!!
It sounds like 8kbps mp3.......with ringing!!

Oh goodness, i've decided to test this for fun, and oh @#$! this is horrible beyond words. Great job!
I too thought this thread is a little, ahm, insane, so didn't reply or try it untill now, but now that the ultimate setting has been found, i had to try it.
Anyway, now i feel like encoding the same songs with MPC to feel emotionally better ...

LOL...they should include this in 3.94 as --preset awful

EDIT: OK, so I just encoded something with these settings just for kicks. Maybe --preset WORSTPOSSIBLEQUALITYEVER is more fitting This beyond bad!

--preset 213374U




edit: --alt-preset bullshit <-- sounds even more 1337.

The smearing gives the encodes a Xing type quality to them. In fact if you lower the bitrate to 128kbps you'd swear you were listening to a real Xing encode with the nice flanging effects. Although amazingly I've downloaded worse sounding MP3s (no doubt the result of transcoding).

*tries settings* X-( HOLY LORD this is awful


save me.

Even worse? It is impossible that something sounds worse........if that's like 8kbps!!!
I've tested the same settings, with the same encoder, but with a front-end, and it encodes fast and better. Front-ends do things bad.
In order this switch to work, it has to be done manually, or with a hand-made .bat file placed in the lame directory. Wave filename must be short, like 01.wav.
If it's working, it MUST encode SLOWLY.
If it encodes fast, the result won't be so bad, something goes wrong. Also the SIZE should the same of a CBR 256 file, not a VBR size file.

I'd call this --alt-preset catastrophe

One funny thing, I've noticed that the addition of "--preset radio" actually can remove some instruments from the music :-)

What about calling our new preset: "--preset grammophone" ?