Shouldn't we upgrade the 20-20 audible range ?!

[pablolie]

Mp3 resasemples, it's graph analyser sean and heard when shown (otherwise you probably won't). I whose less interested in decades old codec to hybrid losy Opus and my starting assumption it will fair much better where wrong. Used Power DB to minimise resasampling artifacts and did transcoding locked to single thread. For polifonic up to 32 bit aprouch and when both codecs are used along with smart bandwidth limit it's better on average 128~160 kbps (including all variants of Farens Mpeg and other encoders and same about AAC). You nead to bare in mind shaping and cutting entirely off are different things. You where lazy too search (so am I). So I went evoluting WavePack in it's hibrid form for the task. Representatively I used 96 KHz 24 bit real master and with specific noise flour and range (including wiled saxophone) & on purpose. Again more than 19 bit we don't need and 16 bit with dithering is fine. Have a nice time.

MP3 discards information that advanced psychoacoustic algortihms deem "perceptually unneccessary". It does NOT cut "everything above 16kHz". At lower bitrates, lossy encoders indeed *do* apply a high-frequency low-pass filter -like you said often around 12-16kHz- to conserve bits but they don't simply hack off the brilliance frequency range.

 

[Sal1950]

IMHO, this is 2026 and we should now be way past the need to save-distribute our music in any lossy form. It's been over a decade (2014) since Tidal opened bandwidth to support the streaming of RedBook or better data rates with all others but Spotify falling in line soon after. Only in very limited cases do we find the need to have to worry over the possibility of inferior SQ due to data loss. I personally find it somewhat comical that in this thread discussing expansion of frequency recording extremes we start to debate about old lossy compression schemes. They're ancient history, send them to the pyramids for archiving.

 

[krabapple]

Lossy perceptual encoding may be effectively obsolete (for stereo; still used for Atmos streaming), but myths about it never seem to die.

 

[ZolaIII]

I assume English isn't your first language, but did you use some translation software?

Anyway, it is not really accurate to state that MP3 (or other perceptual codecs) "resample". They re-encode, into a rather different representation from which the sound waveform is recreated.

If sample rate changes they resasemple and bias errors accure. So you use one you know won't make them and restrict all such tasks to it. It's late, English ain't my first and I don't use translator other than keyboard embedded one. When you say you did it and those where results do you think another language or poetic form would change factical results? If you can see it and hear it (after seeing it) who am the I too disagree? Good night.

 

[Flaesh]

ASR measures distortion/etc at 114dB

Did I miss this?))

 

[RexrothPigeon]

If sample rate changes they resasemple and bias errors accure. So you use one you know won't make them and restrict all such tasks to it. It's late, English ain't my first and I don't use translator other than keyboard embedded one. When you say you did it and those where results do you think another language or poetic form would change factical results? If you can see it and hear it (after seeing it) who am the I too disagree? Good night.

I cannot understand your posts. While that may be my personal failure, I think it would be better for you to write them in your native language and use an LLM or translation app to render them into clear English.

 

[pablolie]

If sample rate changes they resasemple and bias errors accure. So you use one you know won't make them and restrict all such tasks to it. It's late, English ain't my first and I don't use translator other than keyboard embedded one. When you say you did it and those where results do you think another language or poetic form would change factical results? If you can see it and hear it (after seeing it) who am the I too disagree? Good night.

Can we level set? Clearly your post-to-like rate is great. We are not sure -I think- what your exact objections here are. It seems we're aligned, but you end your posts with a bit of a "whatever talk to you whenever" stance? I honestly don't think we disagree other than possibly semantics explained by language.
I just disagreed with the obvious "lossless mean frequencies are cut off at 16k" but did add that psychoacoustic compression indeed often focuses around the 12-18kHz zone. It's where human hearing is fine with some compression of frequencies that overlay on each other and our hearing is fine with that as a general rule. The ultimate SINAD chasers think it crucially relevant, I personally gave up on the chase and just feel privileged to listen to music in the way I do these days.

 

[ZolaIII]

@pablolie I have bad habit of going to details more thrill and sometimes it might sound strange to what it's usually taken as settled up relatively. So everyone will say hearing sensitivity goes up to 110 dB in mids and I will still say 19 bits which is 114 dB but it doesn't make difference and you don't need more. Similar is when I say beyond 12.5 KHz 99% members will say beyond 12, KHz. Of course we don't disagree. SINAD chase... 25 years ago I had a 106 dB SINAD DAC and it's fine now as it whose then. No I don't chase SINAD did persue DR and to a source reproduction to extent low THD to analog transcevers and that they deliver (including room treatment). Have my multimedia and listening room along with great neighbours (rarely hire and not really complaining as they are mostly performers) which I don't use almost for a year and that will probably continue to a next spring. I am lacky also that I have very low environment noise. 20~20 is so ridiculous without psy how we hear and compensation for it.
@RexrothPigeon well other then my bad English it's really me to blame other then that people tend not too read them in order they were written. Person I directly replied usually does and understand as I eventually in furure posts as conversation goes explain if something whosent clear before but I don't explain it all from a beginning again.
In the end modern high completely losy codecs as WavePack can be set to produce results in which you can't hear or see difference using audio annalise graph reading to a master, saving to loseles compression with compression as flac will in such case vary too what the source bit and frequency ranges are and all do it won't be much for red book (about 25% to 45~60 for ones more than 44100 Hz 16 bit). It also can save what ever it shaved off in hybrid form and use it to reconstruct back master as it whose 1:1 so not a bad way too preserve them either. When you are big content on demand provider the savings in overall bandwidth of 25~30% is huge especially for same (in this case not relative) quality and aim is full fidelity regarding me (not them they just want saving). Telling its not relevant anymore has no sense at all. It's more prominent with video and future development of AV1 and it's successors but keep in mind all those videos go with lossy audio and you do watch such most of the time. I don't want to drag anyone in this discussion but those who spend part of their life developing WMA and come to similar conclusions regarding shaving and required bandwidth (old loseles preset in it) certainly won't see it differently then I am.

 

[Julf]

If sample rate changes they resasemple and bias errors accure. So you use one you know won't make them and restrict all such tasks to it. It's late, English ain't my first and I don't use translator other than keyboard embedded one. When you say you did it and those where results do you think another language or poetic form would change factical results? If you can see it and hear it (after seeing it) who am the I too disagree? Good night.

Precise terminology is important. Re-encoding is different from resampling. As to poetic form, you do very well. Have a good day!

 

[Julf]

@pablolie I have bad habit of going to details more thrill and sometimes it might sound strange to what it's usually taken as settled up relatively. So everyone will say hearing sensitivity goes up to 110 dB in mids and I will still say 19 bits which is 114 dB but it doesn't make difference and you don't need more. Similar is when I say beyond 12.5 KHz 99% members will say beyond 12, KHz. Of course we don't disagree. SINAD chase... 25 years ago I had a 106 dB SINAD DAC and it's fine now as it whose then. No I don't chase SINAD did persue DR and to a source reproduction to extent low THD to analog transcevers and that they deliver (including room treatment). Have my multimedia and listening room along with great neighbours (rarely hire and not really complaining as they are mostly performers) which I don't use almost for a year and that will probably continue to a next spring. I am lacky also that I have very low environment noise. 20~20 is so ridiculous without psy how we hear and compensation for it.
@RexrothPigeon well other then my bad English it's really me to blame other then that people tend not too read them in order they were written. Person I directly replied usually does and understand as I eventually in furure posts as conversation goes explain if something whosent clear before but I don't explain it all from a beginning again.
In the end modern high completely losy codecs as WavePack can be set to produce results in which you can't hear or see difference using audio annalise graph reading to a master, saving to loseles compression with compression as flac will in such case vary too what the source bit and frequency ranges are and all do it won't be much for redwood (about 25% to 45~60 for higher ones). It also can save what ever it shaved off in hybrid form and use it to reconstruct back master as it whose 1:1 so not a bad way too preserve them either. When you are big content on demand provider the savings in overall bandwidth of 25~30% is huge especially for same (in this case not relative) quality and aim is full fidelity regarding me (not them they just want saving). Telling its not relevant anymore has no sense at all. It's more prominent with video and future development of AV1 and it's successors but keep in mind all those videos go with lossy audio and you do watch such most of the time. I don't want to drag anyone in thit discussion but those who spend part of their life developing WMA and come to similar conclusions regarding shaving and required bandwidth (old loseles preset in it) certainly won't see it differently then I am.

I assume you mean "red book" (as in the IEC 60908 CD-DA standard) when you write "redwood"?

 

[ZolaIII]

I assume you mean "red book" (as in the IEC 60908 CD-DA standard) when you write "redwood"?

Yes sorry I should really use proper sample bit rate instead. Usually I don't drift that far, it whose late and I whose tired.

 

[Apples]

Most microphones roll off the highs. Took me a while to realize you don't need to do that.

 

[birb]

i dont care about this number, it's the AVERAGE human range, also most of the time the source material is 44.1k which goes up to 22.05khz in the audible range. movies and video use 48k for proper audio/video sync. and unless you have some super crap audio gear (Temu or whatever). it's a non-issue. also i am 30 and can still hear up to 18,500hz so to me it doesnt matter

 

[fpitas]

100dB is 'fairly ok'? Not for anything more than a short time, if you want to keep your hearing intact.

Lol! Still, it might be fun to torture your listeners with supersonic tones. Well, if you hate your listeners.