I need to find a better way to convert lossless audio to MP3. Whats the best MP3 encoder currently available?

I need to find a better way to convert lossless audio to MP3. Whats the best MP3 encoder currently available?

Attached: need a better encoder.jpg (1920x2160, 3.45M)

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mp3-converter.com/encoders/mp3_encoder_list.htm,
lame.sourceforge.net/download.php
twitter.com/NSFWRedditVideo

easy cd-da

>mp3
why

Becuase the USB port in my car will only accept MP3 and WMA (already tried AAC, it wont play it)

.wav

why mp3 and not opus? have you tried ffmpeg?

>ffmpeg for audio conversion
o i am laffing
try sox

car will only play MP3 and WMA

ffmpeg is always the answer.

ffmpeg is shit for audio conversion
use sox

hey dumbass... again, my only options are MP3 or WMA

Fraunhofer's patents on their mp3 encoder were supposed to have expired. FDK-AAC is Fraunhofer as well. It's definitely not free as in freedom, but they're the best.

you realize it uses the same libraries as any other program, you idiot

and it doesn't give you any options for encoding

How is sox better?

dude ffmpeg works with like any format. no need for namecalling.

To use FLAC instead.

You didn't even check what ffmpeg was dumbass

for fucks sake, I would love to keep my shit in lossless format, but my car will only read MP3 or WMA

get a better car dumbass

ffmpeg does mp3. I wouldn't be surprised if it does wma.

based

Holy fuck, I can't believe no one is actually telling you. LAME is currently the best MP3 encoder (and ffmpeg includes it as a library but you can use it standalone).

Then choose the highest bitrate your head unit will support and encode all of the music in that.

Really?
So when I add -c:a libvorbis when making a webm for /gif/ then I am not selecting the audio encoding?

can you change the sample rate with ffmpeg? I don't think so

You can with the -ar option.

No, LAME is a free and open sourced encoder. It is not the best.

>can you * with ffmpeg?
Yes.

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just use audacity.

Is there a website, that lists all encoders/decoders for common codecs and perhaps even compares them? Something like mp3-converter.com/encoders/mp3_encoder_list.htm, but not just for mp3.

Can anyone tell a difference between 16-bit in and 32-bit float PCM?

LAME, latest stable version. It's the longest continuously-developed encoder and it has surpassed every other MP3 encoder (including FhG's own) by a significant degree for over a decade now.

You should probably use -V2, which will equate to a ≈192kbps quality-based VBR and will be perceptually transparent on just about everything that MP3 can be perceptually transparent on.

Hello, 20-year-old opinion which predates the Dibrom tuning and probably worried about Xing and BladeEnc. You can put away Radium's crack of FhG's codec too.

Yes. One has more bits.

>It is not the best.
Are you dumb? Literally everyone in the ripping scene uses it for MP3 because it's considered to be the best.

>my opinions are not up to date
lmao what a clown

Even itune's mp3 encoder sounds better than lame. The absolute state of you goys.

Why don't you use foobar with lame? My wav to mp3 v0 conversions are always as good as they can get with mp3.

ok, just give me a couple thousand dollars and I'll get right on that,

LAME is bad when it comes to CBR. Your claim is bullshit for VBR though.

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>set recorder to sound board
>turn on audio recorder
>play song
>record played song
>save as .mp3
sounds easy, son

You can do anything with ffmpeg. Do "--help full" if you aren't totally retarded.

Sounds like you're messing up your encoding settings. Like said, unless you're using CBR (which you should only do for compability reasons), LAME is the superior solution for MP3.

Lame encoder, faggot.

Can you tell a difference though?

Does v0 actually sound better than 320?

256kbps AAC beats the shit out of 320kbps mp3

LAME VBR is perfectly fine. You must be doing something wrong. For CBR, Fraunhofer is better.

Not him, but its for archiving purpose. 16bit audio can max support upto 48k sample rate while 24bit supports beyond that. Those who can hear beyond 22khz are complete buggers though and half of them will fail audiometry test upto 16khz.

Try Mediacoder.

LAME V0 uses only the bits necessary to encode at the best possible audio quality in any given sample. There is no logical reason to be using 320 Kbps to encode parts of audio files that don't require that kind of bitrate - if something DOES require that kind of bitrate that's what VBR takes care of: using what's required when required.

Imagine a 5 minute song with nothing but silence. The V0 version will be relatively small (not 0 bytes, obviously), but the 320 Kbps version would be quite large for no good reason.

I gave up fucking caring anymore and I have 40,000 songs archived as FLAC, whenever I need them for my portable devices I just encode to Opus 128 Kbps VBR and I'm fucking happy.

Stop worrying so much, and use something better, use Opus.

yes
read the docs
-a:r sets sample rate

actual retard

i use flac on the portable devices too. theres no reason to waste time making lossy files anymore

foobar2k + encoder pack.

Real answer, just use LAME directly from the command line.

lame.sourceforge.net/download.php

Attached: LAME.gif (358x231, 3K)

>sourceforge

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Fucking retard.