MP3 freeformat bitstream support
 

I know it's not a requirement of a decoder to support it, but since it is part of the MP3 spec, will nullsoft's MP3 decoder ever support bitates above 320kbps anytime soon.

I already know about the MAD plugin, but the problem with that is no unicode filename or ID3v2 support at all

Is there any point in encoding about 320kpbs?

Why would you want to go over 320kbps anyway?? It's almost impossible to tell the difference between 256kbps and AudioCD, let alone 320...

Unless you have a new mp3 output system that allows for more than stereo, say surround, there is no point to having an mp3 bitrate of above 256KB/s.

Damn straight. 256kbps has been proven to be CD quality.

"You ain't be needin' no more."

I'm fairly certain that the MP3 spec does not allow for open-ended bitrates, anyway. Where are you going to get an encoder that will produce greater than 320kbps, anyway? I've never seen one. For that matter, how do you know that Nullsoft's decoder doesn't support 740kbps MP3s? It would be a bit difficult to test, with no MP3s to test on.

If you really want to be so unnecessarily anal about quality, you need to look into using a different type of encoding. If you need anything greater than 320 (and you don't), you need to go look at some sort of lossless compression.

MP3 spec goes up to 640kbps, but decoders are only required to work with 320 and below. I already tried to open a 640k mp3 with nullsoft's decoder and it didn't work.

and about not needing anything above 320k, why not, if you want to encode for absolutely perfect archival quality, why not encode to 640k, especially with all encoder filters off (lame -k). without the filters there's no way you can encode a file without a loss in quality at or below 320k, whether you can actually hear it or not.

Suirely, there is utterly no point in encoding an MP3 at 640kpbs unless you have supersonic hearing to pick out the errors?

If you can't hear it, you don't need it. If you do need it, go use some other format. There is ALWAYS quality loss with MP3, no matter what the bitrate is, so if you need perfect quality, then you need something else.

with bitrates like that.... you may as well use FLAC (non lossy audio format)

or ogg vorbis (more than 2 audio channels)

You need to go to these forums and ask there:

http://www.hydrogenaudio.org

If you're archiving music then mp3 is definitely not for you. The folks at hydrogenaudio will give you expert advice.

Good luck!

You people are no help, I ask for a feature request to comply with a fuller mp3 spec, and all I get is criticized for theoretical uses of unneccessary bitrates:D

I think that before implementing a new feature in a program, any programmer must first think about the necessity of including such feature. There's no point in including a feature just for the sake of it or to be fully compliant with the specifications. MP3 is supposed to be a compression format hence meaning a significant reduction of the file size. Encoding at 640 kbps (and even 320 kbps imho) defeats the purpose. If you really want perfect archival quality, then lossless compression is indeed the way to go.
I personally recommend LPAC

considering that uncompressed wav audio at 44100 16bit stereo is 1411kbps (1384kbps plus headers and such), 640kbps is technically compression, of 2.204:1 and just as lossless as the real lossless formats.

plus, hypothetically what happens if you have a 96khz 24bit (~2900-3000kbps) file and want to make an MP3 (or one of the lossless format) from that without resampling and bit-dithering (all contingent upon an encoder actually accepting that format), 320k just won't cut it, but 640kbps would be a ~4.531-4.687:1 compression

and I've used them all already also
Monkey's audio suports any sampling rate, and up to 24bit, but the compression is only ~2:1, no matter the WAV's format, you got 44100 16bit, you get ~700kbps, you got 96000 25bit, you get ~1500kbps
wavepack roughly same compression ratio unless using hybrid mode (making it lossy) and only up to 48khz 16bit
flac roughly same compression as wavepack and MAC unless using hybrid mode (making it lossy) and same format
I'll point you here for a comparison chart of the lossless formatsI'll point you here for a comparison chart of the lossless formats

so, where is the advantage of a 640kbps MP3 again? more actual compression over lossless formats

and there's also OGGVorbis, does nullsoft's decoder work with it's maximum bitrate of 512kbps? and that does cover more than just 2 channels

According to the docs I read on the official Lame website, only one decoder can handle 640 kbps files. It's MAD so I guess you'll have to use that or stick to lower bitrates.
But, as is stated in said docs, decoders are only required to decode up to 320 kbps. :D

For the love of all that is holy, MP3 is never lossless. Never. Never. Never. You always lose data. That's what MP3 does. It compresses and throws away information that's not important. You will never, ever, get an MP3 that is actually of the same quality as the WAV it was produced from. If that's what you need, then you cannot use MP3. If you are only going to get ~2:1 compression with 640kbit MP3, then use a lossless compression. It'll be full quality, and be the same size.

I'm not replying to the rest of that post, because hypothetical situations are crap. I don't know where you would get a 96khz 24bit WAV file, and I don't care. Dither it. Reduce the bitrate. Your ears cannot hear it.

The fact is that you can't distinguish 256/320 kbps mp3 files from audio cd tracks (and therefore files compressed losslessly). Blind tests prove it. These bitrate values represent the threshold of top quality, this is why most decoders are not required to support higher values.
256/320 kbps compression is practically as good as lossless compression, encoding at a higher bitrate just increases the size of the file without adding quality.

I know that. I already said that earlier. But, if someone wants to be anal about quality they can't possibly percieve, then there is no reason to use a lossy format at all.

I second that :)

Btw,
The --freeformat attribute in Lame seems to allow that as stated in the documentation

LAME will produce a fixed bitrate, free format bitstream.
User must specify the desired bitrate in kbps, which can
be any integer between 8 and 640.

Not supported by most decoders. Decoders only required to support
up to 320 kbps.

As to the decoders, Winamp's one can't support more than 320 kbps.
From the Lame website

Decoders which can handle free format:
supports up to
MAD 640 kbps
"lame --decode" 550 kbps
Freeamp: 440 kbps
l3dec: 310 kbps


That being said, I totally agree with George, all this is purely theoretical :) ;)

That's good to know. I actually thought that he was talking about open-ended bitrates at first, when he said freeform. I thought he was saying that any bitrate (320076, even) should be accepted.

I'm glad to know what he really meant, though of course, I still think it's unnecessary. :D

I actually tried it, George :D
I created a 640 kbps with Lame and played it in Winamp with the MAD decoder. Therefore, it is possible to play such files in Winamp. However, the 3-minute song I encoded reached something like 17 Mb. I then encoded it with several lossless compression encoders (LPAC, Wavpack, FLAC) and all of them reached approximately 22 Mb :D

actual lossless ratios against 640kbps MP3 and 512kbps Vorbis using:
song: Natalie Imbruglia - Impressed
source: fressly ripped from CD using EAC

code:


18,197,632B   512kbpe OGGVorbis (OggDropXP at CBR512)    2.792:1

23,052,538B   640kbps MP3 (LAME --freeformat -b640 -k)   2.204:1

37,037,856B   Monkey's Audio Extra High                  1.372:1

37,409,229B   RKAu                                       1.358:1

37,619,663B   WavPack                                    1.350:1

37,757,915B   LPAC                                       1.345:1

40,190,537B   Shorten                                    1.264:1

50,819,296B   Original size of Wave file                 1.000:1

By that, it's obvious that what I said before about the lossless codecs being ~2:1 was way off, they don't come anywhere close. and they all sound absolutely identical and by visual waveform (open in soundforge or something similar and zoom in) they are identical also

Twilightseer:
I know MAD works, but it doesn't come with the package (minor), and it doesn't support unicode filenames or ID3v2 tags (both of which I use, so major)

Well then you're screwed I guess :D
I don't think that the Winamp mp3 decoder will ever support these bitrates. Good luck in finding a solution.

Wow, so you wasted a lot of space. :D

Not quite. I used my 60 Gb HD which currently has 42 Gb of space left.
Not a reason to encode my stuff at 640 kbps though :D

You wasted more space than I did. :p I don't think I'll be switching to anal quality any time soon, either. :D