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Could well be, as the indication methods are differently. - The clip indicator at the peak meter detects a clip when two or more consecutive samples are at 0 dB or above (either positive or negative side). If "adjust for DC" is checked (which I don't recommend, as I can not imagine any...

You can get the lame.exe here: http://www.rarewares.org/mp3-lame-bundle.php

That should do it, and is unlikely to exceed the 10 MB on any style: LAME -b 32 -m j -h -V 0 -B 320 -q 0 -k

As I told you in the other thread, there is a "decode to 32-bit"-option in the mp3 options in the "save"-dialog. Disable it, and then load your mp3's. They will be quantized to 16 bit then, and clip at 0 dB. Usually those GUI's come with LAME. You certainly will notice, when LAME is missing, as...

Then you loaded as 32 bit float, I guess. If you load as 16 bit, or if you convert to 16 bit after loading, it will clip. All my software which uses lame saves mp3 files. No wav files. I didn't even know that there is such an option. Does this happen with RazorLame as well? If it is in fact...

You bet we are. 7 years are a very long time.:)

There is a difference between "support" and "can play". MP3pro is designed compatible to regular mp3. Though, if it is played on a player which doesn't support MP3pro (which there are a lot of, especially hardware players), you'll get a frequency response up to about 10 kHz only. Also MP3pro...

It doesn't matter what bit depth your session files are. You do the mixdown to 32 bit float in any case, and if your other software needs 24 bit files for input, then you dither it to 24 bit.

In this case, the style seems rather forgiving as there are mainly percussive sounds going on at around 10kHz, thus all the mirror frequencies are barely distracting. Also the higher bitrate apparently has the effect that the holes at around 10 kHz are almost gone which are present in your other...

Sorry, I don't get what you are asking now.:confused: Could you give some examples?

Here is an example how it looks like on a spectral view. Looks as bad as it sounds, like a really poor resampling job.

Myspace plays at 22050 Hz. It doesn't sound compressed or limited in general, as I heared some rather dynamic examples there. This one for example. They are merely lacking any high frequencies. Worse than the actual mp3 is the player though, as it does a really poor resampling job to 44100 Hz...

Then just check how far it goes over 0 dB, go back to the source file you encoded, lower it by that amount and reencode. You should peak at exactly 0 dB now. Dithering generally improves the resolution in mid and low frequency range, thus do it. The difference is by no means that drastic as with...

It's "decoding", not "encoding", taht's why I said, it's a bit confusing that this option is with the "save"-options when in fact it influences the loading only. Not surprising, as you did clip at 0 dB before, and now, your peaks don't clip, but actually go over 0 dB. That's why I said, you can...

I used to use Megabass and stuff like that. Until I got my Koss Porta Pro headphones. They are just the best mobile headphones I ever heared. Nothing sounds better then perfectly linear playback with them. Thus, if you think, stuff like Megabass improves anything, you are probably using bad...

Ok, it's a but confusing, since that particular option is in the save-dialog in the advanced mp3-options called "Set all decoding to 32-bit". After you set this option, you can load any mp3 file, and it doesn't clip anymore. If it's too hot, you can just normalize or limit it. That's a...

If it could fix overcompressed and clipped CD's. I'll take it. Seriously, I don't think so. Though, with a completely different purpose. Usually they do this for remix competitions. I think, this is a good thing, and I like to see more of them. I'm sure, every remix, megamix and mashup hobbyist...

As lossless codecs remove bits from the spectrum, the resulting wave can peak higher then the original. Higher quality encoding with higher bitrates can reduce this effect. And of course, you should enable 32-bit-loading (which doesn't clip) for MP3, if you ever should need to use one as a source.

Sure, do what suits you best. Though, mixing outside the computer doesn't mean that you can't master with a computer. Choose a soundcard with a reliable SPDIF and get the mix into your favourite mastering software.

There sure is some processing, as vinyl has some limitations. You need to have the low frequencies in phase, thus put a high pass over the side-signal. Also frequencies bellow 40 Hz are not allowed, thus put another high pass. Then applying the RIAA curve is actually quite a strong sort of...