"MP3Gain: How can it be possible?", It 's indicated that the gain adjustments are lossless Options

[Typhoon859]

So I've been thinking of trying to write a similar program from scratch and there's one main thing that I don't even understand how it's possible, yet alone done. So as is said, the process MP3Gain uses is lossless. Thinking about it, the only way MP3Gain could work where any player would play back the songs with whatever the target volume was would be if the change is present in the waveform. After MP3Gain is applied, if it weren't obvious from the beginning, in any audio editing software, the gain reduction is clearly visible. I could somewhat understand how the process can be reversed with added value, even if the waveform clips, as the information could still somehow be stored (more easily than the other way around). On the other hand, when taken away, don't you permanently lose the dB that you took from the threshold? As an example, if a song starts with some 6 decibel ambient noise and you reduce the song by 6dB, wouldn't that intro just completely disappear? And if the change is undone, wouldn't you not get any of the data back (unless it's stored) and just make the existing data 6dB louder? If that's the case, it isn't really undoing the changes; it's really just adding the difference in value back between the indicated ReplayGain value and what it is now.

Sorry this was kinda long-winded but the last thing though I'd also like to ask about is clipping. If a track's peak values are clipping by default, reducing the loudness now would be too late, wouldn't it? Wouldn't it be clipping no matter what at this point, contrary to what is indicated? The peaks would be chopped off either way since the structure of the waveform is no longer saved after being finalized. And also, the maximized volume indications don't make sense (has to be turned on in the options). For example, I have a file which ReplayGain indicates peaks at about 1.05 (16-bit = 100.8dB) and yet it's marked that only a 1.5dB reduction would be necessary to get it maximized (the loudest point before clipping - 96dB). Is there something I'm missing?

Thanks guys! Answers to these would be extremely helpful.


PS- A lot of the things here indicate to me that the values, whether over or under, remain as part of the data in the container but just doesn't play back, or rather, clips since it's within the 16-bit parameter.

[greynol]

You appear to assume that mp3 data is 16-bit integer in the time domain. This would be completely incorrect as mp3 data is 32-bit float in the frequency domain.

I recommend a little more research, beginning with the term global gain.

[2Bdecided]

I recommend a little more research, beginning with the term global gain.

...with "mp3" in front...
http://lmgtfy.com/?q=mp3+global+gain

Cheers,
David.