"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.

[Typhoon859]

Right, so, there evidently seems to be a lot I don't understood so I was hoping I can be filled in. I know enough to understand what would be explained.

Regarding my knowledge of 16-bit, am I now understanding correctly that this is a decoding limitation and not at all a limit of 96dB dynamic range within the file itself? In other words, if I have a mix where increase everything past 96dB, up to 140 lets say, and then I save it to the parameters of a 16-bit MP3, the information would all still be there and the clipping would take place during the decoding process?

I understood that the indicated adjustment was to get the peak below 96dB, I guess I'm just not reading this correctly. Why is it incorrect to state that 1.0 represents 96dB for a 16-bit file? Furthermore, assuming that is the case, 96 x 1.05 is how I got the 100.8dB value. If that's not how it's calculated then how and why? XD

Just to also make known, I can only guess what you're talking about when you mention time domain and frequency domain although I'm fairly certain I'd understand a brief description of what that is referencing. I know many essential things when it comes to digital waveforms but not how it relates to digital parameters and related formats.

[db1989]

In other words, if I have a mix where increase everything past 96dB, up to 140 lets say, and then I save it to the parameters of a 16-bit MP3

There is no such thing as a 16-bit MP3, as you have already been told.

Why is it incorrect to state that 1.0 represents 96dB for a 16-bit file?

Because 1.0 is an instantaneous point, specifically the maximum, on a waveform with possible amplitudes between -1 and 1, on a linear scale; whereas a decibel is a measure of loudness based on the aggregation of many samples over a period of time, measured logarithmically. A sample at +1 alone does not equal either 96 dB, 0 dB FS, or any other measure of decibels.

Assuming that your reference to 96 dB means dynamic range – rather than, for example, dB SPL, which is not relevant – to have/demonstrate this, the file would need to contain least two waves: one oscillating between 1 and 1, and one between (-1/32768) and (1/32767).

Furthermore, assuming that is the case, 96 x 1.05 is how I got the 100.8dB value. If that's not how it's calculated then how and why? XD

Again, linear vs. logarithmic. Without intending offence, considering this and the fact that you don’t know what the time and frequency domains are, it’s probably time to do some background reading before continuing with this thread.