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.
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?
I recommend a little more research, beginning with the term global gain.
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
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
Since we're not dealing with power, a ~0.2dB increase would take 1.0 to 1.02. It takes a ~0.4dB increase to take 1.0 to 1.05.
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.
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) ... 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.
If a track's peak values are clipping by default, reducing the loudness now would be too late, wouldn't it?
...with "mp3" in front...http://lmgtfy.com/?q=mp3+global+gain
I seem to recall that the dynamic range of the mp3 format is in excess of 200 dB. While this is not technically the same as float 32, it is way more than needed in any real world situation.Edit: The convention is to equate 1.0 with zero dB and anything smaller as negative dB. That way it makes no difference if you are talking about 16 bit integer, 24 bit integer, 32 bit float etc. Full scale for integers is 1.0 = 0 dB, while for float 1.0 is still o dB, it's just not full scale.Edit2: dB is a logarithmic scale. Multiplying or dividing the amplitude by a factor means adding or subtracting the properly scaled logarithm of the factor to the dB.
There is no such thing as a 16-bit MP3, as you have already been told.
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).
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.
MP3s do not have an associated number of bits, or even any specific precision at all. Number of bits is a property of PCM, which MP3 is definitely not.
This isn't something you can learn from an internet forum. You'll need a textbook.
If you're worried that you've lost something on the quiet end by reducing the global gain throughout the file, your decoder could output 24-bit instead of 16-bit. I'm doubtful this would matter in most recordings.
Quote from: Typhoon859 on 31 July, 2012, 07:35:14 PMAnd if the change is undone, wouldn't you not get any of the data back (unless it's stored) ... 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.That's basically it. Each granule in each frame in the MP3 (2 granules per frame) contains frequency & amplitude info for generating brief sine waves. These waves are summed by the decoder to make an equally brief but much more complex composite waveform. Space is saved at encoding time by (among other things) eliminating waves that don't make an audible difference, and by storing the parameters with less-than-perfect precision, and by using standard lossless compression techniques internally. At decoding time, the global gain field of each frame is used to scale each granule's composite wave. So if you modify the global gain fields, only the amplitude changes; the "shape" (frequency content) of the output wave is unaffected, hence the gain adjustment is "lossless".
The MP3's combined waveform data consists of samples which use 32-bit float amplitudes, essentially perfect precision for audio purposes. However, typical audio playback APIs expect LPCM samples which use 16- or 24-bit integer amplitudes, so MP3 decoders convert the 32-bit float to 16-bit signed integer, normally. By definition, ±1.0 in the 32-bit float is the maximum range of the integers you're converting to. If 16-bit, that means -1.0 is -32768 and +1.0 is +32767. As pointed out, technically you can't assign a decibel value to a single point, but that's irrelevant for purposes of detecting clipping; if the float exceeds ±1.0, there's no choice but to clip when converting to LPCM.Hopefully with this explanation you can see how reducing the global gain brings the float32 amplitudes under ±1.0, which in turn prevents clipping, but is essentially "lossless" in the sense that it doesn't change the shape of the waveform, just its amplitude (volume).
Quote from: db1989 on 01 August, 2012, 09:24:35 AM[…] you can patronize me ON A FORUM where people come to inquire for help due to lack of knowledge on a subject. Whichever makes you feel better I guess...
Now, can we keep it cool from here? If you don't feel like answering any of my inquiries because you feel that my knowledge level is way below what is worth your time (however that makes sense), then simply don't respond.
not sure why it was necessary to even go there besides in a false attempt to put yourself on higher grounds.
What's the reason to want to host a forum if it's not for the reason of spreading knowledge and getting pleasure out of informing people? I have a feeling that even if initially there was good intention here, things just got derailed along the way by the people themselves because they begin using this as their personal platform for their own totalitarian ideals or simply conformist mentality - like in most other places.
I wouldn't be at all surprised if this thread were locked
[or] my posts were modified in what they say or edited together into senseless and out of context pieces - sort of like media propaganda, and I was banned with an automated message that I broke the rules.
Why are you the only one that gave a normal and direct response, yet alone a good one? Anyway...
"-32768" and "+32767": what is the unit of measure for these values or to what does it relate?
QuoteThis isn't something you can learn from an internet forum. You'll need a textbook.You're assuming too much about me and in general.
32bit (float) PCM: Audio range from 0.0 to 1.0 (0.5 is the middle).
Quote from: [JAZ] on 02 August, 2012, 08:03:13 AM32bit (float) PCM: Audio range from 0.0 to 1.0 (0.5 is the middle).Ouch! what was i thinking about? Audio range is -1.0 to 1.0 and 0.0 is the middle. The rest still applies.
Interesting how, on a generally friendly board, a conversation can go so wrong.