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There are a few misconceptions here. Normalization, in the most basic sense, raises the gain of the ENTIRE track to a nominal level. What you are trying to do with normalization is maximize signal. The relative dynamics of the track are not changed. There are actually two types of normalization, but the most common is peak normalization, where the loudest ...

This kind of broad question doesn't really imply a specific answer, so I'm going to go with an equally broad answer that, while not precise, might still be helpful. You'll want to know the basic idea behind Fourier's Theorem which is that all waveforms can be expressed as a sum, possibly infinite, of sine/cosine waves. Wikipedia's page on Fourier series ...

Let me give a different answer here: First of all, let's state that 'normalizing' is a destructive signal processing task that can be done during the mastering of a 2 track mixdown. In general: Try to avoid normalization as much as possible. Or, in better words: try to avoid the need for it. Try to record in such a way that you don't let your converters ...

Use dynamic range compression. Set the threshold to kick in only on the louder voice, and adjust the ratio until the levels between the two voices match. Attack and release times of about 100ms should work.

Audio normalization is the process of uniformly increasing (or decreasing) the amplitude of an entire audio signal so that the resulting peak amplitude matches a desired target (the norm). Normalization applies the same amount of gain across the selected region of the recording so that the relative dynamics (and signal to noise ratio) are unchanged. ...

Depends on what you want to do. For ripping old audio CDs for your music collection, do what you want... :-> For a recording scenario, this is surely not feasible. If you normalize every song, you'd have different average loudness per track. You surely do not want that. If you normalize single instrument tracks, you'll win nothing, because you're going to ...

There is Peak normalization and Loudness normalization, not sure which one this is, you might want to check the manual. Peak normalization simply changes the volume of the entire song, this will also amplify quiet stuff too. If you do it on a classical symphony track you will hear the sniffs and AC of the concert hall louder. Loudness normalization will ...

Normalization is the process of both making the loudest peak 0 dB and making all the tracks the same volume. Compression means that you lower the peaks to get a more consistant volume so you can make it louder to get the highest peak at 0 dB. Well, you can't really go over 0 dB. At infinity dB the speaker is in the middle. At 0 dB the speaker is at the far ...

Digitized sound is just a sequence of numbers recording the amplitude of the soundwave at equal intervals (most often 44100 times per second). Applying FFT and finding the strongest frequency, or all frequencies above a treshold sounds like a reasonable way of finding the note. Converting the frequency into a note is easy, once you know that middle A is ...

You need to normalize the audio. There will be nothing you can do about the points in time when A and B are speaking simultaneously, but if there is that much difference between the two people, then normalization will do the trick. There are numerous commercial, shareware, and possibly free applications that can normalize an audio file. One bit of advice. ...

We built a software which does exactly that (and more): auphonic.com It's free, like the levelator, but also performs well on music regions. Actually audio loudness normalization is also a hot topic ATM, see e.g.: https://auphonic.com/blog/2012/08/02/loudness-measurement-and-normalization-ebu-r128-calm-act/ or ...

In the old days a quick and dirty way to get a little more out of audio would use "maximizing" where you found the max top and then scaled all data to the max value within the available bit range (in those days 8-, 12- and 16-bit - today most are 24-bit or floating point). However, this approach is linear and doesn't work so well as we hear sound ...

You didn't supply very much information, such as your OS and your desired output format, so I can not give you a detailed answer. CAF is a container format that can support several audio formats, but Apple Lossless (ALAC) is probably the most common. ffmpeg can decode ALAC in CAF: ffmpeg -i input.caf output.wav This can be turned into a "batch" command ...

I wouldn't rely on any kind of metering if I were you. Different sounds need different levels to properly fit in the mix, and the only reliable way to tell is to actually hear the sound together with all other instruments. So the "old-fashioned way" is just the right one: set up the levels on rehearsals with the whole band. It can't be that time-consuming to ...