I am new to the forum and do appreciate that there is existing stuff about metering and loudness, but I would like just to ask some q’s from my own particular perspective and position.
Although I have doing basic audio editing for many years as a hobby and for the charity with which I volunteer, I am only now starting to learn to do it on a basic professional level, so sort of starting from scratch again really.
I have been an Engineer and Presenter with a Talking Newspaper Charity for 31 years, and I am also an aspiring radio presenter and have my own home studio.
As there’s only me when making my radio shows, I’m trying to (as much as possible) emulate the setup which a Presenter has on their own desk in a radio studio (i.e. minimum need for fader riding, level controls etc in-show).
The main part of this is wanting to learn how to make all my music mp3s and jingles sound the same loudness so that they mix easily into my show. (I do have an outboard compressor to help, but would like where possible to learn how to do this before compression/limiting etc).
Due to my Autism however I have struggled with understanding the difference between metering and loudness, but hopefully I have now picked up the basic idea, in that metering is there to prevent distortion and loudness to standardise the listener’s experience at their chosen volume, but the 2 aspects do still seem to me to crossover a little, which is still providing some confusion.
The problem I am having is that if I rip 2 different tracks from CD and meter them to a fairly standard average peak amplitude (such as the traditional PPM4 or -12 dBFS UK Broadcast Setting), the 2 tracks can sound totally different loudnesses through the monitors, and my thinking is that if i’m having to adjust my monitor volume too often, then so would the listener; so I would like to know the best method and tools to level out the loudness of my tracks.
I know that some use ‘Normalisation’ routines in various software, but i’m not sure if these might not be a bit of a ‘sledgehammer’, and might perhaps cause some hard tops or bottoms on the waveform, which could sound bad to the listener ?
Until my friend started to explain the usual dB scales to me, I had only previously used the free web widget mp3Gain, which appears to work in dB SPL, so that scale is the one I have been used to using for (very roughly) normalising the loudness of my mp3s. This is surprisingly effective for most good quality mp3s encoded from CD, however even with these, I am still finding so many differences in overall loudness (rather than and over/above the wide dynamic ranges which the artist and mastering engineer would have intended).
I do now understand generally that loudness (LUFS aside) is quite subjective, and that the mp3gain widget I have been used to uses the sound pressure reading, and that its (replaygain) algorhythm also takes in all sorts of other factors. I’m still curious to know though, whether there is any mathematical relationship at all between dbFS and dB SPL ? This is because I am so used to thinking in say 83 dB SPL, and switching to thinking in -? dBFS is like speaking another language at the moment, so if there is any sort of correlation between the 2 (even highly approximate), that might help me to get my bearings better.
(Probably for a separate thread, but also interested to know what draws different professionals to different DAWs (Audition and Pro Tools being the 2 favourites as far as I can gather - similar to the Canon v Nikon schools in photography I guess).