Normalization in Audacity

So aside from knowing that normalization helps tracks stay at a stable volume, I know nothing about normalization, nor how to use it in Audacity, or what recommended settings are.

I ask because I am trying (again) to (re)create a complete score for Batman: Arkham Knight and there are several tracks that are a lower volume than others. I wondered if there was any way to correct this, and I just want to generally improve the quality of my mixes.

Avoid normalization at all costs.

You want to obtain the difference in average RMS values between the lower amplitude tracks and similar sounding tracks. After obtaining the numerical difference between the two adjust the amplification of the quieter tracks to an approximate value.

So I’m guessing normalization just does a quick and dirty fix whereas meticulously finding the RMS values and adjusting for them is the proper way to do it?

Yes, normalization is the quick and dirty way. It distorts the values of a waveform completely.

On another note… is there any downside to converting M4A to FLAC? I’m trying to make a FLAC version of the score, and I may have to source certain bits of it with the official release, which is in M4A.

Nope.

Thanks. One last question… based on what I can gather from Spek, my raw ogg files range from 158 kbps to 544 kbps. When converting to MP3, should I just do VBR? I want to avoid upscaling because apparently it damages audio quality.

1. Unless your editor retains bitrate value upon save/export, any edited lossy file should be saved/exported to a lossless format.

2. Under no circumstances do you downsample lossy files.

I want to avoid upscaling because apparently it damages audio quality.

"Upscaling"? M4A -> MP3 damages the audio; M4A -> FLAC does not.

But if you don’t care about losing any audio information, then yes, transcoding to VBR-0 is best.

Always Normalise to 0dB

———- Post added at 12:18 PM ———- Previous post was at 11:31 AM ———-

my raw ogg files range from 158 kbps to 544 kbps. When converting to MP3, should I just do VBR?

Don’t

Ogg to Mp3 (VBR, CBR, AAC, etc) -> Worse quality than source file. Your source is already lossy. lossy to lossy = bad
Ogg to FLAC (Or ALAC, WAV, AIFF, etc) -> Same quality as source

Yeah the thing is my music player only takes MP3 format, much to my chagrin. Henceforth I will be making a FLAC version and a MP3 version for myself.

Yeah the thing is my music player only takes MP3 format, much to my chagrin. Henceforth I will be making a FLAC version and a MP3 version for myself.

Change your music player.

Change your music player.

As soon as I have $1,000 lying around, yes.

Always Normalise to 0dB

That’s for Peak Normalization.
useful if the source is downmixed from multiple channels to fewer channels (ie, 5.1 surround sound to stereo) to prevent clipping due to overflow (over 0dBFS).

If you want peak normalization, a safe way to put it is at -1dB instead of full zero.

RMS normalization works differently than peak normalization.

With Foobar2000, you can use a plugin to achieve EBU R128 loudness equalization. It’s a standard, but others argue there a better standards (more modern).
There’s also ReplayGain you can use to scan for levels and then encode based on those levels, too.

Some reading on normalization:
https://www.wiki.ed.ac.uk/display/ACEMusic/Loudness+Standards+-+EBU+R128+and+ITU+1770
(fancy)

There’s a CLI (Command Line Interface; requires using CMD/Command Prompt) tool that does various normalizations and has presets for different standards.
http://bs1770gain.sourceforge.net/
The ReplayGain 2.0 might be worth testing.

It all requires dedicated time to reading.
If you want to achieve what you want.

bs1770gain might be advanced. And there’s no development for a GUI and no one’s asked about it to be featured in Foobar2000.
And it’s quite unlikely that Audacity will include it.

Yeah the thing is my music player only takes MP3 format, much to my chagrin. Henceforth I will be making a FLAC version and a MP3 version for myself.

What time period is your artifact music player from??

The only Walkmans that I have been able to find that can play lossless music are seriously $1,000. It sucks. I’m a PC person so buying an Apple device isn’t really an option. I suppose I could try my phone but it’s not really that new, and my battery is bad enough as is. I don’t need playing music to make that worse.

Um……I have this to recommend (https://www.amazon.com/Resolution-Lossless-Music-Player-Generation/dp/B01M1KSUJX/ref=sr_1_3?ie=UTF8&qid=1482970427&sr=8-3&keywords=FiiO+X1). Or this for even cheaper. (https://www.amazon.com/s/ref=nb_sb_ss_i_8_9?url=search-alias%3Daps&field-keywords=sandisk+clip&sprefix=Sandisk+c%2Caps%2C214&crid=1IUK90BJCVUXY)

The only Walkmans that I have been able to find that can play lossless music are seriously $1,000. It sucks. I’m a PC person so buying an Apple device isn’t really an option. I suppose I could try my phone but it’s not really that new, and my battery is bad enough as is. I don’t need playing music to make that worse.

iPod/iPhone plays WAV, AIFF, Apple Lossless—all suitable lossless options. Hi-res lossless is another beast, but I assume you are using a standard 16/44.1 spec?

Thanks for the recommendations! Sadly I have zero money right now… but I can’t wait to ditch my crappy player.

roflrick: I have no idea.

My Sony portable can do lossless but it’s overkill, imo.

AAC is quite advanced. Better than Mp3.
Opus is new getting well developed.

If your phone is an Android, I would look into players.
VLC player, even. It’s free and has its own codec library so you can play various formats: Lossless (flac, alac, wav, etc) and Lossy (mp3, aac, opus I think).

I bought the app for Neutron Player Pro (https://play.google.com/store/apps/details?id=com.neutroncode.mp&hl=en) (like $10 or something).
It too can read various formats and support a wide variety of features such as crossfeed (get the most out of headphones), normalizations (using ReplayGain), and various other features.
It has some decent USB/streaming options as well for surround sound playback.
It’s a heavy duty player for Android.
It can process in 64-bit floating point if you want to be paranoid, default is 32-bit floating point (just as good).
It can output audio in 32-bit instead of 16-bit. Separate settings.
It can resample audio with different qualities. Standard resampling is okay. But higher quality version for placebophiles that kill your battery faster.
Automatic Gain Protection (AGP) so that it never exceeds 0dBFS to create clipping (only clipping from poor quality formats and not enough math for floating point <-> integer).
It’s very customizable, in terms of equalizer settings, crossfeed settings, speaker alignment, speed/tempo changing.

Literally the only player I use on Android for music if I’m not using my dated Sony player for AAC.

With codecs today, with software that updates, the encodes should be transparent if you use AAC or Opus.
Much better formats than MP3 and saves a lot of space over placebo-sized qualities for mobile.

iPod/iPhone plays WAV, AIFF, Apple Lossless—all suitable lossless options. Hi-res lossless is another beast, but I assume you are using a standard 16/44.1 spec?

Some snooty people ’round these parts will argue iTunes lossless (ALAC) isn’t as good as FLAC. To me, if it lossless – then it’s lossless. And honestly, my hearing cannot tell the difference between lossy and lossless (I call bs on the people that claim to, even a dog can’t hear that well) but I download and upload lossless anyway, since it’s the superior format.

Fair enough. I’ll be sure to always post lossless music in the future. Oversight on my part.

Fair enough. I’ll be sure to always post lossless music in the future. Oversight on my part.

At least you weren’t previously given scores in a lossless format in private, only to convert to MP3 and then DELETE the lossless -_-

If I could turn back time…..

Ugh that sucks. Yeah I’m currently in the fun situation where my 2TB SSHD decided to crash on me, and my music POOF! gone.

iTunes lossless (ALAC) isn’t as good as FLAC

On a technical level, it’s not nearly as opitmized as FLAC.
ALAC went open-source but no one wants to continue development. It’s abandonware now.

FLAC still receives constant updates to stay optimal.

The real benefit of FLAC is saving space over ALAC.

Especially where files can actually have an average bit depth 17 bits instead of 24 bits.
In multiple cases, where the averate bit depth is between 16 and 24, FLAC will compress accurately to small size. While ALAC will overbloat and give a file as if it were actually 24-bit. Which has no real benefit and doesn’t do anything except waste space. More does not equal more.

High resolution audio is only good for editing.
So for portable listening, anything over 16-bit, it’s best to dither down to 16-bit and resample to lower rates (>48kHz to 44.1kHz).

———- Post added at 06:12 PM ———- Previous post was at 06:07 PM ———-

iTunes lossless (ALAC)

It’s also not strictly limited or started by iTunes.
Apple Lossless Audio Codec was started by Apple in general.
Although they use ALAC in iTunes, it’s open to a lot of software. Any program can use it really.
Associating it to just iTunes is sloppy.

Apple Lossless is more apt.

Apple Lossless is more apt.

True dat…