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Normalizing Music?
Beans asks: "I have a couple classical music CD's which I listen to at work, and use for putting the baby to sleep. I can never find the correct volume, I can't hear soft spots, so I turn it up, only to have a rising crescendo rouse the baby, or at work, have co-workers glace over. What is a good way to normalize them (read on for what I mean by normalize)? All of the normalizing software I have seen uses the entire song for the window of normalzing. Basically makes determines a static value required to get the average volume of the song to the user defined level, then applies that value to the entire song. What I need is something that normalizes over a sliding window, or say 5 seconds, or whatever. In effect making soft spots louder, and crescendo's quieter. Not the way the music was intended to be heard, but perfect for music-at-work, or putting kids to sleep. Does anyone know of any software that does this? On a side note, I work for a Seismic processing company, and we do stuff like this all the time on Seismic waves, not sound waves. If I can't find any canned software to do this, I may modify some of our code to work with WAV files, but I don't want to reinvent the wheel."
They both have a volume equalizer option.
Not a Twitter sockpuppet... but I wish I was.
it seems the music "industry" has been doing a very good job of this. What exactly you want is dynamic compression, not normalization. click here for more information.
I've used normalizer plugins in both XMMS and Winamp. They aren't perfect, but they're generally alright.
Check out http://volnorm.sourceforge.net/ for an XMMS plugin, or one of the many Winamp plugins here.
What you want is a compressor. Audacity (GPL software for Linux, Mac, Windows) includes a simple built-in compressor, and also works with compressor plug-ins like SC4 by Steve Harris. (You can get SC4 by installing the swh-plugins collection on Linux or Unix; it's also included with the Windows version of Audacity).
Yes iTunes calls it "Sound Check", and IIRC it normalizes the entire song.
/* oops I accidentally made a comment, sorry */
That's not normalizing, that's dynamics compression.
Normalizing scales the entire file so that the highest sample is the highest value that can be represented. For many sound formats, this is 32768, for 15 bits. (1 bit left for the sign). This lets the sample use the full range. However, the volume of the file is based on the value of one sample.
What you want is to amplify different parts of the file so that the level of the song is more consistent. For example, sample every 1/4 second and adjust the level based on the average of that sample. A typical value might be 2:1 compression above -10 db. In this case, if the volume is above -10 db, the amount above -10 db is cut in half. -10 stays at -10, but -6 is scaled to -8, 0 to -5. (0 db is the highest value that can be represented.) There is a lot more to it, including look ahead, recovery time, multiple levels, etc.
It is similar to riding the voume control manually and turning it down for the loud parts and back up for the soft parts, but very quickly.
After dynamics adjustment, you usually normalize the sample.
I edit speech audio for my church and deal with this all the time. I use Cool Edit 2000 and it is great at this. However, it is no longer available. But most audio editing programs can do this. Try Audacity.
Classical music with its large dynamic ranges is meant to be loud at times. People think classical is something nice and calm like a little old lady, but much of it is meant to be exciting, and brash.
I'd do something interesting, but my server can't handle a slashdotting.
Give this plug-in for Winamp a try. It's called Rock Steady and I've been using it for a year or two at the office without a hitch.
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What you want is 'compression', not normalization. I don't mean data compression (ie. making smaller file), btw.
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Music studio use "compressors" to reduce the dynamic range of sound. Ones that work in real time have to be a bit sophisticated (sliding window, like you said), but doing it offline is easier, since you can know the max and min sound values beforehand.
I don't know what tools are out there in PC land, hopefully someone will answer. But I will mention that I used to have a portable CD player with a built in compressor (to help keep your music above the ambient background noise, espicially when driving) and it was GREAT! It is a feature that I don't see much in portable devices anymore, which surprises me...
Anyway, hopefully this might give you a new term to google for, at least. Try searching for "dynamic range compressor", or something...
Alright, I found some windows plugins that might help; Anyone with some Linux/MacOs/Unix equivalents?
http://www.divx-digest.com/software/tfm_filter.
http://www.divx-digest.com/software/dedynamic.
"It's overkill, of course. But you can never have too much overkill." - Anonymous Slashdot Coward
You want quiet, listen to Enya. But don't go around raping the masters because of mundane utilitarian reasons.
On my nForce 2 motherboard, the sound applet has a "Night Mode" which um... "compresses" the sound so you don't have to turn it up as loud to hear the soft notes...
It's a pretty niffty feature to have, but I've never seen it anywhere else that I can think of...
Wiwi
"I trust in my abilities,
but I want more then they offer"
Have you tried using Replay Gain? It finds the psychoacoustic level of the music and calculates an appropriate gain correction. Replay gain is supported by foobar2000. MP3Gain is a tool that computes the replay gain for a track and changes the overall gain of the file.
Geez...I'm surprised nobody brought this up yet. MP3gain normalizes MP3's, but it NOT based on an average of the entire song. Read the site for more info. This is exactly the solution you are looking for.
MP3 Gain at Sourceforge
Sangloth
I'd appreciate any comment with a logical basis...it doesn't even have to agree with me.
Some home theater receivers and tv's have "Night Mode" which is used to do this. Feature is great in that you don't have to crank it to hear all of the dynamic ranges of a movie thus not loosing too much when you watch at lower volumes. (perfect for those with kids..)
I think even my DVD player does this.
http://trikuare.cx/code/AudioCompress.html
Works with most players -- xmms plugin as well.
Ratboy
Just another "Cubible(sic) Joe" 2 17 3061
CoolEdit was bought by Adobe and is now available under the name Adobe Audition. Perhaps you were referring to the lack of a budget version of the software, however, in which case, yeah, that kinda sucks.
I recommend the Volume Logic plugin for iTunes. It provides dynamic automatic gain control (AGC) over multiple bands. I find it really brings out the nuances in quiet music without blowing me away, when things get loud.
Michael.
Linux : Mac
What you need to take advantage of is called "clipping" by sound experts.
Let's say your stereo can play a waveform that varies between -1.0 and 1.0. If you send it a peak of +/- 2 for instance, it will clip to +/- 1.
What you need to do is find the lowest peak in your music. Even the "silent" parts have a little noise.
Let's assume that it goes +/- 0.01 in those parts. Find the recipricol (1/0.01) = 100 and set your gain to that.
The quiet parts will range +/- 1, and the loud parts will be clipped to also be +/- 1. Voila! You've "expanded" and "compressed" without all that fancy professional equipment.
Note: if you don't have an easy way to estimate the gain you need, just use a round estimate like 1000. You can find op-amps at your local Radio Shack that will amplify even higher than this! Just be sure to leave off the "feedback" part of the circuit. We all know that feedback is bad, right?
And as an added advantage, your child will grow up with a love of music by the Japanese artist Merzbow. It's a win-win-win.
The audio processing algorithm being requested here is called "compression", and it's a standard feature of most audio editing package such as SoundForge. You can also buy hardware compressors for under $100 (dbx and Alesis make stereo compressors) that are good for hooking into a home stereo/theater system.
I keep an Alesis digital stereo compressor hooked into my home stereo system so that when loud spots (commercials, action scenece, etc) come on it doesn't suddenly get orders of magnitude louder and drive my apartment neighbors nuts.
Some DVD players have audio compression built-in as a feature. Some newer TV sets have it built-in also.
Moderator hint: a comment is neither "Flamebait" nor "Troll" if it is true.
Similarly, you can increase the dynamic range of the sound by putting the bulb in parallel to your headphones. Loud sounds increase the temperature (and thus the resistance) of the bulb filament, which diverts more current through the headphones. Quieter sounds don'y heat the filament as much and allow more current to take that route, thus making them even quieter.
You may have to experiment with different types of lamps to find which works best.
Unknown host pong.
for winamp there is a dsp plugin called tomsteady which does exactly this. i use it when i play mp3's at night.
What you want is a compander. (compressor/expander). It is so called because it compresses or expands the dynamic range of the input audio. Sox includes one, if you like command lines.
.1,.1 -60,-10 0 0 .1
.1s to specify how fast it changes the volume; change the -60,-10 (always negative) to more similar values to make the compression less severe.
For winamp, I've found Rocksteady is beautiful, and does separate compression for several frequency bands (i can't begin to express how good this is for pop/rock).
For XMMS, AudioCompress does a sufficient job, although the windowing is somewhat stupid (not predictive == you'll get pops if things suddenly get loud) and it doesn't separate into multiple frequency bands, so it won't sound so good if you're playing something where the bass is really pushed (like Fiona Apple) next to something where it's not (classical).
Neither allow you specify complex transfer functions (of input volume to output volume).
For your purposes, sox is really the right thing, although it may feel a bit like "ack! drowning in sea of unexplained options!". You could start off trying:
$ sox infile.wav outfile.wav compand
will give you VERY hard compression. Change the
$ man sox
will tell you something more.
- KLS
SOX has a compand operator that lets you tune this stuff and parameters that let you tweak how fast it responds, how hard it attempts to correct the sound, and how fast it "lets go." Put this in line with your CD RIP process after you run a few through a test bed.
Don't forget to specify settings for each channel (normally two).
If you're willing to forego ID3 tags, or can hack them in yourself with mp3info, you can use cdda2wav to do sox processing: cdda2wav -O wav -t 7 -D /dev/cdrom - | sox [sox options] | lame - foo.mp3
I acknowledge that not everybody (including me) is an audiophile, but the recording engineer went to some effort to make a recording that, when played, would reproduce the original performance as closely as possible.
If you ride the volume control (or use automatic gain control to do it for you) or use dynamic range compression (a different animal from digital compression), you're compromising the music.
I'd no sooner do this than bleach my favourite painting.