I have recorded a number of public debates to be posted online. Each recording is a single stereo wave file running about 60 minutes. Typically there are huge differences in volume between different people speaking, and then there are the very faint voices from the audience during Q&A periods. So I'm looking for a way to average out the volume within each file - either by bumping up the quiet parts, or deamplifying the louder parts.
I've attached a screenshot of a snippet of a wave like this - what you see is two people speaking in turn. Of course each file has a lot of such jumps from loud to quiet and back.
Naively I though that is what normalization does, but normalization in Audacity or Sound Forge 10 only works across separate tracks or files. It appears to do nothing when applied to a single file, or more specifically, there seems to be no difference between normalizing and amplifying in this case.
So far the only way I've found is to manually select each of the quiet parts and amplify them, but this is rather tedious and also rather inexact. Is there a better, or more automated method of doing this?
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To even out the differences, you instead want a function called compression or dynamic range reduction.
I think what Lordsmurf was saying was that "ideally" you would want to do it manually, because many of those adjustments ultimately would sound best when gradually adjusted in some sections, quickly adjusted in others, all according to how you want it to be understood easily to the human ear.
What he didn't say, and presumed against, was that you can easily use a COMPRESSOR and/or LIMITER (both "dynamic range reduction" devices) and if you set it carefully and by trial and error sampling, you might be able to get it close enough for your current needs, all automatically. Though, if you've never used them before, you could end up making things sound funky, and possibly much worse.
Compressor/Limiters exist as stand alone programs, or as common plugins (Directshow, as well as other APIs) in most audio editors.
If you understand visuals better, Normalizing raises your picture brightness levels by a constant amount thoughout the picture, whereas compression adjusts the contrast and gamma (with an adjustable centerpoint and "curves").
BTW, MP3Gain IIRC, is more of a normalizing app, and since the OP stated "WAVE", wouldn't even be appropriate.
There is also a Great Program called "MP3 to CD Burner", It has a lot of features. It can do WAV Files, OGG, Vorbis etc.
It can do a lot of things "Audacity" does, but is very user friendly. It keeps a library of all your music files and you can adjust each song by Volume, as well as by using an equalizer on each song to get the sound you like. Free to try, Great Tech Support and it's about 25 Bucks with Lifetime upgrades for free.
moodforaday, do the file(s) have to be in .wav?..If not, I recommend ReplayGain..Try foobar2k and convert the .wav(s) to something lossless like flac..Then in foobar2k right-click the flac file-> Scan per-file track gain, and then update the tag info..Try it..
You can always un-do it, since only the tags are modified and not the original audio data..
http://en.wikipedia.org/wiki/Replay_Gain" Who needs Google, my wife knows everything"
In Soundbooth/Audition, the function you want is "Equalize Volume". I'm not sure what the equivalent function is called in SoundForge
In most audio programs , "Normalize" will amplify, but will look at peaks to avoid clipping. So the amount of maximum gain is limited in tracks that have high peaks such as yours - because they are already near max levels. So the low volume parts will only get amplified by a tiny amount
"Equalize Volume" will differentially amplify low volume sections, and diminish high volume sections, so overall the volume is a more equal level within a track. Attached below is a before & after example of a waveform
You can equalize between separate tracks or "match tracks" as well. For example you might do this if you had a lecture series, where you want similar volume or consistency throughout the series.