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I require that the sound levels of all the songs for my project are uniform. The songs are from different albums and different artists, hence they are not uniform from the perspective of volume level.

I found MP3Gain for MP3 files. But people say it does not work with the WAV file format. There are some problems like reducing audio quality when it comes to Audacity.

Which programs like MP3Gain which can be successfully utilized for wave files?

Can I use SoX for this, and how?

  • 1
    I'm not sure Wav or Aif can hold ReplayGain data; flac, ogg, mp3 & aac can, so you might be better off using one of those formats - en.wikipedia.org/wiki/ReplayGain has an explanation & links to available software. – Tetsujin Sep 25 '15 at 6:57
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You are talking about perceived loudness. This is quite unlike peak level. What you can do is measure the perceived loudness for all your songs and then adjust the gain accordingly. The best way currently available to measure perceived loudness is using the R128 standard. I hope you're on a Mac, because then you can use this free commandline tool:

https://github.com/audionuma/r128x

Perceived loudness according to R128 is measured in LUFS (loudness units full scale), but these are identical to deciBels, except that they are measuring an RMS window over a certain time (which is of course the only way to measure perceived loudness). Now comes the tricky thing: if your song has a loud part and a more quiet part, statistics dictates that the perceived integrated loudness for the the song you're measuring is a function of the entire song. What you can do:

  • Isolate the parts of all your songs that you want to sound the same to the listener. If a song has several drops, you can just take a single portion of a few seconds that is representative of the loudness of the song.

  • Measure all these isolated parts using the above tool or another R128 measuring tool. You will have to differentiate between M, S and I loudness values. You can simply ignore M (momentary) and S (short term) loudness and just look at the I (integrated) level.

  • R128 wants tv programs to all be at -23 LUFS, but you don't care about this. Just make notes of all the integrated levels of your song snippets. As they are digital full scale measurements, the measurements will yield negative results, like -9 LUFS and -15 LUFS.

  • Like we've seen, LUFS are just RMS deciBels, so you can gain scale the entire song using any linear gain plugin or adjustment. Take the softest song, e.g. the one that measures -15 LUFS and use that one as a reference. You can now gain scale a song that measured -9 LUFS using a -6 dB gain adjustment, and the song will then have the same perceived loudness for the parts you selected.

  • Voila, album mastered.

You may want to be sure your tracks are all mastered in terms of dynamics and EQ before you do all of the above. You can even normalize them first, so you know you are not wasting bits because all songs are way too quiet. If you are very technical inclined, you can even measure the LRA (loudness range) using R128 tools and compare those of songs you think should pretty much be in the same loudness range.

If all else fails, I'd be happy to master your songs for you, but it won't be free! Good luck,

Hens Zimmerman http://henszimmerman.audio

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Replaygain is a metadata based system - it analyses the absolute peak value of your audio, then writes a scaling value which the player reads and uses to amplify the audio data when you play it back.

The wave format doesn't natively support the metadata used for it so you have to either alter the actual audio data to normalise it, or use a hack to the format - I refer you to this old hydrogenaudio post: https://www.hydrogenaud.io/forums/index.php?s=&showtopic=54958&view=findpost&p=492968

Can't guarantee interoperability with any of your other apps.

Simplest thing to do would be to use foobar, do a batch convert with a normalisation processing step and save copies of the files for your project use.

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MP3 metadata has a sort of dynamics compression hack where an individual track can say to the player “play me at 110% volume” and another track can say “play me at 90% volume” and the result is that those 2 MP3’s seem to be at the same perceived volume.

With lossless audio, you don’t have this hack. You have to use actual dynamics compression. So the way you compress the dynamics is to run the audio files through the same dynamics compressor and create new lossless audio files that all play at the same perceived volume. This is classic audio editing.

Note that this has nothing to do with normalization. You can normalize 10 WAV files and they can all still have a different perceived volume. Normalizing is math, but perceived volume comes from art. Each song has a different dynamic range because of the way the people who made it expressed themselves artistically. A dynamics compressor is how you can make 2 songs from 2 different sources match.

There is an art to finding the right settings for your dynamics compressor based on your source material. That is why the tools that set one MP3 at 110% and another at 90% do an analysis of the audio first. With lossless audio, you will listen in real-time to the output of the dynamics compressor and make the adjustments you want. Some dynamics compressors also have automatic features that simplify their use and are equivalent to what is done with MP3.

Audacity probably has a built-in dynamics compressor and if not, then you can probably use a plug-in. If you are running on a Mac there is one built-in called “AUDynamicsCompressor.”

If you were only going to have one audio processor, it would be a dynamics processor. It’s the most fundamental of audio processors. It would be hard to imagine any kind of audio editing going on without a dynamics processor.

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If you're needing auto-gain to be applied in realtime then your best bet would be to convert the WAV's in question to MP3 using the highest possible quality settings in Audacity (trust me, if no one is specifically listening for audio artifacts/deformities, no one will notice any change in quality).

If you don't need auto-gain to be applied in realtime you can just use Audacity's Amplify effect and let it automatically determine the exact amount of gain to apply to bring the audio up to 0dB. Proceed to render the WAV to the disk and use where needed.

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    Normalising is not the same as adjusting for replay gain – Tetsujin Sep 25 '15 at 6:55

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