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6

You're right, that's what dialnorm is for; to try to create a standard for average dialogue level (dialogue being king and all). Having Dolby Media Meter helps a lot, but your ears should do a fine job too. I would avoid boosting dialogue in the absence of loud FX. Just because you have the headroom doesn't mean you need to use it. If i'm understanding the ...


5

Normalizing your audio probably won't have the desired effect that you're trying to achieve. Probably what you want to do is load all your audio into a software application, where you can mix the relative levels of the audio files so that they sound good in relation to each other. If you normalize all your audio, the crisp packet rustling along the floor ...


4

Normalization is a very simple process. It finds the highest peak and applies gain (makes everything louder) until the highest peak is at the specified maximum level (-1dB in the picture). There's also the option to remove DC. You can imagine DC as an upward or downward shift in the waveform, making it stay averagely more on the positive or the negative ...


3

In my experience, it all depends on where the audio is to be used next. If there is a limiter of some kind in the next phase of the process. Sometimes a limiter will be applied at varying levels below 0dB to prevent overload and clipping distortion. -o.1dBFS is a peak normalization preset because basically, that is the highest level a sample can be without ...


3

I too agree that the question does not provide the right detail BUT talking about asymmetrical waveforms, the answer lies in phase rotation, the way kahn (or whatever spelling that name has) the radiowaves pioneer managed to squeeze +6 db headroom from male voices, cause this is very often in male vocal recordings (And class A amplifiers when they are driven ...


3

Audacity does normalising but it sounds like what you want is a peak limiter. Try the Yohng W1 Limiter, which is modelled a a well-known and (possibly overused) plug-in. It doesn't look all that, but it does the job nicely. You should also try a compressor, but you'll need to use it really carefully on classical material.


3

Normalisation only raises the overall volume uniformly (the soft parts will be louder, but so will the loud parts). What you're looking for is dynamic-range compression, and I think there's a plug-in for that in Audacity. I'd advise you to learn a bit about the different controls on a compressor before you get stuck in.


3

I agree with @tim that normalizing depends on content. I've written up what I call a "style sheet" for our team. It's guideline of how certain effects should be normalized. It's not rigid, purely a starting point, but is something like this: Specifics: -10 to -3 dBFS (door knob moves on the low side, gunshots on the high end) Atmospheres: -12 to -8 dBFS (...


2

Have you tried Levelator ? (though i have not used it)


2

If you normalize all of your samples to -0.1 dB, then use them in your DAW projects, you'll end up clipping your audio tracks all the time as soon as you use a plug-in. I use Pro Tools where the channel faders default to unity gain in a new project. The channel fader on an audio track is post-plugin. This means if I put a Kick Drum sample on Channel 1, and ...


1

if you dont know how to use a compressor , automate the volume. Some software allows you to draw a volume curve to adjust loud and low parts.


1

This is what a compressor is for. You can reduce the dynamic range of a whole track (making the loud parts quieter) and then adjust the overall volume of the track as you wish. It would be easy to do that in a DAW (like Logic), but I'm not sure if Audacity can do that. After a quick google, I found this in the Audacity wiki: https://manual.audacityteam.org/...


1

ReplayGain tags aren't standard in WAV files, so you have to alter the PCM data with the required gain. As per my reading of the Replaygain specs, a correctly implemented Replaygain scanner will print out the gain required to attain 89 dB SPL (as defined in the specs). FFmpeg has a filter to detect replaygain. You can run ffmpeg -i in.wav -af replaygain -...


1

You can't really do this as you expect. You can increase the volume of the frequency ranges that your voice is in, but this will also increase the volume of anything else in those frequency ranges. You could also use compression, or normalisation, and these may help bring your voice up, but again will increase the volume of anything at lower volume.


1

First question to clarify is do you really need to align loudness level of your audio files ? It really depends on the goal of your project and what subsequent processing do you plan on these files. If you want to align them, do you need to align perceived loudness or normalize the audio signals ? This being said, you could use FreeLCS, an open source tool ...


1

You're not approaching this in the right way. By doing this "iterative normalization" you will be basically changing gain with little to no crossfade between the regions. This is not going to give you a natural sounding result. Sure an overall gain reduction may be necessary, but the initial recording should have enough headroom so that you can process the ...


1

In the old days a quick and dirty way to get a little more out of audio would use "maximizing" where you found the max top and then scaled all data to the max value within the available bit range (in those days 8-, 12- and 16-bit - today most are 24-bit or floating point). However, this approach is linear and doesn't work so well as we hear sound ...


1

Normalisation does not reduce dynamics. It makes the file as loud as possible -> Peak at 0dBFS. So is you have one little Click in your recording it might not get louder with normalisation. The solution might be the clip volume. Just doubleclick a clip to open the clip Editor, then use the volume slider to adjust the waveform by eye. This Way you can bring ...


1

I wouldn't rely on any kind of metering if I were you. Different sounds need different levels to properly fit in the mix, and the only reliable way to tell is to actually hear the sound together with all other instruments. So the "old-fashioned way" is just the right one: set up the levels on rehearsals with the whole band. It can't be that time-consuming to ...


1

Just peak normalizing does not really matter. Normalizing by RMS level could get closer. But ideally the files should be balanced by ear, taken that it's done with care so that you aren't gain riding or equalizing the files excessively to achieve that, but just balancing them mildly without causing some irreversible changes to the sounds that may not be fit ...


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