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

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 ...


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 ...


4

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 -...


4

Better is subjective, and taste differs, but a chain of effects might include the following: High pass filter, maybe set at 20Hz. Depends on the material, but if you have a recording done with microphones the very lowest frequencies often are aeroplanes or lorrys or air condition and not the signal you want. When the signal is spoken word, go higher in ...


3

My suggestion is to do three steps. Step one, using your EARS, adjust the volume of the various portions of your program so that they have the correct relative volume that you want. If you want them to all sound the same, adjust it until they do sound the same regardless of what your meters are telling you. An ordinary peak or even RMS meter will not be a ...


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.


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 ...


2

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


2

The process is identical, but one is tested, the other is not. Gain will not test for overs, it will just clip if you add too much. Normalisation will scan the entire track & only allow gain to be increased until the single loudest sample is at 0dBFS [or very slightly less if you want to avoid clipping on cheaper equipment. I've always used -0.3 as a '...


1

Wav, afaik, has no equivalent soft-gain parameter. You could simply use Audacity to normalise to zero [so you have a level playing field], then again to -92.8dB... but -92.8 is going to be awfully quiet. Are you certain of your figures? It's also pretty much a one-way process because of the noise floor you will generate, so keep your originals in case ...


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

The trouble with Normalisation is it is undiscerning, it simply increases all gain until one peak reaches 0dBFS - & brings the entire noise-floor with it by the same amount. Adding compression is only going to make this worse, as it's limiting the peaks, allowing the noise floor to come up further. The simplest method is going to be to use the '...


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

Best practise these days when considering content normalization is not to normalize to an arbitrary peak level, but to consider programme content and overall loudness when conducting normalization activities. There are two areas of loudness that need to be considered: Integrated loudness - Integrated loudness measurement (in LUFS) made according to ITU-...


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

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

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 ...


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