whats the fastest way to increase the volume of a sample? (before inserting in a daw) ?

 

multiple files?
batch normalize to -1dB (TruePeak) without altering anything else,
in Reaper's batch converter or iZotope RX Advanced batch converter for example

 

add gain...

 

for single files, with the most control; use a Wave Editor. Wavelab, Soundforge, Audacity, RX, etc. You could apply a Limiter or a Clipper, Gain, or Normalize. Any processing like this would depend on the original individual file. Batch conversion is the fastest, but it is cookie-cutter to an extent. That's why you would use it for multiple files. it's not just fast, it will make everything as similar as possible. You can also just load the file into your DAW, apply any processing, and bounce it in place. You don't even need another program.

 

Besides Audacity, there's also this free tool

Spoiler

 

Having Reaper installed just for batch conversion is worth it. You can insert your own plugin chain into the batch converter too. For basic stuff, I like Signet. SoX and ffmpeg if you need advanced operations.

 

Batch conversion normalization, plenty of tools ranging from simple to complex. I tend to use Sound Forge, but that's been my tool for 20-odd years so that's the most comfortable to me. The suggested tools upthread are great too.

Force of habit, but I normalize to -0.3dB to avoid intersample clipping, dunno if it's that needed anymore, but old habits die hard with vengeance.

 

I use ffmpeg for that purpose.
It works very well:

Single File:
ffmpeg -i scrollclick.mp3 -filter:a "volume=2.5" scrollclick.mp3

Batch
(Linux):
mkdir -p gain && for file in *.mp3; do ffmpeg -i "$file" -filter:a "volume=2.5" "gain/$file"; done
(Windows - untested):
mkdir gain
for %%F in (*.mp3) do (ffmpeg -i "%%F" -filter:a "volume=2.5" "gain/%%F")

(The mkdir is for not having to overwrite and retaining the filenames)

Many of the tools mentioned etc most likely use ffmpeg as backend anyways (or sox),
so I usually do everythin by cmd

 

normalize to -0.5dB in Audacity

 

the fastest tool would be an editor which allows you to create droplets.

 

I like CoolEditPro 2.1, I beleive the author made it free when Adobe purchased the code from him.

 

Intersample peaks are only interesting in the final master.
And OT to the OP, -0.5db is not helping much as intersample peaks can be very high. If you are mastering yourself and care about this, use an intersample peak aware limiter.
Intersample peaks are not wrong per se, but if the playback device cant handle them they will cause distortion. A properly designed DA will be capable to play back intersample peaks without issues cause intersample peaks are a given in digital audio design.