Hello,
Is there any way to normalize raw audio bytes of the same sound recorded on different smartphones knowing that there are mixed smartphones and each phone has different or same microphone hardware and each phone has different sensitivity of generating PCM?
I hope I have explained it correctly. I am kind of stuck in this type of normalization.
Any help will be much appreciated.
Regards,
Khubaib
Hi Khubaib, normalising the range of your audio recordings should be pretty easy. Most audio editing tools (Audacity [which is free] or Adove Audition) will let you do this. You can simply load in the WAV (or other format) audio and do it for each file.
Here's the instructions on how to do it in Audacity: https://manual.audacityteam.org/man/normalize.html
Stuart Cunningham, I'll use it and let you know if it works as per my requirements. But could you suggest some Signal processing techniques too?
To normalise you just increase the overall level of the audio file (so the amplitude of all sounds in it) until the highest wave peak is just below 1 or 0dBFS. Although with that said, this thread may be more helpful:
https://www.researchgate.net/post/What-is-the-correct-way-of-normalizing-an-audio-signal
Normalization could be changing all balance of the mix. İf you are not sufficient on the mastering engineering realm you can use. İt's also raising master signal to the highest peak level of the digital signal to 0dbfs.
Ben Mc Hugh and Tarık Balta, I read about RMS normalization (loudness normalization) "https://www.hackaudio.com/digital-signal-processing/amplitude/rms-normalization/". Will it work in a real-time scenario..?