Audio is commonly represented in a format called pulse code modulation (PCM), where the amplitude of the sound is recorded using an integer or a floating point

Audio is commonly represented in a format called pulse code modulation (PCM), where the amplitude of the sound is recorded using an integer or a floating point number (a "sample") anywhere from 12,000 to 192,000 times a second. Each sample is usually linearly proportional to amplitude rather than logarithmically. I am unaware of any audio program which internally represents audio in dB. Waveform displays default to linear scales, though logarithmic scaling is usually available. Just about the only place where logarithmic scales are the default is frequency domain graphical representations like spectrograms.