If I use the simple command ffmpeg -i file.m4a file.wav, the resulting file invariably seems to have a bitrate of around 750kbit/s (resulting in a 500mb file from a 100mb one my case).
I've already tried this answer (tldr: ffmpeg -i song.mp3 -acodec pcm_u8 -ar 22050 song.wav), among others, but the bitrate always turns out to be ridiculously large.
I like my files to be in wav format so that I can easily burn a CD from the file.
wav is simply uncompressed audio. Its bitrate is simply the size that audio would be without some smart algorithm reducing the size using psychoacoustic modelling.
Your files are probably 16-bit 44.1kHz audio. that will end up at 16-bit (2 bytes) x dual channel (2) x 44.1kHz which will give a bitrate of about 1.44Mbit/s. Single channel audio would be half that at 720kBit/s, and similarly reducing 44.1kHz to 22.05kHz keeping 16 bit sampling will again be 720kBit/s.
Without knowing what your goal is, then I'd suggest you leave the files in their original format, or use an actual compressed output such as flac.
WAVs have uncompressed audio, so the file size is a linear function of duration, irrespective of content.
Formula is sample depth x sampling rate x channel count x duration (in seconds).
