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I have a question about the basic wav file data chunk.
I know that each 2 bytes in the data chunk section represent mono sound and left and right channel is alternately stored, but I still have no idea how to understand the 2 byte value.
Is it high and low?
If so, when I want to reduce volume 50%, is all I have to do just classfy low and high
and divive by 2?
WAV files can contain samples at different bit depths and encodings, but one of the most commonly used is 16 bit PCM. With 16 bit PCM data in a WAV file, each two bytes should be interpreted as a short (i.e. a signed, two-byte number). In WAV files, the samples are always “little-endian”. If you tag your question with a language, someone might be able to provide some simple sample code. In C/C++, you would cast your array of data read from the file into a (short *), allowing you to easily access each sample.
To answer the second part of your question, yes you can reduce the volume by halving the value of each sample.