This format is extremely simple (like the BMP case in images). I would say there is no encoding on the data which is represented by PCM (Pulse Coded Modulation). This uncompressed case was good in the early days where the Internet transfer of files wasn’t so regular. With the coming and popularisation of the Internet it was felt to reduce these wave file sizes and thus came the now famous MP3.
A crude measurement of the file size can be done in this way:
Time of audio file (in seconds) x No. of Channels x Samples per second (Sampling Rate) x Bits per sample.
Generally used values for each are 2 channel, 44.1kHz (CD Quality), 48kHz (DVD Quality), 8kHz (decent for only speech) and 8-bit (on the lower side), 16-bit (good), 32-bit (overkill!) per sample PCM code.
Now coming to the part of the bit-stream, this is done by making chunks as mentioned earlier. There is a header which gives all the required details of the file, and is followed by the data chunks. Important in the header are the no. of channels, audio format, the sample rate, and bits per sample.
Its interesting to note that 8-bit samples are stored as unsigned integers (0 to 255) while 16-bit samples are stored as 2’s complement integers from -32768 to 32767. For more information on the byte-structure and bit-stream I would suggest to have a look at this to allow me from copying the same redundant information. 🙂