About the wav data sub-chunk - c

Question

I am working on a project in which I have to merge two 8bits .wav files using C and i still have no clue how to do it.
I have read about wav files and I want to start by reading one of the files.
There's one thing i didn't understand:
Let's say i have an 8bit WAV audio file, And i was able to read (even tho I am still trying to) the Data that starts after the 44 byte, I will get numbers between 0 and 255 logically.
My question is:
What do those numbers mean?
If I get 255 or 0 what do they mean?
Are they samples from the wave?
Can anyone please explain?
Thanks in advance

Answer 1

Assuming we're not dealing with file format issues, getting values between 0 and 255 means that the audio samples are of unsigned eight-bit format, as you have put it.
One way of merging data would consist of reading data from files into buffers, arrays a and b and summing them value by value: c[i] = a[i] + b[i]. By doing so, you'd have to take care of the following:
length of the files may not be equal
on summing the unsigned 8-bit buffers, such as yours will almost certainly overflow
This is usually achieved using a for loop. You first get the sizes of the chunks. Your for loop has to be written in such a way that it neither reads past the array boundary, nor ignores what can be read. For preventing overflows you can either:
divide values by two on reading
or
read (convert) into a format which wouldn't overflow, then normalize and convert the merged data back into the original format or whichever format desired.
For all particulars of reading from and writing to a .wav format file you may use some of the existing audio file libraries, or write your own routine. Dealing with audio file format is not a trivial thing, though. Here's a reference on .wav format.
Here are few audio file APIs worth of looking at:
libsndfile
sndlib
Hope this can help.

Answer 2

See any good guide to WAVE for information on the format of samples in the data chunk, such as this one I found: http://www.neurophys.wisc.edu/auditory/riff-format.txt
Relevant excerpts:
In a single-channel WAVE file, samples are stored consecutively. For
stereo WAVE files, channel 0 represents the left channel, and channel
1 represents the right channel. The speaker position mapping for more
than two channels is currently undefined. In multiple-channel WAVE
files, samples are interleaved.
Data Format of the Samples
Each sample is contained in an integer i. The size of i is the
smallest number of bytes required to contain the specified sample
size. The least significant byte is stored first. The bits that
represent the sample amplitude are stored in the most significant bits
of i, and the remaining bits are set to zero.
For example, if the sample size (recorded in nBitsPerSample) is 12
bits, then each sample is stored in a two-byte integer. The least
significant four bits of the first (least significant) byte is set to
zero.
The data format and maximum and minimums values for PCM waveform
samples of various sizes are as follows:
Sample Size Data Format Maximum Value Minimum Value
One to Unsigned 255 (0xFF) 0
eight bits integer
Nine or Signed Largest Most negative more bits
integer i positive value of i
value of i
N.B.: Even if the file has >8 bits of audio resolution, you should read the file as an array of unsigned char and reconstitute the larger samples manually as per the above spec. Don't try to do anything like reading the samples directly over an array of native C ints, as their layout and size is platform-dependent and therefore should not be relied upon in any code.
Note also that the header is not guaranteed to be 44 bytes long: How can I detect whether a WAV file has a 44 or 46-byte header? You need to read the length and process the header based on that, not any assumption.