Thread: WAV file handling and filtering

Hi guys .. new to this forum .. hoping you can help me out ..

I got this issue to tackle which consists of using an FIR LPF filter to filter a wav file using C programming.

Till now I managed to tackle the filter part following this link:
Implementation of FIR Filtering in C (Part 1) | Shawn's DSP Tutorials
(I have my own filter co-efficients which have been produced using FDAtool on MATLAB and some other minor changes have been applied to match my requirements)

Now, the link shown does not compensate for files having a header. I have three wav files which need to be filtered. Thus I need to first extract the data from the header .. this was done using RIFFpad. Now that I have the data I don't know exactly how to tackle this issue.

Info from RIFFpad:

fmt -
Offset=20
ID=fmt
dwSize = 16
wFormatTag = 1
nChannel = 1 (Mono)
nSamplesPerSec = 16000
nAvgBytesPerSec = 32000
nBlockAlign = 2
wBitsPerSample = 16

Data - (len=537440, off=44)


I've got this hint to start with:

1. Each audio datasample inside the wav_files can be made up of a number of bytes and these bytes arestored in the little-endian order. Inside your program you have to change this to a big-endian order.

Any help?

Would like to thank you for providing me with a link on how to ask questions ... maybe it could come in handy in the future.

If it's dead easy for you, this is a whole new thing for me .. and practically I asked "Any help?" since I need at least to know what to look for to start tackling the issue.

I already searched about the relationship between little and big endian ..
I know that: In big endian the most significant byte is stored at the smallest address.
and that: In little endian, the least significant byte is stored at the smallest address.
but can't figure out how to "fetch" and store them and finally convert them properly.

Hope you get my concern now. Hopefully I get better help so that I can proceed.

If you've gotten to here without being able to figure out "biggest" and "smallest" we've got other problems. What have you tried and how doesn't it work?

Sorry for taking long to reply .. was busy with other things ..
Well to be honest with you I'm totally new to this wav file thing .. as mentioned in the first post, I followed that example and amended it to mach my filter requirements. The filter works fine, but only works for .pcm files (headerless files)

I followed these links, but being new to wav file and practically green in C programming I'm not able to understand everything to make a functional code:
Big and Little Endian
https://ccrma.stanford.edu/courses/4...ts/WaveFormat/
Wave File Format - The Sonic Spot

done:

Code:

#include <stdio.h>


unsignedshort be2le(unsignedshort x) {


    x = ( (((x) >> 8) & 0x00FF) | (((x) << 8) & 0xFF00) );

    return x;
}


int main() {
unsignedshortint a = 0x1234;
unsignedshortint b = be2le(a);
    printf("%hx %hx\n", a, b );
    return 0;
}

so to make sure I got you right:

Now i need to store my wave file as single bytes right? Maybe using an array of "?" size?
Next is using the part code you wrote and implementing it for this array?