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:
- 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?