I am coding a basic frequency analisys of WAVE audio files, but I have trouble when it comes to convertion from WAVE frames to integer.
Here is the relevant part of my code:
track = wave.open('/some_path/my_audio.wav', 'r')
byt_depth = track.getsampwidth() #Byte depth of the file in BYTES
frame_rate = track.getframerate()
buf_size = 512
def byt_sum (word):
#convert a string of n bytes into an int in [0;8**n-1]
return sum( (256**k)*word[k] for k in range(len(word)) )
raw_buf = track.readframes(buf_size)
One frame is a string of n bytes, where n = byt_depth.
For instance, with a 24bits-encoded file, track.readframe(1) could be:
raw_buf[n] returns an int in [0;255]
sample_buf = [byt_sum(raw_buf[byt_depth*k:byt_depth*(k+1)])
- 2**(8*byt_depth-1) for k in range(buf_size)]
It might be because you need to use an unsigned value for representing the 16bit samples. See https://en.wikipedia.org/wiki/Pulse-code_modulation
Try to add 32767 to each sample.
Also you should use the python struct module to decode the buffer.
import struct buff_size = 512 # 'H' is for unsigned 16 bit integer, try 'h' also sample_buff = struct.unpack('H'*buf_size, raw_buf)