my program reduces music speed by 50% but only in one channel

Question

I am using the wave library in python to attempt to reduce the speed of audio by 50%. I have been successful, but only in the right channel. in the left channel it is a whole bunch of static.

import wave,os,math
r=wave.open(r"C:\Users\A\My Documents\LiClipse Workspace\Audio 
compression\Audio compression\aha.wav","r")
w=wave.open(r"C:\Users\A\My Documents\LiClipse Workspace\Audio 
compression\Audio compression\ahaout.wav","w")
frames=r.readframes(r.getnframes())
newframes=bytearray()
w.setparams(r.getparams())
for i in range(0,len(frames)-1):
    newframes.append(frames[i])
    newframes.append(frames[i])
w.writeframesraw(newframes)

why is this? since I am just copying and pasting raw data surely I can't generate static? edit: I've been looking for ages and I finally found a useful resource for the wave format: http://soundfile.sapp.org/doc/WaveFormat/ If I want to preserve stereo sound, it looks like I need to copy the actual sample width of 4 twice. This is because there are two channels and they take up 4 bytes instead of 2.

`import wave
r=wave.open(r"C:\Users\A\My Documents\LiClipse Workspace\Audio 
compression\Audio compression\aha.wav","r")
w=wave.open(r"C:\Users\A\My Documents\LiClipse Workspace\Audio 
compression\Audio compression\ahaout.wav","w")
frames=r.readframes(r.getnframes())
newframes=bytearray()
w.setparams(r.getparams())
w.setframerate(r.getframerate())
print(r.getsampwidth())
for i in range(0,len(frames)-4,4):
    newframes.append(frames[i])
    newframes.append(frames[i+1])
    newframes.append(frames[i+2])
    newframes.append(frames[i+3])
    newframes.append(frames[i])
    newframes.append(frames[i+1])
    newframes.append(frames[i+2])
    newframes.append(frames[i+3])
w.writeframesraw(newframes)`

Edit 2: Okay I have no idea what drove me to do this but I am already enjoying the freedoms it is giving me. I chose to copy the wav file into memory, edit the copy directly, and write it to an output file. I am incredibly happy with the results. I can import a wav, repeat the audio once, and write it to an output file, in only 0.2 seconds. Reducing the speed by half times now takes only 9 seconds instead of the 30+ seconds with my old code using the wav plugin :) here's the code, still kind of un-optimized i guess but it's better than what it was.

import struct
import time as t
t.clock()
r=open(r"C:/Users/apier/Documents/LiClipse Workspace/audio editing 
software/main/aha.wav","rb")
w=open(r"C:/Users/apier/Documents/LiClipse Workspace/audio editing 
software/main/output.wav","wb")
rbuff=bytearray(r.read())
def replacebytes(array,bites,stop):
    length=len(bites)
    start=stop-length
    for i in range(start,stop):
        array[i]=bites[i-start]
def write(audio):
    w.write(audio)
def repeat(audio,repeats):
    if(repeats==1):
        return(audio)
    if(repeats==0):
        return(audio[:44])
    replacebytes(audio, struct.pack('<I', struct.unpack('<I',audio[40:44])
[0]*repeats), 44)
    return(audio+(audio[44:len(audio)-58]*(repeats-1)))
def slowhalf(audio):
    buff=bytearray()
    replacebytes(audio, struct.pack('<I', struct.unpack('<I',audio[40:44])
[0]*2), 44)
    for i in range(44,len(audio)-62,4):
        buff.append(audio[i])
        buff.append(audio[i+1])
        buff.append(audio[i+2])
        buff.append(audio[i+3])
        buff.append(audio[i])
        buff.append(audio[i+1])
        buff.append(audio[i+2])
        buff.append(audio[i+3])
    return(audio[:44]+buff)
rbuff=slowhalf(rbuff)
write(rbuff)
print(t.clock())

I am surprised at how small the code is.

Answer 1

Each of the elements returned by readframes is a single byte, even though the type is int. An audio sample is typically 2 bytes. By doubling up each byte instead of each whole sample, you get noise.

I have no idea why one channel would work, with the code shown in the question it should be all noise.

This is a partial fix. It still intermixes the left and right channel, but it will give you an idea of what will work.

for i in range(0,len(frames)-1,2):
    newframes.append(frames[i])
    newframes.append(frames[i+1])
    newframes.append(frames[i])
    newframes.append(frames[i+1])

Edit: here's the code that should work in stereo. It copies 4 bytes at a time, 2 for the left channel and 2 for the right, then does it again to double them up. This will keep the channel data from interleaving.

for i in range(0, len(frames), 4):
    for _ in range(2):
        for j in range(4):
            newframes.append(frames[i+j])