makewave

#!/data/data/com.termux/files/usr/bin/python

import argparse
import struct
import time
import wave

# For better future proofing, these shall be read from the input file.
sample_rate = 48000.0
CPU_hertz = 6000000
wavename = str(time.time())

parser = argparse.ArgumentParser(description='Generate Wave-file from linkport signals logged from AndieGraph.')
parser.add_argument('name', type=str, nargs='?', default=wavename,
                    help='Name of wave file excluding extension.')

args = parser.parse_args()

# Audio will contain a long list of samples (i.e. floating point numbers describing the
# waveform).  If you were working with a very long sound you'd want to stream this to
# disk instead of buffering it all in memory list this.  But most sounds will fit in 
# memory.
audio = []

noisef = open('/sdcard/a-ti-den/almostaudio.txt', 'rb')
sample_rate = float(noisef.readline().split(b':')[1])
CPU_hertz = int(noisef.readline().split(b':')[1])
noised = noisef.readlines()
dps = []
for dpstring in noised:
    dp = dpstring.split(b" ")
    ts = float(dp[0]) # This one expected uSeconds
    ts = (int(dp[0])*167)/1000 # Clockcycles times 167 NanoSeconds (6Mhz)
    rl = int(dp[1])
    left = int(rl%2)
    right = int((rl/2)%2)
    dps.append((ts, left, right))

# Use microseconds to make it comparable
step_length = 1000000/sample_rate

samplelength=dps[-1][0] - dps[0][0]
print(dps[-1][0], dps[0][0], samplelength)

noofsamples=int(samplelength//step_length)
print(noofsamples)

amps=[-1.0, 1.0]
timepoint = dps[0][0]
i=0
for x in range(noofsamples):
    if not timepoint < dps[i+1][0]:
         i+=1
    audio.append((amps[dps[i][1]],amps[dps[i][2]]))
    timepoint+=step_length

print(len(audio))

def save_wav(file_name):
    # Open up a wav file
    wav_file=wave.open(file_name,"w")

    # wav params
    nchannels = 2

    sampwidth = 2

    nframes = len(audio)
    comptype = "NONE"
    compname = "not compressed"
    wav_file.setparams((nchannels, sampwidth, sample_rate, nframes, comptype, compname))

    # Stereo channels are placed next to each other frame by frame.
    framedata = []
    for sample in audio:
        framedata.append(struct.pack('h', int( sample[0] * 32767.0 )))
        framedata.append(struct.pack('h', int( sample[1] * 32767.0 )))

    frameblob = b''.join(framedata)
    wav_file.writeframes(frameblob)

    wav_file.close()

    return


save_wav("/sdcard/a-ti-den/" + args.name + ".wav")