heres a script to detect and record sound that ive been hacking on for a while. it works great, except that its memory usage grows after every successful detection. what am i missing?
from sys import byteorder
from array import array
from struct import pack
from datetime import datetime
import pyaudio
import wave
import os
import time
THRESHOLD = 6348
MAX_SILENCE = 500
CHUNK_SIZE = 1024
FORMAT = pyaudio.paInt16
RATE = 44100
MAX_LENGTH = 1024
def is_silent(snd_data):
"Returns 'True' if below the 'silent' threshold"
return max(snd_data) < THRESHOLD
def normalize(snd_data):
"Average the volume out"
MAXIMUM = 16384
times = float(MAXIMUM)/max(abs(i) for i in snd_data)
r = array('h')
for i in snd_data:
r.append(int(i*times))
return r
def trim(snd_data):
"Trim the blank spots at the start and end"
def _trim(snd_data):
snd_started = False
r = array('h')
for i in snd_data:
if not snd_started and abs(i) > THRESHOLD:
snd_started = True
r.append(i)
elif snd_started:
r.append(i)
return r
# Trim to the left
snd_data = _trim(snd_data)
# Trim to the right
snd_data.reverse()
snd_data = _trim(snd_data)
snd_data.reverse()
return snd_data
def add_silence(snd_data, seconds):
"Add silence to the start and end of 'snd_data' of length 'seconds' (float)"
silence = [0] * int(seconds * RATE)
r = array('h', silence)
r.extend(snd_data)
r.extend(silence)
return r
def record():
"""
Record a word or words from the microphone and
return the data as an array of signed shorts.
Normalizes the audio, trims silence from the
start and end, and pads with 0.5 seconds of
blank sound to make sure VLC et al can play
it without getting chopped off.
"""
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT, channels=1, rate=RATE,
input=True, output=True,
frames_per_buffer=CHUNK_SIZE)
num_silent = 0
num_snd = 0
num_listening = 0
snd_started = False
r = array('h')
while num_snd < MAX_LENGTH:
# little endian, signed short
snd_data = array('h', stream.read(CHUNK_SIZE, exception_on_overflow = False))
if byteorder == 'big':
snd_data.byteswap()
r.extend(snd_data)
silent = is_silent(snd_data)
if not silent and not snd_started:
snd_started = True
if snd_started:
num_snd += 1
if num_silent > MAX_SILENCE:
break
if silent:
if snd_started:
num_silent += 1
if not snd_started:
num_listening += 1
if num_listening > 4096:
del r[:]
num_listening = 0
sample_width = p.get_sample_size(FORMAT)
stream.stop_stream()
stream.close()
p.terminate()
del r[0:8000]
r = normalize(r)
r = trim(r)
r = add_silence(r, 0.5)
return sample_width, r
def record_to_file(path):
"Records from the microphone and outputs the resulting data to 'path'"
sample_width, data = record()
data = pack('<' + ('h'*len(data)), *data)
wf = wave.open(path, 'wb')
wf.setnchannels(1)
wf.setsampwidth(sample_width)
wf.setframerate(RATE)
wf.writeframes(data)
wf.close()
if __name__ == '__main__':
while True:
print("Ready!")
recorded = datetime.now()
recorded = "testpi1_" + recorded.strftime("%Y-%m-%d--%H-%M-%S") + ".wav"
record_to_file("/motion/" + recorded)
os.system("./convert-audio.py " + recorded)```