I\'m trying to make real-time plotting sound in python. I need to get chunks from my microphone.
Using PyAudio, try to use
import pyaudio
import wav
For me this helped: https://stackoverflow.com/a/46787874/5047984
I used multiprocessing to write the file in parallel to recording audio. This is my code:
recordAudioSamples.py
import pyaudio
import wave
import datetime
import signal
import ftplib
import sys
import os
# configuration for assos_listen
import config
# run the audio capture and send sound sample processes
# in parallel
from multiprocessing import Process
# CONFIG
CHUNK = config.chunkSize
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = config.samplingRate
RECORD_SECONDS = config.sampleLength
# HELPER FUNCTIONS
# write to ftp
def uploadFile(filename):
print("start uploading file: " + filename)
# connect to container
ftp = ftplib.FTP(config.ftp_server_ip, config.username, config.password)
# write file
ftp.storbinary('STOR '+filename, open(filename, 'rb'))
# close connection
ftp.quit()
print("finished uploading: " +filename)
# write to sd-card
def storeFile(filename,frames):
print("start writing file: " + filename)
wf = wave.open(filename, 'wb')
wf.setnchannels(CHANNELS)
wf.setsampwidth(p.get_sample_size(FORMAT))
wf.setframerate(RATE)
wf.writeframes(b''.join(frames))
wf.close()
print(filename + " written")
# abort the sampling process
def signal_handler(signal, frame):
print('You pressed Ctrl+C!')
# close stream and pyAudio
stream.stop_stream()
stream.close()
p.terminate()
sys.exit(0)
# MAIN FUNCTION
def recordAudio(p, stream):
sampleNumber = 0
while (True):
print("* recording")
sampleNumber = sampleNumber +1
frames = []
startDateTimeStr = datetime.datetime.now().strftime("%Y_%m_%d_%I_%M_%S_%f")
for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
data = stream.read(CHUNK)
frames.append(data)
fileName = str(config.sensorID) + "_" + startDateTimeStr + ".wav"
# create a store process to write the file in parallel
storeProcess = Process(target=storeFile, args=(fileName,frames))
storeProcess.start()
if (config.upload == True):
# since waiting for the upload to finish will take some time
# and we do not want to have gaps in our sample
# we start the upload process in parallel
print("start uploading...")
uploadProcess = Process(target=uploadFile, args=(fileName,))
uploadProcess.start()
# ENTRYPOINT FROM CONSOLE
if __name__ == '__main__':
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT,
channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
# directory to write and read files from
os.chdir(config.storagePath)
# abort by pressing C
signal.signal(signal.SIGINT, signal_handler)
print('\n\n--------------------------\npress Ctrl+C to stop the recording')
# start recording
recordAudio(p, stream)
config.py
### configuration file for assos_listen
# upload
upload = False
# config for this sensor
sensorID = "al_01"
# sampling rate & chunk size
chunkSize = 8192
samplingRate = 44100 # 44100 needed for Aves sampling
# choices=[4000, 8000, 16000, 32000, 44100] :: default 16000
# sample length in seconds
sampleLength = 10
# configuration for assos_store container
ftp_server_ip = "192.168.0.157"
username = "sensor"
password = "sensor"
# storage on assos_listen device
storagePath = "/home/pi/assos_listen_pi/storage/"
I worked this on OS X 10.10, Got the same error while trying to get audio from the microphone in a SYBA USB card (C Media chipset), and process it in real time with fft's and more:
IOError: [Errno Input overflowed] -9981
The overflow was completely solved when using a Callback Mode, instead of the Blocking Mode, as written by libbkmz.(https://www.python.org/dev/peps/pep-0263/)
Based on that, the bit of the working code looked like this:
"""
Creating the audio stream from our mic
"""
rate=48000
self.chunk=2**12
width = 2
p = pyaudio.PyAudio()
# callback function to stream audio, another thread.
def callback(in_data,frame_count, time_info, status):
self.audio = numpy.fromstring(in_data,dtype=numpy.int16)
return (self.audio, pyaudio.paContinue)
#create a pyaudio object
self.inStream = p.open(format = p.get_format_from_width(width, unsigned=False),
channels=1,
rate=rate,
input=True,
frames_per_buffer=self.chunk,
stream_callback = callback)
"""
Setting up the array that will handle the timeseries of audio data from our input
"""
self.audio = numpy.empty((self.buffersize),dtype="int16")
self.inStream.start_stream()
while True:
try:
self.ANY_FUNCTION() #any function to run parallel to the audio thread, running forever, until ctrl+C is pressed.
except KeyboardInterrupt:
self.inStream.stop_stream()
self.inStream.close()
p.terminate()
print("* Killed Process")
quit()
This code will create a callback function, then create a stream object, start it and then loop in any function. A separate thread streams audio, and that stream is closed when the main loop is stopped. self.audio is used in any function. I also had problems with the thread running forever if not terminated.
Since Pyaudio runs this stream in a separate thread, and this made the audio stream stable, the Blocking mode might have been saturating depending on the speed or timing of the rest of the processes in the script.
Note that the chunk size is 2^12, but smaller chunks work just as well. There are other parameters I considered and played around with to make sure they all made sense:
Hope that works for someone!
My other answer solved the problem in most cases. However sometimes the error still occurs.
That was the reason why I scrapped pyaudio and switched to pyalsaaudio. My Raspy now smoothly records any sound.
import alsaaudio
import numpy as np
import array
# constants
CHANNELS = 1
INFORMAT = alsaaudio.PCM_FORMAT_FLOAT_LE
RATE = 44100
FRAMESIZE = 1024
# set up audio input
recorder=alsaaudio.PCM(type=alsaaudio.PCM_CAPTURE)
recorder.setchannels(CHANNELS)
recorder.setrate(RATE)
recorder.setformat(INFORMAT)
recorder.setperiodsize(FRAMESIZE)
buffer = array.array('f')
while <some condition>:
buffer.fromstring(recorder.read()[1])
data = np.array(buffer, dtype='f')