Detect Tap with Pyaudio from Live Mic

Detect tap with pyaudio from live mic

One way I've done it:

• read a block of samples at a time,
  say 0.05 seconds worth
• compute the
  RMS amplitude of the block (square
  root of the average of the squares of
  the individual samples)
• if the block's RMS amplitude is greater than a threshold, it's a "noisy block" else it's a "quiet block"
• a sudden tap would be a quiet block followed by a small number of noisy blocks followed by a quiet block
• if you never get a quiet block, your threshold is too low
• if you never get a noisy block, your threshold is too high

My application was recording "interesting" noises unattended, so it would record as long as there were noisy blocks. It would multiply the threshold by 1.1 if there was a 15-second noisy period ("covering its ears") and multiply the threshold by 0.9 if there was a 15-minute quiet period ("listening harder"). Your application will have different needs.

Also, just noticed some comments in my code regarding observed RMS values. On the built in mic on a Macbook Pro, with +/- 1.0 normalized audio data range, with input volume set to max, some data points:

• 0.003-0.006 (-50dB to -44dB) an obnoxiously loud central heating fan in my house
• 0.010-0.40 (-40dB to -8dB) typing on the same laptop
• 0.10 (-20dB) snapping fingers softly at 1' distance
• 0.60 (-4.4dB) snapping fingers loudly at 1'

Update: here's a sample to get you started.

#!/usr/bin/python

# open a microphone in pyAudio and listen for taps

import pyaudio
import struct
import math

INITIAL_TAP_THRESHOLD = 0.010
FORMAT = pyaudio.paInt16 
SHORT_NORMALIZE = (1.0/32768.0)
CHANNELS = 2
RATE = 44100  
INPUT_BLOCK_TIME = 0.05
INPUT_FRAMES_PER_BLOCK = int(RATE*INPUT_BLOCK_TIME)
# if we get this many noisy blocks in a row, increase the threshold
OVERSENSITIVE = 15.0/INPUT_BLOCK_TIME                    
# if we get this many quiet blocks in a row, decrease the threshold
UNDERSENSITIVE = 120.0/INPUT_BLOCK_TIME 
# if the noise was longer than this many blocks, it's not a 'tap'
MAX_TAP_BLOCKS = 0.15/INPUT_BLOCK_TIME

def get_rms( block ):
    # RMS amplitude is defined as the square root of the 
    # mean over time of the square of the amplitude.
    # so we need to convert this string of bytes into 
    # a string of 16-bit samples...

    # we will get one short out for each 
    # two chars in the string.
    count = len(block)/2
    format = "%dh"%(count)
    shorts = struct.unpack( format, block )

    # iterate over the block.
    sum_squares = 0.0
    for sample in shorts:
        # sample is a signed short in +/- 32768. 
        # normalize it to 1.0
        n = sample * SHORT_NORMALIZE
        sum_squares += n*n

    return math.sqrt( sum_squares / count )

class TapTester(object):
    def __init__(self):
        self.pa = pyaudio.PyAudio()
        self.stream = self.open_mic_stream()
        self.tap_threshold = INITIAL_TAP_THRESHOLD
        self.noisycount = MAX_TAP_BLOCKS+1 
        self.quietcount = 0 
        self.errorcount = 0

    def stop(self):
        self.stream.close()

    def find_input_device(self):
        device_index = None            
        for i in range( self.pa.get_device_count() ):     
            devinfo = self.pa.get_device_info_by_index(i)   
            print( "Device %d: %s"%(i,devinfo["name"]) )

            for keyword in ["mic","input"]:
                if keyword in devinfo["name"].lower():
                    print( "Found an input: device %d - %s"%(i,devinfo["name"]) )
                    device_index = i
                    return device_index

        if device_index == None:
            print( "No preferred input found; using default input device." )

        return device_index

    def open_mic_stream( self ):
        device_index = self.find_input_device()

        stream = self.pa.open(   format = FORMAT,
                                 channels = CHANNELS,
                                 rate = RATE,
                                 input = True,
                                 input_device_index = device_index,
                                 frames_per_buffer = INPUT_FRAMES_PER_BLOCK)

        return stream

    def tapDetected(self):
        print("Tap!")

    def listen(self):
        try:
            block = self.stream.read(INPUT_FRAMES_PER_BLOCK)
        except IOError as e:
            # dammit. 
            self.errorcount += 1
            print( "(%d) Error recording: %s"%(self.errorcount,e) )
            self.noisycount = 1
            return

        amplitude = get_rms( block )
        if amplitude > self.tap_threshold:
            # noisy block
            self.quietcount = 0
            self.noisycount += 1
            if self.noisycount > OVERSENSITIVE:
                # turn down the sensitivity
                self.tap_threshold *= 1.1
        else:            
            # quiet block.

            if 1 <= self.noisycount <= MAX_TAP_BLOCKS:
                self.tapDetected()
            self.noisycount = 0
            self.quietcount += 1
            if self.quietcount > UNDERSENSITIVE:
                # turn up the sensitivity
                self.tap_threshold *= 0.9

if __name__ == "__main__":
    tt = TapTester()

    for i in range(1000):
        tt.listen()

How get sound input from microphone in python, and process it on the fly?

If you are using LINUX, you can use pyALSAAUDIO.
For windows, we have PyAudio and there is also a library called SoundAnalyse.

I found an example for Linux here:

#!/usr/bin/python
## This is an example of a simple sound capture script.
##
## The script opens an ALSA pcm for sound capture. Set
## various attributes of the capture, and reads in a loop,
## Then prints the volume.
##
## To test it out, run it and shout at your microphone:

import alsaaudio, time, audioop

# Open the device in nonblocking capture mode. The last argument could
# just as well have been zero for blocking mode. Then we could have
# left out the sleep call in the bottom of the loop
inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE,alsaaudio.PCM_NONBLOCK)

# Set attributes: Mono, 8000 Hz, 16 bit little endian samples
inp.setchannels(1)
inp.setrate(8000)
inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)

# The period size controls the internal number of frames per period.
# The significance of this parameter is documented in the ALSA api.
# For our purposes, it is suficcient to know that reads from the device
# will return this many frames. Each frame being 2 bytes long.
# This means that the reads below will return either 320 bytes of data
# or 0 bytes of data. The latter is possible because we are in nonblocking
# mode.
inp.setperiodsize(160)

while True:
    # Read data from device
    l,data = inp.read()
    if l:
        # Return the maximum of the absolute value of all samples in a fragment.
        print audioop.max(data, 2)
    time.sleep(.001)

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