#!/usr/bin/python


import pyaudio
import aubio
import numpy as np
from time import sleep


seconds = 10 # how long this script should run

bufferSize = 512
windowSizeMultiple = 2 # or 4 for higher accuracy, but more computational cost

audioInputDeviceIndex = 7 # use 'arecord -l' to check available audio devices
audioInputChannels = 1


pa = pyaudio.PyAudio()

info = pa.get_host_api_info_by_index(0)
num_devices = info.get('deviceCount')
for i in range(0, num_devices):
    if (pa.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0:
        print("Input Device id ", i, " - ", pa.get_device_info_by_host_api_device_index(0, i).get('name'))


audioInputDevice = pa.get_device_info_by_index(audioInputDeviceIndex)
audioInputSampleRate = int(audioInputDevice['defaultSampleRate'])

# create the aubio tempo detection:
hopSize = bufferSize
winSize = hopSize * windowSizeMultiple
tempoDetection = aubio.tempo(method='default', buf_size=winSize, hop_size=hopSize, samplerate=audioInputSampleRate)


# this function gets called by the input stream, as soon as enough samples are collected from the audio input:
def readAudioFrames(in_data, frame_count, time_info, status):

    signal = np.frombuffer(in_data, dtype=np.float32)

    beat = tempoDetection(signal)
    if beat:
        bpm = tempoDetection.get_bpm()
        print("beat! (running with "+str(bpm)+" bpm)")

    return (in_data, pyaudio.paContinue)


# create and start the input stream

inputStream = pa.open(format=pyaudio.paFloat32,
                input=True,
                channels=audioInputChannels,
                input_device_index=audioInputDeviceIndex,
                frames_per_buffer=bufferSize,
                rate=audioInputSampleRate,
                stream_callback=readAudioFrames)

while True:
    pass

# because the input stream runs asynchronously, we just wait for a few seconds here before stopping the script:
sleep(seconds)

inputStream.stop_stream()
inputStream.close()
pa.terminate()