realtime_tdoa.py

"""
 Estimate realtime DoA (Direction of Arrival) using two mic
 Copyright (c) 2017 Yihui Xiong

 Licensed under the Apache License, Version 2.0 (the "License");
 you may not use this file except in compliance with the License.
 You may obtain a copy of the License at

     http://www.apache.org/licenses/LICENSE-2.0

 Unless required by applicable law or agreed to in writing, software
 distributed under the License is distributed on an "AS IS" BASIS,
 WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 See the License for the specific language governing permissions and
 limitations under the License.
"""

import pyaudio
import webrtcvad
import numpy as np
import collections
import Queue
import threading
import signal
import sys
import math
import audioop
from gcc_phat import gcc_phat
from vad import vad


class Microphone:

    def __init__(self, rate=16000, channels=2):
        self.pyaudio_instance = pyaudio.PyAudio()
        self.queue = Queue.Queue()
        self.quit_event = threading.Event()
        self.channels = channels
        self.sample_rate = rate

    def _callback(self, in_data, frame_count, time_info, status):
        self.queue.put(in_data)
        return None, pyaudio.paContinue

    def read_chunks(self, size):
        device_index = None
        # for i in range(self.pyaudio_instance.get_device_count()):
        #     dev = self.pyaudio_instance.get_device_info_by_index(i)
        #     name = dev['name'].encode('utf-8')
        #     print(i, name, dev['maxInputChannels'], dev['maxOutputChannels'])
        #     if dev['maxInputChannels'] >= self.channels:
        #         print('Use {}'.format(name))
        #         device_index = i
        #         break

        # if not device_index:
        #     print('can not find input device with {} channel(s)'.format(self.channels))
        #     return

        stream = self.pyaudio_instance.open(
            input=True,
            format=pyaudio.paInt16,
            channels=self.channels,
            rate=self.sample_rate,
            frames_per_buffer=size,
            stream_callback=self._callback,
            input_device_index = device_index,
        )

        while not self.quit_event.is_set():
            frames = self.queue.get()
            if not frames:
                break
            yield frames
        
        stream.close()

    def close(self):
        self.quit_event.set()
        self.queue.put('')


def main():
    sample_rate = 48000
    channels = 2
    N = 4096 * 4

    mic = Microphone(sample_rate, channels)
    window = np.hanning(N)

    sound_speed = 343.2
    distance = 0.14

    max_tau = distance / sound_speed

    def signal_handler(sig, num):
        print('Quit')
        mic.close()

    signal.signal(signal.SIGINT, signal_handler)

    for data in mic.read_chunks(N):
        buf = np.fromstring(data, dtype='int16')
        mono = buf[0::channels].tostring()
        if sample_rate != 16000:
            mono, _ = audioop.ratecv(mono, 2, 1, sample_rate, 16000, None)

        if vad.is_speech(mono):
            tau, _ = gcc_phat(buf[0::channels]*window, buf[1::channels]*window, fs=sample_rate, max_tau=max_tau)
            theta = math.asin(tau / max_tau) * 180 / math.pi
            print('\ntheta: {}'.format(int(theta)))

        
if __name__ == '__main__':
    main()