vadSound.py

# coding=utf-8
import webrtcvad
import collections
import sys
import signal
import pyaudio

from array import array
from struct import pack
import wave
import time

import os

FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 16000
CHUNK_DURATION_MS = 30       # supports 10, 20 and 30 (ms)
PADDING_DURATION_MS = 1500   # 1 sec jugement
CHUNK_SIZE = int(RATE * CHUNK_DURATION_MS / 1000)  # chunk to read
CHUNK_BYTES = CHUNK_SIZE * 2  # 16bit = 2 bytes, PCM
NUM_PADDING_CHUNKS = int(PADDING_DURATION_MS / CHUNK_DURATION_MS)
NUM_WINDOW_CHUNKS = int(240 / CHUNK_DURATION_MS)
# NUM_WINDOW_CHUNKS = int(400 / CHUNK_DURATION_MS)  # 400 ms/ 30ms  ge
NUM_WINDOW_CHUNKS_END = NUM_WINDOW_CHUNKS * 2

START_OFFSET = int(NUM_WINDOW_CHUNKS * CHUNK_DURATION_MS * 0.5 * RATE)
CHUNK = 512


def handle_int(sig, chunk):
    global leave, got_a_sentence
    leave = True
    got_a_sentence = True


def record_to_file(path, data, sample_width):
    "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()


def normalize(snd_data):
    "Average the volume out"
    MAXIMUM = 32767  # 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

signal.signal(signal.SIGINT, handle_int)


def record_sound():

    vad = webrtcvad.Vad(1)
    pa = pyaudio.PyAudio()
    stream = pa.open(format=FORMAT,
                     channels=CHANNELS,
                     rate=RATE,
                     input=True,
                     start=False,
                     # input_device_index=2,
                     frames_per_buffer=CHUNK_SIZE)

    got_a_sentence = False
    leave = False
    no_time = 0

    while not leave:
        ring_buffer = collections.deque(maxlen=NUM_PADDING_CHUNKS)
        triggered = False
        voiced_frames = []
        ring_buffer_flags = [0] * NUM_WINDOW_CHUNKS
        ring_buffer_index = 0

        ring_buffer_flags_end = [0] * NUM_WINDOW_CHUNKS_END
        ring_buffer_index_end = 0
        buffer_in = ''
        # WangS
        raw_data = array('h')
        index = 0
        start_point = 0
        StartTime = time.time()
        print("* recording: ")
        stream.start_stream()

        while not got_a_sentence and not leave:
            chunk = stream.read(CHUNK_SIZE)
            # add WangS
            raw_data.extend(array('h', chunk))
            index += CHUNK_SIZE
            TimeUse = time.time() - StartTime

            active = vad.is_speech(chunk, RATE)

            if not active:
                no_time += 1
            else:
                no_time = 0

            if no_time >= 248:
                stream.close()
                return False

            sys.stdout.write('1' if active else '_')
            ring_buffer_flags[ring_buffer_index] = 1 if active else 0
            ring_buffer_index += 1
            ring_buffer_index %= NUM_WINDOW_CHUNKS

            ring_buffer_flags_end[ring_buffer_index_end] = 1 if active else 0
            ring_buffer_index_end += 1
            ring_buffer_index_end %= NUM_WINDOW_CHUNKS_END

            # start point detection
            if not triggered:
                ring_buffer.append(chunk)
                num_voiced = sum(ring_buffer_flags)
                if num_voiced > 0.8 * NUM_WINDOW_CHUNKS:
                    sys.stdout.write(' Open ')
                    triggered = True
                    start_point = index - CHUNK_SIZE * 20  # start point
                    # voiced_frames.extend(ring_buffer)
                    ring_buffer.clear()
            # end point detection
            else:
                # voiced_frames.append(chunk)
                ring_buffer.append(chunk)
                num_unvoiced = NUM_WINDOW_CHUNKS_END - sum(ring_buffer_flags_end)
                if num_unvoiced > 0.90 * NUM_WINDOW_CHUNKS_END or TimeUse > 10:
                    sys.stdout.write(' Close ')
                    triggered = False
                    got_a_sentence = True

            sys.stdout.flush()

        sys.stdout.write('\n')
        # data = b''.join(voiced_frames)

        stream.stop_stream()
        print("* done recording")
        got_a_sentence = False

        # write to file
        raw_data.reverse()
        for index in range(start_point):
            raw_data.pop()
        raw_data.reverse()
        raw_data = normalize(raw_data)
        record_to_file("temp.wav", raw_data, 2)
        leave = True

    stream.close()

    return True


def play_sound(file_path='temp.wav'):
    # play wave file, used on pc
    wf = wave.open(file_path, 'rb')
    p = pyaudio.PyAudio()

    stream = p.open(format=p.get_format_from_width(wf.getsampwidth()),
                    channels=wf.getnchannels(),
                    rate=wf.getframerate(),
                    output=True)

    data = wf.readframes(CHUNK)

    while data != b'':
        stream.write(data)
        data = wf.readframes(CHUNK)
    stream.stop_stream()
    stream.close()
    p.terminate()
    return