树莓派基于语音活性检测VAD的应用

如果你想尝试用树莓派打造一款语音对话机器人，那么你肯定避免不了一点，录音！

前言：

我之前的文章中提到过alsa的arecord录制，录音时间固定，当程序运行一次后就会出现arecord资源被占用的情况，除非你把这个进程kill掉。然后事实上，你不可能每次运行完就手动kill一次，那样太麻烦了。
后来我用pyaudio，完美解决了上述的问题。直到今天，我遇到了语音活性检测VAD···

科普来了~
语音活性检测 (Voice activity detection，VAD), 也称为 speech activity detection or speech detection, 是一项用于语音处理的技术，目的是检测语音信号是否存在。VAD技术主要用于语音编码和语音识别。
在本篇文章中用到的是WebRTC之VAD算法。
其主要功能：

1. 自动打断
2. 去掉语音中的静音成分
3. 获取输入语音中有效语音
4. 去除噪声，对语音进行增强

# coding=utf-8
import webrtcvad   # 检测判断一组语音数据是否为空语音；
import collections
import sys
import signal
import pyaudio    # 从设备节点读取原始音频流数据，音频编码是PCM格式
 
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       # //检验音频帧长度，只支持10/20/30ms
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
 
 
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)
 
 
"""
当检测到持续时间长度 T1 vad检测都有语音活动，可以判定为语音起始。

当检测到持续时间长度 T2 vad检测都没有有语音活动，可以判定为语音结束。
""" 
 
def record_sound(file_path='record.wav'):
    # 录音，有声音自动写入文件，默认为'record.wav'，声音结束后录音也停止，调用一次，录制一个片段
    vad = webrtcvad.Vad(1) # 这个参数可为1，2，3，可改变灵敏度，越大越粗犷
    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（原作者的名字）
        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)
 
            sys.stdout.write('~' 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
 
            # 开始端点检测
            if not triggered:
                ring_buffer.append(chunk)
                num_voiced = sum(ring_buffer_flags)
                if num_voiced > 0.8 * NUM_WINDOW_CHUNKS:      # 声音起始
                    sys.stdout.write(' Open ')
                    triggered = True
                    start_point = index - CHUNK_SIZE * 20  # start point
                    # voiced_frames.extend(ring_buffer)
                    ring_buffer.clear()
            # 结束端点检测
            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:   # 声音结束
                    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(file_path, raw_data, 2)
        leave = True
 
    stream.close()
 
    return True
 
CHUNK = 512  # 512是树莓派能使用的最大的CHUNK
 
def play_sound(file_path='test.wav'):
    # 播放声音文件，默认为'test.wav'
    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
 

if __name__ == '__main__':
	record_sound('record.wav')
	play_sound('record.wav')

参考：gdjzkj.com/?m=home&c=View&a=index&aid=118