Python在检测到的声音上录制音频

Question

我希望有一个python脚本运行在后台，并使用pyaudio录制声音文件时，麦克风的阈值已达到一定的点。这是给双向无线电网络上的监视器用的。因此，我们只想录制传输的音频。

牢记的任务：

• 在n%门阈值上录制音频输入

• 经过这么多秒的静音后停止录制

• 音频播放后，请将录音保留这么多秒

• 第2阶段：将数据输入MySQL数据库以搜索录制的内容

我在看一个类似的文件结构

/home/Recodings/2013/8/23/12-33.wav将是2013年8月23日12:33.wav的传输记录

我使用了

Detect and record a sound with python

我现在有点不知所措，如果能给我一点指导，我将不胜感激

谢谢你

Answer 1

不久前我写了一些步骤

• Record audio input on a n% gate threshold

A:为“Silence”启动一个布尔变量类型，您可以计算RMS以确定Silence是真是假，设置一个RMS阈值

• stop recording after so many seconds of silence

A:您需要计算一个超时吗？因为它会得到帧速率、块大小和您需要的秒数，来计算您的超时时间（帧速率/块*最大秒数）

• keep recording for so many seconds after audio

A：如果静音为false==（RMS>；阈值）获取音频的最后一块数据（LastBlock），并只保留记录：-）

• Phase 2: input data into MySQL database to search the recordings

A：这一步由你决定

源代码：

import pyaudio
import math
import struct
import wave

#Assuming Energy threshold upper than 30 dB
Threshold = 30

SHORT_NORMALIZE = (1.0/32768.0)
chunk = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 16000
swidth = 2
Max_Seconds = 10
TimeoutSignal=((RATE / chunk * Max_Seconds) + 2)
silence = True
FileNameTmp = '/home/Recodings/2013/8/23/12-33.wav'
Time=0
all =[]

def GetStream(chunk):
    return stream.read(chunk)
def rms(frame):
    count = len(frame)/swidth
    format = "%dh"%(count)
    # short is 16 bit int
    shorts = struct.unpack( format, frame )

    sum_squares = 0.0
    for sample in shorts:
        n = sample * SHORT_NORMALIZE
        sum_squares += n*n
    # compute the rms 
    rms = math.pow(sum_squares/count,0.5);
    return rms * 1000

def WriteSpeech(WriteData):
    stream.stop_stream()
    stream.close()
    p.terminate()
    wf = wave.open(FileNameTmp, 'wb')
    wf.setnchannels(CHANNELS)
    wf.setsampwidth(p.get_sample_size(FORMAT))
    wf.setframerate(RATE)
    wf.writeframes(WriteData)
    wf.close()

def KeepRecord(TimeoutSignal, LastBlock):
    all.append(LastBlock)
    for i in range(0, TimeoutSignal):
        try:
            data = GetStream(chunk)
        except:
            continue
        #I chage here (new Ident)
        all.append(data)

    print "end record after timeout";
    data = ''.join(all)
    print "write to File";
    WriteSpeech(data)
    silence = True
    Time=0
    listen(silence,Time)     

def listen(silence,Time):
    print "waiting for Speech"
    while silence:
        try:
            input = GetStream(chunk)
        except:
            continue
        rms_value = rms(input)
        if (rms_value > Threshold):
            silence=False
            LastBlock=input
            print "hello ederwander I'm Recording...."
            KeepRecord(TimeoutSignal, LastBlock)
        Time = Time + 1
        if (Time > TimeoutSignal):
            print "Time Out No Speech Detected"
            sys.exit()

p = pyaudio.PyAudio()

stream = p.open(format = FORMAT,
    channels = CHANNELS,
    rate = RATE,
    input = True,
    output = True,
    frames_per_buffer = chunk)

listen(silence,Time)

Answer 2

所以你只需要getLevel(data)函数？ 一个简单的方法是：

def getLevel(data):
   sqrsum = 0
   for b in data:
      b = ord(b)
      sqrsum+=b*b
   return sqrsum

这应该会随着音量的增加而增加。通过反复试验适当地设置阈值。

Answer 3

当前的首选答案有些过时，只适用于python 2。这是为python 3更新的版本。它将函数包装成类，并将所有内容打包成一个简单易用的版本。请注意，上面的答案和我的脚本有一个关键的区别：

顶部的脚本记录一个文件，然后停止，而我的脚本则在检测到噪音时保持记录，并在运行时将记录转储到目录中。

两个脚本的主要思想非常相似：

步骤1:“侦听”直到rms大于阈值

第2步：开始录制，设置停止录制的计时器，==超时长度

步骤3：如果在计时器超时之前，rms再次打破阈值，则重置计时器

步骤4：现在计时器已过期，请将录制写入目录并返回步骤1

import pyaudio
import math
import struct
import wave
import time
import os

Threshold = 10

SHORT_NORMALIZE = (1.0/32768.0)
chunk = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 16000
swidth = 2

TIMEOUT_LENGTH = 5

f_name_directory = r'C:\Users\Jason\PyCharmProjects\AutoRecorder\records'

class Recorder:

    @staticmethod
    def rms(frame):
        count = len(frame) / swidth
        format = "%dh" % (count)
        shorts = struct.unpack(format, frame)

        sum_squares = 0.0
        for sample in shorts:
            n = sample * SHORT_NORMALIZE
            sum_squares += n * n
        rms = math.pow(sum_squares / count, 0.5)

        return rms * 1000

    def __init__(self):
        self.p = pyaudio.PyAudio()
        self.stream = self.p.open(format=FORMAT,
                                  channels=CHANNELS,
                                  rate=RATE,
                                  input=True,
                                  output=True,
                                  frames_per_buffer=chunk)

    def record(self):
        print('Noise detected, recording beginning')
        rec = []
        current = time.time()
        end = time.time() + TIMEOUT_LENGTH

        while current <= end:

            data = self.stream.read(chunk)
            if self.rms(data) >= Threshold: end = time.time() + TIMEOUT_LENGTH

            current = time.time()
            rec.append(data)
        self.write(b''.join(rec))

    def write(self, recording):
        n_files = len(os.listdir(f_name_directory))

        filename = os.path.join(f_name_directory, '{}.wav'.format(n_files))

        wf = wave.open(filename, 'wb')
        wf.setnchannels(CHANNELS)
        wf.setsampwidth(self.p.get_sample_size(FORMAT))
        wf.setframerate(RATE)
        wf.writeframes(recording)
        wf.close()
        print('Written to file: {}'.format(filename))
        print('Returning to listening')



    def listen(self):
        print('Listening beginning')
        while True:
            input = self.stream.read(chunk)
            rms_val = self.rms(input)
            if rms_val > Threshold:
                self.record()

a = Recorder()

a.listen()