在 Python 中检测和记录音频

我需要捕获音频剪辑作为 WAV 文件,然后我可以传递给另一位巨蟒处理。问题是,我需要确定什么时候有音频存在,然后记录下来,当它静音时停止,然后将该文件传递给处理模块。

我认为波模块应该可以检测出什么时候有纯粹的沉默,然后丢弃它然后一旦检测到除了沉默之外的东西开始录音,然后当线路再次变得沉默时停止录音。

我还是不太明白,谁能给我举个简单的例子。

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I believe the WAVE module does not support recording, just processing existing files. You might want to look at PyAudio for actually recording. WAV is about the world's simplest file format. In paInt16 you just get a signed integer representing a level, and closer to 0 is quieter. I can't remember if WAV files are high byte first or low byte, but something like this ought to work (sorry, I'm not really a python programmer:

from array import array


# you'll probably want to experiment on threshold
# depends how noisy the signal
threshold = 10
max_value = 0


as_ints = array('h', data)
max_value = max(as_ints)
if max_value > threshold:
# not silence

PyAudio code for recording kept for reference:

import pyaudio
import sys


chunk = 1024
FORMAT = pyaudio.paInt16
CHANNELS = 1
RATE = 44100
RECORD_SECONDS = 5


p = pyaudio.PyAudio()


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


print "* recording"
for i in range(0, 44100 / chunk * RECORD_SECONDS):
data = stream.read(chunk)
# check for silence here by comparing the level with 0 (or some threshold) for
# the contents of data.
# then write data or not to a file


print "* done"


stream.stop_stream()
stream.close()
p.terminate()

You might want to look at csounds, also. It has several API's, including Python. It might be able to interact with an A-D interface and gather sound samples.

As a follow up to Nick Fortescue's answer, here's a more complete example of how to record from the microphone and process the resulting data:

from sys import byteorder
from array import array
from struct import pack


import pyaudio
import wave


THRESHOLD = 500
CHUNK_SIZE = 1024
FORMAT = pyaudio.paInt16
RATE = 44100


def is_silent(snd_data):
"Returns 'True' if below the 'silent' threshold"
return max(snd_data) < THRESHOLD


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


def trim(snd_data):
"Trim the blank spots at the start and end"
def _trim(snd_data):
snd_started = False
r = array('h')


for i in snd_data:
if not snd_started and abs(i)>THRESHOLD:
snd_started = True
r.append(i)


elif snd_started:
r.append(i)
return r


# Trim to the left
snd_data = _trim(snd_data)


# Trim to the right
snd_data.reverse()
snd_data = _trim(snd_data)
snd_data.reverse()
return snd_data


def add_silence(snd_data, seconds):
"Add silence to the start and end of 'snd_data' of length 'seconds' (float)"
silence = [0] * int(seconds * RATE)
r = array('h', silence)
r.extend(snd_data)
r.extend(silence)
return r


def record():
"""
Record a word or words from the microphone and
return the data as an array of signed shorts.


Normalizes the audio, trims silence from the
start and end, and pads with 0.5 seconds of
blank sound to make sure VLC et al can play
it without getting chopped off.
"""
p = pyaudio.PyAudio()
stream = p.open(format=FORMAT, channels=1, rate=RATE,
input=True, output=True,
frames_per_buffer=CHUNK_SIZE)


num_silent = 0
snd_started = False


r = array('h')


while 1:
# little endian, signed short
snd_data = array('h', stream.read(CHUNK_SIZE))
if byteorder == 'big':
snd_data.byteswap()
r.extend(snd_data)


silent = is_silent(snd_data)


if silent and snd_started:
num_silent += 1
elif not silent and not snd_started:
snd_started = True


if snd_started and num_silent > 30:
break


sample_width = p.get_sample_size(FORMAT)
stream.stop_stream()
stream.close()
p.terminate()


r = normalize(r)
r = trim(r)
r = add_silence(r, 0.5)
return sample_width, r


def record_to_file(path):
"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()


if __name__ == '__main__':
print("please speak a word into the microphone")
record_to_file('demo.wav')
print("done - result written to demo.wav")

The pyaudio website has many examples that are pretty short and clear: http://people.csail.mit.edu/hubert/pyaudio/

Update 14th of December 2019 - Main example from the above linked website from 2017:


"""PyAudio Example: Play a WAVE file."""


import pyaudio
import wave
import sys


CHUNK = 1024


if len(sys.argv) < 2:
print("Plays a wave file.\n\nUsage: %s filename.wav" % sys.argv[0])
sys.exit(-1)


wf = wave.open(sys.argv[1], '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 != '':
stream.write(data)
data = wf.readframes(CHUNK)


stream.stop_stream()
stream.close()


p.terminate()

Thanks to cryo for improved version that I based my tested code below:

#Instead of adding silence at start and end of recording (values=0) I add the original audio . This makes audio sound more natural as volume is >0. See trim()
#I also fixed issue with the previous code - accumulated silence counter needs to be cleared once recording is resumed.


from array import array
from struct import pack
from sys import byteorder
import copy
import pyaudio
import wave


THRESHOLD = 500  # audio levels not normalised.
CHUNK_SIZE = 1024
SILENT_CHUNKS = 3 * 44100 / 1024  # about 3sec
FORMAT = pyaudio.paInt16
FRAME_MAX_VALUE = 2 ** 15 - 1
NORMALIZE_MINUS_ONE_dB = 10 ** (-1.0 / 20)
RATE = 44100
CHANNELS = 1
TRIM_APPEND = RATE / 4


def is_silent(data_chunk):
"""Returns 'True' if below the 'silent' threshold"""
return max(data_chunk) < THRESHOLD


def normalize(data_all):
"""Amplify the volume out to max -1dB"""
# MAXIMUM = 16384
normalize_factor = (float(NORMALIZE_MINUS_ONE_dB * FRAME_MAX_VALUE)
/ max(abs(i) for i in data_all))


r = array('h')
for i in data_all:
r.append(int(i * normalize_factor))
return r


def trim(data_all):
_from = 0
_to = len(data_all) - 1
for i, b in enumerate(data_all):
if abs(b) > THRESHOLD:
_from = max(0, i - TRIM_APPEND)
break


for i, b in enumerate(reversed(data_all)):
if abs(b) > THRESHOLD:
_to = min(len(data_all) - 1, len(data_all) - 1 - i + TRIM_APPEND)
break


return copy.deepcopy(data_all[_from:(_to + 1)])


def record():
"""Record a word or words from the microphone and
return the data as an array of signed shorts."""


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


silent_chunks = 0
audio_started = False
data_all = array('h')


while True:
# little endian, signed short
data_chunk = array('h', stream.read(CHUNK_SIZE))
if byteorder == 'big':
data_chunk.byteswap()
data_all.extend(data_chunk)


silent = is_silent(data_chunk)


if audio_started:
if silent:
silent_chunks += 1
if silent_chunks > SILENT_CHUNKS:
break
else:
silent_chunks = 0
elif not silent:
audio_started = True


sample_width = p.get_sample_size(FORMAT)
stream.stop_stream()
stream.close()
p.terminate()


data_all = trim(data_all)  # we trim before normalize as threshhold applies to un-normalized wave (as well as is_silent() function)
data_all = normalize(data_all)
return sample_width, data_all


def record_to_file(path):
"Records from the microphone and outputs the resulting data to 'path'"
sample_width, data = record()
data = pack('<' + ('h' * len(data)), *data)


wave_file = wave.open(path, 'wb')
wave_file.setnchannels(CHANNELS)
wave_file.setsampwidth(sample_width)
wave_file.setframerate(RATE)
wave_file.writeframes(data)
wave_file.close()


if __name__ == '__main__':
print("Wait in silence to begin recording; wait in silence to terminate")
record_to_file('demo.wav')
print("done - result written to demo.wav")
import pyaudio
import wave
from array import array


FORMAT=pyaudio.paInt16
CHANNELS=2
RATE=44100
CHUNK=1024
RECORD_SECONDS=15
FILE_NAME="RECORDING.wav"


audio=pyaudio.PyAudio() #instantiate the pyaudio


#recording prerequisites
stream=audio.open(format=FORMAT,channels=CHANNELS,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)


#starting recording
frames=[]


for i in range(0,int(RATE/CHUNK*RECORD_SECONDS)):
data=stream.read(CHUNK)
data_chunk=array('h',data)
vol=max(data_chunk)
if(vol>=500):
print("something said")
frames.append(data)
else:
print("nothing")
print("\n")




#end of recording
stream.stop_stream()
stream.close()
audio.terminate()
#writing to file
wavfile=wave.open(FILE_NAME,'wb')
wavfile.setnchannels(CHANNELS)
wavfile.setsampwidth(audio.get_sample_size(FORMAT))
wavfile.setframerate(RATE)
wavfile.writeframes(b''.join(frames))#append frames recorded to file
wavfile.close()

I think this will help.It is a simple script which will check if there is a silence or not.If silence is detected it will not record otherwise it will record.