Answers:
作为Nick Fortescue回答的后续措施，下面是一个更完整的示例，说明如何从麦克风进行录音并处理结果数据：
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")xrange也没有range必要add_silence（因此现在不存在了）。我认为Arek可能在这里-从沉默到“单词”的过渡听起来太生涩了。我认为还有其他答案也可以解决。
我相信WAVE模块不支持记录，仅处理现有文件。您可能需要查看PyAudio进行实际录制。WAV是关于世界上最简单的文件格式。在paInt16中，您仅获得一个表示电平的有符号整数，而接近0则更安静。我不记得WAV文件是高字节开头还是低字节开头，但是类似这样的东西应该起作用（对不起，我并不是真正的python程序员：
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保留用于记录的PyAudio代码以供参考：
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()感谢cryo的改进版本，我基于下面的测试代码：
#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")return copy.deepcopy(data_all[_from:(_to + 1)])为copy.deepcopy(data_all[int(_from):(int(_to) + 1)])
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()我认为这会有所帮助，这是一个简单的脚本，它将检查是否存在静音，如果检测到静音，则不会记录，否则会记录。
pyaudio网站上有许多简短明了的示例：http ://people.csail.mit.edu/hubert/pyaudio/
2019年12月14日更新-上述链接网站自2017年以来的主要示例：
"""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()