我已经让 OpenCV 和 PyAudio 都工作了，但是我不确定如何将它们同步在一起。我无法从 OpenCV 获取帧速率并测量帧时时刻刻变化的调用时间。然而，对于 PyAudio 来说，它的基础是获取一定的采样率。我如何将它们同步到相同的速率。我假设编解码器有一些标准或某种方式来做到这一点。 （我已经尝试过谷歌，我得到的只是有关口型同步的信息：/）。
OpenCV 帧率
from __future__ import division
import time
import math
import cv2, cv
vc = cv2.VideoCapture(0)
# get the frame
while True:
before_read = time.time()
rval, frame = vc.read()
after_read = time.time()
if frame is not None:
print len(frame)
print math.ceil((1.0 / (after_read - before_read)))
cv2.imshow("preview", frame)
if cv2.waitKey(1) & 0xFF == ord('q'):
break
else:
print "None..."
cv2.waitKey(1)
# display the frame
while True:
cv2.imshow("preview", frame)
if cv2.waitKey(1) & 0xFF == ord('q'):
break
抓取并保存音频
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"
print "\n\n\n\n\n\n\n\n"
print max(snd_data)
print "\n\n\n\n\n\n\n\n"
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)"
r = array('h', [0 for i in xrange(int(seconds*RATE))])
r.extend(snd_data)
r.extend([0 for i in xrange(int(seconds*RATE))])
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(1024))
if byteorder == 'big':
snd_data.byteswap()
print "\n\n\n\n\n\n"
print len(snd_data)
print snd_data
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 > 1:
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")
