Zim*_*m3r 15 python opencv pyaudio
我已经让OpenCV和PyAudio都工作了但是我不确定如何将它们同步到一起.我无法从OpenCV获得帧速率,并且测量帧的调用时间会随时变化.然而,对于PyAudio,它的基础是获取某个采样率.我如何以相同的速率同步它们.我假设有一些标准或某种方式编解码器做它.(我已经尝试了谷歌,我得到的是唇同步的信息:/).
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
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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")
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小智 2
我认为你最好使用 GSreamer 或 ffmpeg,或者如果你使用的是 Windows,则使用 DirectShow。这些库可以处理音频和视频,并且应该有某种多路复用器来允许您正确混合视频和音频。
但是如果你真的想使用 Opencv 来做到这一点,你应该能够使用它VideoCapture
来获取帧速率,你尝试过使用这个吗?
fps = cv.GetCaptureProperty(vc, CV_CAP_PROP_FPS)
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另一种方法是将帧数除以持续时间来估计 fps:
nFrames = cv.GetCaptureProperty(vc, CV_CAP_PROP_FRAME_COUNT)
cv.SetCaptureProperty(vc, CV_CAP_PROP_POS_AVI_RATIO, 1)
duration = cv.GetCaptureProperty(vc, CV_CAP_PROP_POS_MSEC)
fps = 1000 * nFrames / duration;
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我不确定我是否理解您在这里尝试做什么:
before_read = time.time()
rval, frame = vc.read()
after_read = time.time()
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在我看来,这样做after_read - before_read
只测量 OpenCV 加载下一帧所需的时间,它不测量 fps。OpenCV 并不尝试进行播放,它只是加载帧,并且它会尝试以最快的速度执行此操作,我认为没有办法对其进行配置。我认为waitKey(1/fps)
在显示每一帧后放置一个将达到您所寻找的效果。
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