因为工作中会经常遇到不同采样率的声音文件的问题,特意写了一下重采样的程序。
原理就是把采样点转换到时间刻度之后再进行插值,经过测试,是没有问题的。
#!/usr/bin/env python
# -*- coding: utf-8 -*-
# @Time : 17-7-21 下午2:32
# @Author : Lei.Jinggui
# @Site : http://blog.csdn.net/lccever
# @File : Resample.py
# @Software: PyCharm Community Edition
# @contact: [email protected]
import numpy as np
def Resample(input_signal,src_fs,tar_fs):
'''
:param input_signal:输入信号
:param src_fs:输入信号采样率
:param tar_fs:输出信号采样率
:return:输出信号
'''
dtype = input_signal.dtype
audio_len = len(input_signal)
audio_time_max = 1.0*(audio_len-1) / src_fs
src_time = 1.0 * np.linspace(0,audio_len,audio_len) / src_fs
tar_time = 1.0 * np.linspace(0,np.int(audio_time_max*tar_fs),np.int(audio_time_max*tar_fs)) / tar_fs
output_signal = np.interp(tar_time,src_time,input_signal).astype(dtype)
return output_signal
if __name__ == '__main__':
import wave
import pyaudio
def playSound(audio_data_short, framerate=16000, channels=1):
preply = pyaudio.PyAudio()
# 播放声音
streamreply = preply.open(format=pyaudio.paInt16,
channels=channels,
rate=framerate,
output=True)
data = audio_data_short.tostring()
streamreply.write(data)
streamreply.close()
preply.terminate()
wave_file = 'test.wav'
audio_file = wave.open(wave_file, 'rb')
audio_data = audio_file.readframes(audio_file.getnframes())
audio_data_short = np.fromstring(audio_data, np.short)
src_fs = audio_file.getframerate()
src_chanels = audio_file.getnchannels()
if src_chanels > 1:
audio_data_short = audio_data_short[::src_chanels]
tar_fs = np.int(src_fs * 0.5)
playSound(audio_data_short,framerate=src_fs)
audio_data_short0 = Resample(audio_data_short,src_fs,tar_fs)
playSound(audio_data_short0,framerate=tar_fs)