使用python 實現音頻疊加的方法

#!/usr/bin/env python
# -*- coding:utf-8 -*-
import os
import wave
import numpy as np
import pyaudio
import librosa
import soundfile as sf
import scipy.signal as signal
import struct
# ok，音頻疊加！我這里4.wav和5.wav都是5s的音頻，還沒有測試時長不同的音頻！
# 參考文檔：https://www.cnblogs.com/xingshansi/p/6799994.html
x,_ = librosa.load('D:/4.wav', sr=16000) #需要修改的地方：音頻1
sf.write('t1.wav',x,16000)
y,_ = librosa.load('D:/5.wav', sr=16000) #需要修改的地方：音頻2
sf.write('t2.wav',y,16000)
f1 = wave.open('t1.wav', 'rb')
f2 = wave.open('t2.wav', 'rb')
 
# 音頻1的數據
params1 = f1.getparams()
nchannels1, sampwidth2, framerate1, nframes1, comptype1, compname1 = params1[:6]
print(nchannels1, sampwidth2, framerate1, nframes1, comptype1, compname1)
f1_str_data = f1.readframes(nframes1)
f1.close()
f1_wave_data = np.frombuffer(f1_str_data, dtype=np.int16)
 
# 音頻2的數據
params2 = f2.getparams()
nchannels2, sampwidth3, framerate2, nframes2, comptype2, compname2 = params2[:6]
print(nchannels2, sampwidth3, framerate2, nframes2, comptype2, compname2)
f2_str_data = f2.readframes(nframes2)
f2.close()
f2_wave_data = np.frombuffer(f2_str_data, dtype=np.int16)
 
# 對不同長度的音頻用數據零對齊補位
if nframes1 < nframes2:
    length = abs(nframes2 - nframes1)
    temp_array = np.zeros(length, dtype=np.int16)
    rf1_wave_data = np.concatenate((f1_wave_data, temp_array))
    rf2_wave_data = f2_wave_data
elif nframes1 > nframes2:
    length = abs(nframes2 - nframes1)
    temp_array = np.zeros(length, dtype=np.int16)
    rf2_wave_data = np.concatenate((f2_wave_data, temp_array))
    rf1_wave_data = f1_wave_data
else:
    rf1_wave_data = f1_wave_data
    rf2_wave_data = f2_wave_data
 
# ================================
# 合并1和2的數據
new_wave_data = rf1_wave_data + rf2_wave_data
new_wave_data = new_wave_data*1.0/(max(abs(new_wave_data)))#wave幅值歸一化
new_wave = new_wave_data.tostring()
 
p = pyaudio.PyAudio()
CHANNELS = 1
FORMAT = pyaudio.paInt16
 
# 寫文件
framerate = 44100
time = 10
 
# 產生10秒44.1kHz的100Hz - 1kHz的頻率掃描波。沒用！
t = np.arange(0, time, 1.0/framerate)
wave_data = signal.chirp(t, 100, time, 1000, method='linear') * 10000
wave_data = wave_data.astype(np.short)
 
# 打開WAV文檔
f = wave.open(r"D:\6.wav", "wb") # 需要修改的地方：輸出音頻
 
# 配置聲道數、量化位數和取樣頻率
nchannels = 1 #單通道為例
sampwidth = 2
data_size = len(new_wave_data)
framerate = 16000 # 設置為44100就是1s，設置為8000就是10s，只有16000才是5s是對的。這里還沒搞懂！
nframes = data_size
comptype = "NONE"
compname = "not compressed"
f.setparams((nchannels, sampwidth, framerate, nframes, comptype, compname))
# 將wav_data轉換為二進制數據寫入文件
# f.writeframes(new_wave)
for v in new_wave_data:
    f.writeframes(struct.pack('h', int(v * 64000 / 2)))
f.close()
 
# 實現錄音,暫時用不到。
def record(re_frames, WAVE_OUTPUT_FILENAME):
    print("開始錄音")
    wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
    wf.setnchannels(CHANNELS)
    wf.setsampwidth(p.get_sample_size(FORMAT))
    wf.setframerate(RATE)
    wf.writeframes(re_frames)
    wf.close()
    print("關閉錄音")