用matlab让高低音频重叠,音频分帧，叠加后仍有不连续

我还是把我的代码发上来(完整代码见我传上的文档)

滤波器系数(这里给的是7介的我用的是127介的，太长了占地方，就用7介示意了 是S16Q15格式)

#if (FILTER == LPF_7_TAP)

// Fs = 48KHz, Fc = 2KHz, Hamming window 7 taps

Int16 coeffs[LPF_7_TAP] = {

0x00c5, 0x0328, 0x081f, 0x0aab, 0x081f, 0x0328, 0x00c5

};

对滤波器做FFT变换函数

/* FFT_Coeffs: Calculate and store the FFT of the filter coefficients */

void FFT_Coeffs()

{

Uint16 i, fft_flag, scale_flag, out_sel;

Int32 *scratch, *data_br, *coeffs_fft, coeffSum;

scratch = scratch_buf;

data_br = data_br_buf;

coeffs_fft = coeffs_fft_buf;

// put coeffs into re|im 32-bit words

// zero-pad to FFT_LENGTH

coeffSum = 0; // make sure sum <= 1.0 for COLA properties

for(i=0; i

{

if(i

{

scratch= ((Int32)coeffs<<16);

coeffSum += coeffs;

}

else

{

scratch= 0;

}

}

// FFT and store in coeffs_fft[]

fft_flag = FFT_FLAG;

//scale_flag = NOSCALE_FLAG;  //why is this necessary

scale_falg = SCALE_FLAG;

hwafft_br(scratch, data_br, FFT_LENGTH);  // bit-reverse zero-padded coefficients //位倒序

out_sel = (*Hwafft_Func)(data_br, coeffs_fft, fft_flag, scale_flag); // perform FFT //这个函数是TI写好的硬件FFT函数 if(out_sel == OUT_SEL_DATA)

{

buff_copy32(data_br, coeffs_fft, FFT_LENGTH);

}

}

//对输入信号做l滤波

/* FFT_Filter:

* FFT the current window of samples

* Complex Multiply FFT output with FFT of filter coefficients

* (Multiplication in Frequency Domain = Convolution in Time Domain

* IFFT Complex Product to obtain filtered window plus ringing

*/</