我有一个包含音频样本的双精度阵列,以44100的采样率直接从麦克风传来。我想要得到基频(样本包含幅度)。在自相关页面的维基百科上,我找到了基于维纳 - 钦钦定理的解决方案的描述,我通过对互联网的更多研究完成了该算法,并最终编写了下面的代码,但我不确定它是否正确:
private double determineFrequency(double[] signal) {
//Get a FastFourierTransformer instance (Apache library)
FastFourierTransformer fft = new FastFourierTransformer(DftNormalization.STANDARD);
//The size of the array used by the fft must be a power of two, wrapping
//the original array in a bigger one padded to zero
//NOTE: Here I assume that the input array is smaller than 8192
double[] paddedSignal = new double[8192];
System.arraycopy(signal, 0, paddedSignal, 0, signal.length);
//First fft (forward) to switch from amplitude domain to the frequency domain
Complex[] transformed = fft.transform(paddedSignal, TransformType.FORWARD);
// Calculate the conjugate of the complex array
for (int i=0; i
transformed[i] = transformed[i].conjugate();
//Second fft (inverse) to complete the autocorrelation
transformed = fft.transform(transformed, TransformType.INVERSE);
//Calculate the array of corresponding real values to switch
// from the frequency domain to the amplitude domain
double[] autocorrelationMatrix = new double[transformed.length];
for (int i=0; i
if (Double.isNaN(transformed[i].abs()) || Double.isInfinite(transformed[i].abs()))
autocorrelationMatrix[i] = 0;
else
autocorrelationMatrix[i] = transformed[i].abs();
}
//Get the index of the max amplitude
Integer indexOfMax = Utils.indexOfMax(autocorrelationMatrix);
return transformed[indexOfMax].getReal()*audioFormat.getSampleRate()/transformed.length;
}