通过实施核心音频后处理的低通滤波器 [英] Implementing a post-processed low-pass filter using core audio

问题描述

我已经实现使用基于时间值一个基本低通滤波器。这是好的，但试图找到正确的时间片是猜测，并给出了基于不同的输入音频文件不同的结果。以下是我现在有：

I have implemented a rudimentary low-pass filter using a time based value. This is ok, but trying to find the correct time slice is guess work, and gives different results based on different input audio files. Here is what I have now:

- (void)processDataWithInBuffer:(const int16_t *)buffer outBuffer:(int16_t *)outBuffer sampleCount:(int)len {   
    BOOL positive;
    for(int i = 0; i < len; i++) {
        positive = (buffer[i] >= 0);
        currentFilteredValueOfSampleAmplitude = LOWPASSFILTERTIMESLICE * (float)abs(buffer[i]) + (1.0 - LOWPASSFILTERTIMESLICE) * previousFilteredValueOfSampleAmplitude;
        previousFilteredValueOfSampleAmplitude = currentFilteredValueOfSampleAmplitude; 
        outBuffer[i] = currentFilteredValueOfSampleAmplitude * (positive ? 1 : -1);
    }
}

我能做些什么来转换这个code到code，让我通过一定的分贝水平削减超过一定频率赫兹？

What can I do to convert this code into code that will allow me to cut frequencies over a certain hz by a certain db level?

推荐答案

我强烈建议数字食谱ç。以外的是，我不知道我可以帮你。

I strongly recommend numerical recipes in c. Outside of that, I'm not sure I can help you.

当你设计一个过滤器，需要根据频率来计算滤波器的系数，所以你几乎需要一个类来处理它，而不是只是一个函数。

When you design a filter, you need to calculate the coefficients of that filter based on the frequency so you almost need a class to handle it, not just a function.

这是在 C ++ 但它应该得到你开始。对不起，我不能提供一个具体的答案。

This is in C++ but it should get you started. Sorry I can't provide a concrete answer.

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