Implementing audio effects using comb filters in C++

Audio effects are essential for enhancing the quality and creativity of audio recordings. One popular technique used in audio processing is the implementation of comb filters. Comb filters manipulate audio signals by creating a series of delayed and attenuated copies of the original sound, which results in interesting echo and reverberation effects. In this article, we will explore how to implement comb filters in C++ and apply them to audio signals.

Understanding Comb Filters

Before diving into the implementation, let’s briefly understand the concept of comb filters. A comb filter consists of a feedback loop, where a delayed version of the input signal is mixed back with the original signal. This feedback creates a series of repetitions or echoes of the original sound, spaced at regular intervals known as the comb spacing.

The attenuation factor and the delay time determine the characteristics of the comb filter. Higher attenuation values result in shorter reverberation tails, while longer delay times create more pronounced echoes.

Implementing Comb Filters in C++

To implement comb filters in C++, we can start by defining a class that represents the comb filter. Here’s an example implementation:

class CombFilter {
public:
    CombFilter(int delay, float attenuation) : delay(delay), attenuation(attenuation) {
        buffer.resize(delay);
    }
  
    float process(float input) {
        float output = input + buffer[readIndex] * attenuation;
        buffer[writeIndex] = input + buffer[readIndex] * attenuation;
      
        if (++readIndex >= delay)
            readIndex = 0;
      
        if (++writeIndex >= delay)
            writeIndex = 0;
      
        return output;
    }
  
private:
    int delay;
    float attenuation;
    std::vector<float> buffer;
    int readIndex = 0;
    int writeIndex = 0;
};

In this implementation, the CombFilter class takes two parameters - delay and attenuation. The delay represents the length of the buffer and determines the delay time of the comb filter. The attenuation factor controls the strength of the feedback signal.

The process method applies the comb filter to an input sample. It calculates the output by adding the input sample with the delayed and attenuated sample from the buffer. Then, it updates the buffer with the current input and continues to the next index in the buffer.

Applying Comb Filters to Audio Signals

To apply the comb filter to an audio signal, we can process each sample in a loop. Here’s an example code snippet that demonstrates this process:

// Create a comb filter with a delay of 500 samples and attenuation of 0.5
CombFilter combFilter(500, 0.5);

// Process each sample in the audio signal
for (int i = 0; i < audioSignal.size(); i++) {
    float inputSample = audioSignal[i];
    float outputSample = combFilter.process(inputSample);
    processedSignal.push_back(outputSample);
}

In the above code, we create an instance of the CombFilter class with a delay of 500 samples and an attenuation factor of 0.5. Then, we process each sample in the audioSignal vector by passing it to the process method of the comb filter. The resulting output samples are stored in the processedSignal vector.

Conclusion

Comb filters offer a versatile way to manipulate audio signals and create interesting echo and reverberation effects. By implementing comb filters in C++, we can apply these effects to audio recordings or real-time audio streams. Experimenting with different parameters, such as delay and attenuation, can unlock a wide range of creative possibilities in sound design and audio processing.

#AudioEffects #CombFilters