In digital audio processing, reverb is an essential effect used to simulate the acoustic characteristics of a physical space, such as a concert hall. One popular type of reverb is the hall reverb, which provides a rich and immersive sound experience. In this blog post, we will explore how to implement hall reverbs using C++.

What is Hall Reverb?

Hall reverb is a type of reverb that simulates the sound reflections and reverberations that occur in a large concert hall. It creates a sense of spaciousness and depth, making audio recordings or synthesized sounds more realistic and engaging. Hall reverbs are commonly used in music production, audio mixing, and sound design applications.

Implementing Hall Reverb in C++

To implement hall reverbs in C++, we can use techniques such as convolution and feedback delay networks (FDN). Convolution involves convolving the input audio signal with an impulse response, which captures the characteristics of the desired reverb space. FDN involves feeding the output of the reverb unit back into its inputs with multiple delay lines.

Here’s a simplified example of a hall reverb implementation using C++:

#include <iostream>

// Parameters
const int numDelayLines = 4;
const float delayTimes[] = { 0.03f, 0.06f, 0.09f, 0.12f };
const float gainCoeffs[] = { 0.4f, 0.5f, 0.6f, 0.7f };

// Delay lines
float delayLines[numDelayLines][MAX_NUM_SAMPLES];
int writeIndices[numDelayLines] = { 0 };

// Process audio through hall reverb
void processHallReverb(float* audioIn, float* audioOut, int numSamples)
{
    for (int i = 0; i < numSamples; i++)
    {
        float inputSample = audioIn[i];
        float outputSample = 0.0;
        
        for (int j = 0; j < numDelayLines; j++)
        {
            // Compute read index based on delay time
            int readIndex = writeIndices[j] - static_cast<int>(delayTimes[j] * sampleRate);
            
            // Circular buffer wrap-around
            if (readIndex < 0)
                readIndex += MAX_NUM_SAMPLES;
            
            // Read and scale delayed sample
            float delayedSample = delayLines[j][readIndex] * gainCoeffs[j];
            
            // Accumulate delayed samples
            outputSample += delayedSample;
            
            // Store input sample in delay line
            delayLines[j][writeIndices[j]] = inputSample;
            
            // Update write index
            writeIndices[j]++;
            
            // Circular buffer wrap-around
            if (writeIndices[j] >= MAX_NUM_SAMPLES)
                writeIndices[j] -= MAX_NUM_SAMPLES;
        }
        
        // Output sample
        audioOut[i] = outputSample;
    }
}

int main()
{
    // TODO: Read and process audio
    return 0;
}

This is a basic implementation of hall reverbs using FDN. It uses a set of delay lines with different delay times and gain coefficients to create the reverb effect. The input audio is processed through the delay lines, and the delayed samples are accumulated to create the output audio signal.

Note that this example is simplified and does not include details such as buffer sizes, interpolation, and filtering. A complete hall reverb implementation would require additional considerations to achieve high-quality audio effects.

Conclusion

Implementing audio effects like hall reverbs in C++ can enhance the overall sound quality of your audio applications. By leveraging techniques such as convolution and FDN, you can simulate the acoustic characteristics of different spaces and create immersive audio experiences. Experimenting with different parameters and optimizations can help achieve desired results.

#audioeffects #hallreverb