Improve Your C++ Skills by Building an Audio Plugin (3-Band Compressor)

As a full-stack developer, one of the best ways to level up your C++ programming skills is by taking on a challenging real-world project. Building an audio plugin, like a 3-band compressor, is an excellent way to gain practical experience with C++ while also learning about digital signal processing (DSP) and audio programming concepts.

In this expert-level guide, we‘ll walk through the process of building a professional-grade audio plugin using modern C++ and the JUCE framework. By the end, you‘ll have a fully-functional 3-band compressor plugin with an integrated spectrum analyzer that works seamlessly in popular digital audio workstations (DAWs) on both Windows and macOS.

What is an Audio Plugin?

Before we dive into the code, let‘s clarify what an audio plugin actually is. In music production and audio engineering, a plugin is a piece of software that extends the functionality of a host application, such as a DAW. Plugins come in various types, including virtual instruments, effects processors, and analysis tools.

Audio plugins are an integral part of modern music production workflows. They allow producers and engineers to shape and manipulate audio signals in creative ways, whether it‘s applying compression to even out dynamics, using EQ to balance frequency content, or adding spatial effects like reverb and delay.

Under the hood, audio plugins are essentially miniature applications that communicate with the host software via a standardized plugin API. The most common plugin APIs are Steinberg‘s VST (Virtual Studio Technology), Apple‘s AU (Audio Units), and Avid‘s AAX (Avid Audio eXtension). To enable cross-platform compatibility, we‘ll be using the JUCE framework to develop our plugin.

Improving C++ Skills with Audio Programming

You might be wondering, why build an audio plugin to improve your C++ skills? Aren‘t there simpler projects to start with? While there are certainly easier ways to practice C++, building an audio plugin offers several unique benefits:

1. Real-time performance: Audio plugins must process audio data in real-time with minimal latency. This requires efficient, optimized code that takes advantage of modern C++ features and best practices.

2. Complex algorithms: Implementing DSP algorithms like filters, compressors, and Fourier transforms is a great way to deepen your understanding of advanced mathematical and computational concepts.

3. Cross-platform development: Using a framework like JUCE allows you to write a single codebase that compiles to multiple plugin formats and works on different operating systems. This exposes you to the challenges (and solutions) of cross-platform C++ development.

4. Graphical interfaces: Audio plugins typically have interactive GUIs that control their parameters. Building a custom GUI from scratch is an excellent opportunity to practice working with graphics frameworks and event-driven programming.

5. Practical applications: Unlike many contrived programming examples, an audio plugin is a real-world tool that you can actually use in your own music productions or share with other artists and producers.

So, not only will building an audio plugin help you become a better C++ programmer, but it will also equip you with valuable skills and knowledge in the field of audio software development.

Introducing the JUCE Framework

To streamline the plugin development process, we‘ll be using the JUCE framework. JUCE (Jules‘ Utility Class Extensions) is a powerful cross-platform application framework designed for creating audio plugins and standalone audio applications.

Some of the key features of JUCE include:

• Cross-platform compatibility (Windows, macOS, Linux, iOS, Android)
• Support for multiple plugin formats (VST, AU, AAX, RTAS)
• Built-in classes for DSP, MIDI, graphics, and GUI components
• Integrated IDE (the Projucer) for generating project files and managing builds
• Extensive documentation and active community support

By leveraging the JUCE framework, we can focus on writing the actual DSP and GUI code for our plugin without worrying about the low-level details of the plugin APIs or build process.

Building the 3-Band Compressor: DSP

Now that we have a basic understanding of audio plugins and the JUCE framework, let‘s start building our 3-band compressor!

We‘ll begin by implementing the DSP (digital signal processing) portion of the plugin, which is responsible for processing the incoming audio signal. Here‘s a high-level overview of the steps involved:

1. Creating the processor class: In JUCE, the audio processing code lives in a class that inherits from the AudioProcessor base class. This is where we‘ll define our plugin‘s parameters and implement the core DSP algorithms.

2. Designing the compressor algorithm: A compressor is a dynamics processor that attenuates the level of an audio signal when it exceeds a certain threshold. We‘ll need to implement the basic compressor parameters (threshold, ratio, attack time, release time) and the gain computation logic.

3. Splitting the signal into frequency bands: To create a 3-band compressor, we first need to split the incoming audio signal into three frequency ranges (low, mid, high) using a crossover filter. We‘ll use a 4th-order Linkwitz-Riley filter for this purpose.

4. Applying compression to each band: Once we have the split signal, we‘ll apply our compressor algorithm to each frequency band independently. This allows us to control the dynamics of each band separately, which is useful for tasks like taming harsh high frequencies or adding punch to the low end.

5. Recombining the compressed bands: After processing each frequency band, we‘ll sum the compressed signals back together to obtain the final output signal. We‘ll also add some makeup gain to compensate for the level reduction introduced by the compressor.

6. Implementing additional features: To make our plugin more versatile, we‘ll add some extra features like a bypass switch, solo/mute controls for each band, and input/output gain knobs. These will allow users to fine-tune the compressor‘s behavior and audition individual frequency bands.

Here‘s a code snippet that demonstrates the basic structure of the compressor processor class:

class CompressorProcessor : public AudioProcessor
{
public:
    CompressorProcessor()
    {
        // Constructor code here
    }

    ~CompressorProcessor() {}

    // AudioProcessor overrides
    void prepareToPlay(double sampleRate, int samplesPerBlock) override;
    void releaseResources() override;
    void processBlock(AudioBuffer<float>& buffer, MidiBuffer& midiMessages) override;

    // Compressor parameters
    AudioParameterFloat* thresholdParam;
    AudioParameterFloat* ratioParam;
    AudioParameterFloat* attackParam;
    AudioParameterFloat* releaseParam;

private:
    // Compressor state variables
    float compGain;
    float compReduction;

    // Crossover filter coefficients
    IIRCoefficients lowCoefs, midCoefs, highCoefs;

    // Filter state variables
    IIRFilter lowFilter, midFilter, highFilter;
};

In the prepareToPlay() method, we‘ll initialize the filter coefficients based on the sample rate and desired crossover frequencies. The processBlock() method is where the actual audio processing happens – this is where we‘ll apply the filters, compute the compressor gain, and apply the gain to each band.

We‘ll also need to define the parameters (threshold, ratio, attack, release) as AudioParameterFloat objects so they can be automated and controlled via the host DAW.

Building the 3-Band Compressor: GUI

With the DSP code in place, the next step is to create a graphical interface for our plugin. The GUI allows users to interact with the plugin‘s parameters and visualize the audio signal as it‘s being processed.

For our 3-band compressor, we‘ll create a custom interface using JUCE‘s graphics and GUI classes. Here‘s a rough sketch of what the interface might look like:

+--------------------------------------------------+
|  3-Band Compressor                   [Bypass]   |
|                                                  |
|  +------------++------------++------------+      |
|  |   Low      ||    Mid     ||   High     |      |
|  | [Thresh]   || [Thresh]   || [Thresh]   |      |
|  | [Ratio]    || [Ratio]    || [Ratio]    |      |
|  | [Attack]   || [Attack]   || [Attack]   |      |
|  | [Release]  || [Release]  || [Release]  |      |
|  | [Solo]     || [Solo]     || [Solo]     |      |
|  | [Mute]     || [Mute]     || [Mute]     |      |
|  +------------++------------++------------+      |
|                                                  |
|  +------------------------------------------+    |
|  |                 Analyzer                |    |
|  +------------------------------------------+    |
|                                                  |
|  [Input]                              [Output]   |
+--------------------------------------------------+

The interface is divided into several sections:

1. Global controls: At the top, we have a bypass button that allows users to enable or disable the entire plugin. This is useful for comparing the compressed signal with the original unprocessed signal.

2. Band controls: The main section of the interface is split into three columns, one for each frequency band (low, mid, high). Each band has its own set of controls for adjusting the compressor parameters (threshold, ratio, attack, release) as well as solo and mute buttons for auditioning individual bands.

3. Spectrum analyzer: Below the band controls, we have a spectrum analyzer that displays the frequency content of the input signal in real-time. This provides visual feedback to the user and helps them identify which frequency ranges are being affected by the compressor.

4. Input/output gain: At the bottom, we have knobs for adjusting the input and output gain of the plugin. These allow users to compensate for any level changes introduced by the compressor and ensure optimal signal levels.

To implement this interface in JUCE, we‘ll create a new class called CompressorEditor that inherits from the AudioProcessorEditor base class. This class will contain all the GUI components (sliders, buttons, labels, etc.) and handle user input events.

Here‘s a simplified version of the CompressorEditor class:

class CompressorEditor : public AudioProcessorEditor
{
public:
    CompressorEditor(CompressorProcessor& p)
        : AudioProcessorEditor(&p), processor(p)
    {
        // Add GUI components
        addAndMakeVisible(bypassButton);
        bypassButton.onClick = [this] { processor.setBypass(bypassButton.getToggleState()); };

        addAndMakeVisible(lowThresholdSlider);
        lowThresholdSlider.onValueChange = [this] { processor.setLowThreshold(lowThresholdSlider.getValue()); };

        // ... (add remaining sliders, buttons, etc.)

        addAndMakeVisible(analyzer);

        setSize(600, 400);
    }

    void paint(Graphics& g) override
    {
        g.fillAll(Colours::darkgrey);
    }

    void resized() override
    {
        auto bounds = getLocalBounds();

        bypassButton.setBounds(bounds.removeFromTop(30).withSizeKeepingCentre(100, 30));

        auto bandBounds = bounds.removeFromTop(200).withTrimmedTop(10);
        auto lowBounds = bandBounds.removeFromLeft(bandBounds.getWidth() / 3);
        auto midBounds = bandBounds.removeFromLeft(bandBounds.getWidth() / 2);
        auto highBounds = bandBounds;

        lowThresholdSlider.setBounds(lowBounds.removeFromTop(30));
        // ... (set bounds for remaining low-band components)

        midThresholdSlider.setBounds(midBounds.removeFromTop(30));
        // ... (set bounds for remaining mid-band components)

        highThresholdSlider.setBounds(highBounds.removeFromTop(30));
        // ... (set bounds for remaining high-band components)

        analyzer.setBounds(bounds.removeFromTop(100).reduced(10));
    }

private:
    CompressorProcessor& processor;

    ToggleButton bypassButton;

    Slider lowThresholdSlider, lowRatioSlider, lowAttackSlider, lowReleaseSlider;
    ToggleButton lowSoloButton, lowMuteButton;

    // ... (declare remaining sliders and buttons for mid and high bands)

    SpectrumAnalyzer analyzer;

    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR(CompressorEditor)
};

In the constructor, we create and add all the GUI components to the editor. We also attach lambda functions to the sliders and buttons to update the corresponding parameters in the processor when their values change.

The resized() method is where we set the positions and sizes of the GUI components within the editor‘s bounds. We use a combination of removeFromTop(), removeFromLeft(), and reduced() to divide the available space into smaller rectangles for each section of the interface.

Finally, we need to modify our CompressorProcessor class to create an instance of the CompressorEditor when the host requests it:

class CompressorProcessor : public AudioProcessor
{
    // ...

    AudioProcessorEditor* createEditor() override
    {
        return new CompressorEditor(*this);
    }

    // ...
};

With that, we have a fully functional 3-band compressor plugin with a custom graphical interface! Of course, there are many more details and improvements we could add (e.g., parameter smoothing, proper gain staging, responsive analyzer display), but this covers the essential components.

Conclusion and Next Steps

Building an audio plugin like this 3-band compressor is an excellent way to put your C++ skills to the test and gain practical experience with real-time DSP programming. By working through each stage of the development process – from designing the algorithm to implementing the user interface – you‘ll encounter a wide range of challenges that will help you grow as a programmer.

But this is just the beginning! There are endless possibilities for further customization and experimentation. Here are a few ideas to continue your audio programming journey:

• Add more parameters to the compressor (knee, lookahead, sidechain input)
• Implement different crossover filter types (Butterworth, Chebyshev, etc.)
• Extend the interface with metering and visual feedback (gain reduction, input/output levels)
• Optimize the DSP code for better performance and lower latency
• Port the plugin to other formats (VST3, AUv3) and platforms (Linux, iOS)
• Build a completely different type of processor (EQ, reverb, distortion)

The world of audio software development is vast and constantly evolving, so there‘s always something new to learn and explore. By combining your passion for music with your C++ programming skills, you can create powerful tools that enhance the creative process and inspire others to do the same.

So what are you waiting for? Fire up your IDE, put on your headphones, and start coding! The next game-changing audio plugin is just a few keystrokes away.