book

Designing Audio Effect Plugins in C++, 2nd Edition

Name: Designing Audio Effect Plugins in C++, 2nd Edition
Author: Will Pirkle
ISBN: 9780429954313

by Will Pirkle

May 2019

Intermediate to advanced

704 pages

21h 21m

English

Routledge

Read now

Unlock full access

Cover
Half Title
Title
Copyright
Dedication
Contents
List of Figures
List of Tables
Preface
Chapter 1: Introduction
1.1 Using This Book1.2 Fundamentals of Audio Signal Processing1.2.1 Acquisition of Audio Samples1.3 Reconstruction of the Analog Signal1.4 Numerical Representation of Audio Data1.5 Analytical DSP Test Signals1.5.1 DC and Step (0 Hz)1.5.2 Nyquist1.5.3 ½ Nyquist1.5.4 ¼ Nyquist1.5.5 Impulse1.6 Signal Processing Algorithms1.6.1 Bookkeeping1.6.2 The One-Sample Delay1.6.3 Multiplication With a Scalar Value1.6.4 Addition and Subtraction1.6.5 Some Algorithm Examples and Difference Equations1.6.5.1 Gain, Attenuation and Phase Inversion1.7 1st Order Feed-Forward and Feedback Algorithms1.8 Bibliography

Chapter 2: Anatomy of an Audio Plugin
2.1 Plugin Packaging: Dynamic-Link Libraries (DLLs)2.2 The Plugin Description: Simple Strings2.2.1 The Plugin Description: Features and Options2.3 Initialization: Defining the Plugin Parameter Interface2.3.1 Initialization: Defining Channel I/O Support2.3.2 Initialization: Sample Rate Dependency2.4 Processing: Preparing for Audio Streaming2.4.1 Processing: Audio Signal Processing (DSP)2.5 Mixing Parameter Changes With Audio Processing2.5.1 Plugin Variables and Plugin Parameters2.5.2 Parameter Smoothing2.5.3 Pre and Post-Processing Updates2.5.4 VST3 Sample Accurate Updates2.5.5 Multi-Threaded Software2.6 Monolithic Plugin Objects2.7 Bibliography
Chapter 3: VST3 Programming Guide
3.1 Setting Up the VST3 SDK3.1.1 VST3 Sample Projects3.1.2 VST3 Documentation3.2 VST3 Architecture and Anatomy3.2.1 Single vs. Dual Component Architectures3.2.2 VST3 Base Classes3.2.3 MacOS Bundle ID3.2.4 VST3 Programming Notes3.2.5 VST3 and the GUID3.2.6 VST3 Plugin Class Factory3.3 Description: Plugin Description Strings3.4 Description: Plugin Options/Features3.4.1 Side Chain Input3.4.2 Latency3.4.3 Tail Time3.4.4 Custom GUI3.4.5 Factory Presets and State Save/Load3.4.6 VST3 Support for 64-bit Audio3.5 Initialization: Defining Plugin Parameters3.5.1 Thread-Safe Parameter Access3.5.2 Initialization: Defining Plugin Channel I/O Support3.5.3 Initialization: Channel Counts and Sample Rate Information3.6 The Buffer Process Cycle3.6.1 Processing: Updating Plugin Parameters From GUI Controls3.6.2 Processing: Resetting the Algorithm and Preparing for Streaming3.6.3 Processing: Accessing the Audio Buffers3.6.4 Processing: Writing Output Parameters3.6.5 Processing: VST3 Soft Bypass3.7 Destruction/Termination3.8 Retrieving VST3 Host Information3.9 Validating Your Plugin3.10 Using ASPiK to Create VST3 Plugins3.11 Bibliography
Chapter 4: AU Programming Guide
4.1 Setting Up the AU SDK4.1.1 AU Sample Projects4.1.2 AU Documentation4.2 AU Architecture and Anatomy4.2.1 AU Base Classes4.2.2 MacOS Bundle ID4.2.3 AU Programming Notes4.3 Description: Plugin Description Strings4.4 Description: Plugin Options/Features4.4.1 Side Chain Input4.4.2 Latency4.4.3 Tail Time4.4.4 Custom GUI4.4.5 Factory Presets and State Save/Load4.5 Initialization: Defining Plugin Parameters4.5.1 Thread-Safe Parameter Access4.5.2 Initialization: Defining Plugin Channel I/O Support4.5.3 Initialization: Channel Counts and Sample Rate Information4.6 The Buffer Process Cycle4.6.1 Processing: Updating Plugin Parameters From GUI Controls4.6.2 Processing: Resetting the Algorithm and Preparing for Streaming4.6.3 Processing: Accessing the Audio Buffers4.6.4 Processing: Writing Output Parameters4.7 The AU/GUI Connection4.7.1 Cocoa’s Flat Namespace4.7.2 The AU Event Listener System4.7.2.1 The Event Lister Dispatch Callback Function4.7.2.2 Creating the AU Event Listener4.7.2.3 Destroying the Event Listener4.8 Destruction/Termination4.9 Retrieving AU Host Information4.10 Validating Your Plugin4.11 Using ASPiK to Create AU Plugins4.12 Bibliography
Chapter 5: AAX Native Programming Guide
5.1 Setting Up the AAX SDK5.1.1 AAX Sample Projects5.1.2 AAX Documentation5.2 AAX Architecture and Anatomy5.2.1 AAX Model-Algorithm Synchronization5.2.2 AAX Base Classes5.2.3 MacOS Bundle ID5.2.4 AAX Programming Notes5.2.5 AAX Class Factory5.2.6 AAX Effect Categories5.2.7 AAX Algorithms: Channel-Processing Functions5.2.8 AAX Algorithm Data5.2.9 Algorithm Data Contents5.3 Description: Plugin Description Strings5.3.1 Description: Defining AAX Algorithms5.4 Description: Plugin Options/Features5.4.1 Side Chain Input5.4.2 Latency5.4.3 Tail Time5.4.4 Custom GUI5.4.5 Factory Presets and State Save/Load5.4.6 AAX Notification System5.4.7 AAX Custom Data5.4.8 AAX EQ and Dynamics Curves5.4.9 AAX Gain Reduction Meter5.5 Initialization: Defining Plugin Parameters5.5.1 Thread-Safe Parameter Access5.5.2 Initialization: Defining Plugin Channel I/O Support5.5.3 Initialization: Channel Counts and Sample Rate Information5.6 The Buffer Process Cycle5.6.1 Processing: Updating Plugin Parameters From GUI Controls5.6.2 Processing: Resetting the Algorithm and Preparing for Streaming5.6.3 Processing: Accessing the Audio Buffers5.6.4 Processing: Writing Output Parameters5.6.5 Processing: AAX Soft Bypass5.7 Destruction/Termination5.8 Retrieving AAX Host Information5.9 Validating Your Plugin5.10 Using ASPiK to Create AAX Plugins5.11 Bibliography
Chapter 6: ASPiK Programming Guide
6.1 Plugin Kernel Portability and Native Plugin Shells6.2 Organizing the SDKs: AAX, AU, and VST6.2.1 Your C++ Compiler6.2.2 Setting Up the AAX SDK6.2.3 Setting Up the AU SDK6.2.4 Setting Up the VST SDK6.2.5 Creating the Universal SDK Folder Hierarchy6.2.6 Adding the VSTGUI4 Library6.2.7 CMake6.3 Creating a Plugin Project With ASPiKreator: IIRFilters6.3.1 ASPiK Project Folders6.3.2 Running CMake6.4 Adding Effect Objects to the PluginCore6.4.1 The PluginCore Constructor6.4.2 IIRFilters: GUI Parameter Lists6.4.3 Parameter Smoothing6.4.4 Handling the String-List Parameters6.4.5 IIRFilters: Declaring Plugin Variables6.4.6 Parameter Object Enumerations for Attributes6.4.6.1 Continuous Floating Point Parameters and Discrete Integer  Parameters6.4.6.2 String-List Parameters6.4.7 IIRFilters: Object Declarations and Reset6.4.8 IIRFilters: GUI Parameter Updates6.4.9 IIRFilters: Processing Audio Data6.4.10 Buffer Pre-Processing6.4.11 Buffer Post-Processing6.4.12 Buffer vs. Frame Processing6.4.13 processAudioFrame: Information About the Frame6.4.14 processAudioFrame: Input and Output Samples6.5 Defining Factory Presets6.6 Basic Plugin GUI Design With ASPiK’s PluginGUI6.7 GUI Design With VSTGUI46.7.1 Modifier Keys6.7.2 Zooming (Scaling the GUI)6.7.3 Reserved Control Tags6.7.4 VSTGUI4 Objects6.7.5 Creating a GUI With VSTGUI6.7.6 Important GUI Designer Terms6.8 VSTGUI C++ Objects6.8.1 Basic GUI Design6.8.2 The GUI Designer Workspace6.8.3 Changing Your GUI Canvas Size6.8.4 Setting Up the Control Tags6.8.5 Importing the Graphics Files6.8.6 Assembling the GUI6.8.7 Setting the Background6.8.8 Adding the GUI Elements6.8.9 Saving and Re-Building6.8.10 Scaling the GUI6.8.11 More ASPiK Features6.9 Bibliography
Chapter 7: Using RackAFX to Create ASPiK Projects
7.1 Installing RackAFX7.2 Getting Started With RackAFX7.3 Setting Up Your Project Preferences and Audio Hardware7.4 Installing VSTGUI47.5 Creating a Project and Adding GUI Controls7.5.1 Numerical Continuous Controls7.5.2 String-List Controls7.5.3 Meters7.6 Anatomy of Your RackAFX Project7.7 Testing Audio Algorithms With RackAFX7.8 RackAFX Impulse Convolver and FIR Design Tools7.9 Designing Your Custom GUI7.10 Exporting Your ASPiK Project7.11 Bibliography
Chapter 8: C++ Conventions and How to Use This Book
8.1 Three Types of C++ Objects8.1.1 Effect Objects Become Framework Object Members8.1.2 All Effect Objects and Most DSP Objects Implement Common Interfaces8.1.3 DSP and Effect Objects Use Custom Data Structures for Parameter  Get/Set Operations8.1.4 Effect Objects Accept Native Data From GUIs8.1.5 Effect Objects Process Audio Samples8.1.5.1 Series Objects8.1.5.2 Parallel Objects8.1.6 Effect Objects Optionally Process Frames8.2 Book Projects8.2.1 ASPiK Users8.2.2 JUCE and Other Non-ASPiK Users8.2.3 A Sample Plugin Project: GUI Control Definition
Chapter 9: How DSP Filters Work (Without Complex Math)
9.1 Frequency and Phase Response Plots9.2 Frequency and Phase Adjustments From Filtering9.3 1st Order Feed-Forward Filter9.4 1st Order Feedback Filter9.5 Final Observations9.6 Homework9.7 Bibliography
Chapter 10: Basic DSP Theory
10.1 The Complex Sinusoid10.2 Complex Math Review10.3 Time Delay as a Math Operator10.4 The Sampled Sinusoid10.5 1st Order Feed-Forward Filter Revisited10.5.1 Negative Frequencies10.6 Evaluating the Transfer Function H(ω)10.6.1 DC (0 Hz)10.6.2 Nyquist (π)10.6.3 ½ Nyquist (π/2)10.6.4 ¼ Nyquist (π/4)10.7 Evaluating e jω10.8 The z Substitution10.9 The z Transform10.10 The z Transform of Signals10.11 The z Transform of Difference Equations10.12 The z Transform of an Impulse Response10.13 The Zeros of the Transfer Function10.14 Estimating the Frequency Response: Zeros10.15 Filter Gain Control10.16 1st Order Feedback Filter Revisited10.16.1 Step 1: Take the z Transform of the Difference Equation10.16.2 Step 2: Fashion the Difference Equation into a Transfer Function10.16.3 Step 3: Factor out a0 as the Scalar Gain Coefficient10.17 The Poles of the Transfer Function10.17.1 Step 4: Estimate the Frequency Response10.17.2 Step 5: Perform Direct Evaluation of the Frequency Response10.17.3 DC (0 Hz)10.17.4 Nyquist (π)10.17.5 ½ Nyquist (π/2)10.17.6 ¼ Nyquist (π/4)10.18 2nd Order Feed-Forward Filter10.18.1 Step 1 and Step 2: Take the z Transform of the Difference Equation  and Fashion it into a Transfer Function10.18.2 Step 3: Factor Out a0 as the Scalar Gain Coefficient10.18.3 Step 4: Estimate the Frequency Response10.18.4 Step 5: Direct Evaluation10.18.5 DC (0 Hz)10.18.6 Nyquist (π)10.18.7 ½ Nyquist (π/2)10.18.8 ¼ Nyquist (π/4)10.18.9 z Transform of Impulse Response10.19 2nd Order Feedback Filter10.19.1 Step 1, Step 2, and Step 3: Take the z Transform of the Difference  Equation to Get the Transfer Function, Then Factor Out a0 as the Scalar Gain Coefficient10.19.2 Step 4: Estimate the Frequency Response10.19.3 Step 5: Direct Evaluation10.19.4 DC (0 Hz)10.19.5 Exercise10.20 1st Order Pole/Zero Filter: The Shelving Filter10.20.1 Step 1, Step 2, and Step 3: Take the z Transform of the Difference  Equation to Get the Transfer Function, Then Factor Out a0 as the Scalar Gain Coefficient10.20.2 Step 4: Estimate the Frequency Response10.20.3 Step 5: Direct Evaluation10.20.4 DC (0 Hz)10.20.4.1 Exercise10.21 The Biquadratic Filter10.21.1 Step 1, Step 2, and Step 3: Take the z Transform of the Difference  Equation to Get the Transfer Function, Then Factor Out a0 as the Scalar Gain Coefficient10.21.2 Step 4: Plot the Poles and Zeros of the Transfer Function10.21.3 The aN and bM Coefficient Naming Conventions10.22 Other Biquadratic Structures10.23 C++ DSP Object: Biquad10.23.1 Biquad: Enumerations and Data Structure10.23.2 Biquad: Members10.23.3 Biquad: Programming Notes10.24 Homework10.25 Bibliography
Chapter 11: Audio Filter Designs: IIR Filters
11.1 Direct z-Plane Design11.1.1 Simple Resonator11.1.2 Smith-Angell Resonator11.2 Analog Filter to Digital Filter Conversion11.2.1 Example11.3 Audio Biquad Filter Designs11.3.1 The Audio Biquad Filter Structure11.3.2 Classical Filters11.3.2.1 1st Order LPF and HPF11.3.2.2 2nd Order LPF and HPF11.3.2.3 2nd Order BPF and BSF11.3.2.4 2nd Order Butterworth LPF and HPF11.3.2.5 2nd Order Butterworth BPF and BSF11.3.2.6 2nd Order Linkwitz-Riley LPF and HPF11.3.2.7 1st and 2nd Order APF11.3.2.8 1st Order Shelving Filters11.3.2.9 Low Shelving11.3.2.10 High Shelf11.3.2.11 2nd Order Parametric EQ Filter: Non-Constant-Q11.3.2.12 2nd Order Parametric EQ Filter: Constant-Q11.4 Poles and Zeros at Infinity11.4.1 1st Order All Pole Filter11.4.2 2nd Order All Pole Filter: The MIDI Manufacturer’s Association LPF11.4.3 Vicanek’s Analog Matched Magnitude 2nd Order LPF11.4.4 Vicanek’s Analog Matched Magnitude 2nd Order BPF11.5 The Impulse Invariant Transform Method11.5.1 Impulse Invariant 1st Order LPF11.5.2 Impulse Invariant 2nd Order LPF11.6 C++ Effect Object: AudioFilter11.6.1 AudioFilter: Enumerations and Data Structure11.6.2 AudioFilter: Members11.6.3 AudioFilter: Programming Notes11.7 Chapter Plugin: IIRFilters11.7.1 IIRFilters: GUI Parameters11.7.2 IIRFilters: Object Declarations and Reset11.7.3 IIRFilters: GUI Parameter Update11.7.4 IIRFilters: Process Audio11.8 Homework11.9 Bibliography
Chapter 12: Audio Filter Designs: Wave Digital and Virtual Analog
12.1 Wave Digital Filters12.1.1 Scattering Parameters and WDFs12.1.2 Simulating WDF Components12.1.2.1 WDF Resistor12.1.2.2 WDF Inductor12.1.2.3 WDF Capacitor12.1.3 Simulating WDF Component Interconnections12.2 WDF Adaptors12.2.1 Series Adaptors12.2.2 Parallel Adaptors12.2.3 More Component Combinations12.2.4 Signal Flow Through a WDF Circuit12.2.5 Ladder Filter WDF Library Conventions12.2.6 Filter Source/Termination Impedance Matching12.2.7 Bilinear Transform Frequency Warping12.3 Designing Digital Ladder Filters With the WDF Library12.3.1 Components and Component Combinations12.3.2 Adaptors12.3.3 WDF Ladder Filter Design: 3rd Order Butterworth LPF12.3.4 WDF Ladder Filter Design: 3rd Order Bessel BSF12.3.5 WDF Ladder Filter Design: 6th Order Constant-K BPF12.3.6 WDF Ladder Filter Design: Ideal 2nd Order RLC Filters12.4 Zavalishin’s Virtual Analog Filters12.4.1 1st Order VA Filters12.4.2 2nd Order State Variable VA Filter12.5 C++ DSP Object: ZVAFilter12.5.1 ZVAFilter: Enumerations and Data Structure12.5.2 ZVAFilter: Members12.5.3 ZAFilter: Programming Notes12.6 C++ DSP Objects: WDF Ladder Filter Library12.6.1 WDFIdealRLCxxx: Enumerations and Data Structure12.6.2 WDFIdealRLCxxx: Members12.6.3 WDFIdealRLCxxx: Programming Notes12.7 Chapter Plugin: RLCFilters12.7.1 RLCFilters: GUI Parameters12.7.2 RLCFilters: Object Declarations and Reset12.7.3 RLCFilters: GUI Parameter Update12.7.4 RLCFilters: Process Audio12.8 Homework12.9 Bibliography
Chapter 13: Modulators: LFOs and Envelope Detectors
13.1 LFO Algorithms13.1.1 The IAudioSignalGenerator Interface13.1.2 C++ DSP Object: LFO13.1.3 LFO: Enumerations and Data Structure13.1.4 LFO: Members13.1.5 LFO: Programming Notes13.2 Envelope Detection13.2.1 C++ DSP Object: AudioDetector13.2.2 AudioDetector: Enumerations and Data Structure13.2.3 AudioDetector: Members13.2.4 AudioDetector: Programming Notes13.3 Modulating Plugin Parameters13.3.1 Modulation Range, Polarity, and Depth13.3.2 Modulation With the Envelope Detector13.4 C++ Effect Object: EnvelopeFollower13.4.1 EnvelopeFollower: Enumerations and Data Structure13.4.2 EnvelopeFollower: Members13.4.3 EnvelopeFollower: Programming Notes13.5 Chapter Plugin 1: ModFilter13.5.1 ModFilter: GUI Parameters13.5.2 ModFilter: Object Declarations and Reset13.5.3 ModFilter: GUI Parameter Update13.5.4 ModFilter: Process Audio13.6 The Phaser Effect13.6.1 C++ Effect Object: PhaseShifter13.6.2 PhaseShifter: Enumerations and Data Structure13.6.3 PhaseShifter: Members13.6.4 PhaseShifter: Programming Notes13.7 Chapter Plugin 2: Phaser13.7.1 Phaser: GUI Parameters13.7.2 Phaser: Object Declarations and Reset13.7.3 Phaser: GUI Parameter Update13.7.4 Phaser: Process Audio13.8 Homework13.9 Bibliography
Chapter 14: Delay Effects and Circular Buffers
14.1 The Basic Digital Delay14.2 Digital Delay With Wet/Dry Mix14.3 An Efficient Circular Buffer Object14.3.1 C++ DSP Object: CircularBuffer With Fractional Delay14.3.2 CircularBuffer: Enumerations and Data Structure14.3.3 CircularBuffer: Members14.3.4 CircularBuffer: Programming Notes14.4 Basic Delay Algorithms14.4.1 Stereo Delay With Feedback14.4.2 Stereo Ping-Pong Delay14.5 C++ Effect Object: AudioDelay14.5.1 AudioDelay: Enumerations and Data Structure14.5.2 AudioDelay: Members14.5.3 AudioDelay: Programming Notes14.6 Chapter Plugin: StereoDelay14.6.1 StereoDelay: GUI Parameters14.6.2 StereoDelay: Object Declarations and Reset14.6.3 StereoDelay: GUI Parameter Update14.6.4 StereoDelay: Process Audio14.6.5 Synchronizing the Delay Time to BPM14.6.5.1 ASPiK Users: BPM and Time Signature Information14.7 More Delay Algorithms14.7.1 Analog Modeling Delay14.7.2 Multi-Tap Delay14.7.3 LCR Delay14.7.4 TC Electronics TC-2290 Dynamic Delay14.8 Homework14.9 Bibliography
Chapter 15: Modulated Delay Effects
15.1 The Flanger/Vibrato Effect15.1.1 Stereo Flanger15.2 The Chorus Effect15.2.1 Stereo Chorus15.3 C++ Effect Object: ModulatedDelay15.3.1 ModulatedDelay: Enumerations and Data Structure15.3.2 ModulatedDelay: Members15.3.3 ModulatedDelay: Programming Notes15.4 Chapter Plugin: ModDelay15.4.1 ModDelay: GUI Parameters15.4.2 ModDelay: Object Declarations and Reset15.4.3 ModDelay: GUI Parameter Update15.4.4 ModDelay: Process Audio15.5 More Modulated Delay Algorithms15.5.1 Korg Stereo Cross-Feedback Flanger/Chorus15.5.2 Sony DPS-M7 Multi-Flanger15.5.3 Bass Chorus15.5.4 Dimension-Style Chorus (Roland Dimension D)15.5.5 Sony DPS-M7 Deca Chorus15.6 Homework15.7 Bibliography
Chapter 16: Audio Filter Designs: FIR Filters
16.1 The Impulse Response Revisited: Convolution16.2 FIR Filter Structures16.3 Generating Impulse Responses16.3.1 Impulse Responses of Acoustic Environments16.3.2 Impulse Responses of Speakers and Cabinets16.3.3 Impulse Responses by Frequency Sampling16.3.4 Sampling Arbitrary Frequency Responses16.3.4.1 Linear Phase FIR Using the Frequency Sampling Method16.3.5 Sampling Analog Filter Frequency Responses16.3.6 Sampling Ideal Filter Frequency Responses16.4 The Optimal/Parks-McClellan Method16.5 Other FIR Design Methods16.6 C++ DSP Function: FreqSample16.7 C++ DSP Function: CalculateAnalogMagArray16.7.1 calculateAnalogMagArray: Enumerations and Data Structure16.7.2 calculateAnalogMagArray: Calculations16.8 C++ DSP Object: LinearBuffer16.8.1 LinearBuffer: Enumerations and Data Structure16.8.2 LinearBuffer: Members16.8.3 LinearBuffer: Programming Notes16.9 C++ DSP Object: ImpulseConvolver16.9.1 ImpulseConvolver: Enumerations and Data Structure16.9.2 ImpulseConvolver: Members16.9.3 ImpulseConvolver: Programming Notes16.10 C++ Effect Object: AnalogFIRFilter16.10.1 AnalogFIRFilter: Enumerations and Data Structure16.10.2 AnalogFIRFilter: Members16.10.3 AnalogFIRFilter: Programming Notes16.11 Chapter Plugin: AnalogFIR16.11.1 AnalogFIR: GUI Parameters16.11.2 AnalogFIR: Object Declarations and Reset16.11.3 AnalogFIR: GUI Parameter Update16.11.4 AnalogFIR: Process Audio16.12 Homework16.13 Bibliography
Chapter 17: Reverb Effects
17.1 Anatomy of a Room Impulse Response17.1.1 RT6017.2 Echoes and Modes17.3 The Comb Filter Reverberator17.4 The Delaying All-Pass Reverberator17.4.1 Alternate and Nested Delaying APF Structures17.5 Schroeder’s Reverberator17.6 The LPF-Comb Reverberator17.7 The Absorbent-APF Reverberator17.8 The Modulated Delay APF17.9 Moorer’s Reverberator17.10 Dattorro’s Plate Reverb17.11 The Spin Semiconductor® Reverb Tank17.12 Generalized Feedback Delay Network Reverbs17.12.1 Searching for FDN Coefficients17.13 C++ DSP Objects: Reverb Objects17.13.1 C++ DSP Object: SimpleDelay17.13.2 SimpleDelay: Custom Data Structure17.13.3 SimpleDelay: Members17.13.4 SimpleDelay: Programming Notes17.13.5 C++ DSP Object: SimpleLPF17.13.6 SimpleLPF: Custom Data Structure17.13.7 SimpleLPF: Members17.13.8 SimpleLPF: Programming Notes17.13.9 C++ DSP Object: CombFilter17.13.10 CombFilter: Custom Data Structure17.13.11 CombFilter: Members17.13.12 CombFilter: Programming Notes17.13.13 C++ DSP Object: DelayAPF17.13.14 DelayAPF: Custom Data Structure17.13.15 DelayAPF: Members17.13.16 DelayAPFParameters: Programming Notes17.13.17 C++ DSP Object: NestedDelayAPF17.13.18 NestedDelayAPF: Custom Data Structure17.13.19 NestedDelayAPF: Members17.13.20 NestedDelayAPF: Programming Notes17.13.21 C++ DSP Object: TwoBandShelvingFilter17.13.22 TwoBandShelvingFilter: Custom Data Structure17.13.23 TwoBandShelvingFilter: Members17.13.24 TwoBandShelvingFilter: Programming Notes17.14 C++ Effect Object: ReverbTank17.14.1 ReverbTank: Enumerations and Data Structure17.14.2 ReverbTank: Members17.14.3 ReverbTank: Programming Notes17.15 Chapter Plugin: Reverb17.15.1 Reverb: GUI Parameters17.15.2 Reverb: Object Declarations and Reset17.15.3 Reverb: GUI Parameter Update17.15.4 Reverb: Process Audio17.16 Homework17.17 Bibliography
Chapter 18: Dynamics Processing
18.1 Compressor Output Calculation18.1.1 Hard-Knee Compressor and Limiter18.1.2 Soft-Knee Compressor and Limiter18.2 Downward Expander Output Calculation18.2.1 Hard-Knee Expander and Gate18.2.2 Soft-Knee Expander18.3 Final Gain Calculation18.4 Stereo-Linked Dynamics Processor18.5 Spectral Dynamics Processing18.6 Parallel Dynamics Processing18.7 Look-Ahead Processing18.8 External Keying18.8.1 ASPiK Users: Side Chain Code18.9 Gain Reduction Metering18.10 Alternate Side Chain Configurations18.11 C++ DSP Object: LRFilterBank18.11.1 LRFilterBank: Enumerations and Data Structure18.11.2 LRFilterBank: Members18.11.3 LRFilterBank: Programming Notes18.12 C++ Effect Object: DynamicsProcessor18.12.1 DynamicsProcessor: Enumerations and Data Structure18.12.2 DynamicsProcessor: Members18.12.3 DynamicsProcessor: Programming Notes18.13 Chapter Plugin: Dynamics18.13.1 Dynamics: GUI Parameters18.13.2 Dynamics: Object Declarations and Reset18.13.3 Dynamics: GUI Parameter Update18.13.4 Dynamics: Process Audio and External Keying18.13.5 Stereo Linking the DynamicsProcessor Objects18.13.6 ASPiK Users: Enabling the Special Pro Tools Gain Reduction Meter18.14 Homework18.15 Bibliography
Chapter 19: Nonlinear Processing: Distortion, Tube Simulation, and HF Exciters
19.1 Frequency Domain Effects of Nonlinear Processing19.2 Vacuum Tubes19.3 Solid State Distortion19.4 Bit Crushers19.5 High Frequency Exciters19.6 Virtual Bass19.7 Ring Modulation19.8 Nonlinear Processing Functions19.8.1 Asymmetrical Waveshaping19.9 C++ DSP Object: BitCrusher19.9.1 BitCrusher: Enumerations and Data Structure19.9.2 BitCrusher: Members19.9.3 BitCrusher: Programming Notes19.10 C++ DSP Object: DFOscillator19.10.1 DFOscillator: Enumerations and Data Structure19.10.2 DFOscillator: Members19.10.3 DFOscillator: Programming Notes19.11 C++ DSP Functions: Waveshapers19.12 C++ DSP Object: TriodeClassA19.12.1 TriodeClassA: Enumerations and Data Structure19.12.2 TriodeClassA: Members19.12.3 TriodeClassA: Programming Notes19.13 C++ Effect Object: ClassATubePre19.13.1 ClassATubePre: Enumerations and Data Structure19.13.2 ClassATubePre: Members19.13.3 ClassATubePre: Programming Notes19.14 Chapter Plugin: TubePreamp19.14.1 TubePreamp: GUI Parameters19.14.2 TubePreamp: Object Declarations and Reset19.14.3 TubePreamp: GUI Parameter Update19.14.4 TubePreamp: Process Audio19.15 Bonus Plugin Projects19.16 Homework19.17 Bibliography
Chapter 20: FFT Processing: The Phase Vocoder
20.1 The Fourier Series20.2 Understanding How the Fourier Kernel Works20.2.1 Windowing DFT Input Data20.3 The Complete DFT20.4 The FFT20.4.1 Overlap/Add Processing20.4.2 Window Gain Correction20.4.3 FFT and IFFT Magnitude and Phase20.4.4 Using Phase Information20.4.5 Phase Deviation20.4.6 Phase Vocoder Coding20.5 Some Phase Vocoder Effects20.5.1 Robot and Simple Noise Reduction20.5.2 Time Stretching/Shrinking20.5.3 Pitch Shifting20.5.4 Phase Locking20.6 Fast Convolution20.7 Gardner’s Fast Convolution20.8 Chapter Objects and Plugins20.9 C++ DSP Object: FastFFT20.9.1 FastFFT: Enumerations and Data Structure20.9.2 FastFFT: Members20.9.3 FastFFT: Programming Notes20.10 C++ DSP Object: PhaseVocoder20.10.1 PhaseVocoder: Enumerations and Data Structure20.10.2 PhaseVocoder: Members20.10.3 PhaseVocoder: Programming Notes20.10.3.1 PhaseVocoder Example: Robotization20.10.3.2 PhaseVocoder Example: NoiseReduction20.11 C++ DSP Object: FastConvolver20.11.1 FastConvolver: Members20.11.2 FastConvolver: Programming Notes20.12 C++ Effect Object: PSMVocoder20.12.1 PSMVocoder: Enumerations and Data Structure20.12.2 PSMVocoder: Members20.12.3 PSMVocoder: Programming Notes20.13 Chapter Plugin: PitchShifter20.13.1 PitchShifter: GUI Parameters20.13.2 PitchShifter: Object Declarations and Reset20.13.3 PitchShifter: GUI Parameter Update20.13.4 PitchShifter: Process Audio20.14 Third Party C++ DSP Object: TwoStageFFTConvolver20.15 Homework20.16 Bibliography
Chapter 21: Displaying Custom Waveforms and FFTs
21.1 Custom Views for Plotting Data21.1.1 ASPiK: The GUI Lifecycle and Messaging21.1.2 Multi-threading: The Lock-Free Ring Buffer21.1.3 A Waveform Histogram Viewer21.1.4 An Audio Spectrum Analyzer View
Chapter 22: Sample Rate Conversion
22.1 Interpolation: Overview22.1.1 Interpolation: Operations22.1.2 Interpolation: Polyphase Decomposition22.2 Decimation: Overview22.2.1 Decimation: Operations22.2.2 Decimation: Polyphase Decomposition22.3 Polyphase Decomposition Math22.3.1 Type-1 Decomposition22.3.2 Type-2 Decomposition22.3.2.1 What Is the Meaning of “Polyphase?”22.4 C++ DSP Objects: Interpolator and Decimator22.4.1 C++ DSP Object: Interpolator22.4.2 Interpolator: Enumerations and Data Structure22.4.3 Interpolator: Members22.4.4 Interpolator: Programming Notes22.4.5 C++ DSP Object: Decimator22.4.6 Decimator: Enumerations and Data Structure22.4.7 Decimator: Members22.4.8 Decimator: Programming Notes22.5 Chapter Plugin: TubePreamp Revisited22.6 Homework22.7 Bibliography
Index

Content preview from Designing Audio Effect Plugins in C++, 2nd Edition

Become an O’Reilly member and get unlimited access to this title plus top books and audiobooks from O’Reilly and nearly 200 top publishers, thousands of courses curated by job role, 150+ live events each month,
and much more.

Read now

Unlock full access

More than 5,000 organizations count on O’Reilly

O’Reilly covers everything we've got, with content to help us build a world-class technology community, upgrade the capabilities and competencies of our teams, and improve overall team performance as well as their engagement.

Julian F.

Head of Cybersecurity

I wanted to learn C and C++, but it didn't click for me until I picked up an O'Reilly book. When I went on the O’Reilly platform, I was astonished to find all the books there, plus live events and sandboxes so you could play around with the technology.

Addison B.

Field Engineer

I’ve been on the O’Reilly platform for more than eight years. I use a couple of learning platforms, but I'm on O'Reilly more than anybody else. When you're there, you start learning. I'm never disappointed.

Amir M.

Data Platform Tech Lead

I'm always learning. So when I got on to O'Reilly, I was like a kid in a candy store. There are playlists. There are answers. There's on-demand training. It's worth its weight in gold, in terms of what it allows me to do.

Mark W.

Embedded Software Engineer

Publisher Resources

ISBN: 9780429954313

Cloud Computing

Data Engineering

Data Science

AI & ML

Programming Languages

Software Architecture

IT/Ops

Security

Design

Business

Soft Skills

Designing Audio Effect Plugins in C++, 2nd Edition

by Will Pirkle

Become an O’Reilly member and get unlimited access to this title plus top books and audiobooks from O’Reilly and nearly 200 top publishers, thousands of courses curated by job role, 150+ live events each month,
and much more.