👁 Image
SoundFlow.Extensions.WebRtc.Apm 1.4.0

<div align="center"> <img src="https://raw.githubusercontent.com/LSXPrime/SoundFlow/refs/heads/master/logo.png" alt="Project Logo" width="256" height="256">

SoundFlow - Audio Processing Module Extension (WebRTC)

WebRTC Audio Processing Module (Google & PulseAudio) Integration for SoundFlow

👁 License: MIT
👁 NuGet
👁 SoundFlow Main Repository

</div>

Introduction

SoundFlow.Extensions.WebRtc.Apm is an official extension package for the SoundFlow (.NET) audio engine. It integrates a native library based on the high-quality WebRTC Audio Processing Module (APM), bringing advanced voice processing capabilities to your SoundFlow audio pipelines.

The WebRTC APM is widely used in real-time communication applications (like WebRTC itself) to enhance audio quality by addressing common issues such as echo, background noise, and varying audio levels.

Features

This extension provides the following WebRTC APM features, configurable and applicable within your SoundFlow projects:

• Acoustic Echo Cancellation (AEC): Effectively cancels echoes that occur when audio played through speakers is picked up by the microphone. Essential for speakerphone or conference call scenarios.
• Noise Suppression (NS): Reduces unwanted steady-state background noise (e.g., fan noise, hum) from the audio signal, improving clarity.
• Automatic Gain Control (AGC): Dynamically adjusts the microphone input volume to ensure a consistent and appropriate level, preventing audio that is too quiet or clipped.
• High Pass Filter (HPF): Removes low-frequency components below ~80 Hz, helping to eliminate DC offset, rumble, and plosives.
• Pre-Amplifier: Applies a configurable gain to the audio signal before other APM processing steps.

These features are primarily applied via a single, unified WebRtcApmModifier component designed for real-time audio graph processing. The package also includes a NoiseSuppressor component for convenient offline/batch processing of ISoundDataProvider sources.

Getting Started

Installation

This package requires SoundFlow (the core library). You can install it via NuGet:

NuGet Package Manager:

Install-Package SoundFlow.Extensions.WebRtc.Apm

.NET CLI:

dotnet add package SoundFlow.Extensions.WebRtc.Apm

Usage

The primary way to use the WebRTC APM features in a real-time SoundFlow graph is by adding the WebRtcApmModifier to a SoundComponent, typically one that represents your audio input source (e.g., a component connected to a MicrophoneDataProvider).

using SoundFlow.Abstracts;
using SoundFlow.Backends.MiniAudio;
using SoundFlow.Components;
using SoundFlow.Providers;
using SoundFlow.Enums;
using SoundFlow.Extensions.WebRtc.Apm.Modifiers;

// Initialize the audio engine with mixed capability
var audioEngine = new MiniAudioEngine(48000, Capability.Mixed, channels: 1);


// --- Real-time Processing ---
// For real-time audio processing and voice communication applications:, add the WebRtcApmModifier to a SoundComponent.

// Create the microphone data provider and sound player
var microphoneDataProvider = new MicrophoneDataProvider();
var soundPlayer = new SoundPlayer(microphoneDataProvider);

// Create and configure the WebRTC APM (Audio Processing Module) modifier
var apmModifier = new WebRtcApmModifier(
 // Echo Cancellation (AEC) settings
 aecEnabled: true, // Enable echo cancellation to remove feedback from speakers
 aecMobileMode: false, // Use desktop-optimized AEC (false) vs mobile-optimized (true)
 aecLatencyMs: 40, // Expected system latency in milliseconds for echo cancellation

 // Noise Suppression settings
 nsEnabled: true, // Enable background noise reduction
 nsLevel: NoiseSuppressionLevel.High, // Aggressiveness of noise suppression

 // Automatic Gain Control (AGC) settings - Version 1
 agc1Enabled: true, // Enable legacy AGC (Automatic Gain Control)
 agcMode: GainControlMode.AdaptiveDigital, // AGC operating mode
 agcTargetLevel: -3, // Target output level in dBFS (-31 to 0) for Adaptive mode
 agcCompressionGain: 9, // Fixed gain in dB (0-90) for FixedDigital mode
 agcLimiter: true, // Enable output limiter to prevent clipping

 // AGC Version 2 settings (newer algorithm)
 agc2Enabled: false, // Disable newer AGC implementation

 // Filter settings
 hpfEnabled: true, // Enable high-pass filter to remove low-frequency noise
 preAmpEnabled: false, // Disable pre-amplifier stage
 preAmpGain: 1.0f, // Default gain (1.0 = no amplification)

 // Multi-channel audio processing settings
 useMultichannelCapture: false, // Process capture audio as mono/stereo (false)
 useMultichannelRender: false, // Process render audio as mono/stereo (false)
 downmixMethod: DownmixMethod.AverageChannels // How to mix channels when downmixing
);

// Add the modifier to the sound player
soundPlayer.AddModifier(apmModifier);

// Add sound player to the master mixer
Mixer.Master.AddComponent(soundPlayer);

// Start capturing and playing audio
microphoneDataProvider.StartCapture();
soundPlayer.Play();

Console.WriteLine("Audio processing with WebRTC APM is running.");
Console.WriteLine("Press keys to modify settings:");
Console.WriteLine(" [A] Toggle AEC");
Console.WriteLine(" [M] Toggle AEC Mobile Mode");
Console.WriteLine(" [N] Toggle Noise Suppression");
Console.WriteLine(" [L] Cycle Noise Suppression Level");
Console.WriteLine(" [1] Toggle AGC1");
Console.WriteLine(" [2] Toggle AGC2");
Console.WriteLine(" [H] Toggle High Pass Filter");
Console.WriteLine(" [P] Toggle Pre-Amplifier");
Console.WriteLine(" [+] Increase Pre-Amp Gain");
Console.WriteLine(" [-] Decrease Pre-Amp Gain");
Console.WriteLine(" [ESC] Exit");

bool running = true;
while (running)
{
 var key = Console.ReadKey(true).Key;

 switch (key)
 {
 case ConsoleKey.A:
 apmModifier.EchoCancellation.Enabled = !apmModifier.EchoCancellation.Enabled;
 Console.WriteLine($"AEC Enabled: {apmModifier.EchoCancellation.Enabled}");
 break;

 case ConsoleKey.M:
 apmModifier.EchoCancellation.MobileMode = !apmModifier.EchoCancellation.MobileMode;
 Console.WriteLine($"AEC Mobile Mode: {apmModifier.EchoCancellation.MobileMode}");
 break;

 case ConsoleKey.N:
 apmModifier.NoiseSuppression.Enabled = !apmModifier.NoiseSuppression.Enabled;
 Console.WriteLine($"Noise Suppression Enabled: {apmModifier.NoiseSuppression.Enabled}");
 break;

 case ConsoleKey.L:
 var currentLevel = (int)apmModifier.NoiseSuppression.Level;
 var nextLevel = (NoiseSuppressionLevel)((currentLevel + 1) % Enum.GetValues(typeof(NoiseSuppressionLevel)).Length);
 apmModifier.NoiseSuppression.Level = nextLevel;
 Console.WriteLine($"Noise Suppression Level: {apmModifier.NoiseSuppression.Level}");
 break;

 case ConsoleKey.D1:
 apmModifier.AutomaticGainControl.Agc1Enabled = !apmModifier.AutomaticGainControl.Agc1Enabled;
 Console.WriteLine($"AGC1 Enabled: {apmModifier.AutomaticGainControl.Agc1Enabled}");
 break;

 case ConsoleKey.D2:
 apmModifier.AutomaticGainControl.Agc2Enabled = !apmModifier.AutomaticGainControl.Agc2Enabled;
 Console.WriteLine($"AGC2 Enabled: {apmModifier.AutomaticGainControl.Agc2Enabled}");
 break;

 case ConsoleKey.H:
 apmModifier.HighPassFilterEnabled = !apmModifier.HighPassFilterEnabled;
 Console.WriteLine($"High Pass Filter Enabled: {apmModifier.HighPassFilterEnabled}");
 break;

 case ConsoleKey.P:
 apmModifier.PreAmplifierEnabled = !apmModifier.PreAmplifierEnabled;
 Console.WriteLine($"Pre-Amplifier Enabled: {apmModifier.PreAmplifierEnabled}");
 break;

 case ConsoleKey.Add:
 case ConsoleKey.OemPlus:
 apmModifier.PreAmplifierGainFactor += 0.5f;
 Console.WriteLine($"Pre-Amp Gain: {apmModifier.PreAmplifierGainFactor}");
 break;

 case ConsoleKey.Subtract:
 case ConsoleKey.OemMinus:
 apmModifier.PreAmplifierGainFactor = Math.Max(0.5f, apmModifier.PreAmplifierGainFactor - 0.5f);
 Console.WriteLine($"Pre-Amp Gain: {apmModifier.PreAmplifierGainFactor}");
 break;

 case ConsoleKey.Escape:
 running = false;
 break;
 }
}

// Clean up
microphoneDataProvider.StopCapture();
soundPlayer.Stop();
Mixer.Master.RemoveComponent(soundPlayer);
microphoneDataProvider.Dispose();
audioEngine.Dispose();

Console.WriteLine("Audio processing stopped.");

// --- Offline Processing ---
// For batch processing of an entire audio source without the real-time graph:

using SoundFlow.Extensions.WebRtc.Apm.Components; // For NoiseSuppressor
ISoundDataProvider sourceForOffline = ...; // e.g., new AssetDataProvider(...) or ChunkedDataProvider(...)

// Ensure source data matches the sampleRate and channels used here!
try
{
 using var offlineSuppressor = new NoiseSuppressor(sourceForOffline, 48000, 1, NoiseSuppressionLevel.VeryHigh); 
 var encoder = AudioEngine.Instance.CreateEncoder(CleanedFilePath, "wav", SampleFormat.F32, 1, 48000);

 // Option 1: Process all into a single array (for smaller files) - See Samples\SoundFlow.Samples.NoiseSuppression
 // float[] cleanedAudio = offlineSuppressor.ProcessAll(); 

 // Option 2: Process chunk-by-chunk (for large files)
 offlineSuppressor.OnAudioChunkProcessed += (chunk) =>
 {
 if (encoder.IsDisposed == false)
 encoder.Encode(chunk.ToArray()); // Handle the processed 'chunk' (ReadOnlyMemory<float>), encode to file
 };
 offlineSuppressor.ProcessChunks(); // This method blocks until done
 encoder.Dispose(); // Dispose encoder to save the cleaned audio
}
catch (Exception ex)
{
 Console.Error.WriteLine($"Offline processing failed: {ex.Message}");
}

Sample

A sample application demonstrating the usage of the NoiseSuppressor component for offline audio file processing can be found here:

SoundFlow.Samples.NoiseSuppression (Note: This link assumes the sample is in the main SoundFlow repository)

Origin and Licensing

This SoundFlow.Extensions.WebRtc.Apm package consists of C# code (wrapper, integration logic, modifiers, components) and relies on a separate native binary derived from the WebRTC Audio Processing Module.

• The C# code within this SoundFlow.Extensions.WebRtc.Apm package is licensed under the MIT License.
• The native webrtc_audio_processing library wrapped by this package is based on the efforts of the PulseAudio project to extract the APM from the full WebRTC source tree into a standalone, buildable library.
• The underlying WebRTC Audio Processing Module code is typically licensed under the BSD 3-Clause "New" or "Revised" License.

Users of this package must comply with the terms of BOTH the MIT License (for the C# wrapper) and the BSD 3-Clause License (for the native library). The BSD license generally requires including the copyright notice and license text of the WebRTC code. Please consult the native library's specific distribution for the exact licensing requirements.

Contributing

Contributions to the SoundFlow.Extensions.WebRtc.Apm package are welcome! Please open issues or submit pull requests to this repository following the general SoundFlow Contributing Guidelines.

Acknowledgments

We gratefully acknowledge the work of the original WebRTC team for developing the powerful Audio Processing Module, and the PulseAudio project for their valuable effort in extracting and maintaining a standalone version of the APM library, which makes this integration possible.

License

The C# code in SoundFlow.Extensions.WebRtc.Apm is licensed under the .