Welcome to the first lesson of our course on handling large and long-form audio and video files in C#. In this course, you will learn how to process, transform, and manage media files using modern tools and libraries. Working with audio and video is a common requirement in many real-world applications, such as transcription services, podcast platforms, and video streaming sites. However, dealing with large files can be challenging due to their size and complexity.

In this lesson, you will be introduced to FFmpeg, a powerful tool for media processing, and learn how to use it in C# through the Xabe.FFmpeg library. You will also explore practical audio preprocessing steps—such as normalization and format conversion—through a hands-on code example. By the end of this lesson, you will be ready to start working with audio files in your own C# projects.

FFmpeg is a free, open-source software suite for handling video, audio, and other multimedia files and streams. It is widely used in the industry for tasks such as converting file formats, extracting audio from video, compressing files, and applying filters or effects. FFmpeg works from the command line and supports almost every audio and video format you can think of.

The flexibility and performance of FFmpeg make it essential for developers dealing with podcasts, speech recognition, or any scenario where media quality and compatibility matter. For example, you might use FFmpeg to convert uploaded audio files to a standard format, normalize their volume for transcription, or split recordings into smaller, manageable segments for analysis.

While FFmpeg is powerful, its command-line interface can be intimidating for beginners or cumbersome in automated workflows. This is where Xabe.FFmpeg comes in. Xabe.FFmpeg is a C# library that acts as a wrapper around FFmpeg, letting you use its features directly from your C# code. This makes integrating complex media processing into your applications much easier and far more maintainable.

With Xabe.FFmpeg, you can perform format conversion, extract and preprocess audio, or apply effects—all using clear and modern C# syntax. The library manages running FFmpeg and parsing its output, so you can focus on your application logic instead of low-level scripting. While Xabe.FFmpeg simplifies the interaction, developers still need to implement robust error handling, logging, and manage the asynchronous nature of media processing operations within their C# code.

To have Xabe.FFmpeg automatically download the FFmpeg executables at runtime, you can use the FFmpegDownloader.GetLatestVersion() method. This method downloads the latest FFmpeg binaries and sets the path automatically, so you don’t need to manually manage the executables. Here’s how you can do it:

You can place this call in your application’s startup logic or inside your AudioProcessor.Initialize() method. For example:

This approach is especially useful for development, CI/CD pipelines, or when you want to ensure your application always uses the latest FFmpeg version without manual setup. For production environments, you may still prefer to manage the binaries yourself for stability and compliance reasons.

Alternatively, you can manually download the FFmpeg executables from the official FFmpeg website and place them in a directory accessible to your application. Then, use FFmpeg.SetExecutablesPath("your_ffmpeg_directory") to specify the path.

Once you have the executables, to use Xabe.FFmpeg in your own C# projects, you install it via NuGet:

After installing the package, you also need to make sure that the FFmpeg executables (the actual programs that do the work) are available to your application. Xabe.FFmpeg can download these for you, or you can provide your own.

To initialize Xabe.FFmpeg in your code, you set the path to the FFmpeg executables. This tells the library where to find the FFmpeg tools it needs to run. Here is how you might do this:

This line sets the path to a folder named ffmpeg_binaries where the FFmpeg executables are stored in our case.

To keep your code modular and reusable, it’s a good idea to place audio processing logic in a separate file. In the next header there we'll start building an AudioProcessor class that you would put in its own file, AudioProcessor.cs. This class will handle initializing FFmpeg and preprocessing audio files by normalizing their volume and converting them to a mono, 16kHz WAV format—an ideal input for tasks such as speech recognition and transcription.

Next, let’s look at how to initialize this processor so it can be used throughout your application.

Before you can preprocess audio files, you need to ensure that your application knows where to find the FFmpeg executables. The following code demonstrates how to set up the AudioProcessor class to initialize FFmpeg by specifying the path to the required binaries. This initialization step is essential for enabling all subsequent audio processing operations.

The Initialize() method is declared as static because the FFmpeg executable path is a global setting for the entire application process—it only needs to be set once, regardless of how many AudioProcessor instances you create. This pattern is appropriate for most real-world applications, as it avoids redundant configuration and ensures all audio processing operations use the same FFmpeg setup.

If your application ever needs to support different FFmpeg versions or paths simultaneously, you would need a more advanced approach. For most scenarios, static initialization is simple, safe, and effective.

Once FFmpeg is initialized, you can use the NormalizeAudioAsync method to preprocess your audio files. The code below shows how this method takes an input audio file, applies volume normalization, converts it to mono, sets the sample rate to 16kHz, and outputs a WAV file in a widely compatible format.

Here are the key parameters used in this method:

• -i "{inputPath}": Specifies the input audio file.
• -af loudnorm: Applies the "loudnorm" audio filter, which normalizes the volume of the audio.
• -ar 16000: Sets the audio sample rate to 16,000 Hz (16kHz), which is a common requirement for speech recognition systems.
• -ac 1: Converts the audio to a single (mono) channel.
• -acodec pcm_s16le: Sets the audio codec to 16-bit signed little-endian PCM, which is a standard uncompressed WAV format.
• "{outputPath}": Specifies the output file path for the processed audio.

This preprocessing ensures your audio is ready for tasks like transcription or further analysis by standardizing the format and quality of the audio data.

Here is the complete AudioProcessor class, ready to be used in your project. This class handles both the initialization of FFmpeg and the normalization of audio files:

This class can be placed in its own file, for example, AudioProcessor.cs, and reused throughout your application for consistent audio preprocessing.

Once you have a reusable processor class, using it in your application becomes very straightforward. We'll also add a TranscriptionService that combines the logic to transcribe the processed audio, reusing code and techniques you learned during the previous course. You'll see this service in action in the upcoming practice section.

The following example code, placed in your Program.cs file, shows how to initialize the processor, run preprocessing on an audio file, and then transcribe the normalized audio:

When you run this code, you’ll see output similar to:

This demonstrates a practical workflow: first, the audio is normalized and converted, then the resulting file is sent to the transcription service, and finally, the transcribed text is displayed. This modular approach makes it easy to extend your pipeline with additional processing or analysis steps.

In this lesson, you learned about the importance of media processing and how FFmpeg is used in real-world applications. You were introduced to the Xabe.FFmpeg library, which makes it easy to use FFmpeg in C#. You saw how to set up Xabe.FFmpeg both in your own projects and in the CodeSignal environment, and you explored how to preprocess audio—normalizing volume and converting format—to prepare files for transcription or analysis.

You are now ready to practice these concepts with hands-on exercises. In the next section, you will get a chance to work with audio files yourself, using the tools and techniques you have just learned. This will help you build confidence and prepare you for more advanced media processing tasks later in the course. Good luck, and enjoy experimenting with audio processing in C#!