What is low-latency-audio-pipeline?

This is a C++ project that plays audio through your computer's speakers with as little delay as possible, using techniques normally found in professional audio software. The author built it as a learning project after studying operating systems and wanted to apply low-level performance concepts to something audible. The program generates a continuous 440 Hz sine wave tone (the musical note A) and sends it to the speakers in real time. It splits this work across two threads running in parallel: one thread does the math to generate the audio data, and another (managed by a library called Miniaudio) reads that data and sends it to the hardware. The two threads share data through a ring buffer, which is a fixed-size circular queue that keeps getting reused without allocating new memory while the audio plays. The ring buffer is the technical focus of the project. It is lock-free, meaning neither thread ever stops and waits for the other using a traditional lock mechanism. Instead, it uses low-level CPU instructions called atomics to coordinate access safely without pausing. The buffer is also designed to avoid false sharing, a subtle performance problem where two CPU cores accidentally contend over the same small piece of cache memory. These optimizations together are what keep audio output smooth and glitch-free even under load. The project is built with C++20 and primarily targets Windows, where the author has confirmed it works with Visual Studio. The README notes that Linux and macOS support may require adjustments to the audio driver code. It is a student-level learning project, not a general-purpose audio library, but the ring buffer implementation is well-documented and demonstrates several real techniques used in low-latency software.