Real-Time Voice AI Platform
Conversational voice treated as a real-time system rather than a sequential speech pipeline — bounded memory, predictable execution, and interruption handled at the audio layer.
The problem
The project explores how conversational voice AI can operate as a real-time system rather than a conventional sequential speech pipeline. The current implementation focuses on establishing reliable low-level audio transport, bounded concurrency and testable runtime foundations.
Build and validate the technical foundations for a responsive real-time voice system, keeping latency, memory ownership, interruption handling and system safety explicit architectural concerns.
The challenges
- Designing the audio path around bounded memory and predictable execution needed deliberate ownership, queueing and concurrency decisions rather than generic application patterns.
- Real-time interruption and streaming behaviour has to be coordinated across audio transport, session state and generation boundaries, without letting stale work reach playback.
- Probabilistic model output and deterministic control are separated by design, so actions and critical values pass a validation boundary before they take effect.
The solution
- Core C++ audio primitives: PCM validation, audio frame handling, jitter buffering and fixed-capacity SPSC queues.
- A working TCP/WebSocket listener feeding incoming PCM audio into the real-time processing path.
- Deterministic ASR and TTS test doubles, so the pipeline can be exercised without waiting on model integrations.
- Bounded per-producer media queues built around single-producer, single-consumer ownership.
- An engineering workflow of small tickets, explicit invariants and automated validation, so implementation status is evidence-based.
Technology
C++17TEN FrameworkWebSocketsTCPCMake
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