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Real-Time Audio Systems Engineer (Real-Time Browser Audio)

Roles & Responsibilities

  • Strong experience with WebSockets and browser-based real-time audio
  • Deep understanding of Web Audio API, audio buffers, sample rates, streams, and playback pipelines
  • Hands-on experience with streaming STT/TTS and integrating LLMs
  • Proficiency in Node.js or Python for real-time backend services
  • Knowledge of DSP basics (noise suppression, VAD), latency optimization, and audio pipeline debugging
  • Nice to have: WebRTC audio; wake-word engines (Picovoice, Porcupine); Alexa/Siri-like conversational agents

Requirements:

  • Build real-time browser audio clients using Web Audio API, MediaRecorder, AudioContext, AudioWorklets, and WebSockets
  • Implement bi-directional audio streaming (microphone, backend, speaker) with 500ms latency targets
  • Integrate streaming STT (AssemblyAI, OpenAI Realtime, Transcribe, or similar)
  • Implement streaming TTS playback, including chunked audio decoding and buffered playback
  • Develop backend streaming servers (Node.js/Python) for handling audio frames, STT events, LLM routing, and TTS responses
  • Build and optimize wake-word workflows and VAD (voice activity detection)
  • Handle audio formats codecs (PCM, Opus, WAV) and real-time transcoding
  • Ensure reliability: reconnection logic, jitter buffers, packet timing, audio frame synchronization

Job description

Job Description Real-Time Audio Systems Engineer (Real-Time Browser Audio)

We are seeking a highly skilled Real-Time Audio Systems Engineer with strong expertise in browser-based audio streaming, WebSockets, and real-time voice bot architecture. The ideal candidate will design and implement low-latency, full-duplex audio pipelines that run directly in the browser and integrate with modern STT, LLM, and TTS systems.

Responsibilities

  • Build real-time browser audio clients using Web Audio API, MediaRecorder, AudioContext, AudioWorklets, and WebSockets.

  • Implement bi-directional audio streaming (microphone backend speaker) with <500ms latency targets.

  • Integrate streaming STT (AssemblyAI, OpenAI Realtime, Transcribe, or similar).

  • Implement streaming TTS playback, including chunked audio decoding and buffered playback.

  • Develop backend streaming servers (Node.js/Python) for handling audio frames, STT events, LLM routing, and TTS responses.

  • Build and optimize wake-word workflows and VAD (voice activity detection).

  • Handle audio formats & codecs (PCM, Opus, WAV) and real-time transcoding.

  • Ensure reliability: reconnection logic, jitter buffers, packet timing, audio frame synchronization.

Requirements

  • Strong experience with WebSockets and browser-based real-time audio.

  • Deep understanding of Web Audio API, audio buffers, sample rates, streams, and playback pipelines.

  • Hands-on experience with streaming STT/TTS and integrating LLMs.

  • Proficiency in Node.js or Python for real-time backend services.

  • Knowledge of DSP basics (noise suppression, VAD), latency optimization, and audio pipeline debugging.

Nice to Have

  • Experience with WebRTC audio.

  • Familiarity with wake-word engines (Picovoice, Porcupine).

  • Experience building Alexa/Siri-like conversational agents.

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