Speech Software Engineer

What you'll do

• Speech Model Optimization & Applied Research:
Tune and optimize ASR and TTS models for real-world call center environments, improving transcription accuracy, noise robustness, and speaker variability
Improve spoken output naturalness by refining prosody, pacing, number and spelling pronunciation, and conversational flow
Balance latency vs. quality tradeoffs in streaming speech pipelines to maintain real-time responsiveness
Evaluate and integrate emerging speech technologies (e.g., noise suppression, voice activity detection, diarization) to measurably improve performance
• Voice Infrastructure & Systems Engineering
Architect and modernize a scalable, high-availability voice infrastructure that replaces legacy systems
Build multi-threaded, low-latency server frameworks capable of handling thousands of concurrent real-time audio streams
Design and operate streaming ASR → LLM → TTS pipelines that power live AI-driven customer conversations
Develop robust media stream handling to ensure reliable audio flow between telephony providers, clients, and ML services
• Evaluation, Observability & Quality
Define and implement speech quality evaluation frameworks, including WER/CER analysis, latency tracking, and perceptual TTS metrics
Build tooling and dashboards to monitor production performance and detect regressions in accuracy, latency, or naturalness
Create load-testing and simulation tools to model high-concurrency, real-world voice traffic
• Cross-Functional Collaboration
Partner with Speech Scientists and ML Researchers to productionize new ASR and TTS models
Work with Security and Compliance teams to ensure voice data handling meets enterprise and regulatory standards
Collaborate with Product teams to translate conversational quality requirements into measurable system improvements

What you'll need

• Core Engineering Background
5+ years of software engineering experience building and operating production-grade distributed systems
Strong proficiency in Golang or Python (or willingness to become an expert quickly)
Experience designing low-latency, high-concurrency systems, ideally involving real-time media or streaming data
• Speech & Audio Expertise
Practical experience working with ASR and/or TTS systems in applied or production environments
Understanding of how to adapt and tune speech models for domain-specific use cases
Familiarity with speech quality metrics such as WER, CER, MOS, latency, and streaming stability
Strong grasp of audio fundamentals, including sample rates, codecs (Opus, G.711), buffering, packet loss, and jitter
• Applied ML for Speech
Experience evaluating model performance and running structured experiments to improve transcription accuracy and speech naturalness
Comfort working with modern ML tooling and model APIs to fine-tune, adapt, or post-process speech model outputs
Ability to make pragmatic tradeoffs between model quality, compute cost, and real-time constraints

What we'd like to see

• Experience with noise reduction, echo cancellation, VAD, diarization, or other speech enhancement technologies
• Familiarity with forced alignment techniques or phoneme/word-level timing models
• Hands-on experience deploying ML services with Kubernetes, Docker, and cloud platforms (AWS/GCP/Azure)
• Knowledge of event-driven and asynchronous systems (e.g., async I/O, event loops, streaming frameworks)
• Experience analyzing large-scale speech or conversation datasets to drive model or system improvements