Audio & Voice Analysis

Build speech-to-text applications that feel human, with the lowest-in-class time-to-first-token, fastest streaming output, and stable throughput under concurrency.

problem

Latency disrupts audio analysis and voice agents

Unstable inference disrupts real-time voice and audio interactions

Voice and audio applications rely on steady token generation and multi-step reasoning. Bursty or inconsistent streaming causes poor user experience.

Any delay breaks the conversation flow

Even sub-second latency creates pauses users perceive as broken. Voice and audio applications demand response times that feel instant.

Scaling to millions of hours is challenging

Infrastructure that throttles under sustained, high-concurrency workloads can't keep up with production voice and audio pipelines as workload volume grows.

Throughput degrades under sustained concurrency

Thousands of simultaneous sessions introduce queuing and latency spikes, breaking real-time performance when demand is highest.

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solution

FriendliAI delivers the speed and reliability that voice and audio analysis demand

Real-time inference for voice and audio pipelines

Consistent token streaming and high-throughput inference keep multi-step voice and audio interactions smooth and uninterrupted under load.

Ultra-low latency streaming

Friendli TCache prefix caching and custom GPU kernels reduces time-to-first-token latency, keeping the gap between user speech and AI response imperceptible.

Effortless scaling to millions of hours

Continuous batching and autoscaling sustain throughput across simultaneous jobs, keeping voice and audio pipelines as workload volume grows to millions of hours.

Stable throughput under high concurrency

Continuous batching absorbs thousands of simultaneous sessions without queuing degradation, maintaining consistent quality as load scales.

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