Why Speech Recognition API Requires a Different Architecture
Speech Recognition API: Streaming, WebSockets and Latency A speech recognition API that accepts a file and returns a transcript is a solved problem.The architecture is simple because the constraint...

Source: DEV Community
Speech Recognition API: Streaming, WebSockets and Latency A speech recognition API that accepts a file and returns a transcript is a solved problem.The architecture is simple because the constraints are simple. Real-time transcription is different. The audio doesn't exist yet when processing needs to begin. The user is still speaking while the system needs to be building a hypothesis about what they said. The application needs a partial answer now, not a complete answer in two seconds. These constraints change the architecture at every layer, from how audio is captured and transmitted to how the recognition model processes it and how results flow back to the client. This piece walks through that architecture end to end. Not as an API reference, but as an explanation of what is actually happening inside a streaming speech recognition system and why each component is designed the way it is. The fundamental problem with batch transcription for real-time use Before looking at how streaming