The preceding chapters have detailed the components of an ASR system, from audio processing to the combination of acoustic and language models. The fundamental goal is to find the word sequence that maximizes the probability for a given audio observation , often expressed as:
This chapter moves from the mechanics of that equation to its practical application. The focus shifts to using established tools that have already implemented these components, allowing you to build functional applications without training models from scratch.
You will use popular Python libraries to construct a complete speech-to-text program. We will cover setting up your environment, loading pre-trained models, and writing scripts to transcribe audio. You will work with both pre-recorded audio files and live input from a microphone. The chapter concludes with a hands-on exercise to build a simple voice-activated command tool, solidifying your ability to integrate speech recognition into an application.
6.1 Introduction to Speech Recognition APIs and Libraries
6.2 Setting Up Your Python Environment
6.3 Using a Pre-trained Model for Transcription
6.4 Transcribing Audio from a File
6.5 Capturing and Transcribing Microphone Input in Real-Time
6.6 Handling API Responses and Errors
6.7 Practice: Build a Simple Voice Command Tool