Skill UI

This guide provides comprehensive instructions for integrating the Wispr Flow API for advanced voice-to-text transcription. It details both WebSocket streaming (recommended for low latency) and REST API endpoints, covering authentication setup, API key configuration, and accessing short-lived client access tokens. Ideal for building real-time, voice-powered applications, such as code editors or dictation tools.

This guide provides detailed instructions for setting up a local development loop using the Wispr Flow voice-to-text API. It covers connecting via WebSocket (recommended for low latency) or REST API, handling authentication, and optimizing audio input. Ideal for building voice-powered applications, dictation tools, or integrating sophisticated voice command features into existing SaaS products.

Provides comprehensive guidance for optimizing the performance of the Wispr Flow voice-to-text API. This skill covers best practices for integrating dictation, streaming, and building robust voice-powered applications. It details optimal WebSocket usage, recommended audio formats (16kHz PCM), context setting for specialized fields like programming, and robust error handling strategies (e.g., 401 Unauthorized, network disconnects).

This comprehensive checklist provides guidance for integrating the Wispr Flow voice-to-text API into a production environment. It covers best practices for both WebSocket and REST API usage, addressing critical elements like authentication, robust error handling (e.g., 401 Unauthorized or network disconnections), and optimizing performance. It is essential for building reliable, voice-powered applications, especially those handling complex contexts like programming code and CLI commands.

This comprehensive guide provides developers with best practices and detailed instructions for integrating the Wispr Flow voice-to-text API. It covers critical topics such as managing rate limits, choosing between WebSocket streaming and REST endpoints, handling common error codes (e.g., 401 Unauthorized), and optimizing context awareness, particularly for programming code and developer jargon. Essential for building reliable, low-latency voice-powered applications.

This guide provides the reference architecture and best practices for integrating Wispr Flow's voice-to-text API. It details how to utilize both WebSocket (for low-latency streaming) and REST APIs, covering essential concepts like audio format requirements, context awareness (especially for programming code), and comprehensive error handling strategies.

This pattern provides comprehensive guidance for integrating the Wispr Flow voice-to-text API. It covers best practices for both low-latency WebSocket streaming and simpler REST API usage. Key topics include authentication, recommended audio formats, robust error handling, and context awareness (e.g., understanding code or CLI commands), making it ideal for building advanced voice-powered applications.

This guide provides essential security and technical guidance for integrating Wispr Flow's voice-to-text API. It covers best practices for authentication (API keys, access tokens), utilizing streaming methods like WebSockets, handling common API errors, and optimizing context awareness for high-accuracy transcription in various applications.

A comprehensive guide for migrating and integrating with the Wispr Flow voice-to-text API. This skill details best practices for both WebSocket (low-latency streaming) and REST API methods. It covers authentication, optimal audio formatting (16kHz mono PCM), context awareness for specialized jargon (e.g., code), and robust error handling across multiple platforms (Mac, iOS, browser).

A comprehensive guide for integrating the Wispr Flow voice-to-text API using webhooks. This skill covers best practices for real-time speech recognition, including low-latency WebSocket streaming and robust REST API calls. It is essential for developers building voice-powered applications, dictation tools, or any system requiring advanced audio processing and context-aware transcription.

This skill provides comprehensive testing for WebSocket APIs, covering protocol basics, Cross-Site WebSocket Hijacking (CSWSH), and common security flaws. Use it when testing applications that rely on real-time communication, chat features, or persistent data streams. It validates Origin checks, session binding, and message integrity.

This skill guides developers on building real-time, bidirectional, and multimodal streaming applications using the Gemini Live API. It covers low-latency voice, video, and text streaming over WebSockets, incorporating advanced features like Voice Activity Detection (VAD), function calling, and secure session management. Ideal for conversational AI, interactive video conferencing, and complex real-time user experiences.