Topic

AssemblyAI Website

All digests tagged AssemblyAI Website

Workshop: Building and optimizing dictation features thumbnail

· 36:06

Workshop: Building and optimizing dictation features

This workshop details the architecture for building low-latency dictation features using AssemblyAI's APIs. The core strategy involves leveraging the Sync API—a single POST request—to achieve speed superior to traditional async or streaming methods. The process is optimized through three key stages: using key terms prompting for accuracy, implementing connection warming to minimize network overhead (DNS, TCP, TLS), and running a cleanup pass via LLM Gateway (e.g., Qwen3.5 4B Fast) to refine raw transcripts into polished, intent-preserving text. The goal is to deliver results on screen in under one second.

Key takeaways

  1. Sync API Preference 5:35

    The Sync API is preferred for dictation because it avoids the overhead of maintaining a WebSocket connection (streaming) and bypasses the inherent latency floor of traditional async endpoints, which is unsuitable for short, burst dictation sessions.

  2. Latency Optimization via Warming 11:40

    To minimize latency, it is critical to call a warm endpoint (`client.sync.warm`) while the user is recording. This pre-pays the networking costs (DNS, TCP, TLS handshake), ensuring the subsequent transcription request goes straight to inference.

  3. Cleanup Pass with LLM Gateway 18:20

    A cleanup step using an LLM (like Qwen3.5 4B Fast) is necessary to transform raw, spoken text (e.g., 'I think we should meet in 5 minutes') into polished, corrected text while preserving the original intent. This is achieved by prompting the model to act as a function in a pipeline, not an assistant.

  4. Upcoming Dictation API 30:00

    AssemblyAI is rolling out a dedicated Dictation API (`client.dictation.describe`) that will wrap the entire loop (STT + Cleanup) into a single, simplified call, making the process easier for developers.

Watch on YouTube Full article

Universal 3.5 Pro Demo: Smarter Speech-to-Text with Contextual Awareness thumbnail

· 10:07

Universal 3.5 Pro Demo: Smarter Speech-to-Text with Contextual Awareness

This demo introduces Universal 3.5 Pro, an advanced Speech-to-Text (STT) model designed to significantly boost transcription accuracy through enhanced contextual awareness. Key features include passing domain-specific prompts (e.g., 'cardiology consultation'), applying context to key terms to prevent misapplication, and supporting dynamic mid-call prompt updates via API calls. Furthermore, the model retains conversation history (agent context), allowing it to accurately transcribe user input even in poor audio conditions by understanding the situational flow of a voice agent interaction.

Key takeaways

  1. Contextual Prompting for Domain Accuracy

    Passing detailed information about the audio content (e.g., 'cardiology consultation between Dr. Smith and elderly patient regarding chest pain...') dramatically improves model accuracy within specific domains. The more specific the prompt, the better the results.

  2. Contextual Key Terms 2:00

    Unlike previous methods where key terms were applied blindly, Universal 3.5 Pro allows users to define what a key term represents (e.g., 'The user's name is Zachary Klebanoff'). This prevents the model from incorrectly applying terminology based solely on acoustic similarity.

  3. Dynamic Mid-Call Prompt Updates 2:55

    The prompt can be updated in real time via the API (not available in the playground demo). This is crucial for voice agents, allowing tool calls or external data to adjust the model's context mid-conversation.

  4. Conversation/Agent Context 3:30

    The model retains previous transcriptions and accepts LLM-generated responses from a voice agent as context. This provides situational awareness, improving accuracy even in poor audio conditions and reducing the Word Error Rate (WER) on voice agent datasets.

Watch on YouTube Full article