we have integrated livekit, when we show customer ...
# ask-ai
c
we have integrated livekit, when we show customer demo on google meet, the audio track merges and the agent doesn't pick the voice at all
t
Thanks for reaching out, Arjun. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Could you please share a bit more about how you're using LiveKit during the Google Meet demo? For instance, are you embedding LiveKit in a web page that you view during the call, using LiveKit Meet alongside Google Meet, or maybe screen-sharing a LiveKit application? Understanding this will really help us find the best solution for you. Sources: src/room/Room.ts | livekit-android-sdk/src/main/java/io/livekit/android/LiveKitOverrides.kt | CHANGELOG.md | Understanding "silence detected on local audio track" warning
c
For the text that appears below as one talks, the transcript generated for what I said seemed very wrong a lot of the time. Yet, the agent seemed to have heard what I said accurately and gave competent answers. If you looked at the text below as it went though, the thing looked pretty embarrassing. Is there a fix coming for that?
t
Improving transcription accuracy in LiveKit is definitely achievable by fine-tuning the configuration of your Speech-to-Text (STT) provider. Let me walk you through some helpful steps: • Choose an accurate STT provider: Providers like Google Cloud STT and Deepgram are great choices known for delivering high accuracy in real-time transcription. • Enable enhanced accuracy settings: For instance, Speechmatics offers an
operating_point
setting that lets you prioritize accuracy over latency, which can make a real difference. • Use custom vocabulary: Adding domain-specific terms with the
additional_vocab
parameter can really boost recognition of those unique or uncommon words. • Optimize audio quality: Clear audio is key! Using noise cancellation or tweaking noise reduction settings in your STT plugin can help a lot. • Adjust audio chunking: Finding the right frame size and encoding (like 20ms frames) helps balance latency and accuracy smoothly. If you’re working with a particular STT provider, I’d be more than happy to guide you through configuring these settings step-by-step. Just let me know if you'd like some extra help with implementation or testing—I’m here to support you! Sources: livekit-plugins/livekit-plugins-openai/livekit/plugins/openai/stt.py | Speechmatics integration guide | LiveKit Docs | Live Conversations with AI using ChatGPT and WebRTC | Live Conversations with AI using ChatGPT and WebRTC | README.md
c
we are using realtime models
t
Since you're working with real-time models for transcription, I’d love to share some friendly tips to help improve accuracy: • Provider Selection: It’s great to choose a provider that’s really tuned for real-time transcription, like Google Cloud STT or Deepgram—they’re known for being both fast and accurate. • Enhanced Accuracy Settings: If you’re using Speechmatics, try setting the
operating_point
to
ENHANCED
. It might add a little latency, but it usually boosts accuracy nicely. • Custom Vocabulary: Adding domain-specific terms through the
additional_vocab
parameter can really help the system recognize specialized words or phrases better. • Audio Quality: Using noise cancellation and ensuring your audio input is clear can make a big difference. If your STT provider supports noise reduction settings, configuring those can reduce errors too. • Audio Chunking: Choosing the right frame sizes—like around 20ms—helps balance latency and accuracy during real-time streaming. If you’d like, I’m here to help you set these up for your specific provider or walk you through testing the changes. Just let me know!
c
yes, share the code config
t
I'm happy to share an example configuration that can help improve your transcription accuracy using Speechmatics and Google Cloud STT. Let's walk through it together: Speechmatics Configuration
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const config = {
  operating_point: 'ENHANCED', // Prioritize accuracy over latency
  additional_vocab: ['custom', 'terms', 'specific', 'to', 'your', 'domain'], // Add domain-specific vocabulary
  max_delay: 500 // Adjust delay for better accuracy
};
You’ll want to pass this configuration when you initialize the Speechmatics transcription service. Google Cloud STT Configuration
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const config = {
  encoding: 'LINEAR16',
  sampleRateHertz: 16000,
  languageCode: 'en-US',
  enableAutomaticPunctuation: true, // Improves readability
  model: 'default', // Use 'video' or 'phone_call' for specific use cases
  useEnhanced: true // Enable enhanced models for better accuracy
};
This configuration should be passed when setting up your Google Cloud STT client. If you’re working with a different provider or would like some help integrating these settings into your application, please don’t hesitate to reach out—I’m here to help!