How do I lower the voice chat quality to save on bandwidth?
Avatar voice comment says " (stream will automatically resample to 48000 before encoding with OPUS)"
4 Replies
Hi @Mischa. I'm kapa.ai, your friendly support bot. I will try to answer your question by searching through the knowledge base. :wave:
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Short answer: there isn’t a documented knob to lower RealtimeAvatarVoice quality. Normcore uses the Opus codec and automatically adapts bitrate/quality based on network conditions, targeting high-quality, low-latency voice. The stream is resampled to 48 kHz before Opus encoding, and the adaptive bitrate is managed internally. [High Quality A/V; Voice Chat]
What you can do today:
- Rely on built‑in adaptive bitrate and congestion control; Normcore/WebRTC will scale down when bandwidth is constrained. [WebRTC congestion; High Quality A/V]
- Optimize overall bandwidth elsewhere (e.g., transform updates) to leave more headroom for voice. [Network profiler; Optimizing bandwidth]
If you need explicit control (e.g., force a lower bitrate or different audio settings), the knowledge sources don’t provide a public API for that in Normcore Public. I’m sorry, but there’s no information about a supported setting to directly lower voice bitrate/sample rate via RealtimeAvatarVoice. You could fork/modify RealtimeAvatarVoice if you’re integrating at the code level, but this isn’t documented here. [Voice Chat]
For stricter control (custom media settings), consider Normcore Private’s MediaStream API (private preview) and talk to your account manager. [Media streams]
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<@847936237003604048> Kapa.ai is still learning and improving, please let me know how I did by reacting below.
There is no setting to reduce the quality unfortunately.
Some folks use Vivox which sounds considerably worse but it is generally free for VR scale games
We do love the high quality audio Normcore gives us!