Hey everyone, I’ve been using MeetGeek for a few weeks now to record and transcribe our team’s weekly syncs. Overall, it’s been super helpful for pulling out action items. But I’ve run into a persistent issue and I’m wondering if anyone else has figured this out.
The audio for two of my remote team members consistently comes through very quiet in the recordings. Everyone else sounds fine. It’s not their hardware—I’ve checked, and they sound normal on Zoom directly and in other apps. The MeetGeek transcript for their speech also has more errors, presumably because the source audio is too low.
I’m connecting MeetGeek to Google Meet. I’ve looked through the settings, but I can’t find any per-participant audio gain controls. Is there something in the setup I might have missed? Here’s how I have it configured:
```json
// My current bot settings (from the API perspective)
{
"integration": "google_meet",
"auto_join": true,
"record_video": false,
"audio_processing": "standard"
}
```
Could this be related to how MeetGeek captures the individual audio streams? Maybe those two participants are joining from a different client (like a phone?) and MeetGeek is handling that stream differently? Any troubleshooting steps or workarounds would be awesome. I really want the transcripts to be accurate for everyone.
You've hit on a likely culprit with the different client idea. MeetGeek, like many recording bots, receives a separate audio stream from each participant via the conferencing platform's API. If those two participants are joining from the Google Meet mobile app or a browser with specific audio codec settings, their stream's input level might be normalized lower at the source before it ever reaches the bot.
The `"audio_processing": "standard"` setting is probably applying a uniform gain stage to all incoming streams. Since there's no per-participant control in the settings, you'd need to address it upstream. Ask those two members to check their Google Meet audio settings *within the meeting*: click the three dots menu on their own video tile, select 'Audio settings,' and look for an input volume slider. They should max that out.
Also, have them test from Chrome if they're using something else, as Safari and Firefox can handle WebRTC audio levels differently. The transcript errors are a direct symptom; the signal-to-noise ratio is too poor for the ASR engine. If level adjustments don't work, you're looking at a platform limitation and might need to log a feature request for per-stream gain controls with MeetGeek support.