Initial commit: HPR Knowledge Base MCP Server
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hpr_transcripts/hpr1181.txt
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Episode: 1181
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Title: HPR1181: Mumble Audio Issues
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Source: https://hub.hackerpublicradio.org/ccdn.php?filename=/eps/hpr1181/hpr1181.mp3
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Transcribed: 2025-10-17 21:06:58
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---
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Greetings, Delan here again.
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After getting feedback on the last mumble episode, I thought it might be worthwhile to take
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a few minutes to talk about some of the specific audio issues people encountered and what
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can be done about them.
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There seem to be four main issues that people typically experience, and we're going to
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be working in the audio input setting section with advanced options toggled on.
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The checkbox for the advanced options is located in the lower left corner of the configured
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screen.
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Going through the audio wizard does many of these things automatically, but if you're
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going to try and fine tune your audio, I think it's simpler to do it this way.
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The first and most common issue is static or peeking, particularly when speaking loudly,
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and it sounds something like this.
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Okay, with luck, this should be badly overdriven.
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The first step in addressing this is to decrease your amplification in your mumble audio
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settings.
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If you have decreased the amplification within mumble to 1.0, it's lowest setting, but
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you're still experiencing peeking, then you'll need to adjust your microphone input level
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in your sound card mixer settings.
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The second problem is quiet audio, which I suggest handling in the opposite direction.
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First, adjust your mixer settings and raise your mic level as high as you're comfortable
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with, and then try the amplification settings within mumble.
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The third is distorted or warbling audio, which might sound something like this.
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And this should be what it sounds like when noise suppression is too high.
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This is generally caused by the noise suppression settings.
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Noise suppression attempts to act as a filter to remove background noise, but when it is set
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too high, it can distort your audio and make it sound almost watery for lack of a better
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term.
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Mumble defaults to negative 30 decibels, but I find that negative 15 to negative 20 tends
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to offer a better balance between distortion and background filtering.
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Results can vary pretty wildly here, and it all depends on the noise level in your environment
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and the microphone you're using.
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The fourth is choppy audio, like this.
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This is typically a result of a poor internet connection rather than audio settings.
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The first thing to try is to turn off the quality of service settings in mumble, which
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are located in the network section.
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It's not likely that this will help.
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It seems to be more affixed for when your client is being repeatedly dropped by the server,
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but turning it off will help eliminate one possible source of trouble.
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Just try adjusting your audio per packet in quality settings, found in the advanced audio
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input section.
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I recommend increasing your audio per packet settings, and then decreasing your quality
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settings.
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But be aware that once you drop below about 45 kilobits per second, you'll be using
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speaks rather than kelt as your audio codec.
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Mumble will tell you which codec you're trying to use under the audio per packet slider,
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so it will let you know once you've crossed that threshold.
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I should also note here that mumble 1.2.3 is starting to ship without speaks support
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in some distros, and once 1.2.4 drops, the preferred codec will be opus.
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If you run into an issue where people can't hear one another, this is almost certainly
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the cause.
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The answer for the time being is to try to get everyone on kelt.
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Yes, this is a bit of a pain in the ass.
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As a year ago, speaks was the codec all installations had in common.
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And yes, we're probably going to go through this rigmarole again once 1.2.4 comes out,
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at least until everyone is fully on board with opus.
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And in the last episode I did about mumble, I mentioned that you can turn on loopback either
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locally or from the server in order to hear your own audio.
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This works well, but it requires you to listen to your voice as you're speaking, which
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can be a little confusing.
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However, there is another option which might work better for you.
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Evebot was written to capture audio from one channel and then transmitted into another
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channel after a delay, mainly used during game tournaments to match team communications
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with the delayed video feed.
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However, you can also use it as a delayed loopback for an audio testing channel.
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When you give the bot the command to connect, tell it to use the same channel in the eavesdrop
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and relay options.
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It's smart enough to avoid going into a mimic spawning loop.
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I like to set the delay to 10 seconds, as a shorter delay doesn't give me enough time
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to say enough to get a decent idea of what the audio is actually sounding like.
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You can find Evebot at Freemaster, f-r-y-m-a-s-t-e-r dot 1-2-7-001 dot org slash mumble.
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You have been listening to Hacker Public Radio, or Tacker Public Radio does our.
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