Kazam does not utilize the pulseaudio default-sample-rate setting
| Affects | Status | Importance | Assigned to | Milestone | |
|---|---|---|---|---|---|
| Kazam Screencaster |
New
|
Undecided
|
Unassigned | ||
Bug Description
I've taken the trouble to adjust my pulseaudio default-sample-rate setting, adjusting it to 48000 from the default 44100, and have confirmed this setting is being picked up by pulseaudio by running: pulseaudio --dump-conf. Whenever I record a screencast, the audio sample rate remains at 44100 (confirmed by editing the resuting AVI file with Audacity), which is not correct for my sound card hardware. The result is that the quality of audio tracks for my screencasts is noticeably degraded. The worst symptom is a persistent whooshing/static sound. Now, I *can* filter this sound out with Audacity and get good results, but I really shouldn't have to. I've had a look at the Kazam source code, but I can't really see how the sample rate is set... I suspect it's inherited from a library you're using. I think this may be the cause of many of the sound quality issues that have been mentioned on conjunction with Kazam. Fixing this should be an easy win for you, and will be a nice improvement for Kazam.
| Hardy Pottinger (pottingerhj) wrote | #1 |
Alternatively, you can add a configuration option to set the audio sample rate, though just using the configuration from pulseaudio would probably be a better choice