Call audio quality on PinePhonePro is problematic for others
I've been trying to make PinePhonePro my primary phone for a while (currently on Manjaro ARM + Phosh Beta 22). Worked through some issues and most things are awesome, it's like having a laptop in my pocket -- but using it as an actual phone is socially awkward, to the point where I can't have real conversations with friends and family. Sadly, the audio quality is noisy and distorted for the person on the other end. I can hear them loud and clear, but they hear my voice as noisy and muffled. One friend said it's about a 4 out of 10 where 1 is can't understand words at all and 10 is clear. Connecting a Bluetooth headset may potentially work around this, but the settings get all confused and don't seem to respect manually configured inputs/outputs during a call.
I wish I could help directly with working through this but there seems to be many interconnected software layers. From what I can tell, it's using Pipewire, GStreamer, maybe also PulseAudio too, all together? More moving parts (especially new parts like Pipewire) means more chance of failure, and currently failure is the experience. I know this can be worked out, and I look forward to that day -- maybe with some guidance I could even contribute...
I have used ALSA alone successfully for years with pro audio, including real-time DSP. It's very efficient and low latency. I hope this project will consider a simplified direct ALSA audio flow as an alternative to Pipewire -- at least for call audio. Can someone help me understand this a bit better so we can try moving toward good call quality? Would JACK be an option? I've used it with much success also, though it's not as efficient as ALSA. Are codecs the problem, or maybe the PinePhonePro hardware itself?