Hi Folks,
Since I obviously can’t keep my hands to myself, I like to tinker and make things gooder on my node. And lately I have been archiving the node traffic. I use Opus, as it’s light and small and sounds good at a reasonable sample rate and bitrate. But I have to rsync it offsite to do that and then warehouse it.
Now that I have seemed to have solved the riddle of using improved codecs between my nodes (better quality/higher sample rate than ulaw), I have turned my attention to the archiving utility in ASL3.
It seems to use a GSM codec wrapped in a WAV wrapper (at least, according to ffmpeg). I wonder if this GSM/WAV solution can’t be taken apart and at least configuration flags be put in to specify a destination codec, sample rate and bitrate (within logical reason).
Does anyone have a clue as to where this particular routine lives?
Carl/K6CRS