I finally got my hdmi2csi board, and have had a lot of success with everything including audio. I’m now stuck with 1 issue, and I hope someone can point me in the right direction.
When I capture audio using arecord, all is well… but when I switch to a gstreamer pipeline, it seems to get stuck at 44100hz audio, even though the source is 48000hz. The result is that the audio is pitch-shifted down making voices sound like they’re all sucking on some sulfur hexafluoride. Does anyone know how to get alsasrc to override the rate?
Capturing the audio to a file using arecord looks like this:
arecord -f S16_LE -c 2 -r 48000 -d 5 -D hw:1,0 test.wav
My pipeline looks like this:
gst-launch-1.0 alsasrc ! 'audio/x-raw, format=S16LE, rate=48000, channels=2' ! autoaudiosink
It doesn’t seem to matter what I put into the “rate” field. I would have expected putting bad rates in there to pitch-up or pitch-down the audio. I have tried to force the rate up & down using “audiorate” in the pipeline, and this does start to sound correct… except that it starts to get static then dies.
Anyone have any ideas?