Hello Simon,
I had a very similar (perhaps the same) issue when using PJSIP on
my Raspberry Pi.In my case, the problem was: I was using ALSA
directly (no PulseAudio) and didn't have a .asoundrc file in my
home directory with "plug" devices defined (type plug). You need
to define these plug devices as default ALSA devices - and I think
the exact reason was that PJSIP tries to play audio in mono mode
(1 channel), but only 2 channels were natively supported by my
hardware. PJSIP failed to notice this and streamed the audio in
mono mode nonetheless, which resulted in that garbled audio
output. I think there was also some way to tell PJSIP to use
stereo mode for playback and that also made things work again-
even without the .asoundrc. The plug devices in .asoundrc solve
the issue, since they will convert the audio output from mono to
stereo if required. This also solved a couple of other issues I
had when using other apps that were using audio. I have attached
my .asoundrc.
Best regards,
Christian
Hello,
I am currently fighting with static noise whenever call is trying to be started. I think its due to clockrate, but I am unable to fix it.
Its not just the call issue, as when I try to go with the list of troubleshooting audio, the issue also appears when I try to playback a WAV file. (there is a noise and then there is the WAV file in the background being played).
Here is the full log of the ./playfile sample: https://paste.brcb.eu/wajyxurele.apache. When I tried the sipecho or recording a file. It all worked fine. But just to note, when the recording was running, it reported a different sample rate: amp.rate=44100 (the playback showed amp.rate=22050). Here is the full log of recording: https://paste.brcb.eu/herexyxixa.apache).
I also read (here: https://docs.pjsip.org/en/latest/specific-guides/audio-troubleshooting/checks/problematic_clock_rate.html) that it can be an issue with clock rate. But on the doc page, there is no way how to fix it. So I want to kindly ask for your help, how to get this working.
My setup is: Raspberry Pi Zero 2W with ReSpeaker 2mics hat running Raspi OS (legacy buster based) with PJSIP 2.10 (i know, older version, but specified by my school, cant change that).
Also, my main usage is a custom script using PJSIP as a doorbell, so can you please help me to get the settings right at least when using my script? Here it is: ttps://paste.brcb.eu/yrygofukud.md. (there is also a modified call.py file: https://paste.brcb.eu/gywequhaqu.rb)
Thank you!
Kind regards
Simon Cechacek
_______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list -- pjsip@xxxxxxxxxxxxxxx To unsubscribe send an email to pjsip-leave@xxxxxxxxxxxxxxx
pcm.cirrus {
type hw
card 0
device 0
}
pcm.myhdmi {
type hw
card 1
device 0
}
pcm.cirrusmixer {
type dmix
ipc_key 800
ipc_perm 0666 # allow other users
ipc_key_add_uid true # for testing; see https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
slave {
pcm "hw:0,0"
period_time 0
period_size 1024
buffer_size 4096
rate 48000
# channels 2
}
bindings {
0 0
1 1
}
}
pcm.cirrusmic {
type dsnoop
ipc_key 801
ipc_perm 0666 # allow other users
ipc_key_add_uid true # for testing; see https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
slave {
pcm "hw:0,0"
period_time 0
period_size 1024
buffer_size 4096
rate 48000
channels 2
}
bindings {
0 0
1 1
}
}
#pcm.!default {
# type plug
# slave.pcm {
# @func getenv
# vars [ DEFAULT_PCM_DEVICE ]
# default "cirrusmixer"
# }
#}
pcm.!default {
type asym
capture.pcm {
type plug
slave.pcm {
@func getenv
vars [ DEFAULT_CAPTURE_DEVICE ]
default "cirrusmic"
}
}
playback.pcm {
type plug
slave.pcm {
@func getenv
vars [ DEFAULT_PCM_DEVICE ]
default "cirrusmixer"
}
}
}
ctl.!default {
type hw
card 0
}
pcm.dsp0 {
type plug
slave.pcm "cirrus"
}
ctl.mixer0 {
type hw
card 0
}
_______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list -- pjsip@xxxxxxxxxxxxxxx To unsubscribe send an email to pjsip-leave@xxxxxxxxxxxxxxx
