<newbie alert, be gentle> I'm using the latest (from gitlab) linphone-sdk on my nanopi neo air ARM board, and when a call is established (to another sip phone connected to my asterisk server), I get very poor audio performance, impossible to understand. Trying the same with linphonec, same results.
I see this: 2021-02-01 18:30:35:070 mediastreamer-error-snd_pcm_avail_update: Broken pipe 2021-02-01 18:30:35:074 mediastreamer-error-*** alsa_can_read fixup, trying to recover I have restricted codecs to ulaw/pcmu and alaw/pcma. aplay and arecord both work fine, e.g. like this: arecord -d30 -f S16_LE -c1 -r8000 attempt47.wav nano3:/home/per # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Codec [H3 Audio Codec], device 0: CDC PCM Codec-0 [CDC PCM Codec-0] Subdevices: 1/1 Subdevice #0: subdevice #0 I have already tried altering ALSA_PERIOD_SIZE in mediastreamer2/src/audiofilters/alsa.c but it made no difference. I have built the sdk with these options: -DENABLE_LIME=no -DENABLE_OPUS=no -DENABLE_VIDEO=no -DENABLE_JPEG=no -DENABLE_SOUND=yes -DENABLE_BV16=no -DENABLE_ZLIB=no -DENABLE_LIME_X3DH=no -DENABLE_SPEEX=no -DENABLE_ADVANCED_IM=no -DENABLE_VCARD=no Any hints or suggestions? </newbie alert> -- Per Jessen, Zürich (6.5°C) _______________________________________________ Linphone-users mailing list [email protected] https://lists.nongnu.org/mailman/listinfo/linphone-users
