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Re: [Linphone-developers] MediaStreamer/Alsa misconfiguration on an Imx6


From: Thomas Dalmayrac
Subject: Re: [Linphone-developers] MediaStreamer/Alsa misconfiguration on an Imx6 based product
Date: Wed, 7 Feb 2018 10:37:23 +0100
User-agent: Mozilla/5.0 (Windows NT 10.0; WOW64; rv:52.0) Gecko/20100101 Thunderbird/52.6.0

Hi, news from my bug :

I found that when I play with ALSA_PERIODS and ALSA_PERIOD_SIZE in alsa.c it changes the output of my sound. The sound is still bad but I clearly see that it changes the frequency of the stutter/distortion.

So my bug is clearly situated in mediastreamer. When I launch linphone (verbose) I can see that log :

Strange, sound card aic31xx-audio does not seems to be capable of anything, retrying with plughw...

And in the code there is a comment saying that mediastreamer trys to use plughw to workaround an alsa bug. So my question is simple : What can I do with my alsa driver to avoid this comment ?

What do you think of OSS ?

Thanks !


Le 01/02/2018 à 12:10, Thomas Dalmayrac a écrit :

Hey !

I discussed about my trouble on chat.freenode.net #linphone yesturday and I had many ideas from there. One of the ideas is to revert the mediastreamer library, what do you think of that ?

Thomas.


Le 30/01/2018 à 15:55, Thomas Dalmayrac a écrit :

Hi everybody !

I'm working on a on an i.MX6 (freescale) based product under a buildroot personnalized linux, and I try to use linphonec with it, but I've got troubles when I'm in call (voice is hatched). See the record I've made from the call : https://vocaroo.com/i/s1RrgXBSXIaT. The wireshark : https://www.cloudshark.org/captures/9fd82b8b4e60

To test and isolate the issue of hatched voice I played an example sound from an asterisk server - I correctly establish a call but every ~35sec the voice is hatched for 5sec. The codec used for the call is PCMU, and the soundcard is an I2S with a codec TI TLV320AIC3110 (http://www.ti.com/product/TLV320AIC3110).

  • The sound is correctly received by the softphone. We put linphone in file mode (without sound card - "soundcard use files" "record") to record the audio received by the phone before it is sent to the lower layers. It is about the 5th second that we hear a fault on the loudspeakers (like the sound given in intro see https://vocaroo.com/i/s1RrgXBSXIaT), while on the recorded sample, we do not hear any defect .
  • The disturbance occurs every 35 seconds or so.
  • The disturbance is insensitive to the CPU load (tested with stress), or the network load (tested with iperf)
  • The sound is correctly transported by the network, there are no losses and if we sniff the packets received by the handset we can play the sound without worry via wireshark (https://www.cloudshark.org/captures/9fd82b8b4e60)
  • The codecs are correctly managed (we will use the PCMU G711-Ulaw, tested G722 and PCMA ok)
  • The same example sound is correctly decoded / played by aplay if we download it.

Perhaps an important fact : Everytime I start linphonec I got these messages :

ALSA lib conf.c:4738:(snd_config_expand) Unknown parameters 0
ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL default:0
ALSA lib conf.c:4738:(snd_config_expand) Unknown parameters 0
ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM default:0
ALSA lib conf.c:4738:(snd_config_expand) Unknown parameters 0
ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM default:0
ALSA lib conf.c:4738:(snd_config_expand) Unknown parameters 1
ALSA lib control.c:954:(snd_ctl_open_noupdate) Invalid CTL default:1
ALSA lib conf.c:4738:(snd_config_expand) Unknown parameters 1
ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM default:1
ALSA lib conf.c:4738:(snd_config_expand) Unknown parameters 1
ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM default:1

We think that the concern comes from the lower layers used by the linphone. Can there be incompatibilities with MediaStreamer2? Does all this bring up something that could help us solve our problem?

Thank you for every single idea you'll give to us :)

--
Thomas Dalmayrac


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