Hi, the problem with the assertion error when using alsa is caused by a bug in alsa_dev.c. I'm attaching a patch that solves the issue. Regards, Dan On Thu, 2012-05-10 at 21:04 +0400, Igor Kravets wrote: > It is 2.x release active now and the problem with "ALSA native > support" in pjsua application still remains. > > While auddemo application (compiled with ALSA native support) works > great both recording and playing sound through > my aliased "duplex" alsa device (see asound.conf below) simultaneosly > with other application > > Pjsua with same compilation option DOES NOT WORK, breaking with: > >>> pjsua8: ../src/pjmedia/conference.c:1795: get_frame: Assertion > `frame->size == conf->samples_per_frame * conf->bits_per_sample / 8' > failed. > > > It is worth noting that pjsua built with portaudio DOES WORK, BUT > works unstable and sound stops playing after 10-30 seconds of > playback! > Using latest external libportaudio works a little bit more stable and > sound plays a little bit longer and then also stops. > > > A guy called Damir also had that problem and it seems that he could > fix things up by asound.conf magic. > That solution doesn't work for my hardware. I've played a lot > with asoundrc options and aliases. Nothing helps. > > here's Damirs post: > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2011-February/012591.html > > > Some other guys recommend me using external sound mixing server, which > doesn't suite my task very well: > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2010-April/010721.html > > > Problem exits. Noone yet had offered a fix, only workarounds. Maybe > guys that created alsa_dev.c can shed some light on the problem? > I really need help with this :( > > The only thing that works with ALSA and shared soundcard is still > pjsua version 1.0.3. > It also trends to just stop playing sound but really really rare > (twice a week). > > > == Here's how pjsua crashes after local echotest (cc 0 0): == > > TC421:~# ./pjsua8 > 20:22:01.147 os_core_unix.c !pjlib 2.0.0-beta for POSIX initialized > 20:22:01.148 sip_endpoint.c .Creating endpoint instance... > 20:22:01.151 pjlib .select() I/O Queue created (0x81bae18) > 20:22:01.151 sip_endpoint.c .Module "mod-msg-print" registered > 20:22:01.151 sip_transport. .Transport manager created. > 20:22:01.151 pjsua_core.c .PJSUA state changed: NULL --> CREATED > 20:22:01.151 sip_endpoint.c .Module "mod-pjsua-log" registered > 20:22:01.151 sip_endpoint.c .Module "mod-tsx-layer" registered > 20:22:01.151 sip_endpoint.c .Module "mod-stateful-util" registered > 20:22:01.151 sip_endpoint.c .Module "mod-ua" registered > 20:22:01.151 sip_endpoint.c .Module "mod-100rel" registered > 20:22:01.151 sip_endpoint.c .Module "mod-pjsua" registered > 20:22:01.151 sip_endpoint.c .Module "mod-invite" registered > 20:22:01.213 alsa_dev.c ..ALSA driver found 8 devices > 20:22:01.213 alsa_dev.c ..ALSA initialized > 20:22:01.213 pjlib ..select() I/O Queue created (0x81e94bc) > 20:22:01.251 sip_endpoint.c .Module "mod-evsub" registered > 20:22:01.251 sip_endpoint.c .Module "mod-presence" registered > 20:22:01.251 sip_endpoint.c .Module "mod-mwi" registered > 20:22:01.251 sip_endpoint.c .Module "mod-refer" registered > 20:22:01.252 sip_endpoint.c .Module "mod-pjsua-pres" registered > 20:22:01.252 sip_endpoint.c .Module "mod-pjsua-im" registered > 20:22:01.252 sip_endpoint.c .Module "mod-pjsua-options" registered > 20:22:01.252 pjsua_core.c .1 SIP worker threads created > 20:22:01.252 pjsua_core.c .pjsua version 2.0.0-beta for > Linux-2.6.32.9/i686/glibc-2.9 initialized > 20:22:01.252 pjsua_core.c .PJSUA state changed: CREATED --> INIT > 20:22:01.252 sip_endpoint.c Module "mod-default-handler" registered > 20:22:01.252 pjsua_core.c SIP UDP socket reachable at > 192.168.4.190:5060 > 20:22:01.252 udp0x81f09a0 SIP UDP transport started, published > address is 192.168.4.190:5060 > 20:22:01.252 pjsua_acc.c Adding account: > id=<sip:192.168.4.190:5060> > 20:22:01.252 pjsua_acc.c .Account <sip:192.168.4.190:5060> added > with id 0 > 20:22:01.252 pjsua_acc.c Acc 0: setting online status to 1.. > 20:22:01.253 tcplis:5060 SIP TCP listener ready for incoming > connections at 192.168.4.190:5060 > 20:22:01.253 pjsua_acc.c Adding account: > id=<sip:192.168.4.190:5060;transport=TCP> > 20:22:01.253 pjsua_acc.c .Account > <sip:192.168.4.190:5060;transport=TCP> added with id 1 > 20:22:01.253 pjsua_acc.c Acc 1: setting online status to 1.. > 20:22:01.253 pjsua_core.c PJSUA state changed: INIT --> STARTING > 20:22:01.253 pjsua_media.c ..NAT type detection failed: Invalid STUN > server or server not configured (PJNATH_ESTUNINSERVER) > 20:22:01.253 sip_endpoint.c .Module "mod-unsolicited-mwi" registered > 20:22:01.253 pjsua_core.c .PJSUA state changed: STARTING --> > RUNNING > >>>> > Account list: > [ 0] <sip:192.168.4.190:5060>: does not register > Online status: Online > *[ 1] <sip:192.168.4.190:5060;transport=TCP>: does not register > Online status: Online > Buddy list: > -none- > > +=============================================================================+ > | Call Commands: | Buddy, IM & Presence: | > Account: | > | | | > | > | m Make new call | +b Add new buddy .| +a Add > new accnt | > | M Make multiple calls | -b Delete buddy | -a Delete > accnt. | > | a Answer call | i Send IM | !a Modify > accnt. | > | h Hangup call (ha=all) | s Subscribe presence | rr > (Re-)register | > | H Hold call | u Unsubscribe presence | ru > Unregister | > | v re-inVite (release hold) | t ToGgle Online status | > Cycle > next ac.| > | U send UPDATE | T Set online status | < Cycle > prev ac.| > | ],[ Select next/prev call > +--------------------------+-------------------+ > | x Xfer call | Media Commands: | Status & > Config: | > | X Xfer with Replaces | | > | > | # Send RFC 2833 DTMF | cl List ports | d Dump > status | > | * Send DTMF with INFO | cc Connect port | dd Dump > detailed | > | dq Dump curr. call quality | cd Disconnect port | dc Dump > config | > | | V Adjust audio Volume | f Save > config | > | S Send arbitrary REQUEST | Cp Codec priorities | > | > +-----------------------------------------------------------------------------+ > | q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT > type | > +=============================================================================+ > You have 0 active call > >>> cc 0 0 > 20:22:08.552 pjsua_media.c Conf connect: 0 --> 0 > 20:22:08.552 pjsua_media.c .Set sound device: capture=-1, > playback=-2 > 20:22:08.552 pjsua_app.c ..Turning sound device ON > 20:22:08.552 pjsua_media.c ..Opening sound device PCM at 16000/1/20ms > 20:22:08.565 ec0x81c4070 ...AEC created, clock_rate=16000, > channel=1, samples per frame=320, tail length=200 ms, latency=99 ms > 20:22:08.565 conference.c .Port 0 (duplex) transmitting to port 0 > (duplex) > Success > >>> pjsua8: ../src/pjmedia/conference.c:1795: get_frame: Assertion > `frame->size == conf->samples_per_frame * conf->bits_per_sample / 8' > failed. > > > > > > == my asound.conf == > > pcm.dmixer { > type dmix > ipc_key 1024 > slave { > pcm "hw:0,0" > period_time 0 > rate 44100 > period_size 1024 > buffer_size 8192 > } > bindings { > 0 0 > 1 1 > } > } > > > pcm.conv_dmixer { > type plug > slave.pcm "dmixer" > } > > pcm.conv_record { > type plug > slave.pcm "hw:0,0" > } > > pcm.duplex { > type asym > playback.pcm "conv_dmixer" > capture.pcm "conv_record" > > hint { > show on > description "Duplex device" > } > > } > > pcm.!default { > type plug > slave.pcm "duplex" > } > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- A non-text attachment was scrubbed... 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