Le 16/05/2012 04:09, Benny Prijono a ?crit : > Thanks Dan! I created https://trac.pjsip.org/repos/ticket/1510 for > this, I will integrate the patch as soon as possible. > > Best regards, > Benny > > On Tue, May 15, 2012 at 12:07 AM, Igor Kravets <megaboots at mail.ru > <mailto:megaboots at mail.ru>> wrote: > > Thanks Dan! The problem seems to be resolved! > The sound card is shared between pjsua and other applications and > works fine!!! > And even without asound.conf file and all those creepy asym/dmix > magic! > > I'll check everything more carefully in the evening, i'm too > excited now!!! Thanks! :))) > > > Mon, 14 May 2012 09:36:23 +0200 ?? Dan Arrhenius <dan at keystream.se > <mailto:dan at keystream.se>>: > > 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 <http://192.168.4.190:5060> > > > 20:22:01.252 udp0x81f09a0 SIP UDP transport started, published > > > address is 192.168.4.190:5060 <http://192.168.4.190:5060> > > > 20:22:01.252 pjsua_acc.c Adding account: > > > id=<sip:192.168.4.190:5060 <http://192.168.4.190:5060>> > > > 20:22:01.252 pjsua_acc.c .Account <sip:192.168.4.190:5060 > <http://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 <http://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 <http://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 <mailto:pjsip at lists.pjsip.org> > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > This body part will be downloaded on demand. I'm sorry but how do you apply this patch ? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120622/77b7d71e/attachment-0001.html>
