Hi, It's not *my* library and I'm also relatively new actually :) Yes the log tells that there is still resampling involved. It may be the conference bridge that works in 16 kHz (default clock rate). Please try pjsua with the following param: --null-audio --clock-rate 8000 or apply it directly to the pjsua media config setting if you build your own application using pjsua-lib. Regards, nanang On Mon, Sep 8, 2008 at 12:32 PM, Fr??d??ric CLEMENT <fclement at viatelecom.com> wrote: > Hi Nanang, > > As I am a very new user of your library, I only use the high-level API > (pjsua). > Would you have some sample code using the methods you describe ? > > I can see following in the logs : > > 06:42:44.886 resample.c resample created: high qualiy, large filter, > in/out rate=8000/16000 > 06:42:44.886 resample.c resample created: high qualiy, large filter, > in/out rate=16000/8000 > > My Wav file has following format : > > Input Filename : Acc.wav > Sample Size : 16-bits > Sample Encoding: signed (2's complement) > Channels : 1 > Sample Rate : 8000 > > What does it mean ? > > Regards, > > Frederic > > CLEMENT Fr?d?ric a ?crit : > > HI, > > Thanks very much. > I will test this tomorrow. > > Regards, > > Frederic > > On Thu, 4 Sep 2008 01:59:08 +0700, "Nanang Izzuddin" <nanang at pjsip.org> > wrote: > > > Hi, > > Sorry, it should be master port, one of pjmedia clock generator, here > is the doc: > http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__MASTER__PORT.htm > > There is no default clock rate for wav player, wav player clock rate > follows the wav file clock rate. So if you are using G711, please use > 8000 Hz wav file to avoid resampling. > > Btw, replacing wav player with mem player may increase the > performance, not sure though. > > Regards, > nanang > > > On 04/09/2008, Fr??d??ric CLEMENT <fclement at viatelecom.com> wrote: > > > /* > > > - use master clock (not sound device) for the clock source, since > sound > > > device may be disturbed by high CPU load and CPU spikes. > > > - use clock rate > > > 8000 Hz for the master clock (or conf bridge, if you > > > use it) to avoid > > > resampling. > */ > > Thanks very much for this complete answer. > > I didn't understand what you mean with 'use master clock'. > I don't user any hardware but only your wav_player methods. > The clock rate is 8000 Hz by default, isn't it ? > > > Regards, > > Frederic > > > Nanang Izzuddin a ?crit : > Hi, > > > G711 is very low complexity codec > > > (see > > > http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS, stream > > > with G711 in P3 > > > 700MHz only cost < 0.1% CPU), so the codec shouldn't be the > > > problem. > > > Also your CPU utilization seems to be relatively low (although > > > it > > > supposed to be lower than 30%). So it's not the CPU limitation. > > Please > > > see this, in case you > haven't: > > > https://trac.pjsip.org/repos/wiki/FAQ#high-perf. > And > > > also please try this, in case you haven't: > > > - use master clock (not sound > > > device) for the clock source, since > > > sound device may be disturbed by high > > > CPU load and CPU spikes. > > > - use clock rate 8000 Hz for the master clock (or > > > conf bridge, if you > > > use it) to avoid resampling. > > Btw, is there any chance > > > that the problem is in the caller/tester > > > (e.g: the app tester can only > > > handle 20 concurrent calls, so you > > > should run 2 instances of app tester on 2 > > > different machines)? > > > Regards, > nanang > > > On 03/09/2008, Fr??d??ric CLEMENT > > > <fclement at viatelecom.com> wrote: > > > My Load average is about 1.00 (34 % CPU utilization). > > > The codec I am using > > > is Alaw (g711). > > > Can I expect to handle 100 calls with a Core2Duo or QuadCore > > > Xeon ? > > > Should I user another codec ? My switch have many implemented. > > > (Dialogic > > > Cantata IMG). > > Regards, > > Frederic > > > Nanang Izzuddin a ?crit : > Hi, > > > Which codec are you using? How about the CPU utilization? Since > > > it's > > > may be caused by CPU limitation in doing media processing > > > (especially > > > encoding). > > > Regards, > nanang > > > On 03/09/2008, Fr??d??ric CLEMENT > > > > <fclement at viatelecom.com> wrote: > > > Hi again all, > > > I am still building an IVR server and testing the max > > > possible account of > > > calls that can be handled by pjsip. > > > I set > > > > PJSUA_MAX_CALLS to 100 > > > The value inf config_site.h for > > > PJMEDIA_SOUND_BUFFER_COUNT is set to 16. > > > My application just take calls and > > > play a sound file looped. I use the pjsua > > > high level methods > > > (pjsua_player_create ....) > > > The machine is a Pentium IV 1.9 Ghz with enough > > > memory. > > > When I send 20 calls, the sound stays being correct. > > > If I send 30 > > > > simultaneous calls, I get the same result than described on > > > your site > > > : > > > http://trac.pjsip.org/repos/wiki/audio-problem-dropouts > > > The > > > sound I have is the same than the stutter.wav found > > > there. > > > > http://trac.pjsip.org/repos/attachment/wiki/audio-problem-dropouts/stutter.wav > > > Wich > > > optimisations can be done ? My goal is about 100 simultaneous > > > calls. > > > > Regards, > > > Frederic > > (Sorry again for my very bad > > > > english) > > > _______________________________________________ > > > 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 > > > > > > _______________________________________________ > > > 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 > > > _______________________________________________ > > > 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 > > > > > > _______________________________________________ > > > 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 > > > _______________________________________________ > 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 > > > > > > _______________________________________________ > 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 > > > _______________________________________________ > 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 > > > _______________________________________________ > 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 > > -- Regards, nanang
