Hi Salahuddin, Thanks so much for the input. I think calling PlayCb() after RecCb() may be a good alternative for CPU idle/normal condition, since in this condition it seems there is no clock drift/skew between mic & spk (at least in E65) so the delay should be kept constant and no breaking of underflow. However I experienced temporary clock drift when there is a CPU load spike (like printing log in console, in a pjsua call using speex) and it took few seconds to recover the synchronized clock rate (clock drift symptom is analyzed by observing delay buffer's log messages, i.e: periodically generating/reducing samples), this temporary drift may result either underflow/breaking or the delay increasing badly (I read somewhere that the APS message queue size is 100, so the speaker delay can reach max 100 frames == 2 seconds for 20ms ptime). Actually I was looking for discussion topics or articles about where/when to call PlayCb() 'correctly' in APS, but still no luck. After a few experiments (I've also desperately tried PlayCb controlled by RTimer), I decided to put the PlayCb() there (in the EPlayCommQueue) since the voice result was good and it seems to be safe when clock drift occurs. However, your case is still a mystery why the voice was breaking ONLY when the mic receive no sound/silence. Perhaps you can analyze on the frame size and type (the first 2 bytes) from mic when silence, or how many times the RecCb and PlayCb was called in silence (they should be same or a small diff). Regards, nanang 2008/8/1 Salahuddin Ahmed <bd.rubel at gmail.com>: > Hi Nanang, > Thanks For your reply. I am sorry for late reply. Actually I am not > so much matured in codec area. > Breaking is not 10 times in a second. it is 1/2 times in a second but > the length of breaking duration is more. > I make loopback(mic->spk) test this is alright(getting no break). I > make another wild test :) I call play_cb after recb_cb and comment > play_cb from EPlayCommQueue. This make little bit delay but the > breaking reduce much more. I guess it is not the perfect possition to > call play_cb. Another thing When I make continuous sound in mic then I > didnt get any break in play stream. > > My and ur Environment are same. I use E65, 3rd Ed MR SDK, APS 2.43. > > Thanks, > Salahuddin > > On Thu, Jul 31, 2008 at 8:04 PM, Nanang Izzuddin <nanang at pjsip.org> wrote: >> Hi Salahuddin, >> >> How frequent is the breaking noise (e.g: about ten times in a second)? >> Or completely noise? >> >> It can be the frames from mic are normal G.711 frames when not >> silence, but it returns non-normal frames (e.g: CNG frames or perhaps >> lower rates) when silence (non-normal frames are not handled by now). >> However this isn't supposed to happen since VAD and CNG are disabled. >> Could you specify your environment (e.g: application, device, SDK >> version, APS version)? Was that in a call or looped back mic->spk? I >> will try to build a 'similar' environment if possible and start >> 'debugging' around, since the problem doesn't occur in my current >> environment (symsndtest/symbian_ua/symbian_ua_gui, E65, S60 3rd ed MR >> SDK, APS 2.43). >> >> To be honest, I don't have a lot of experience with APS and know >> almost nothing on APS issues/tricks. So opinions or contributions on >> the APS integration are greatly appreciated. >> >> Regards, >> nanang >> >> >> 2008/7/31 Salahuddin Ahmed <bd.rubel at gmail.com>: >>> Hello Nanang, >>> >>> Firstly so many thanks for your APS integration. I can successfully >>> install and use it But I got an problem... The play is not continuous. >>> It contain so many breaks if I dont say anything in mic. If I make >>> some sound in mic then the play will continuous. I can't understand >>> what is problem. >>> >>> thanks >>> >>> On Wed, Jul 30, 2008 at 7:13 PM, Nanang Izzuddin <nanang at pjsip.org> wrote: >>>> Hi Karthik, >>>> >>>> Was the problem in running or installation? Any error message/code >>>> issued? The APS code is actually experimental, however it seems to >>>> work smoothly on E65. Please make sure you have the APS server >>>> installed on your device and use the correct target device of SDK API >>>> Plug-In. Please also see http://trac.pjsip.org/repos/wiki/APS. >>>> Feedbacks on this are very welcomed. >>>> >>>> Btw, could you share some hints regarding Audio Routing API >>>> integration here? May it be useful for the others. >>>> >>>> Thanks & regards, >>>> nanang >>>> >>>> >>>> 2008/7/30 Karthik Babu <cytrion at gmail.com>: >>>>> Hello Nannang , >>>>> >>>>> I am able to use the Audio Routing API successfully on my n95 , but this is >>>>> limited to fp1 and fp2 phones . >>>>> >>>>> So I planned to use APS (for non fp1 and fp2 devices) and I did notice >>>>> that the APS is available in the current svn trunk . I encounetered few >>>>> issues with this but could finally make a build. But the exe fails on my N95 >>>>> . >>>>> >>>>> Can you please advice ? >>>>> >>>>> >>>>> >>>>> -- >>>>> Karthik >>>>> >>>>> _______________________________________________ >>>>> 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 >>>> >>> >>> >>> >>> -- >>> Salahuddin Ahmed >>> Software Engineer >>> Genuity Systems Ltd. >>> www.genuitysystems.com >>> Tel. 88-02-8057038-9, 88-02-8079997 >>> Sip address: sip:86233 at iptel.org >>> Skype : bdrubel >>> LinkedIn: www.linkedin.com/in/salahuddin >>> >>> _______________________________________________ >>> 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 >> > > > > -- > Salahuddin Ahmed > Software Engineer > Genuity Systems Ltd. > www.genuitysystems.com > Tel. 88-02-8057038-9, 88-02-8079997 > Sip address: sip:86233 at iptel.org > Skype : bdrubel > LinkedIn: www.linkedin.com/in/salahuddin > > _______________________________________________ > 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 >
