symbian port

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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
>>>>
>>>> _______________________________________________
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>>>>
>>>>
>>>
>>> _______________________________________________
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>>>
>>
>>
>>
>> --
>> 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



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