Hi,
Not really sure what was going on. Please try to put the code of
changing output route *after* connecting wav player to the sound
device, just to make sure that the pjsua_snd_set_setting() is executed
after the audio device is opened.
BR,
nanang
On Mon, Apr 19, 2010 at 3:44 AM, Ricardo <ricardo2500 at hotmail.com> wrote:
> Nanang Izzuddin <nanang <at> pjsip.org> writes:
>
>>
>> Hi,
>>
>> I have no idea about CMdaAudioPlayerUtility API. Just FYI, pjsua
>> sample app implements ringtone by creating wave player port and
>> register it to the conference bridge (or audio switch when
>> APS/VAS-Direct is used). Connecting this port to port 0 will play the
>> ringtone, disconnecting it will stop the ringtone.
>>
>> BR,
>> nanang
>>
>> On Thu, Feb 25, 2010 at 5:15 AM, Ricardo Antonio Ponce Gomez
>> <ricardo2500 <at> hotmail.com> wrote:
>> > Hi all.
>> > I'm trying to play a ring tone when an incomming call is there. I'm using
>> > the CMdaAudioPlayerUtility API and it just work fine the first time,?i can
>> > hear a .wav audio file in the loudspeaker, but after I finish with this
>> > call, and there's a second incomming call my ring tone it's gone. I'm dong
>> > something like this:
>> > _LIT(tono,"ring.wav");
>> > CPlayer *player = CPlayer::NewL(tono);
>> > //on_incoming_call callback
>> > ....
>> > pjsua_call_answer(call_id, 180, NULL, NULL);
>> > player->Play();
>> > ....
>> > //on_call_state callback
>> > ......
>> > if (ci.state == PJSIP_INV_STATE_DISCONNECTED) {
>> > if (call_id == g_call_id){
>> > ?? ? ? ?g_call_id = -1;
>> > ?? ? ? ?player->Stop();
>> > ?? ? }
>> > .....
>> > Do you think i'm missing something? Is there any other better option to do
>> > this?
>> > I'm using VAS.
>> > Thanks & Regards
>> > ________________________________
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>> >
>> >
>>
>>
> Hi,
> Thanks Nanang for your help. Now I'm facing a similar problem creating the wav
> player in this way:
>
> pjsua_player_id player_id;
>
> in on_incoming_call callback.
> ....
> ....
> pj_str_t awav = pj_str ("C:\\Data\\ring.wav");
> ? ?status =pjsua_player_create(&awav,0,&player_id);
> if (status != PJ_SUCCESS)
> ? ?console->Printf(_L("\n->error creando player\n"));
> status=pjsua_conf_connect(pjsua_player_get_conf_port(player_id),0);
>
>
> And then I disconnect the conference in on_call_state and in on_call_media_state
> callbacks.
>
> With this code I can hear the ringtone when I get an incomming call, the problem
> is that when the call is finished the next time I get an incomming call my
> ringtone is played in the earpiece instead of the Loudspeaker. Then, if ?I hang
> up the call before answer it, the next time I have an incomming call the
> Ringtone is played in the Loudspeaker again.
>
> So I tried adding this before I connect the wav player to the conference:
>
> pjmedia_aud_dev_route route;
> route = PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER;
> status =
> pjsua_snd_set_setting(PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE,&route,PJ_FALSE);
> if (status != PJ_SUCCESS)
> ? ?console->Printf(_L("\nError cambiando a altavoz\n"));
>
> But I allways get the same result. Do you have any idea about what could be my
> problem?
>
> Thanks in advance for any help.
>
>
>
>
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