Re: [Linphone-developers] Problem when call from an IP phone to linphonec

[Simon Morlat]

Hi,

Thank you very much for reporting this problem and the solution !
I look at the code and it seems it only produces when using the OSS output, 
not ALSA, that 's why most users (and me) didn't see it before you.
I've applied your patch. Unfortunately it is not in 1.5.1 (released yesterday) 
but I'll upload a 1.5.2 soon.

Simon

Le mardi 14 novembre 2006 09:11, Roy Huang a écrit :
> Hi,
>
> I have a Polycom IP phone, model SoundPoint IP 300SIP. I download
> linphone 1.5.0 to my PC, compile it without gnome and install it. Then
> I start linphonec.
>
> If call from linphonec to IP phone, it works fine, If call from IP
> phone to linphonec, the sounds are terrible. After study log file, I
> find the root cause: When IP phone call linphonec, linphonec play ring
> file /usr/share/sounds/linphone/rings/oldphone.wav. This is a stereo
> file, so Ringstream set sound card in stereo mode. After answer to
> accept the call, linphonec will start audio stream, but it don't reset
> the mode and it still keep stereo mode setting. But audio stream only
> offer mono audio to sound.
>
> I confirm that by remove rings/oldphone.wav or change oldphone.wav to
> a mono file. The call will become normal.
>
> The following patch seems also fix this problem. Please have a look.
>
> diff -ur linphone-1.5.0.orig/mediastreamer2/src/audiostream.c
> linphone-1.5.0/mediastreamer2/src/audiostream.c
> --- linphone-1.5.0.orig/mediastreamer2/src/audiostream.c      2006-10-04
> 18:01:11.000000000 +0800
> +++ linphone-1.5.0/mediastreamer2/src/audiostream.c   2006-11-14
> 15:58:28.000000000 +0800
> @@ -221,7 +221,12 @@
>       /* give the sound filters some properties */
> 
>       
> ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&pt->clo
>ck_rate);
> ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&pt->clo
>ck_rate); -
> +     if (outfile==NULL) {
> +             int tmp = 1;
> +
> +             
> ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_NCHANNELS, &tmp);
> +     }
> +
>       /* give the encoder/decoder some parameters*/
> 
>       
> ms_filter_call_method(stream->encoder,MS_FILTER_SET_SAMPLE_RATE,&pt->clock
>_rate);
> ms_filter_call_method(stream->encoder,MS_FILTER_SET_BITRATE,&pt->normal_bit
>rate);
>
>
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