[Freeswitch-users] Portaudio Improvements

[João Mesquita]

Mitch, I will work on continuing FSComm for a thick client multi-platform
solution but one of the big show stoppers for me was the lack of AEC on
mod_pa.

I tried the preprocessors embedded on the core by Tony that are using the
speexdsp but I got no luck. The main tests I made were on a mac using the
speakerphone. Do you have any experience with this type of technology? It
seems to me that the only AEC available is the one implemented on speexdsp
but I am completely new to this.

Regards,
João Mesquita


On Fri, Jan 21, 2011 at 2:00 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> I have submitted a ticket with a patch for portaudio and my improvements.
> This includes the improvements I had discussed previously during my call for
> input on the mailing list and is available at:
> http://jira.freeswitch.org/browse/FS-3006
>
> It was only tested in windows, however most of the changes should not
> effect default behavior (but please test if you can).
>
> To test you will want to enable some things in the portaudio config
> including:
>                     <param name="live-stream-switch" value="true" />
>                     <param name="no-auto-resume-call" value="true" />
>                     <param name="no-ring-during-call" value="true" />
>
> you will then be able to use the new features fully.
>
> Many of these changes were made to add better support for softphone's using
> freeswitch, I will be releasing my embedded freeswitch phone (FSClient) for
> the conference call in a week and a half  it is .net/c# but WPF so sadly
> windows only currently.  jlink and drk are helping so we should end up with
> a nice installer also for it.     If you would like to test it please let me
> know.
>
> ~Mitch
>
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