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@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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