On 2/14/08, Keith Kyzivat <[EMAIL PROTECTED]> wrote: > On Thu, Feb 14, 2008 at 7:45 AM, Karsten Schubotz <[EMAIL PROTECTED]> wrote: >> The main reason why I would like to use per-call volume is to prioritise >> calls during a conference call. >> This is something that would be done inside the bridge of the media layer.. >> (MprBridge). > > While I'm pretty sure prioritization (who is the active speaker) is present > in the current bridge,
Sadly there is no such ability in current code. So, you have to implement it by yourself. To do this you have to look at sipXmediaLib level and modify flowgraph behaviour - create Voice Activity Detection resource (MpResource child), put it before Bridge on every incoming connection, and then somehow convert this information into Bridge mix weights. > not sure if code to do gain adjustments during the > mixing of the different call streams is present, and I'm quite sure that > hooks up to the mediaAdapter layer and sipXtack lib layers are present. > > Take a look at MprBridge in the sipXmediaLib project -- > svn co http://scm.sipfoundry.com/rep/sipXtapi > edit sipXtapi/sipXmediaLib/include/mp/MprBridge.h > sipXtapi/sipXmediaLib/src/mp/MprBridge.cpp > >> Mono data would be OK, but I would like to put the mono audio data either >> to only one (left or right) or both of my two speakers. So If I have several >> calls active in a conference call, I want to decide which call to put on >> which speaker. Perhaps I have the opportunity to use the SPEAKER_TYPE >> "RINGER" in sipXtapi as my second speaker. Is it possible ? Or are there >> any alternatives to get such kind of function? (using of GIPS?). > > This is definitely not possible. sipXtapi is entirely mono -- single audio > stream passing through. > We just finished up work to make mediaLib support wideband sample rates > (i.e. 16kHz, 32kHz, 48kHz). Previously sipXtapi was fixed at 8kHz. But > there is no plans currently to add stereo support for sipXmediaLib. Well, data may be leaved mono in sipXmediaLib. Rather you could create second output from Bridge and pass it second output device. This second audio device will be other speaker. But you have to modify sipXmediaLib's output device driver (MpodWinMM) to support channel separation. That is so it will be able to write to only one channel. -- Regards, Alexander Chemeris. SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000 _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
