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savoirfairelinux
jami-daemon
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c766b7371e6e70089aed88ffced2681784016e64
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Created with Raphaël 2.2.0
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28
Fix SIPCallClose() function, recorded file now close properly
Function terminateSIPCall added in sipvoiplink and managerimpl
Add a library manager + exception handling;
Fix thread destructor
Fix bug in sipvoiplink.cpp
Clean up in the working directory
Remove zeroconf from the build system
Merge branch 'master' into recording
Merge branch 'debug_threads'
Merge branch 'master' of git+ssh://repos-sflphone-git@sflphone.org/~/sflphone
Better handling of capture XRUNs
setRecordingOption function implement in audiorecord
Restore mic adjust volume on ALSA layer
Protect device ALSA operation if not opened
FIx the switching layer bug
Use dynamic_cast<> to use audiolayer-specific methods
Record now implemented in Call class
Open the audio devices only once at startup
Record interface complete (on hold erase previous recording)
Refactoring of the ALSA part
Functional plug-in manager
Merge branch 'plug-in' of ssh://manu@192.168.1.105/~/dev/sflphone into plugins
Use a C++ thread to handle tones and DTMF in ALSA
Added recButton in client
Restore IAXVoIPLink, restore Mutex
Make the plugins registering against the plugin manager
Merge branch 'threads' into plug-in
SIP part redesign almost done
Added: record button related icons
Record button added
Migrate to 1->N relationship between voiplink and accounts
api plugin for registration
Overload AudioRecord::recData to get mic and speaker data mixed
Recording now in audiortp::run() method
Use c++ thread in SIP, move everything in sipvoiplink;
Audio recording working in AudioRTP: receiveSessionForSpeaker
Update rpm spec file
Open/close a wave file when pulse audio stream start/stop
Merge commit 'origin/plugins' into plug-in
Continue plugin API definition
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