- 07 Aug, 2009 1 commit
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Alexandre Savard authored
Call someone, add incoming calls to the conference using the new conference button Signed-off-by:
Alexandre Savard <alexandre.savard@savoirfairelinux.net>
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- 06 Aug, 2009 1 commit
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Alexandre Savard authored
Which send a signal through DBUS
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- 05 Aug, 2009 4 commits
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
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- 04 Aug, 2009 1 commit
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Alexandre Savard authored
test for multiple conference participants MainBuffer::getData MainBuffer::putData MainBuffer::discard MainBuffer::flush
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- 03 Aug, 2009 1 commit
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Alexandre Savard authored
Now, creation and deletion of rngbuffers is automated. It fixes the problem concerning discarding the data
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- 31 Jul, 2009 5 commits
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
create, remove, get, store ReadPointer
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Alexandre Savard authored
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- 30 Jul, 2009 3 commits
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Alexandre Savard authored
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Alexandre Savard authored
Data captured during call dialing is now flushed when rtp thread is started (like pulse audio) However, Audio stream for both Alsa and PulseAudio should be stopped see
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Alexandre Savard authored
Currently we are pooling the playback and the capture devices in the same function Idealy we should use the Alsa interuption mechanism by implementing playback_callback and capture_playback SEPARATELY There is still a problem with the capture as it start to fill the ring buffer at the same time than the playback (i.e. as soon as the tone occurs) which introduces latency during the call.
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- 29 Jul, 2009 7 commits
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Alexandre Savard authored
However, Microphone Capture using alsa cannot be copied inot ring buffer. The application was by passing the micRingBuffer in this case...
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
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Alexandre Savard authored
Conflicts: sflphone-common/src/sipvoiplink.cpp sflphone-common/src/sipvoiplink.h
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- 28 Jul, 2009 8 commits
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Alexandre Savard authored
This buffer is not one coresponding to the call ID since we do not want to send back the incoming data In case of a conference, we just need to extend this id to a set of ids and mix them together if the set is greater than one
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Emmanuel Milou authored
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Alexandre Savard authored
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Alexandre Savard authored
Only the pointer from the ringbuffer map were erased, not the ring buffer itself...
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Alexandre Savard authored
The audio frame is copied into the right ring buffer given the CallID as an argument
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Alexandre Savard authored
Put(int), Get(int), Put(float), Get(float), using two pointer (which should be in AudioLayer and in AudioRtp)
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Alexandre Savard authored
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Alexandre Savard authored
Since the AudioRTP pointer is now in SIPcall, we need to pass the call object as a parameter.
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- 27 Jul, 2009 3 commits
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Emmanuel Milou authored
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pierre-luc authored
Conflicts: sflphone-client-gnome/src/config/configwindow.c sflphone-client-gnome/src/menus.c
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SFLphone Project authored
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- 24 Jul, 2009 6 commits
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Emmanuel Milou authored
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Jérémy Quentin authored
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Jérémy Quentin authored
[redmine_down] crash when hanging up a dialing call because tries to add it to history whereas no starttime
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Jérémy Quentin authored
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Jérémy Quentin authored
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Jérémy Quentin authored
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