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Fixes prevent attempting to start playout/recording before the devices
are set up, to prevent restarting playout/recording, to prevent
attempts to stop when not playing/recording, and so on...
This should address the case where audio device changes can cause
an assert. It should also address the case where audio was unnecessarily played
or transmitted when connecting.
And, when voice is disabled, the audio devices are not set up to play/record
so there should be no disruption of bluetooth music from other apps.
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WebRTC logs now pass out of the webrtc library into a logging sink,
which converts them into SecondLife.log compatable logging calls.
This includes fatal errors and asserts, which are now logged into
SecondLife.log, and should be available in the crash logger.
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Previously, there were two places audio gain could be controlled:
- the device manager
- the audio track
The device manager audio gain control sets the system gain for all applications,
not just the webrtc application.
The audio track gain happens well after the audio processing where we want it to happen.
So, gain control was added to the existing custom audio processor, which previously only
handled calculating and retrieving the audio levels.
After these changes, the microphone gain slider does impact the audio volume heard by peers.
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The simulator will send a chatterbox notification that
voice is no longer in use for a given channel, and
the viewer should take that as a case where the peer
does not want voice, hence it's a decline.
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Windows and Mac/Linux behave slightly differently with respect
to Default devices, in that mac/linux (I think) simply assumes
the device at index 0 is the default one, and windows has a
separate API for enabling the default device.
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* sampling rate was set to 8khz for audio processing, which was
causing a 'bands' mismatch with the echo cancler.
* Some funnybusiness with lambdas and captures and such was causing
a heap crash with respect to function parameters.
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Plumb audio settings through from webrtc to the sound preferences
UI (still needs some tweaking, of course.)
Also, choose stun servers based on grid. Ultimately, the stun
stun servers will be passed up via login or something.
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This refactor fixed a few bugs. There is an annoying 'click' when
changing devices, however. This will be addressed in the future.
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reason
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will happen after AGC
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Also, start/stop recording depending on whether WebRTC has negotiated.
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for all applications. Instead, modify the volume on the various streams.
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This is useful for cross-region voice, quick voice switching, etc.
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This commit includes code to allow the llwebrtc.dll/dylib to allow
multiple connections at once.
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connected
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