Muting was a bit random in the code, so it's now been straightened out and should
prevent echo.
Also, code was added to not attempt connection to non-webrtc regions in the webrtc code.
When transitioning from mic-on hands-free mode to mic off,
it's expected that the audio stream would return to stereo.
Inproper logic in the mac device code in webrtc was preventing
that.
The microphone issue was causing a short moment of sound, and was
causing bluetooth headsets to switch to hands-free/one channel mode
which is disruptive.
Also, update webrtc to deal with issue where airpods were garbled
after coming out of hands-free mode.
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.
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.
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.
When creating a new connection, the viewer builds a data channel interface.
It then gets a new one, which is a proxy. The viewer uses the new one,
and therefore must unregister the callbacks from the old one.
Also, update the position data before sending it after the join is sent.
When parcel voice permissions and region/parcel-only voice
settings change, a callback will be made to the viewer with
new voice credential information. For webrtc, this means
either just the uuid of the voice channel, or nothing if
voice is disabled.
This change looks at that callback and the channel id,
and sets the appropriate flags on the parcel/region as needed
which will cause voice to be renegotiated.
Also, there was a race condition if the voice connect attempt
was made before caps were retrieved, which would have resulted
in full renegotiate attempts. Now, just wait until the cap
comes in and continue.
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.
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.
* 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.
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.
Muting using the device module microphone mute was muting other
applications, speakers, and so on. Instead, we mute by enabling/disabling
the input and output streams.
Roxie Linden