* [WebRTC] Rework device handling sequence so that we can handle unplugging/re-plugging devices
The device handling was not processing device updates in the proper sequence as
things like AEC use both input and output devices. Devices like headsets are both
so unplugging them resulted in various mute conditions and sometimes even a crash.
Now, we update both capture and render devices at once in the proper sequence.
Test Guidance:
* Bring two users in the same place in webrtc regions.
* The 'listening' one should have a headset or something set oas 'Default'
* Press 'talk' on one, and verify the other can hear.
* Unplug the headset from the listening one.
* Validate that audio changes from the headset to the speakers.
* Plug the headset back in.
* Validate that audio changes from speakers to headset.
* Do the same type of test with the headset viewer talking.
* The microphone used should switch from the headset to the computer (it should have one)
Do other various device tests, such as setting devices explicitly, messing with the device selector, etc.
* Fix race condition when multiple change device requests might come in at once
* Update to m137
The primary feature of this commit is to update libwebrtc from m114
to m137. This is needed to make webrtc buildable, as m114 is not buildable
by the current toolset.
m137 had some changes to the API, which required renaming or changing namespace
of some of the calls.
Additionally, this PR moves from a callback mechanism for gathering the energy
levels for tuning to a wrapper AudioDeviceModule, which gives us more control
over the audio stream.
Finally, the new m137-based webrtc has been updated to allow for 192khz audio
streams.
* Properly pass the observer setting into the inner audio device module
* Update to m137 and get rid of some noise
This change updates to m137 from m114, which required a few API changes.
Additionally, this fixes the hiss that happens shortly after someone unmutes: https://github.com/secondlife/server/issues/2094
There was also an issue with a slight amount of repeated after unmuting if there was audio right before unmuting. This is because
the audio processing and buffering still had audio from the previous speaking session. Now, we inject nearly a half second of
silence into the audio buffers/processor after unmuting to flush things.
* Install nsis on windows
* Use the newer digital AGC pipeline
m137 improved the AGC pipeline and the existing analog style is going away
so move to the new digital pipeline.
Also, some tweaking for audio levels so that we don't see inworld bars when tuning,
so one's own bars seem a reasonable size, etc.
* Install NSIS during windows sisgning and package build step
* Try pinning the packaging to windows 2022 to deal with missing nsis
* Adjust gain calculation and audio level calculations for tuning and peer connections
* Update with mac universal webrtc build
* Tuning of voice indicators for both tuning mode and inworld for self.
* Redo device deployment to handle cases where multiple deploy requests pile up
Also, mute when leaving webrtc-enabled regions or parcels,
and unmute when voice comes back.
* pre commit issue
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.
Crash points at enableDisableCallBtn, but button's floater was cleaned
earlier and has a removeObserver call so an observer record should be
long gone. Likely something else is going on.
Went over various callbacks and made sure they are cleaned. But in case
floater somehow remained, added mSession = nullptr.
This fixes#1436, an issue where the 'call button' sometimes got greyed
out when it shouldn't have been.
Vivox is now 'working' (and the call button can be enabled) when it's logged in.
WebRTC is 'working' (and the call button can be enabled) when the coroutine is active.
The voice server sends up the float power level of peers as an integer
multiplied by 128, in order to save character count as the voice power level
will likely be only 3 digits, instead of many for a full float.
The client was not taking this into account.
In a few locations, there were cases where connection shutdown
would stall, leaving the connection in place. This was due to
bad handling of the outstanding operations counter.
There were changes in atlasaurus that resulted in a hang for incoming p2p
and group calls which throw up dialogs. The changes revolved around mutex,
coroutines, job queues, and such.
The fix was to do any processing that may result in callbacks from
the webrtc code in a queued job instead of a coroutine.
For issue #2064
The connection to the voice server was not upgraded/downgraded to primary/secondary when crossing
region boundaries, so the server sent the wrong value and the viewer chose not to display a voice dot.
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.
Jonathan "Geenz" Goodman