* Fix indexing problem with mac devices
This resulted in the wrong device being selected.
Also, fix a shutdown crash where recording was not being stopped, hence the recording
thread was still running on shutdown and crashed because it lost access to resources.
Fix an issue with p2p calls where they're coming up muted even though the button indicates
they are unmuted.
* Always refresh device list on notification of device changes
Even when the selected device doesn't change, we need to
re-deploy it as it might have had characteristics (sampling rate, etc.) changed.
Also, we need to redeploy when the Default device has changed
* [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
This fixes a specific issue that has existed for a while so it is not a full fix for FIRE-35987. It will however reduce overfetch and increased data/cache/RAM usage as a result of the overfetch.
Reported issue to LL GitHub and FS JIRA FIRE-36003
LL Issue: https://github.com/secondlife/viewer/issues/4794
Linked to FS issue and cherry picked to pull in fix.
Inital Issue:
When you look at the SceneLoadRearMaxRadiusFraction value stored in the settings.xml, you will notice that the value is set to 75.0 for a float value. And the comments mention it is suppose to be fraction.
The code that uses it is for scaling the amount of the scene behind the visible camera is loaded. The issue is because the calculation no longer is divided by 100.0f, the draw distance is multiplied by the value and then clamped at the draw distance value. So it is effectively only ever the draw distance and not the default 75% of it. So a 256 draw distance is locked at 256 meters instead of 192.
This causes more objects to load behind the user, especially very far objects which can cause more RAM and VRAM usage, especially on teleport.
Both collada and gltf have a node and a mesh. Collada uses node-name, gltf was using mesh-name.
GLTF format permits reusing single mesh for multiple nodes, but nodes are warrantied to not be reused.
Switch to using node-names for better dupplicate avoidance and to be more in line with collada.
Fixes texture transforms being reset when switching from Blinn-Phong to PBR materials and between PBR. Previously, custom scale, offset, and rotation settings would be lost, making it tedious to switch between PBR materials.
Hecklezz