Commented out a block of new code recently added which was also being called by one of the callback functions.
This had the side effect of having 2 LLWebRTCDevicesObserver listening to the hardware removal events and cause both to trigger events at the same time, which may be the source of the queue being locked up in a race condition.
Was able to reproduce issue 2x without code change by moving mouse around a lot while on rezing and then pulling USB headset out of the USB port. Direct motherboard USB port seemed to work the best as could not trigger while headset was in USB hub.
Re-adding the feature to store and restore texture boost levels when a user edit's an object.
When you right click on an object. All the textures assigned to it get their boost level set to BOOST_SELECTED.
For LOD textures, once the texture's ProcessStats is completed, it resets the boost level to BOOST_NONE. This can cause the texture to then be subject to Texture Bias when in low memory, where before it was possibly protected as being a higher priority texture.
Now when the boost levels is about to change to BOOST_SELECTED, it gets stored and in ProcessStats, the boost level is restored.
Issue was fixed before but was rolled back. This is just restoring this one fix.
Add UI for controlling the settings, add new styles.
Snapshot guide with golden ratio options
Adds golden ratio composition guides to the snapshot tool, including options for different orientations (top-left, top-right, bottom-left, bottom-right).
Also refactors the snapshot guide settings into a separate floater for better organization.
* 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
Ansariel