Release/2025.07 #4654
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Release/2025.07 #4654
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Fix disabling renderdoc support Improve ll_aligned_alloc functions on darwin for 32 and 64byte aligned by utilizing posix_memalign
Port Fontcache from Firestorm.
…tibility with modern libraries
… have it's own cache/cookie folder underneath the parent cef_cache folder. The whole cef_cache folder structure is purged at startup (before the parent being created at the first media instance creation)
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llkdu
llmath
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llprimitive
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I have read the CLA Document and I hereby sign the CLA 9 out of 10 committers have signed the CLA. |
Improvements found in reviewing mac_arm branch #4629
* Remove GLM sse flag from cmake that was moved to llpreprocessor.h * Further reduce performance loss of HDR and Sharpening on bandwith-constrained gpu by combining gamma correction into tonemap/sharpening shader passes * Update SSE2NEON to 1.8.0 to fix random render nans * Fix occasional startup crash from LLCachedControl being declared in global scope
…t upload location
Temporary until we can better parallelize architecture specific builds.
…ging/re-plugging devices (#4593) * [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: secondlife/server#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
isSystemMemoryLow() and factor check were too agressive for draw range.
resume() was trigggering sOnCurrentChannelChanged which was wiping participant list with no follow up updates.
* 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
for diagnostic purposes
NOTES Anti-virus false positive report:
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Dan-Linden
requested review from
bennettgoble
and removed request for
bennettgoble
|
Continues over there -> #4714 |
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relnotes:
2025.07