Age | Commit message (Collapse) | Author | |
---|---|---|---|
2024-09-16 | Merge branch 'main' into 2024.08-DeltaFPS | Erik Kundiman | |
2024-09-16 | WebRTC on Fedora & openSUSE without breaking CEF | Erik Kundiman | |
Thanks to the Linux x86-64 WebRTC binary from Zenichi Amano (crow-misia). https://megapahit.com/show_bug.cgi?id=64 Haven't been tested on openSUSE, but it should work. | |||
2024-09-01 | Merge remote-tracking branch 'secondlife/release/2024.08-DeltaFPS' into ↵ | Erik Kundiman | |
2024.08-DeltaFPS | |||
2024-08-20 | Merge remote-tracking branch 'origin/release/2024.06-atlasaurus' into develop | Brad Linden | |
# Conflicts: # autobuild.xml # indra/newview/llvoicewebrtc.cpp | |||
2024-08-20 | Merge remote-tracking branch 'secondlife/release/2024.06-atlasaurus' into ↵ | Erik Kundiman | |
2024.06-atlasaurus | |||
2024-08-17 | Fixes to managing device start/stop playout/recording. | Roxie Linden | |
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. | |||
2024-08-02 | Merge remote-tracking branch 'secondlife/release/webrtc-voice' into webrtc-voice | Erik Kundiman | |
2024-08-01 | Merge remote-tracking branch 'origin/release/2024.06-atlasaurus' into develop | Brad Linden | |
2024-07-31 | Implement a Logging Sink for WebRTC | Roxie Linden | |
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. | |||
2024-07-29 | Fix trailing whitespaces in webrtc code to pass pre-commit | Andrey Lihatskiy | |
2024-07-29 | Update boost to 1.85 and fix deprecation warnings | Rye Mutt | |
2024-07-18 | Fix trailing whitespaces in webrtc code to pass pre-commit | Andrey Lihatskiy | |
2024-07-17 | Reenable OnDevicesUpdated on all platforms | Erik Kundiman | |
but not as "override" on the less supported ones. | |||
2024-07-14 | Disable audio device sink related code on some | Erik Kundiman | |
It's the custom part of LL's WebRTC fork, and I haven't got the resources to build LL's fork for the platforms unsupported by them. And for those less supported platforms, we're using binaries from https://github.com/crow-misia/libwebrtc-bin | |||
2024-06-24 | [WebRTC] control microphone gain via custom audio processor. | Roxie Linden | |
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. | |||
2024-05-17 | Clean up some shutdown code. | Roxie Linden | |
2024-04-28 | Reconnects to the voice server weren't happening. | Roxie Linden | |
2024-04-07 | Show 'decline' when peer declines p2p voice | Roxie Linden | |
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. | |||
2024-04-01 | Fix "default" audio device handling. | Roxie Linden | |
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. | |||
2024-03-30 | Fix windows crashes | Roxie Linden | |
* 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. | |||
2024-03-30 | Add UI for managing echo cancellation, AGC, and noise control. | Roxie Linden | |
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. | |||
2024-03-19 | Simplify workqueue calls. Fix issue with webrtc blocking on destruction. | Roxie Linden | |
2024-03-14 | Refactor device selection logic | Roxie Linden | |
This refactor fixed a few bugs. There is an annoying 'click' when changing devices, however. This will be addressed in the future. | |||
2024-03-13 | some comments; allow proximal channel to retry when it drops | Roxie Linden | |
2024-03-10 | Remove trailing spaces. Other code cleanup. | Roxie Linden | |
2024-03-09 | code beautification/comments | Roxie Linden | |
2024-03-05 | The response from the provision account call was being called twice for some ↵ | Roxie Linden | |
reason | |||
2024-02-22 | put observer-based tuning audio level calculation back | Roxie Linden | |
2024-02-22 | Use a custom audio processor to pull data for level determinations, which ↵ | Roxie Linden | |
will happen after AGC | |||
2024-02-22 | Handle 'device changed' callback | Roxie Linden | |
2024-02-22 | New WebRTC with echo cancellation fix. | Roxie Linden | |
Also, start/stop recording depending on whether WebRTC has negotiated. | |||
2024-02-22 | fix device selection (hopefully) | Roxie Linden | |
2024-02-22 | missed file | Roxie Linden | |
2024-02-22 | Using the device module to set speaker/mic volume set the system mic/volume | Roxie Linden | |
for all applications. Instead, modify the volume on the various streams. | |||
2024-02-22 | Refactor/clean-up WebRTC voice to handle multiple voice streams | Roxie Linden | |
This is useful for cross-region voice, quick voice switching, etc. | |||
2024-02-22 | comment fixes | Roxie Linden | |
2024-02-22 | SL-20543 - voice over region boundaries. | Roxie Linden | |
This commit includes code to allow the llwebrtc.dll/dylib to allow multiple connections at once. | |||
2024-02-22 | Fix race in initialization. Fix failure to send ice candidates to janus. | Roxie Linden | |
2024-02-22 | fix device selection while speaking. | Roxie Linden | |
2024-02-22 | Improve reconnection logic and allow device setting when connected or not ↵ | Roxie Linden | |
connected | |||
2024-02-22 | Smooth voice power level reporting. | Roxie Linden | |
2024-02-22 | Fix shutdown crash issue. | Roxie Linden | |
2024-02-22 | Stream audio levels to and from viewers via DataChannels | Roxie Linden | |
2024-02-22 | add datachannel support | Roxie Linden | |
2024-02-22 | Fix voice device settings | Roxie Linden | |
2024-02-22 | Hook up speaker volume. | Roxie Linden | |
2024-02-22 | Fix windows pragma error | Roxie Linden | |
2024-02-22 | Updates to build on mac. | Roxie Linden | |
2024-02-22 | Checkpoint WebRTC Voice | Roxie Linden | |