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2024.06-atlasaurus
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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.
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2024.06-atlasaurus
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release/2024.06-atlasaurus
# Conflicts:
# indra/newview/llpanelpeople.cpp
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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.
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release/2024.06-atlasaurus
# Conflicts:
# .github/workflows/build.yaml
# indra/newview/CMakeLists.txt
# indra/newview/llspeakers.cpp
# indra/newview/llvoicechannel.cpp
# indra/newview/llvoicechannel.h
# indra/newview/llvoiceclient.cpp
# indra/newview/llvoiceclient.h
# indra/newview/llvoicewebrtc.cpp
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bar webrtc's coroutines from necromancy
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release/2024.06-atlasaurus
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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.
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2024.06-atlasaurus
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release/2024.06-atlasaurus
# Conflicts:
# autobuild.xml
# indra/newview/llvoicechannel.cpp
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GCC would treat it as an error.
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At least not in my debootstrapped Debian 12.5 system.
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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.
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other jobs might be using it.
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This is because the button enable state was updated before the
call had fully established.
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When creating a new connection, the viewer builds a data channel interface.
It then gets a new one, which is a proxy. The viewer uses the new one,
and therefore must unregister the callbacks from the old one.
Also, update the position data before sending it after the join is sent.
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When teleporting or moving around, connections to regions are shut down.
We need to track which participants are associated with the given
connections and remove those participants when the connection is
shut down.
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The server will send an update with new voice credentials
when another peer comes in requiring vivox, if the channel
was initially webrtc.
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Before login, when the user shows voice device preferences,
the dropdowns were not populated. This is because WebRTC's view
of the main queue was not configured.
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When parcel voice permissions and region/parcel-only voice
settings change, a callback will be made to the viewer with
new voice credential information. For webrtc, this means
either just the uuid of the voice channel, or nothing if
voice is disabled.
This change looks at that callback and the channel id,
and sets the appropriate flags on the parcel/region as needed
which will cause voice to be renegotiated.
Also, there was a race condition if the voice connect attempt
was made before caps were retrieved, which would have resulted
in full renegotiate attempts. Now, just wait until the cap
comes in and continue.
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voice
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When the server sends up a notification that a peer is not doing
voice in adhoc-style p2p chat, shut down the voice call and
notify the user the peer has declined.
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When declining a P2P voice call for webrtc, instead of relying
on vivox to stop "ringing," we need to send an explicit decline
message from the peer through the server infrastructure.
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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.
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into roxie/webrtc-voice
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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.
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