When joining/returning breakouts, audio would always connect
with full audio. This can lead to a performance problem, once
all listenonly users would join full audio, increasing the
number of streams in FreeSWITCH.
We now have a consistent behavior, which is:
1 - The choice made by the user in the main room is predominant:
if mic is active in main room, user will automatically
join mic in breakout room. When returning from breakout
room, user will also join with mic again.
2 - Changes made in breakout room won't have effect when
returning to the main room. This means if user, for example,
change from listenonly to mic in breakout room, the returning
will consider the option choosen previously (listenonly) and
listenonly will be active again in the main room.
3 - If user didn't join audio in the main room, the audio modal
will be prompted when joining the breakout room (this is
a special case of (1))
The following is some technicall information:
InputStreamLiveSelector (component.jsx) now calls
'handleLeaveAudio' function, which is the default
function when user leaves audio (also used when
dynamic devices are inactive).
We now store information about user's choice (mic or listenonly)
using local storage, instead of the previous cookie method (this
was triggering some warnings in browser's console).
Also did a small refactoring to match eslint rules.
Fixes#11662.
This commit contains three fixes: one already reported and two detected
during the investigation of the solution.
This started as a fix for firefox (#12023), but i also fixed the muted
alert/banner when device changes: the banner wasn't detecting device changes,
unless audio was deactived/actived.
There's another fix for the microphone stream: we now keep sender's track
disabled if it was already disabled for the sender's track of the previous
selected device.
Also did small refactor for eslint checking.
Some technical information: in sip bridge (bridge/sip.js), setInputStream and
liveChangeInputDevice function were both fully turned into promises, which
guarantees we have everything ready when it resolves to the respective values.
This helps AudioManager (audio-manager/index.js) to sequentially sets and
tracks the state of the current microphone stream (inputStream), when calling
liveChangeInputDevice function: we first set the current stream to null,
creats a new one and then set it to the newly created value - this is needed
because MutedAlert (muted-alert/component.jsx) can then gracefully
allocate/deallocate the cloned stream when it is set to a non-null/null value
(the cloned stream is used for speech detection with hark).
In MutedAlert we also make sure to enable the cloned stream's audio
tracks, just in case the user change the device when muted (audio track is
disabled in this case), which also leaves the cloned stream muted (we then
enable the track to allow speech detection).
Closes#12023
