For scenarios where streams are produced in AudioSettings (local echo,
volume meter), force gUM resolution before devices are enumerated.
This effectively guarantees that all devices are present, labelled and
with deviceIds.
public.media.showVolumeMeterInSettings => public.media.showVolumeMeter
public.media.simplifiedEchoTest => public.media.localEchoTest.enabled
Initial hearing state can be configured in public.media.localEchoTest.initialHearingState
New features:
- A simplified echo test mode that only does a local loopback (instead of
going to FS and back)
- A volume meter for microphone streams to the AudioSettings view
Those two features are experimental and disabled by default; see
public.app.media.simplifiedEchoTest and public.app.media.showVolumeMeter configs
Collateral changes:
- fix: localize fallback device strings in AudioSettings/DeviceSelector
- Refactor on some media stream utils to be re-usable across components
- Refactor in AudioSettings to keep gUM #uses stable.
* TODO: need to pass streams through AudioManager to avoid the surplus gUM.
- fix(audio): drop ScriptProcessorNode usage (deprecated)
* Used in volume meter for tracking - use hark instead
We are now leaving the check for the minBrowserVersions object in settings.yml
If the settings enables chrome iOS, audio should allow users to be joining
with audio.
This is related to recent Chrome update (iOS 14.3+) that now allows
camera/microphone to be captured. We are looking for enabling this for
Chrome 93 in iOS (chromeMobileIOS version in settings.yml)
When listenOnlyMode=false, skipCheck=true and skipCheckOnJoin=true, the
audio tries to start a session more than one time, causing it to fail
at the first one (and reconnect after that).
Now we check if user is already connecting before trying to start a new
audio session.
Added some info in settings.yml for the options related to this commit
Closes#12190
When setting skipCheckOnJoin to true, an extra audio channel is created in
FreeSWITCH, after user accepts the echo test. The extra channel is removed
when user leaves the room, but this still may affect performance.
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.
* add param to force echo test when user joins audio after init
* fix UI stuck on connecting when userdata-bbb_auto_join_audio=false
* fix conditions for joinFullAudioImmediately and joinFullAudioEchoTest | remove old format
* remove extra param in getItem
* recover audioLocked | only set getEchoTest if doesnt exist
When user joins audio and for some reason an error (such as 1001, 1002,...), happens, the user is not able to click "Mic" and "Listen Only Buttons"; except if the audio window is closed and oppened again.
When getting disconnected with 1001 ("websocket closed unexpectedly" error) we were creating a new SIP session, therefore a new FreeSWITCH channel.
While reconnecting the socket, instead of closing the SIP session, we keep it alive during reconnection (audio should keep working in the meantime). When reconnected we keep using this same session (avoiding the creation of an extra one).
We also better handle WebSocket error codes from SIP.js.
FF immediately closes websocket when unloading page, so we now to stop user agent when 'beforeunload' event is triggered, to avoid leaving open sessions in FreeSWITCH when user leaves page.
When refusing ("thumbs down" button) echo test, user is able to select a different input device. This should work fine for chrome, firefox and safari (once user grants permission when asked by html5client).
For output devices, we depend on setSinkId function, which is enabled by default on current chrome release (2020) but not in Firefox (user needs to enable "setSinkId in about:config page). This implementation is listed as (?) in MDN.
In other words, output device selection should work out of the box for chrome, only.
When selecting an outputDevice, all alert sounds (hangup, screenshare , polling, etc) also goes to the same output device.
This solves #10592
This considerably changes the way we process audio signaling and start audio elements in user's browser.
We now avoid using AudioContext element for both microphone and listenonly calls, once it is unstable for some iOS devices (cracky audio, user stops hearing audio after a while).
Increased default value for listenOnlyCallTimeout: this avoids activating FreeSWITCH's fallback when ICE negotiation takes longer than 15sec (tested on DO).
Increased listenonly logs.
This fixes#8133#10388
