I/O device IDs are logged in some specific logCodes, but they aren't too
useful on their own without the rest of the MediaDeviceInfo object. We
need that extra data (label, group) to be able to better investigate
incorrect device issues and NotFoundError occurrences.
Register full I/O device info whenever the client fetches them and add
those, unfiltered, to the following logCodes:
- audiomanager_error_getting_device
- audiomanager_error_device_not_found
- audiomanager_error_unknown
- audio_joined
- audio_ended
- audio_failure
- audiomanager_input_live_device_change_failure
- audiomanager_output_device_change_failure
In BBB ≤ 2.7, a procedure monitored system audio device changes, updating
the device list and assigning a fallback device if the current one was removed.
This procedure was removed in 3.0 during the migration of the
input-stream-live-selector component to TypeScript (reasons unknown), causing
the device list to become outdated and leaving the user's client without audio
input if their current device is disconnected.
This commit restores the `devicechange` event handler in the input-stream-live-
selector, ensuring that the device list is updated properly and fallback devices
are assigned when necessary.
There's a regression in 3.0's I/O device selector where default output
devices are not marked as selected in the input-stream-live-selector
component unless the user explicitly selects them. This issue can also affect
input devices, although less commonly than output due to the system's ability
to infer the selected input device ID after the user joins audio.
When a device is the first in the list and no currentDeviceId is set in
the client, treat the first device returned by enumerateDevices as the
system default and hence selected, in accordance with the "Media Capture
and Streams API", Section 9.2, enumerateDevices algorithm.
When going from "no mic" -> mic via the unmute action, the client isn't
unmuting itself after confirming the change. This is caused by not
waiting the liveChangeInputDevice method (which is a Promise) to be
fully executed before unmounting the AudioSettings modal -- the one
responsible for triggering the unmute. Since it unmounts before the
device is changed, the unmute action will be ignored because the device
is still "listen-only" (no mic).
Properly unmute audio when transitioning from "no mic" -> "mic" via the
unmute trigger by waiting for liveChangeInputDevice to resolve.
Additionally, some general improvements to UI/UX:
- Display the AudioSettings modal title when gUM is on prompt mode
- Add specific subtitles to the AudioSettings modal to 1) warn that no
mic is selected 2) Give a hint that the user can test their devices
- Always honor settings.yml's "initialHearingState" state (whether
local echo feedback should be played by default in AudioSettings)
This is a rework of the audio join procedure whithout the explict listen
only separation in mind. It's supposed to be used in conjunction with
the transparent listen only feature so that the distinction between
modes is seamless with minimal server-side impact. An abridged list of
changes:
- Let the user pick no input device when joining microphone while
allowing them to set an input device on the fly later on
- Give the user the option to join audio with no input device whenever
we fail to obtain input devices, with the option to try re-enabling
them on the fly later on
- Add the option to open the audio settings modal (echo test et al)
via the in-call device selection chevron
- Rework the SFU audio bridge and its services to support
adding/removing tracks on the fly without renegotiation
- Rework the SFU audio bridge and its services to support a new peer
role called "passive-sendrecv". That role is used by dupled peers
that have no active input source on start, but might have one later
on.
- Remove stale PermissionsOverlay component from the audio modal
- Rework how permission errors are detected using the Permissions API
- Rework the local echo test so that it uses a separate media tag
rather than the remote
- Add new, separate dialplans that mute/hold FreeSWITCH channels on
hold based on UA strings. This is orchestrated server-side via
webrtc-sfu and akka-apps. The basic difference here is that channels
now join in their desired state rather than waiting for client side
observers to sync the state up. It also mitigates transparent listen
only performance edge cases on multiple audio channels joining at
the same time.
The old, decoupled listen only mode is still present in code while we
validate this new approach. To test this, transparentListenOnly
must be enabled and listen only mode must be disable on audio join so
that the user skips straight through microphone join.
* refactor(storage): replace Tracker.Dependency with observer hook
* fix(storage): set initial value
* refactor(storage): stop using Meteor's Session singleton
Mobile users have no way to change I/O devices after joining audio.
The removal of the audio options chevron in mobile browsers was supposed
to be replaced by something else - in this case, by the dedicated
leave/join audio button. That didn't happen, leave/join audio button
retained the old behavior.
Review device selection in mobile endpoints via two UI/UX changes:
- Fix an styling issue where the mute and listen only buttons were
crammed together
- Restore the device selection chevron/icon in mobile endpoints
- Override the leave/join button action in mobile endpoints so that it
opens the device selection contextual menu, which also includes the
"Leave audio" option. This retains the old behavior (leaving audio)
while also providing an way for users to change devices mid-call in
mobile browsers.
* test: fix shortcuts, add flaky flag for test requiring graphql data, fix slide change for tldraw v2
* test: properly set the execution mode
* test: use isMultiUser parameter inside options obj
* test: fix banner color test
* test: increase breakout test timeouts for user joining room
* test: redo the change in the hide presentation on join test
* test: change hide presentation steps and add flaky flag on it
