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
* Update en.json
* Update settings.yml
* Create transcriptionLocale.ts
* Update component.tsx
* Update component.tsx
* Revert IN -> ID
Because it will be fixed in the main repository
* let -> const message
British -> GB
* Refactor audio captions messages and locales to fix issues reported by typescript code validation
---------
Co-authored-by: Ramón Souza <contato@ramonsouza.com>
This commit refactors the label generation for audio captions in the audio-captions button and captions components. Instead of directly using the intl formatMessage function, the label is now generated using the intlMessages object and the getLocaleName function from the audio-captions service. This change prevents an issue with custom locales.
Commit 325887e325 split the local echo audio
element from the main audio element to allow concurrent playback without the
risk of interfering with one another.
This introduced a regression where local echo doesn't track output device
changes. The main audio element (i.e. the meeting's audio) is not affected by
this regression.
This commit ensures local echo reacts to output device changes as needed.
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the device list to become outdated and leaving the user's client without audio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.
UI team suggested a few adjustments to the audio settings modal:
- Larger (24px/1.5rem) margin between content and headers
- Rephrasing of modal title, subtitle and volume indicator label
- Change the "audio feedback" button to an outline or link styled
button (there are currently two primary buttons and we want users to
focus on the "Join audio" one)
Implement the suggested changes. The approach for the audio feedback
button is link-styled.
Safari may enter a microphone permission check loop due to buggy behavior
in the Permissions API. When permission isn't permanently denied, gUM
requests fail with a NotAllowedError for a few seconds. During this time,
the permission state remains 'prompt' instead of transitioning to 'denied'
and back to 'prompt' after the timeout.
This leads to an issue where, on retrying while in 'prompt' + blocked,
the client loops through gUM checks via: 1) checking permission in the API,
2) receiving 'prompt', so trying gUM, 3) gUM fails, 4) returning to the
modal and checking permission again because the API still says 'prompt'.
Additionally, the `isUsingAudio` flag incorrectly counts the local echo
test/audio settings modal as "using audio," which toggles the flag on/off,
triggering the useEffect that causes the loop more frequently.
To fix this, remove the unnecessary AudioModal permission check that
causes the loop. Also, exclude "isEchoTest" from the `isUsingAudio` flag.
Firefox incorretly displays placeholder audio device labels in the audio
settings/echo test modal when audio is disconnected. This issue arises
due to two quirks:
- Firefox does not support the 'microphone' query from the Permissions
API, causing a fallback gUM permission check.
- Firefox omits device labels from `enumerateDevices` if no streams
are active, even if gUM permission is granted. This behavior differs
from other browsers and causes our `enumerateDevices` handling to
assume that granted permission implies labels are present. This
failed since we clear streams before resolving the fallback gUM.
We now run an additional `enumerateDevices` call in `AudioSettings` when
a selected input device is defined. This ensures `enumerateDevices` is
re-run when a new stream is active, adding the correct device labels in
Firefox and improving device listings in all browsers. We've also
enhanced error handling in the enumeration process and fixed a false
positive in `hasMicrophonePermission`.
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 `listenOnlyMode` is `false` and the audio dialog's "Cancel" action is
clicked, the modal incorrectly re-renders instead of closing. Additionally,
the "Cancel" action is mislabeled as "Back."
This fix ensures the audio dialog closes properly when there are no options
to select (i.e., `listenOnlyMode=false`). The `skipAudioOptions` method is
revised to consider `listenOnlyMode` and ignore the "content" state.
Ignoring the "content" state allows options to be skipped even if a subscreen
is rendered (e.g., returning from the AudioSettings modal). The check for
`content == null` combined with `skipAudioOptions` is only necessary when
rendering the main modal. The `content == null` check has been moved to
the relevant section.
- Move askForConfirmationOnLeave into AudioContainer
- Get rid of the unstable useMuteMicrophone hook, which returns a new reference every time the user gets muted/unmuted. Use the stable useToggleVoice hook instead.
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)
We are missing a way to select transcription languages in some
scenarios, e.g.: listenOnlyMode=false. The audio settings UI is also not
handling item disposition very well on smaller devices.
This commit does the following to improve those blind spots:
- Add the transcription language selector to it whenever applicable
- Add proper styling to the transcription selector
- Handle small screens by changing the disposition of elements to
portrait mode
- Improve how elements are disposed to a more familiar view: Mic ->
Activity Indicator; Speaker -> Speaker test. This is more in line
with how other platforms do audio configuration/pre flight screens.
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.
