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.
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.
* fix incorrect prop type
* remove duplicated code
* fix Unexpected constant truthiness on the left-hand side of a `&&` expression.
* fix Correct one of the identical sub-expressions on both sides of operator
* refactor(storage): replace Tracker.Dependency with observer hook
* fix(storage): set initial value
* refactor(storage): stop using Meteor's Session singleton
The audio troubleshooting modal has very microphone-specific strings,
which might confuse users trying to join listen only.
Review the Help screen so that listen only scenarios are more generic.
As a bonus, review the unknownError locale with a more actionable text.
In scenarios where the join audio flow skips echo test, NotAllowedError
(and any other errors) are all being mashed together under a generic
MEDIA_ERROR object.
Properly handle specific errors in audio-manager so they're correctly
render in the audio modal help screen.
- Adds a new Help view for unknown error codes
- Correctly detect NotAllowedError (permissions) - they are currently
being treated like unknown errors in the Help modal
- Rephrase NotAllowedError help text; make it more succint and direct
- Rephrase the unknown error help text; make it more succint and direct
- Add error code and message to that view
- Add public.media.audioTroubleshootingLinks to allow referencing KB
links on the Help modal
- See inline docs
There's no rollback procedure in case a device switch fails right now,
nor does the code entrypoints that call the switching procedures wait
for resolution or failure before marking the new device as chosen. That
may cause inconsistent states in a couple of ways:
- No rollback: switch fails, audio is still on but no actual
microphone input is being transmitted
- Not waiting for resolutions: inconsistent chosen devices on failures
Device switching errors are also not surfaced to the end user
This commit:
- Adds device rollback and proper resolution/failure response
awaits to try and make the state a bit more consistent.
- Centralizes the input device switching code to be reused between
different bridges
- Centralizes device ID state management in audio-manager to try and
mantain them a bit more consistent across the board
- Surface device switching failures to the end user
- Guarantee device IDs are set to the session storage on all
appropriate scenarios
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)
