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
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.
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.
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`.
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.
* refactor(storage): replace Tracker.Dependency with observer hook
* fix(storage): set initial value
* refactor(storage): stop using Meteor's Session singleton
There are some situations where previously set deviceIds (
local/session storage) may become stale. This causes an unexpected
behavior where audio is temporarily borked until the user clears their
local storage.
This issue has been seen more recently on Safari endpoints when switching
back-and-forth breakout rooms in environments running under iframes.
Also seen randomly on endpoints with virtual input devices.
This centralizes audio gUM calling into a single method that retries the
gUM procedure without pre-set deviceIds only if the initial call fails
due with an OverconstrainedError - hopefully circumventing the issue.
Extract the deviceId again from the stream to guarantee consistency
between stream DID vs chosen DID. That's necessary in scenarios where,
eg, there's no default/pre-set deviceId ('') and the browser's
default device has been altered by the user (browser default != system's
default).
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
The new local echo view doesn't block the "Join audio" button while
awaiting for getUserMedia permission to be granted/denied. That may
cause unexpected behavior when unattentive users just click "Join audio"
without granting or denying gUM.
This commit accounts for gUM resolution when deciding whether to block
the "Join audio" button. It also includes an extra "isConnecting" check
to it to avoid spam-clicking issues.
removing border and implementing box-shadow
adding transparent border
passing styles to common buttons
adding secundary color to component
updating color components
The initial selected output device in AudioSettings could be the wrong one if
the user's session had an output device ID already stored, but is joining on a
new session. That would cause the remote-media tag not to be updated with the
correct output device ID when it should (the service.js change)
The issue is tackled by guaranteeing the output device ID is set on all ends
when AudioSettings/AudioModal mounts.
The initial selected input device in AudioSettings could be the wrong one if
- 1) gUM outputs an user-selected device rather than the default
- 2) no previous device was selected for that domain and the enumeration
list order caused the default not to be the first
The issue is tackled re-extracting the deviceId from an input stream if it
exists and making the DeviceSelector value follow what is defined in the client
(audio-manager) via a trackable prop
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
