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
Mobile endpoints are flaky with the WebSpeechAPI:
- iOS versions that support it are borking our outbound audio when it's
enabled
- Android speech recognition has flaky locale detection and speech
transcription
Additionally: the support check is not checking the WebSpeechAPI
availability properly, so older devices (eg iOS 12) are flagged as
supported even though they aren't.
This commit adds a configuration flag (public.audioCaptions.mobile) to
control transcription availability on mobile. False by default.
Also extends the setSpeechVoices support check and
hasSpeechRecognitionSupport method to prevent false positives.
Adds two new flags to the settings file which change the way the locale
flag is used:
- forceLocale: (true/false) => If true, enforces the transcription
language to be the locale content field and jumps the language
selector
in audio modal.
- defaultSelectLocale: (true/false) => If true, the default selected
value in the dropdown language selector in audio modal will be defined
by the locale content field.
In any case, if the locale flag holds an invalid value, it defaults to
disabled.
Move the language collection to the HTML settings file. This data defines
the available languages available for the speech API.
These language tags are used to filter SpeechSynthesis' API `getVoices`
result. Tags must use BCP 47 format.
https://developer.mozilla.org/en-US/docs/Web/API/SpeechSynthesisVoice/lang
Avoid enable audio transcription if the browser's vendor does not provide
voices data.
This should prevent false positives for browsers such as Chromium and
Brave.
Parse the audio transcript before broadcasting it's content back to the
client and the recording actor. Limiting by 8 words per line and max of
2 lines to avoid CPU intensive operations over this recurring event.
Replace Calibri font family with Verdana to improve character spacing,
add relative sizing to the text content and a background padding.
Add a server-side app for the audio captions feature and record proto-events
for this data.
As it is, only behaves as a pass-through module. The idea is to include all
the business intelligence in this app.
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 talking-indicator, emoji-button and input-live-stream-selector
components are passing props downstream that weren't omitted or handled
by inherited components (Button, Icon). That causes a handful of error
logs to be spammed in the console of dev environments, which is
annoying.
This commit addresses the issue by:
- Making the talking, spoke, muted and isViewer props transient
(styled-components) - which means they won't reach the DOM (as
expected since they're style-only)
- Omitting the EmojiButton `rotate` prop in the Icon component itself
* Made that instead of transient because might be useful to migrate
the rotate code to the Icon component?
Move device selection dropdown to microphone button
Move option 'leave audio' to an option within the dropdown
Remove the audio exit button when device !== mobile
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
