Smart layout (et al) presumes screen sharing will always use 100%
width of the media area. That causes cameras to always be positioned on
top, which is not always the optimal position depending on the viewport
and stream aspect ratio/resolution - so space is wasted.
This commit uses the actual screen sharing video size as provided by
HTMLVideo's videoWidth/videoHeight properties. The calculation uses the
same logic as the one used for presentation/slides, which should make it
a bit familiar.
There's also a handler for HTMLVideo's `resize` event for those browsers
that support it - which enables handling of variable-sized screen
sharing streams. That handler is debounced at 500 ms to prevent
excessive CPU use.
Extra testing is needed with the widest range possible of
browsers/environments and feature combinations.
Only smart layout takes screen sharing/external video states in account
when populating its initial state. The others don't, and that causes
some weird issues when switching back-and-forth between layout types due
to their input states becoming inconsistent - ie having an active screen
sharing and transitioning from Smart -> Custom would mark it as false
(due to its absence from the initial state) and pollute its state for
subsequent layouts.
This commit guarantees those features are taken into account when
populating initial input states for Focus On*/Custom layouts.
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.
There's a VoiceUser cleanup procedure bound to the User's cleanup
routine in Meteor's server-side. That cleanup is _silent_ and does not
use a dedicated modifier from voice-user et al, which is not
straightforward and might waste a few minutes of understanding what's
happening when debugging audio collections.
This commit centralizes that cleanup in a new clearVoiceUser modifier in
voice-user as well as logs when it works.
Firefox doesn't fire the ended evt/onended callback for live
MediaStreamTrack(s). We rely on that event.
Manually emit the ended event which works with the onended callback
when a track is stopped
kurento-utils is unmaintained. It's served us well, but its age
shows. We need to transition to something else if we want to
have better maintainability and include simulcast, multistream, ...
This introduces a simplified/leaner wrapper kit that's almost
API-compatible with what we use right now - so widespread changes
are minimal). It's easier to maintain/read/transition from. This
can be read as an intermediate step to transitioning to
something definitive (ie mediasoup-client).
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.
The 'inactive' event is fired whenever the stream gets inactive (ie it
cannot be used anymore), and there are scenarios where that is
unexpected behavior and must be handled accordingly.
The main example of that is when gUM permissions are revoked by the user
via the browser's permission management panel.
Since MediaStream/Track inactive events aren't being handled in such
scenarios, what actually happens is that the camera just freezes without
further indication why.
This commit handles those scenarios in both video-preview and
video-provider by:
- 1) correctly stopping the camera (provider)
- 2) surfacing a toast (provider) or error indication (preview)
