After a homeserver stall Element Call showed "Connection lost" while the
call bar still looked joined. Handle the fork's io.lotus.call_error:
show "Call disconnected" with a Reconnect button (io.lotus.reconnect),
and reconnect automatically once sync is back.
Fixes#318
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_017PCb59uY3UKqDqt7B6hzd5
The denoise worklets/wasm/models were only copied into dist/ by the
lotus-denoise closeBundle hook, so on the dev server the call frame got
the SPA fallback for them and silently used the raw mic. The asset list
is now shared (denoiseSources) between that copy and a dev-server
middleware serving the same paths. Also send lotusTheme=true with
lotusTransparent, which the fork warned about in every call.
Fixes#293
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_017PCb59uY3UKqDqt7B6hzd5
deviceIds are hashed per frame/origin, so the ids the menu got from the
chat page's enumerateDevices() were unknown inside the call iframe: the
fork fell back to the default output while the menu showed a tick on
the chosen device. Handle io.lotus.audio_output_state (also silencing
the fork's "host did not ack" log) and build the menu from the fork's
list and effective selection; older forks fall back to the old list.
Fixes#310
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_017PCb59uY3UKqDqt7B6hzd5
CallEmbed asked Element Call to send a call-start notification for every
non-DM call. In persistent voice channels and space rooms it carries an
@room mention: every member got a highlighted ping (and phone push)
whenever a moderator or admin joined first, and other clients showed it
as a missed call. Now only rooms that can ring (isRingableRoom: DMs and
invite-only groups outside a space) get ring/notification; everything
else passes sendNotificationType=none (element-call-embedded 0.25.0-lotus.23).
Verified with a local build of the fork: an admin joining the Voice Lounge
sends none, and Bob gets no new notification (before: +1 highlighted).
Fixes#295
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_017PCb59uY3UKqDqt7B6hzd5
Groundwork for serving the call page from its own origin
(call.chat.lotusguild.org). Inert until config.json sets `elementCallUrl`:
without it the bundled same-origin page is used exactly as today.
- callPageUrl: resolves `elementCallUrl` — absolute https only (http only on
localhost for development); anything else, and the desktop app, fall back
to the bundled page so a bad value can't break calls. Set once from the
loaded client config.
- CallEmbed builds the widget URL from it; the widget origin (used by the
message guard and Capability Delegation) follows automatically.
- Soundboard: a host blob: URL can't be fetched from another origin, so
io.lotus.inject_audio now also carries the clip's bytes (`audio`). Forks
that predate it ignore the field and use `url`, so this is safe on the
released fork.
Needs element-call's lotus-call-origin branch (host-origin message check +
inject_audio bytes) released and pinned before `elementCallUrl` is set.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
matrix-widget-api's host transport handled a message from ANY window on the
page as long as it carried the widget's id; its strictOriginCheck only
compares with the host's own origin and is off by default. The call's id is
the fixed 'call-embed', so any other frame (a room widget, a URL-preview
embed) could post fromWidget actions as the call. Reproduced locally: an
opaque-origin frame posting one io.lotus.hotkey keydown for the PTT key
turned a push-to-talk user's mic on ("● Live").
restrictWidgetMessages() swaps each ClientWidgetApi transport's listener
for one that requires ev.source === the widget iframe's window and
ev.origin === the widget's origin. Applied to the call and to room widgets
(so one widget can't impersonate another). Verified: the spoof no longer
opens the mic; PTT/deafen from inside the call, screenshare, speaking
indicator and room widgets (capability prompt, send, live events) unchanged.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
Firefox, Safari and the WebKitGTK desktop app can't hand the user's click to
the call frame (no Capability Delegation), and getDisplayMedia needs it. The
host used to click EC's hidden footer button through the DOM instead, which
dies with same-origin. Now (pins element-call-embedded 0.25.0-lotus.20):
- those engines get `lotusFrameScreenshare`, the fork shows EC's own
screenshare button in the frame, and the call bar and status bar hide
theirs once controls_state reports `frameScreenshare`; the
screenshare-audio mute stays;
- the room's call policy is pushed with io.lotus.set_frame_screenshare, so
the frame button hides where the server would refuse a share, like ours;
- Chromium keeps the delegated io.lotus.set_screenshare from the host bar.
Removed the fallbacks for forks older than lotus.14, which read or clicked
EC's DOM: the screenshare/layout/settings/reactions/leave button lookups
and their MutationObservers, the frame-window hotkey binding, and the
speaking/muted tile scrape in useCallSpeakers (io.lotus.call_state is the
only source now). getCallDocument is gone; the host's only handle on the
frame is postMessage.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
Pins @lotusguild/element-call-embedded 0.25.0-lotus.19.
- Styles: the fork now hides its own footer (`lotusHostControls=1`) and
sets its root color-scheme from the theme, so the host no longer
injects `#lotus-ec-styles` or sets inline styles on EC's DOM. The two
other injected rules matched nothing in EC 0.25 (dead). The
transparent background was already the fork's (`lotusTransparent`).
- Hotkeys: PTT / deafen keys pressed with focus inside the call frame now
arrive as `io.lotus.hotkey` (the host sends the codes via
`io.lotus.set_hotkeys`), instead of listeners on the frame's window.
The window binding stays only for a fork that doesn't report `hotkeys`.
- Fixes (with lotus.19): pressing the deafen key M with focus in the call
also hit EC's own "M = toggle mic" shortcut, so the first press turned
the mic ON instead of deafening — even in push-to-talk mode.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
While unmuted, the microphone button (call view and the bottom call bar)
shows three small green bars in the icon's corner driven by the fork's
io.lotus.mic_level (element-call v0.25.0-lotus.16), so you can see the mic is
actually picking you up. Nothing is drawn while muted or silent; the button's
label is unchanged (the bars are decorative).
Your own name no longer appears in the "… is speaking" line: that is now for
everyone else, and your voice is on the meter. Avatar speaking rings still
include you, and the "You're muted" talking-while-muted notice is unchanged.
Verified in a local call with a fake-tone mic: data-mic-level cycles 3/2/1
with the tone on both buttons, disappears on mute and returns on unmute; the
status line reads "bob is speaking..." only.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
Once the fork reports io.lotus.controls_state (element-call
v0.25.0-lotus.14), the call bar's Spotlight/Grid View, Reactions and Settings
send io.lotus.set_layout / toggle_reactions / open_settings instead of
clicking Element Call's hidden buttons, and the call bar's screenshare and
layout state come from that report instead of a MutationObserver on EC's DOM.
Older forks never send the report and keep the DOM path.
Screensharing itself is still started/stopped via EC's button: it needs the
user's click to reach the frame (Capability Delegation, step 3 of #43).
Verified in a two-person local call (Synapse + LiveKit) against a locally
built fork: controls_state arrives on join; Spotlight View / Grid View flip
EC's layout and the reported layout; Reactions opens EC's reactions menu;
Settings opens EC's settings modal; the four actions go over postMessage;
starting and stopping a screenshare flips the host's button via the report.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
Event handlers typed as React.MouseEvent, join errors as Error, the
service-worker session setter as strings, the UIA policy map and the
webkitAudioContext probe as narrow shapes. No behaviour change.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
One line in the existing toast style when a call you were in ends: the
duration from our own join clock, plus the fork's io.lotus.call_summary
readout (fork ≥ 0.25.0-lotus.10) when it arrives — "connection was
good", "3 reconnects", "connection was poor for 4 min". Nothing is
stored or sent; the summary is one postMessage at hangup. Without the
fork summary the toast still shows the duration.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
Host half of the talking-while-muted feature. The fork (0.25.0-lotus.9)
reports speakingWhileMuted for the local participant in io.lotus.call_state —
a level tap on the muted track, so cinny captures no audio. useMutedTalkWarning
polls that flag while muted; after 1.5 s of it being true it shows one sticky
toast, "You're muted — click to unmute", whose click unmutes; it fires at most
once per mute session (re-arms after unmute → mute), is retired the moment the
mic comes back on by any means, and never runs in Push-to-Talk mode. Setting
'Warn me when I talk while muted' (default on) sits next to AFK Auto-Mute.
Verified headless with a tone mic: no toast at +1 s, toast at +4 s, click
unmutes and clears it, second mute re-arms, unmute via the button clears it,
PTT mode → nothing.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
When the Element Call frame's navigation fails (offline, blocked) it becomes a
cross-origin error page and `iframe.contentWindow.document` throws a
SecurityError. Every DOM-driven call hook (CallControl selectors, deafen DOM
fallback, useCallSpeakers/useRemoteAllMuted observers, CallEmbed theme
injection) read it unguarded, so the moment the 25 s load watchdog fired the
page logged a burst of uncaught SecurityErrors. Reproduced headless by aborting
the EC index request. One `getCallDocument()` helper now treats "can't read" as
"not loaded yet"; the watchdog overlay itself was already correct.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
- io.lotus.request_state: when the fork's lotus handlers (re)register
(an EC-side remount that doesn't unmount us) we re-send deafen, quality
and the focus pin, and the decoration pusher re-pushes its roster —
decorations and the pin no longer vanish for the rest of the call
(element-call#17).
- focus_participant carries the per-device media id from call_state
(speaking device preferred) so a multi-device user pins the right
device (element-call#30).
- injectAudio returns the fork's reply; when it refuses with
reason:"muted" the soundboard shows "Unmute your microphone…" instead
of playing the clip locally as if it went out (element-call#13).
All backwards compatible with the 0.25.0-lotus.1 bundle (unknown action
is acked; missing reply fields default to "played").
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
- C-H1: forceState only on FIRST join; on EC reconnect re-arm the fork handlers
(resendForkState — deafen+quality only) instead of clobbering live mic/video/
deafen back to the join-time snapshot.
- C-H2: AFK auto-mute reads the fork's io.lotus.call_state VAD of the LOCAL
published track instead of getUserMedia on the browser DEFAULT mic (which could
measure silence while the user spoke on another device → auto-mute an active
speaker). Fails safe (never mutes) when call_state is null OR empty.
- C-H3: control observer re-binds after EC re-renders (body subtree:true + 100ms
debounce) with an early-return so unchanged state doesn't re-render.
- C-M3 setQuality join-gated; C-M4 hangup 4s fallback dispose (idempotent);
C-M5 PTT no longer silently un-deafens; C-M6 screenshare-audio mute resets on
stop; C-L4 deafen key works in the iframe; C-L6 setState-after-unmount guards.
Reviewed (C-H2 [] fail-safe + C-H3 re-render guard applied). tsc/eslint/prettier
clean, build OK, 677 tests.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
HEAD-checks the copied denoise worklet/wasm/model assets for the selected model
and console.warns a single line listing anything missing — a silent asset skew
between the EC fork's expectations and vite's copied files would otherwise
disable noise suppression with no signal. Fire-and-forget; never blocks call
setup.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- CallEmbed sets `autoGainControl=false` for the ML noise-suppression tier so
the browser's auto gain control doesn't fight the in-source ML model; the
browser/off tiers keep AGC on.
- Docs: refresh the LOTUS_FEATURES noise-suppression section (browser-native
default, quality-ordered dropdown, DFN3 ML default, attenuation floor,
gate-after-ML, DFN level 60, AGC-off, the reliability fixes) and LOTUS_TODO
P5-30 (mark tuning/reliability/AGC done; record GTCRN as researched-and-deferred).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- Model dropdown is now ordered by quality/CPU, best first (DeepFilterNet 3 →
DTLN → RNNoise → Speex); fix RNNoise's inaccurate "High" voice-quality label.
- When a user opts into the ML tier, default to the highest-quality model
(DeepFilterNet 3). The tier default stays browser-native (known-good, best
perceived in testing so far).
- Wire the "Series Suppression" (native-NS-before-ML) toggle into the real call
path — it was applied only in the settings tester, so the tester could sound
better than the actual call. Default it OFF (a single NS stage is best
practice; it's an opt-in test aid).
- isMLDenoiseSupported now also requires WebAssembly, so ML isn't offered on
strict-CSP shells where it would silently fall back to the raw mic.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Element Call is now consumed as our self-built fork
(@lotusguild/element-call-embedded); wire up its previously-dormant
capabilities and document the fork as live.
Soundboard (P5-15): a call-bar button plays user-uploaded audio clips into the
call as a real published track (io.lotus.inject_audio) plus local playback.
Clips are uploadable like emoji/sticker packs, stored in io.lotus.soundboard
account data (synced across devices). Gated by a Settings toggle + volume.
Quality controls (P5-31): per-user mic/screenshare bitrate + screenshare
framerate (Settings -> Calls), applied via io.lotus.set_quality clamped to any
room cap. Room admins set caps and hard call-permissions (allow_screenshare /
allow_camera) in Room Settings -> Voice; the call bar hides blocked buttons.
- New: CallSoundboard, useSoundboard, soundboardClips; RoomQuality,
useCallQuality, callQuality (+ unit tests).
- Optimistic-write RoomQuality admin UI (no stale-state clobber).
- Docs: mark EC fork live across README/FEATURES/TODO/BUGS/TESTING; add D2
manual-test steps.
Numeric quality caps are client-cooperative; screenshare/camera permissions are
hard-enforced server-side (see LotusGuild/matrix voice-limit-guard).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Resolve the eslint/prettier failures from the previous commit (non-blocking
in CI, but real): drop the banned `void` operator on fire-and-forget
transport.send().catch() calls, prefix the now-unused _denoiseNativeNS
param, and run prettier on the touched files.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Switch to @lotusguild/element-call-embedded@0.20.1-lotus.1 (our self-built
fork) and turn on the source-level features it adds:
- #1 denoise CUTOVER: in-source ML denoise (lotusDenoiseSource=1) replaces
the build-time getUserMedia shim — removed the shim injection from
vite.config.js (denoise/ assets still shipped; the processor loads them).
Survives reconnects (fixes A7).
- #2 call-state: CallEmbed consumes io.lotus.call_state; useCallSpeakers /
useRemoteAllMuted prefer it over scraping EC's DOM (DOM fallback kept;
empty payloads ignored).
- #4 focus: CallControl.focusCameraParticipant sends io.lotus.focus_participant
(works during screenshare), replacing the DOM tile-click hack.
- #5 theming: lotusTransparent=1 (native transparent background).
- #6 decorations: LotusDecorationPusher sends each member's decoration URL
via io.lotus.decorations -> rendered on in-call tiles.
#3 soundboard / #7 quality ship dormant (EC-ready; no host UI sends them yet).
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- CallEmbed watchdog now SELF-HEALS: a genuine ready/joined signal arriving
after the 25s timeout clears the error and notifies subscribers with
undefined, so a slow-but-successful EC load no longer strands the user on
the recovery screen over a live call. Listener dispatch wrapped in try/catch.
- ringtones: synth notes route through a per-session master gain; stop() ramps
it to 0 so the ring is silenced instantly on answer instead of letting the
last scheduled phrase ring out over call audio.
- IncomingCallBanner: ping fires exactly once per incoming call (guarded by
refEventId) instead of re-pinging when ringtone settings change mid-banner.
- focusCameraParticipant: try multiple tile selectors (EC labels vary by
version), defer the tile click past EC's async spotlight layout switch
(rAF x2), and dev-warn when no tile matches so testers get signal.
- uploadContent: a cancelled upload (mx.cancelUpload -> AbortError) is no
longer treated as retryable — previously the retry loop could resurrect an
upload the user just cancelled. Also retry on 408.
- addRoomIdToMDirect/removeRoomIdFromMDirect: guard against a corrupt m.direct
whose values aren't arrays.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
- CallEmbed: 25s load watchdog that fails fast on iframe error / preparing-error /
timeout instead of hanging on a permanent spinner; additive onLoadError API,
cleared on ready/capabilities/joined.
- CallView: user-visible "call failed to load" overlay with Retry/Leave (folds +
tokens) via a new useCallLoadError hook.
- CallMemberCard: wrap the participant avatar in AvatarDecoration so decorations
render in the call roster (the tile rendered UserAvatar bare while member lists
already wrapped it).
Addresses LOTUS_BUGS item 3 (avatar decorations in calls) and EC iframe failure monitoring.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Pure formatting reflows (multi-line wrapping of long lines/imports/tables);
no behavior change. Clears the working tree of pending prettier diffs.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Replace raw error object logging (which may contain Matrix event
payloads, user IDs, or message bodies) with e.message-only strings
in three files:
- CallEmbed.ts: state update and event widget feed errors
- msgContent.ts: image/video element load failures and thumb errors
- RoomInput.tsx: GIF send failure
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Implement a flexible, multi-model noise suppression pipeline for Element Call/LiveKit integration:
- ML Engines: Added support for RNNoise, Speex, DTLN, and DeepFilterNet 3 models.
- Pipeline Architecture: Implemented modular audio processing in lotus-denoise.js, supporting 'Series Suppression' (running browser-native NSNet2 before ML) and a hardware-style Noise Gate.
- UI & UX Enhancements:
- Settings UI: Added model comparison chart with CPU/Quality metadata.
- Tuning: Added Live Microphone Meter for calibrating Noise Gate thresholds.
- Reporting: Added LotusToast system to alert users when ML suppression fails or falls back to raw input.
- Robustness & Quality:
- Capture Fidelity: Removed forced 48kHz capture constraints to allow native-rate capture (solving static issues with high-end audio interfaces).
- Performance: Added WASM SIMD detection with transparent fallback.
- Capability Detection: Added browser feature detection to disable unsupported ML modes.
- Build Integration: Updated Vite config to self-host all model WASM/tflite assets in /denoise/ directory.
Replace the boolean call noise-suppression setting with a 3-way control
(Off / Browser-native / ML beta) in Settings -> General -> Calls.
- Off: noiseSuppression=false to Element Call
- Browser-native: EC's built-in WebRTC suppressor (prior default)
- ML (beta): on-device RNNoise (@sapphi-red/web-noise-suppressor)
Element Call captures the mic inside its iframe and publishes to LiveKit,
so the host can't reach that track; LiveKit's Krisp filter is Cloud-only
(we self-host the SFU) and EC's own RNNoise PR #3892 is unmerged. The ML
tier instead injects a same-origin pre-init shim into the vendored EC
index.html (build/lotus-denoise.js, wired by the lotusDenoise vite plugin)
that patches getUserMedia and routes the captured mic through an RNNoise
AudioWorklet before LiveKit sees it -- the same post-capture pipeline as
#3892, with no EC fork/AGPL/rebase burden. Falls back to the raw mic if
setup fails; keeps echoCancellation/AGC on the raw capture.
- settings.ts: callNoiseSuppression -> 'off'|'browser'|'ml' + legacy
boolean migration (true->browser, false->off)
- CallEmbed/useCallEmbed: tier maps to noiseSuppression param and appends
lotusDenoise=ml (native suppressor off in ML mode)
- vite.config.js: copy RNNoise worklet/wasm + shim into the EC bundle and
inject the shim <script> before EC's module entry
- docs: LOTUS_FEATURES.md, LOTUS_TODO.md (P5-30 done)
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Omitting sendNotificationType for call rooms caused Element Call to
default to ring behavior. Now all starting-call events explicitly set
notification (or ring for DMs). Voice channels always get notification.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
downloadMedia: on 401 (SW session race or allow_redirect hop stripping
auth), retry via /_matrix/media/v3/ which is public on this homeserver
(allow_public_access_to_media_repo: true). Fixes images not loading
after sending, and avatar 401s in call prescreen tiles.
CallEmbed: inject flex-centering CSS for EC 0.19.4 participant avatar
container so the initial letter is correctly centered in its circle.
CSS class names are scoped to _avatarContainer_1mrho_40 in EC 0.19.4.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
- CallEmbed: inject :root { color-scheme } into iframe so EC respects Cinny
theme regardless of OS preference (fixes white background in dark mode)
- CallEmbed: store themeKind, update color-scheme CSS on live setTheme() calls
- CallEmbed: catch transport.send() rejection in setTheme() to prevent
unhandled promise rejection when widget not ready yet (fixes REACT-8)
- CallEmbed: html + body both set to background:none so wallpaper shows through
- CallEmbedProvider: apply chatBackground wallpaper style to call embed
container in full-view mode (not PiP) -- wallpapers carry over to calls
- useCallEmbed: pass themeKind through to CallEmbed constructor
- index.tsx: ignoreErrors: [Request timed out] to suppress matrixRTC
heartbeat timeouts (REACT-9) from Sentry noise
- README: document 0.19.4, positioning fix, dark mode fix, wallpaper,
millify Rolldown interop fix, Sentry noise filter
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Previously the listenAction wrapper only called preventDefault() to stop
the switch default from firing an error, but it never sent a reply.
The widget transport would then wait for a response until it timed out.
Now the wrapper also calls transport.reply(ev.detail, {}) to return an
immediate success, fixing io.element.join, io.element.device_mute, and
set_always_on_screen.
The base WidgetDriver throws Failed to override function for these
methods. ClientWidgetApi routes update_delayed_event widget actions to
cancelScheduledDelayedEvent, restartScheduledDelayedEvent, or
sendScheduledDelayedEvent. Without these overrides every delayed-event
refresh from element-call fails, causing MembershipManager to drop the
call after retries.
Also make listenAction auto-call preventDefault so io.element.join and
other custom widget actions return success. Add set_always_on_screen
handler so element-call PiP requests are acknowledged.
- CallEmbed: fix memory leak — mx event listeners were never removed
because dispose() called .bind(this) again, creating new function
objects. Now uses arrow class fields so start()/dispose() share the
exact same reference.
- callPreferences: toggleVideo is a no-op when cameraOnJoin=false,
preventing internal state drift from the returned value.
- CallControls: PTT key guard now blocks on SELECT elements and walks
the DOM for inherited contentEditable to prevent key interception
inside dropdowns and custom editors.
- RoomInput: GIF fetch validates Giphy CDN domain allow-list,
HTTP Content-Type header, and enforces 20 MB size cap.
- Camera no longer starts enabled when user disables it in prescreen
- When PTT mode is enabled, call starts muted so PTT works immediately
without requiring a manual mute first
- CallControlState also updated to match the forced-off audio for PTT
EC 0.16.x ignores io.element.device_mute for initial state at startup,
so audio= and video= URL params are the only reliable way to set the
initial device state before the call begins.
- Push to Talk: keydown/keyup binds mic to configurable key (default Space)
with visual PTT indicator and key-binding UI in Settings > General > Calls
- Camera always defaults OFF on join; cameraOnJoin setting for explicit opt-in
- Deafen button tooltip corrected to Deafen/Undeafen instead of Turn Off/On Sound
- Screenshare confirmation dialog before broadcasting to call participants
- Noise suppression toggle wired from settings through CallEmbed URL params
- CallControl.setMicrophone() public method for programmatic mic control
- Calls settings section added to General settings page
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
* add option to start video all in DM
* show speaker icon for dm's in call status name
* show call view if call is active in room
* add Atria call ringtone
* update element call and widget api
* add option to start voice/video call in dms
* only show call button if user have permission
* allow call widget to send call notification event
* show incoming call dialog and play sound
* fix call permission checks
* allow option to start call in all rooms
* send notification when starting call in non-voice rooms
* hide header call button from voice rooms
* prevent call join if call not supported and started by other party
* update call menu style
* show call not supported message on incoming call notification
* improve the incoming call layout
* video call with right click without opening menu
* allow call widget to fetch media url
* add webRTC missing error
* improve call permission label
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Co-authored-by: Krishan <33421343+kfiven@users.noreply.github.com>
* add option to start video all in DM
* show speaker icon for dm's in call status name
* show call view if call is active in room
* add Atria call ringtone
* update element call and widget api
* add option to start voice/video call in dms
* only show call button if user have permission
* allow call widget to send call notification event
* show incoming call dialog and play sound
* fix call permission checks
* allow option to start call in all rooms
* send notification when starting call in non-voice rooms
* hide header call button from voice rooms
* prevent call join if call not supported and started by other party
* update call menu style
* show call not supported message on incoming call notification
* improve the incoming call layout
* video call with right click without opening menu
* allow call widget to fetch media url
* add webRTC missing error
* improve call permission label
---------
Co-authored-by: Krishan <33421343+kfiven@users.noreply.github.com>
* allow user to end call if error when loading
* show call support missing error if livekit server is not provided
* prevent joining from nav item double click if no livekit support
* allow user to end call if error when loading
* show call support missing error if livekit server is not provided
* prevent joining from nav item double click if no livekit support
Lotus CI
jaredandClaude Opus 5
KrishanandAjay Bura
Lotus BotandClaude Sonnet 4.6
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