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
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
---------
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
* add mutation observer hok
* add hook to read speaking member by observing iframe content
* display speaking member name in call status bar and improve layout
* fix shrining
* add joined call control bar
* remove chat toggle from room header
* change member speaking icon to mic
* fix joined call control appear in other
* show spinner on end call button
* hide call statusbar for mobile view when room is selected
* make call statusbar more mobile friendly
* fix call status bar item align
* add mutation observer hok
* add hook to read speaking member by observing iframe content
* display speaking member name in call status bar and improve layout
* fix shrining
* add joined call control bar
* remove chat toggle from room header
* change member speaking icon to mic
* fix joined call control appear in other
* show spinner on end call button
* hide call statusbar for mobile view when room is selected
* make call statusbar more mobile friendly
* fix call status bar item align
* add mutation observer hok
* add hook to read speaking member by observing iframe content
* display speaking member name in call status bar and improve layout
* fix shrining
* add mutation observer hok
* add hook to read speaking member by observing iframe content
* display speaking member name in call status bar and improve layout
* fix shrining
* Add users on the nav to showcase call activity and who is in the call
* add check to prevent DCing from the call you're currently in...
* Add avatar and username for the space (needs to be moved into RoomNavItem proper)
* Add background variant to buttons
* Update hook to keep method signature (accepting an array of Rooms instead) to support multiple room event tracking of the same event
* Add state listener so the call activity is real time updated on joins/leaves within the space
* Add RoomNavUser for displaying the user avatar + name in the nav for a visual of call activity and participants
* rename CallNavBottom to CallNavStatus
* Rename callnavbottom and fix linking implementation to actually be correct
* temp fix to allow the status to be cleared in some way
* re-add background to active call link button
* prepare to feed this to child elements for visibility handling
* loosely provide nav handling for testing refactoring
* Add CallView
* Update to funnel Outlet context through for Call handling (might not be the best approach, but removes code replication in PersistentCallContainer where we were remaking the roomview entirely)
* update client layout to funnel outlet the iframes for the call container
* funnel through just iframe for now for testing sake
* Update room to use CallView
* Pass forward the backupIframeRef now
* remove unused params
* Add backupIframeRef so we can re-add the lobby screen for non-joined calls (for viewing their text channels)
* Remove unused imports and restructure to support being parent to clientlayout
* Re-add layout as we're no longer oddly passing outlet context
* swap to using ref provider context from to connect to persistentcallcontainer more directly
* Revert to original code as we've moved calling to be more inline with design
* Revert to original code as we've moved the outlet context passing out and made more direct use of the ref
* Fix unexpected visibility in non-room areas
* correctly provide visibility
* re-add mobile chat handling
* Improve call room view stability
* split into two refs
* add ViewedRoom usage
* Disable
* add roomViewId and related
* (broken) juggle the iframe states proper... still needs fixing
* Conditionals to manage the active iframe state better
* add navigateRoom to be in both conditions for the nav button
* Fix the view to correctly display the active iframe based on which is currently hosting the active call (juggling views)
* Testing the iframe juggling. Seems to work for the first and second joins... so likely on the right path with this
* add url as a param for widget url
* fix backup iframe visibility
* Much closer to the call state handling we want w/ hangups and joins
* Fix the position of the member drawer to its correct location
* Ensure drawer doesn't appear in call room
* Better handling of the isCallActive in the join handler
* Add ideal call room join behavior where text rooms to call room simply joins, but doesn't swap current view
* Fix mobile call room default behavior from auto-join to displaying lobby
* swap call status to be bound to call state and not active call id
* Remove clean room ID and add default handler for if no active call has existed yet, but user clicks on show chat
* Applies the correct changes to the call state and removes listeners of old active widget so we don't trigger hang ups on the new one (the element-call widget likes to spam the hang up response back several times for some reason long after you tell it to hang up)
* Remove superfluous comments and Date.now() that was causing loading... bug when widgetId desynced
* Remove Date.now() that was causing widgetId desync
* add listener clearing, camel case es lint rule exception, remove unneeded else statements
* Remove unused
* Add widgetId as a getWidgetUrl param
* Remove no longer needed files
* revert ternary expression change and add to dependency array
* add widgetId to correct pos in getWidgetUrl usage
* Remove CallActivation
* Move and rename RoomCallNavStatus
* update imports and dependency array
* Rename and clean up
* Moved CallProvider
* Fix spelling mistake
* Fix to use shorthand prop
* Remove unneeded logger.errors
* Fixes element-call embedded support (but it seems to run poorly)
* null the default url so that we fallback to the embedded version (would recommend hosting it until performance issue is determined)
* Fix vite build to place element-call correctly for embedded npm package support
* add vite preview as an npm script
* Move files to more correct location
* Add package-lock changes
* Set dep version to exact
* Fix path issue from moving file locations
* Sets initial states so the iframes don't cause the other to fail with the npm embedded package
* Revert navitem change
* Just check for state on both which should only occur at initial
* Fixes call initializing by default on mobile
* Provides correct behavior when call isn't active and no activeClientWidgetApi exists yet
* Corrects the state for the situations where both iframes are "active" (not necessarily visible)
* Reduce code reuse in handleJoin
* Seems to sort out the hangup status button bug the occurred after joining a call via lobby
* Re-add the default view current active room behavior
* Remove repetitive check
* Add storing widget for comparing with (since we already store room id and the clientWidgetApi anyway)
* Update rendering logic to clear up remaining rendering bug (straight to call -> lobby of another room and joining call from that interface -> lobby of that previous room and joining was leading to duplication of the user in lobbies. This was actually from listening to and acknowledging hangups from the viewed widget in CallProvider)
* Prevent null rooms from ever rendering
* This seems to manage the hangup state with the status bar button well enough that black screens should never be encountered
* Remove viewed room setting here and pass the room to hang up (seems state doesn't update fast enough otherwise)
* Remove unused
* Properly declare new hangup method sig
* Seems to avoid almost all invalid states (hang up while viewing another lobby and hitting join seems to black screen, sets the active call as the previous active room id, but does join the viewed room correctly)
* Fix for cases where you're viewing a lobby and hang up your existing call and try to join lobby (was not rendering for the correct room id, but was joining the correct call prior)
* Re-add intended switching behavior
* More correct filter (viewedRoom can return false on that compare in some cases)
* Seems to shore up the remaining state issues with the status bar hangup
* Fix formatting
* In widget hang up button should be handled correct now
* Solves the CHCH sequence issue, CLJH remains
* Fixes CLJH, found CCH
* Solves CCH. Looks like CLCH left
* A bit of an abomination, but adds a state counter to iteratively handle the diverse potential states (where a user can join from the nav bar or the join button, hang up from either as well, and account for the juggling iframes)
Black screens shouldn't be occurring now.
* Fix dependency array
* Technically corrects the hangup button in the widget, should be more precise though
* Bind the on messaging iframe for easier access in hangup/join handling
* Far cleaner and more sensible handling of the call window... I just really don't like the idea of sending a click event, but right now the element-call code treats preload/skipLobby hangups (sent from our end) as if they had no lobby at all and thus black screens. Other implementation was working around that without just sending a click event on the iframe's hangup button.
* Fixes a bug where if you left a call then went to a lobby and joined it didn't update the actual activeCallRoomId
* Fixes complaints of null contentDocument in iframe
* Update to use new icons (thank you)
* Remove unneeded prop
* Re-arrange more options and add checks for each option to see if it is a call room (probably should manage a state to see if a header is already on screen and provide a slightly modified visual based on that for call rooms)
* Invert icons to show the state instead of the action they will perform (more visual clarity)
* Update src/app/features/room-nav/RoomCallNavStatus.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomCallNavStatus.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomCallNavStatus.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomCallNavStatus.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomCallNavStatus.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomCallNavStatus.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room/RoomViewHeader.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavItem.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavUser.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavUser.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room-nav/RoomNavUser.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/pages/client/space/Space.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/call/CallView.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/call/CallView.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/call/CallView.tsx
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* Update src/app/features/room/RoomView.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room/RoomView.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* adjust room header for calling
* Remove No Active Call text when not in a call
* update element-call version
* Update src/app/features/room/RoomViewHeader.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room/RoomViewHeader.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room/RoomViewHeader.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room/RoomViewHeader.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Update src/app/features/room/RoomViewHeader.tsx
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
* Revert most changes to Space.tsx
* Show call room even if category is collapsed
* changes to RoomNavItem, RoomNavUser and add useCallMembers
* Rename file, sprinkle in the magic one line for matrixRTCSession. and remove comment block
* swap userId to callMembership as a prop and add a nullchecked userId that uses the membership sender
* update references to use callMembership instead
* Simplify RoomNavUser
Discard future functionality since it probably won't exist in time for merging this PR
* Simplify RoomViewHeader.tsx
Remove unused UI elements that don't have implemented functionality. Replace custom function for checking if a room is direct with a standard hook.
* Update Room.tsx to accomodate restructuring of Room, RoomView and CallView
* Update RoomView.tsx to accomodate restructuring of Room, RoomView and CallView
* Update CallView.tsx to accomodate restructuring of Room, RoomView and CallView + suggested changes
* add call related permissions to room permissions
* bump element call to 0.16.3, apply cinny theme to element call ui, replace element call lobby (backup iframe) with custom ui and only use element call for the in-call ui
* update text spacing
* redo roomcallnavstatus ui, force user preferred mute/video states when first joining calls, update variable names and remove unnecessary logic
* set default mic state to enabled
* clean up ts/eslint errors
* remove debug logs
* format using prettier rules from project prettierrc
* fix: show call nav status while active call is ongoing
* fix: clean up call nav/call view console warnings
* fix: keep call media controls visible before joining
* fix: restore header icon button fill behavior
Fixes regression from b074d421b66eb4d8b600dfa55b967e6c4f783044.
* style: blend header and room input button styles in call nav
* fix page header background color on room view header
* fix: permissions and room icon resolution (#2)
* Initialize call state upon room creation for call rooms, remove subsequent useless permission
* handle case of missing call permissions
* use call icon for room item summary when room is call room
* replace previous icon src resolution function with a more robust approach
* replace usages of previous icon resolution function with new implementation
* fix room name not updating for a while when changed
* set up framework for room power level overrides upon room creation
* override join call permission to all members upon room creation
* fix broken usages of RoomIcon
* remove unneeded import
* remove unnecessary logic
* format with prettier
* feat: show connected/connecting call status
* fix: preserve navigation context when opening non-call rooms
* fix: reset room name state when room instance changes
* feat: Disable webcam by default using callIntent='audio'
* Add channel type selecor
* Add option for voice rooms, which for now sets the default selected
option in the creation modal
* Add proper support for room selection from the enu
* Move enums to `types.ts` and change icons selection to use
`getRoomIconSrc`
* fix: group duplicate conditions into one
* fix: typo
* refactor: rename kind/voice to access/type and simplify room creation
- rename CreateRoomVoice to CreateRoomType and modal voice state to type
- rename CreateRoomKind to CreateRoomAccess and KindSelector to AccessSelector
- propagate access/defaultAccess through create room and create space forms
- set voice room power levels via createRoom power_level_content_override
* refactor: unify join rule icon mapping and update call/space icons
- bump folds from 2.5.0 to 2.6.0
- replace separate room/space join-rule icon hooks with useJoinRuleIcons(roomType)
- route join-rule icons through getRoomIconSrc for consistent room type handling
- simplify getRoomIconSrc by removing the locked override path
- use VolumeHighGlobe for public call rooms and VolumeHighLock for private call rooms
* chore(deps): bump matrix-widget-api to 1.17 and remove react-sdk-module-api
* fix: adapt SmallWidget to matrix-widget-api 1.17.0 API
* fix: render call room chat only when chat panel is open
* fix(permissions): show call settings permissions only for call rooms
* refactor: remove redundant room-nav props/guards and minor naming cleanup
* fix: use PhoneDown icon for hang up action
* chore(hooks): remove unused useStateEvents hook
* fix(room): enable members drawer toggle in desktop call rooms
- show filled User icon when the drawer is open
* Revert "fix: adapt SmallWidget to matrix-widget-api 1.17.0 API"
This reverts commit a4c34eff8a.
* fix: semi-revert matrix-widget-api 1.17 bump and migrate to 1.13 API
* fix(call): wait for Element Call contentLoaded before widget handshake
- fixes not working on firefox
* fix missing imports
* improve create room type design and add beta badge for voice room
* add beta badge for voice room in space lobby
* fix create room modal title
* pass missing roomType param to roomicon component
* add roomtype
* Add deafen functionality (#2695)
* feat:(deafen functionality)
* feat:(reworked voice controls for deafen)
* ref:(use muted instead of volume for deafen)
* fix:(backpedal audio_enabled rename)
* ref:(renaming of deafened vars)
* add stack avatar component
* add call status bar - WIP
* remove call status from navigation drawer
* fix deprecated method use in use call members hook
* render new call status bar
* move call widget driver to plugins
* remove old status bar usage from navigation drawer
* add call session and joined hook
* remove unknown changes
* upgrade widget api
* add element call embed plugin
* remove unknown change
* add call embed atom
* add call embed hooks and context
* add call embed provider
* replace old call implementation
* stop joining other call on second click if already in a call
* refactor embed placement hook
* add merge border prop to sequence card
* add call preferences
* add prescreen to call view - WIP
* prevent joining new call if already in call
* make call layout adaptive
* render call chat as right panel
* show call members in prescreen
* render call join leave event in timeline
* remove unknown rewrite in docker-nginx file
* render call event without hidden event enable
---------
Co-authored-by: Gigiaj <gigiaboone@yahoo.com>
Co-authored-by: Jaggar <18173108+GigiaJ@users.noreply.github.com>
Co-authored-by: Gimle Larpes <97182804+GimleLarpes@users.noreply.github.com>
Co-authored-by: Gimle Larpes <gimlelarpes@gmail.com>
Co-authored-by: YoJames2019 <jamesclark1700@gmail.com>
Co-authored-by: YoJames2019 <yobiscuit0@gmail.com>
Co-authored-by: hazre <mail@haz.re>
Co-authored-by: haz <37149950+hazre@users.noreply.github.com>
Co-authored-by: Ajay Bura <32841439+ajbura@users.noreply.github.com>
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Co-authored-by: James Reilly <jreilly1821@gmail.com>
Co-authored-by: Tymek <vonautymek@gmail.com>
Co-authored-by: Thedustbuster <92692948+Thedustbustr@users.noreply.github.com>
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