merge: fold in lotus commits e36aef8a..cbd42bf9 landed during the v0.25.0 sync

Six fix(lotus) commits reached `lotus` while the upstream merge was in
progress (deafen via global output mute, denoise race/fallback fixes,
strictOriginCheck on the widget transport, call_state standalone skip,
set_quality null-clears-cap, plus four new src/lotus/*.test.ts files).
Merged them on top of the v0.25.0 sync so this branch is a superset of
current `lotus`.

One conflict, src/lotus/lotusDenoiseProcessor.ts: the sync commit had
dropped a meaningless `void` (oxlint no-meaningless-void-operator) on a
line the incoming commit also touched; kept the void-less form so oxlint
stays clean.

Verification after the fold-in (Node 24.11.1 / pnpm 11.21.0): tsc clean;
oxlint clean; oxfmt --check clean; knip exit 0; vitest unit 88 files /
639 passed / 9 skipped; build:embedded OK, staged to embedded/web/dist,
all six io.lotus.* actions present in the bundle.

Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA

src/lotus/lotusCallState.ts

@@ -14,6 +14,7 @@ import {
 } from "rxjs/operators";
 
 import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
+import { widget } from "../widget";
 import { LotusWidgetActions, lotusFlag, lotusSendToHost } from "./lotusWidget";
 
 interface ParticipantState {
@@ -37,6 +38,10 @@ interface ParticipantState {
  */
 export function startLotusCallState(vm: CallViewModel): () => void {
   if (!lotusFlag("lotusCallState")) return () => undefined;
+  // [lotus] Standalone (non-widget) mode has no host to send state to;
+  // skip building the whole stream pipeline, mirroring lotusFocus.ts /
+  // lotusDecorations.ts.
+  if (!widget) return () => undefined;
 
   const sub: Subscription = vm.userMedia$
     .pipe(
@@ -67,11 +72,12 @@ export function startLotusCallState(vm: CallViewModel): () => void {
               ),
             ),
       ),
-      // `speaking` flips rapidly; cap the send rate and drop no-op repeats.
-      // 250ms is plenty for speaking rings / mute badges and keeps the
-      // request/response widget traffic modest.
-      throttleTime(250, undefined, { leading: true, trailing: true }),
+      // `speaking` flips rapidly; drop no-op repeats BEFORE throttling so
+      // the throttle window isn't spent re-emitting an unchanged value, then
+      // cap the send rate. 250ms is plenty for speaking rings / mute badges
+      // and keeps the request/response widget traffic modest.
       distinctUntilChanged((a, b) => JSON.stringify(a) === JSON.stringify(b)),
+      throttleTime(250, undefined, { leading: true, trailing: true }),
     )
     .subscribe((participants) => {
       lotusSendToHost(LotusWidgetActions.CallState, { participants });

src/lotus/lotusDeafen.test.ts

@@ -0,0 +1,154 @@
+/*
+Copyright 2026 Lotus Guild
+
+SPDX-License-Identifier: AGPL-3.0-only OR LicenseRef-Element-Commercial
+Please see LICENSE in the repository root for full details.
+*/
+
+import { EventEmitter } from "events";
+import { afterEach, beforeEach, expect, test, vi } from "vitest";
+import { of } from "rxjs";
+import { Track } from "livekit-client";
+
+import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
+import { setAudioEnabled$ } from "../controls";
+import { startLotusDeafen } from "./lotusDeafen";
+import { LotusWidgetActions } from "./lotusActions";
+
+const lazyActions = new EventEmitter();
+
+vi.mock("../widget", () => ({
+  widget: {
+    api: { transport: { reply: vi.fn().mockResolvedValue(undefined) } },
+    // Getter: `vi.mock` factories run at import time, before the const above.
+    get lazyActions(): EventEmitter {
+      return lazyActions;
+    },
+  },
+}));
+
+/** Minimal CallViewModel stub: no connections, so no livekit rooms. */
+function mockVm(participants: unknown[] = []): CallViewModel {
+  const livekitRoom = {
+    remoteParticipants: new Map(participants.map((p, i) => [String(i), p])),
+    on: vi.fn(),
+    off: vi.fn(),
+  };
+  return {
+    allConnections$: of({ getConnections: () => [{ livekitRoom }] }),
+  } as unknown as CallViewModel;
+}
+
+function send(data: unknown): void {
+  lazyActions.emit(LotusWidgetActions.SetDeafen, {
+    detail: { data },
+  });
+}
+
+/** Records everything pushed to the global audio-output subject. */
+let emissions: boolean[];
+let sub: { unsubscribe: () => void };
+
+beforeEach(() => {
+  emissions = [];
+  sub = setAudioEnabled$.subscribe((v) => emissions.push(v));
+});
+
+afterEach(() => {
+  sub.unsubscribe();
+  lazyActions.removeAllListeners();
+});
+
+test("deafen mutes, and undeafen unmutes, EC's global audio output", () => {
+  const stop = startLotusDeafen(mockVm());
+
+  send({ deafened: true, screenshareAudioMuted: false });
+  expect(emissions).toEqual([false]);
+
+  send({ deafened: false, screenshareAudioMuted: false });
+  expect(emissions).toEqual([false, true]);
+
+  stop();
+  expect(emissions).toEqual([false, true]);
+});
+
+test("re-sending the same state is idempotent (host resendForkState)", () => {
+  const stop = startLotusDeafen(mockVm());
+
+  send({ deafened: true, screenshareAudioMuted: false });
+  send({ deafened: true, screenshareAudioMuted: false });
+  send({ deafened: true, screenshareAudioMuted: false });
+  expect(emissions).toEqual([false]);
+
+  stop();
+  // Teardown restores the user's own state, which was "audio enabled".
+  expect(emissions).toEqual([false, true]);
+});
+
+test("undeafen does not re-enable audio the user had muted themselves", () => {
+  const stop = startLotusDeafen(mockVm());
+
+  // The user mutes all audio through EC's own control first.
+  setAudioEnabled$.next(false);
+  expect(emissions).toEqual([false]);
+
+  send({ deafened: true, screenshareAudioMuted: false });
+  send({ deafened: false, screenshareAudioMuted: false });
+  // Nothing further was pushed: audio was already off and stays off.
+  expect(emissions).toEqual([false]);
+
+  stop();
+  expect(emissions).toEqual([false]);
+});
+
+test("deafen never touches per-participant volume", () => {
+  const participant = { setVolume: vi.fn() };
+  const stop = startLotusDeafen(mockVm([participant]));
+
+  send({ deafened: true, screenshareAudioMuted: false });
+  send({ deafened: false, screenshareAudioMuted: false });
+  expect(participant.setVolume).not.toHaveBeenCalled();
+
+  stop();
+});
+
+test("screenshare-audio mute is applied per source and only undone for participants we muted", () => {
+  const participant = { setVolume: vi.fn() };
+  const stop = startLotusDeafen(mockVm([participant]));
+
+  // Not muted yet: no volume writes at all.
+  send({ deafened: false, screenshareAudioMuted: false });
+  expect(participant.setVolume).not.toHaveBeenCalled();
+
+  send({ deafened: false, screenshareAudioMuted: true });
+  expect(participant.setVolume).toHaveBeenCalledWith(
+    0,
+    Track.Source.ScreenShareAudio,
+  );
+
+  participant.setVolume.mockClear();
+  send({ deafened: false, screenshareAudioMuted: false });
+  expect(participant.setVolume).toHaveBeenCalledWith(
+    1,
+    Track.Source.ScreenShareAudio,
+  );
+
+  // Releasing again writes nothing: we no longer own that participant.
+  participant.setVolume.mockClear();
+  send({ deafened: false, screenshareAudioMuted: false });
+  expect(participant.setVolume).not.toHaveBeenCalled();
+
+  stop();
+});
+
+test("a partial payload only moves the flag it names", () => {
+  const stop = startLotusDeafen(mockVm());
+
+  send({ deafened: true });
+  expect(emissions).toEqual([false]);
+  send({ screenshareAudioMuted: true });
+  // Still deafened: no change to the global mute.
+  expect(emissions).toEqual([false]);
+
+  stop();
+});

src/lotus/lotusDeafen.ts

@@ -15,29 +15,52 @@ import { logger } from "matrix-js-sdk/lib/logger";
 import { type IWidgetApiRequest } from "matrix-widget-api";
 
 import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
+import { setAudioEnabled$ } from "../controls";
 import { widget } from "../widget";
 import { LotusWidgetActions } from "./lotusActions";
 
 /**
- * Handle the host's `io.lotus.set_deafen` toWidget action: silence remote audio
- * at the LiveKit source, replacing cinny's brittle iframe-DOM `.muted` hack
- * (which fought MatrixAudioRenderer and broke on re-render / late tracks).
+ * Handle the host's `io.lotus.set_deafen` toWidget action, replacing cinny's
+ * brittle iframe-DOM `.muted` hack (which fought MatrixAudioRenderer and broke
+ * on re-render / late tracks).
  *
- * `deafened` mutes every remote participant's microphone AND screenshare audio;
- * `screenshareAudioMuted` mutes only the screenshare audio (so the host can
- * drop shared-tab/game audio while still hearing voices). Volume is set PER
- * SOURCE via `RemoteParticipant.setVolume(volume, source)` — whose verified
- * signature in livekit-client ^2.18.1 is
- * `setVolume(volume, source?: Track.Source.Microphone | Track.Source.ScreenShareAudio)`
- * and DEFAULTS `source` to `Microphone` (NOT "all audio"), so each source must
- * be set explicitly. setVolume records the value in the participant's
- * `volumeMap`, so a track that (re)subscribes later re-applies it automatically.
+ * `deafened` drives EC's OWN global audio output mute: `setAudioEnabled$` feeds
+ * `muteAllAudio$` (`src/state/MuteAllAudioModel.ts`), which `InCallView` already
+ * passes as `muted` to every `LivekitRoomAudioRenderer` (and hence
+ * `MatrixAudioRenderer`), plus `CallEventAudioRenderer` and
+ * `ReactionsAudioRenderer`. That silences EVERY remote source — microphone,
+ * screenshare audio and `Track.Source.Unknown` soundboard clips — and needs no
+ * per-participant bookkeeping for late joiners or reconnects.
  *
- * State is closure-scoped (per invocation, matching the sibling lotus modules)
- * and re-applied to every current room's participants on change, and to LATE
- * JOINERS via `RoomEvent.ParticipantConnected`. The cinny host re-sends the
- * current state on every call join (CallControl.forceState), so a fresh call
- * never inherits a previous call's deafen state.
+ * This deliberately does NOT use `RemoteParticipant.setVolume` for deafen any
+ * more: EC's per-participant volume slider / per-tile mute writes the same
+ * `volumeMap` from `createVolumeControls` (`src/state/VolumeControls.ts`)
+ * whenever its `sink$` re-emits (every join, every participant re-resolution),
+ * so a `setVolume(0)` deafen was silently undone for anyone joining while
+ * deafened, and an undeafen `setVolume(1)` clobbered the user's own per-tile
+ * volume/mute state.
+ *
+ * `screenshareAudioMuted` is a NARROWER, independent host control (drop shared
+ * tab/game audio while still hearing voices). EC has no global per-source mute,
+ * so that one still has to go through
+ * `RemoteParticipant.setVolume(volume, Track.Source.ScreenShareAudio)` — whose
+ * verified signature in livekit-client ^2.18.1 is
+ * `setVolume(volume, source?: Track.Source.Microphone | Track.Source.ScreenShareAudio)`.
+ * It is re-applied to late joiners via `RoomEvent.ParticipantConnected`, and we
+ * only ever restore participants we muted ourselves. Known limitation: EC's own
+ * screenshare volume slider writes the same `volumeMap`, so a user who moves
+ * that slider while screenshare audio is host-muted wins; the mic path (the
+ * actual deafen) is no longer affected by that race at all.
+ *
+ * Undeafen restores the user's OWN output-enabled state as it was before the
+ * deafen (and never touches the `mute-all-audio` setting), so a user who had
+ * already muted all audio themselves stays muted.
+ *
+ * State is closure-scoped (per invocation, matching the sibling lotus modules).
+ * Applying the same state twice is a no-op, so the host's
+ * `CallControl.resendForkState()` after a reconnect is safe. The host re-sends
+ * the current state on every call join (CallControl.forceState), so a fresh
+ * call never inherits a previous call's deafen state.
  *
  * No effect unless the host sends the action. Returns a teardown function.
  */
@@ -48,32 +71,69 @@ export function startLotusDeafen(vm: CallViewModel): () => void {
   let deafened = false;
   let screenshareAudioMuted = false;
 
+  // The user's own audio-output state, tracked from `setAudioEnabled$` so an
+  // undeafen restores it rather than blindly enabling output. Defaults to
+  // `true`, matching `muteAllAudio$`'s `startWith(true)`.
+  let userAudioEnabled = true;
+  // Ignore our own (synchronous) emissions while tracking the user's state, and
+  // skip pushes that would not change anything so re-applying the same state
+  // (the host's `resendForkState()` after a reconnect) is a no-op.
+  let selfEmitting = false;
+  let lastApplied: boolean | null = null;
+
+  const applyGlobalMute = (): void => {
+    const desired = deafened ? false : userAudioEnabled;
+    // `lastApplied` is our own override, if any; otherwise the live state is
+    // whatever the user last set.
+    const current = lastApplied ?? userAudioEnabled;
+    lastApplied = desired;
+    if (desired === current) return;
+    selfEmitting = true;
+    try {
+      setAudioEnabled$.next(desired);
+    } finally {
+      selfEmitting = false;
+    }
+  };
+
+  const audioSub = setAudioEnabled$.subscribe((enabled) => {
+    if (selfEmitting) return;
+    userAudioEnabled = enabled;
+    if (deafened && enabled) {
+      // Something else (the native output controls) re-enabled audio while
+      // deafened — re-assert the mute instead of silently losing deafen.
+      lastApplied = true;
+      applyGlobalMute();
+    }
+  });
+
+  // Participants whose screenshare audio WE turned down, so undoing the host's
+  // screenshare mute never writes a volume to anyone else.
+  const screenshareMuted = new Set<RemoteParticipant>();
+
   const applyToParticipant = (p: RemoteParticipant): void => {
-    p.setVolume(deafened ? 0 : 1, Track.Source.Microphone);
-    p.setVolume(
-      deafened || screenshareAudioMuted ? 0 : 1,
-      Track.Source.ScreenShareAudio,
-    );
-    // NOTE: injected/soundboard audio (published as `Track.Source.Unknown`) is
-    // deliberately NOT silenced here. The verified `setVolume` type signature
-    // only accepts `Track.Source.Microphone | Track.Source.ScreenShareAudio`,
-    // so passing `Unknown` would fail the fork's `tsc` gate and require an
-    // unsafe cast. Soundboard clips are short, host-triggered content the host
-    // already controls at the inject source, so leaving them audible is the
-    // type-clean, safe choice (a full-parity "mute everything" is not needed
-    // for the deafen semantics: don't-hear-other-people's-voices).
+    if (screenshareAudioMuted) {
+      p.setVolume(0, Track.Source.ScreenShareAudio);
+      screenshareMuted.add(p);
+    } else if (screenshareMuted.delete(p)) {
+      p.setVolume(1, Track.Source.ScreenShareAudio);
+    }
   };
 
   const applyToRoom = (room: LivekitRoom): void =>
     room.remoteParticipants.forEach(applyToParticipant);
 
   // Per-room ParticipantConnected listeners, so LATE JOINERS pick up the
-  // current deafen state the moment they connect.
+  // current screenshare-audio mute the moment they connect. Drive off the
+  // local participant's connection(s), not `livekitRoomItems$` — that stream is
+  // empty until a matrix-validated REMOTE member resolves, so listeners would
+  // be attached too late for the first joiner (the sibling lotus modules all
+  // use `allConnections$` for the same reason).
   const roomListeners = new Map<LivekitRoom, () => void>();
   let rooms: LivekitRoom[] = [];
 
-  const sub = vm.livekitRoomItems$.subscribe((items) => {
-    const next = items.map((i) => i.livekitRoom);
+  const sub = vm.allConnections$.subscribe((data) => {
+    const next = data.getConnections().map((c) => c.livekitRoom);
     rooms = next;
     // Detach listeners for rooms that went away.
     for (const [room, off] of roomListeners) {
@@ -107,6 +167,7 @@ export function startLotusDeafen(vm: CallViewModel): () => void {
     logger.debug(
       `[lotus] set_deafen: deafened=${deafened} screenshareAudioMuted=${screenshareAudioMuted}`,
     );
+    applyGlobalMute();
     rooms.forEach(applyToRoom);
   };
 
@@ -115,6 +176,13 @@ export function startLotusDeafen(vm: CallViewModel): () => void {
     sub.unsubscribe();
     for (const off of roomListeners.values()) off();
     roomListeners.clear();
+    // Leave the user's own output state as they had it before deafen.
+    if (deafened) {
+      deafened = false;
+      applyGlobalMute();
+    }
+    screenshareMuted.clear();
+    audioSub.unsubscribe();
     w.lazyActions.off(LotusWidgetActions.SetDeafen, handler);
   };
 }

src/lotus/lotusDenoise.test.ts

@@ -0,0 +1,141 @@
+/*
+Copyright 2026 Lotus Guild
+
+SPDX-License-Identifier: AGPL-3.0-only OR LicenseRef-Element-Commercial
+Please see LICENSE in the repository root for full details.
+*/
+
+import { afterEach, beforeEach, describe, expect, test, vi } from "vitest";
+import { ParticipantEvent, Track } from "livekit-client";
+
+import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
+import { startLotusDenoise } from "./lotusDenoise";
+
+// Track constructed instances so tests can assert exactly one processor was
+// built across racing apply() calls, and can assert the pending one is
+// destroyed on early teardown. `vi.hoisted` is required because `vi.mock`
+// factories are hoisted above this file's other top-level statements.
+const instances = vi.hoisted(
+  () => [] as { destroy: ReturnType<typeof vi.fn> }[],
+);
+vi.mock("./lotusDenoiseProcessor", () => ({
+  LotusDenoiseProcessor: class {
+    public destroy = vi.fn().mockResolvedValue(undefined);
+    public constructor() {
+      instances.push(this);
+    }
+  },
+}));
+
+/** A promise plus externally-callable resolve, for controlling ordering. */
+function deferred<T>(): {
+  promise: Promise<T>;
+  resolve: (v: T) => void;
+} {
+  let resolve!: (v: T) => void;
+  const promise = new Promise<T>((r) => {
+    resolve = r;
+  });
+  return { promise, resolve };
+}
+
+function makeRoomAndVm(mic: {
+  getProcessor: () => unknown;
+  setProcessor: ReturnType<typeof vi.fn>;
+}): {
+  vm: CallViewModel;
+  room: { localParticipant: Record<string, unknown> };
+  firePublished: () => void;
+} {
+  const handlers = new Map<string, () => void>();
+  const localParticipant = {
+    getTrackPublication: (
+      source: Track.Source,
+    ): { track: typeof mic } | undefined =>
+      source === Track.Source.Microphone ? { track: mic } : undefined,
+    on: (event: string, cb: () => void): Map<string, () => void> =>
+      handlers.set(event, cb),
+    off: (event: string): boolean => handlers.delete(event),
+  };
+  const room = { localParticipant };
+  const vm = {
+    allConnections$: {
+      subscribe: (
+        cb: (data: {
+          getConnections: () => { livekitRoom: unknown }[];
+        }) => void,
+      ) => {
+        cb({ getConnections: () => [{ livekitRoom: room }] });
+        return { unsubscribe: (): void => undefined };
+      },
+    },
+  } as unknown as CallViewModel;
+  return {
+    vm,
+    room,
+    firePublished: () => handlers.get(ParticipantEvent.LocalTrackPublished)?.(),
+  };
+}
+
+beforeEach(() => {
+  instances.length = 0;
+  // `lotusParam`/`lotusFlag` cache the URL params at first read; seed the hash
+  // before startLotusDenoise() so the dedicated `lotusDenoiseSource` flag reads on.
+  window.location.hash = "#/room?lotusDenoiseSource=1";
+});
+
+afterEach(() => {
+  vi.resetModules();
+});
+
+describe("startLotusDenoise", () => {
+  test("racing LocalTrackPublished events only construct one processor", async () => {
+    const setProcessorDeferred = deferred<void>();
+    let attached: unknown;
+    const mic = {
+      getProcessor: (): unknown => attached,
+      setProcessor: vi.fn(async (p: unknown) => {
+        await setProcessorDeferred.promise;
+        attached = p;
+      }),
+    };
+    const { vm, firePublished } = makeRoomAndVm(mic);
+
+    startLotusDenoise(vm);
+    // Simulate a second LocalTrackPublished (e.g. camera) firing before the
+    // first setProcessor() has resolved.
+    firePublished();
+
+    expect(mic.setProcessor).toHaveBeenCalledTimes(1);
+    expect(instances).toHaveLength(1);
+
+    setProcessorDeferred.resolve();
+    await Promise.resolve();
+    await Promise.resolve();
+
+    // Once attached, a further publish should be a no-op (mic.getProcessor()
+    // is now set).
+    firePublished();
+    expect(mic.setProcessor).toHaveBeenCalledTimes(1);
+  });
+
+  test("destroys the in-flight processor if torn down before setProcessor resolves", async () => {
+    const setProcessorDeferred = deferred<void>();
+    const mic = {
+      getProcessor: (): unknown => undefined,
+      setProcessor: vi.fn(async () => setProcessorDeferred.promise),
+    };
+    const { vm } = makeRoomAndVm(mic);
+
+    const teardown = startLotusDenoise(vm);
+    expect(instances).toHaveLength(1);
+
+    teardown();
+
+    expect(instances[0].destroy).toHaveBeenCalledTimes(1);
+
+    // Resolving afterwards must not throw/reject unhandled.
+    setProcessorDeferred.resolve();
+    await Promise.resolve();
+  });
+});

src/lotus/lotusDenoise.ts

@@ -97,13 +97,29 @@ export function startLotusDenoise(vm: CallViewModel): () => void {
     room.localParticipant.getTrackPublication(Track.Source.Microphone)
       ?.track as LocalAudioTrack | undefined;
 
+  // [lotus] `mic.getProcessor()` only becomes set once `setProcessor()`
+  // resolves — i.e. after the whole wasm/model load. LiveKit fires
+  // `LocalTrackPublished` once per local track (mic, then camera on join with
+  // video), so two calls to `apply()` can both observe `!mic.getProcessor()`
+  // and race to construct a second `LotusDenoiseProcessor` (a second
+  // AudioContext + model load) before the first has attached. Track an
+  // in-flight setProcessor per room and skip `apply()` while one is pending.
+  const pendingProcessors = new Map<LivekitRoom, LotusDenoiseProcessor>();
+
   const apply = (room: LivekitRoom): void => {
     const mic = micOf(room);
-    if (mic && !mic.getProcessor()) {
+    if (!mic || mic.getProcessor() || pendingProcessors.has(room)) return;
+    const processor = new LotusDenoiseProcessor(config);
+    pendingProcessors.set(room, processor);
     void mic
-        .setProcessor(new LotusDenoiseProcessor(config))
-        .catch((e) => logger.warn("[lotus] denoise setProcessor failed", e));
-    }
+      .setProcessor(processor)
+      .catch((e) => logger.warn("[lotus] denoise setProcessor failed", e))
+      .finally(() => {
+        // Only clear if we're still the pending entry (a teardown that ran
+        // while this was in flight may have already replaced/removed it).
+        if (pendingProcessors.get(room) === processor)
+          pendingProcessors.delete(room);
+      });
   };
 
   const roomListeners = new Map<LivekitRoom, () => void>();
@@ -149,6 +165,15 @@ export function startLotusDenoise(vm: CallViewModel): () => void {
     for (const room of rooms) {
       const mic = micOf(room);
       if (mic?.getProcessor()) void mic.stopProcessor();
+      else {
+        // [lotus] A setProcessor() call may still be in flight (mid wasm/model
+        // load) when teardown runs, in which case `mic.getProcessor()` is
+        // still undefined and `stopProcessor()` above is a no-op. Destroy the
+        // pending processor directly so its AudioContext/graph don't leak.
+        const pending = pendingProcessors.get(room);
+        if (pending) void pending.destroy().catch(() => undefined);
       }
+    }
+    pendingProcessors.clear();
   };
 }

src/lotus/lotusDenoiseProcessor.test.ts

@@ -0,0 +1,55 @@
+/*
+Copyright 2026 Lotus Guild
+
+SPDX-License-Identifier: AGPL-3.0-only OR LicenseRef-Element-Commercial
+Please see LICENSE in the repository root for full details.
+*/
+
+import { describe, expect, test } from "vitest";
+import { type AudioProcessorOptions } from "livekit-client";
+
+import { LotusDenoiseProcessor } from "./lotusDenoiseProcessor";
+
+function makeProcessor(): LotusDenoiseProcessor {
+  return new LotusDenoiseProcessor({
+    model: "rnnoise",
+    assetBase: "https://example.invalid/denoise/",
+    gate: false,
+    gateThreshold: -45,
+    floor: 0.15,
+  });
+}
+
+describe("LotusDenoiseProcessor.restart", () => {
+  test("falls back to processedTrack = undefined (never the raw LiveKit track) when graph rebuild fails", async () => {
+    const processor = makeProcessor();
+
+    // Stub out the AudioContext/graph plumbing: pretend the context is fine
+    // but the graph rebuild (wasm load / worklet construction) throws, which
+    // is the path this test targets.
+    // eslint-disable-next-line @typescript-eslint/no-explicit-any
+    (processor as any).ensureContext = async (): Promise<void> => {
+      await Promise.resolve();
+    };
+    // eslint-disable-next-line @typescript-eslint/no-explicit-any
+    (processor as any).buildGraph = async (): Promise<never> => {
+      await Promise.resolve();
+      throw new Error("simulated graph build failure");
+    };
+
+    const rawTrack = {
+      stop: (): void => undefined,
+    } as unknown as MediaStreamTrack;
+
+    await processor.restart({
+      track: rawTrack,
+    } as unknown as AudioProcessorOptions);
+
+    // Must NOT be the raw, LiveKit-owned track: LiveKit's
+    // internalStopProcessor() calls `processor.processedTrack?.stop()` then
+    // re-publishes the same `_mediaStreamTrack` object, which would kill the
+    // live mic on the next stopProcessor()/teardown if we handed it back here.
+    expect(processor.processedTrack).toBeUndefined();
+    expect(processor.processedTrack).not.toBe(rawTrack);
+  });
+});

src/lotus/lotusDenoiseProcessor.ts

@@ -177,10 +177,16 @@ export class LotusDenoiseProcessor implements TrackProcessor<
       this.processedTrack = next.track;
     } catch (e) {
       // Never go silent on the A7/device-switch path: fall back to raw audio.
+      // [lotus] IMPORTANT: never assign `opts.track` (LiveKit-owned) here.
+      // LiveKit's internalStopProcessor() does `processor.processedTrack?.stop()`
+      // then re-publishes `_mediaStreamTrack` — the SAME object if we set it as
+      // processedTrack — which kills the live mic on the next stopProcessor()/
+      // teardown. Leaving processedTrack undefined makes LiveKit fall through to
+      // its own `_mediaStreamTrack` instead.
       logger.warn("[lotus] denoise restart failed; using raw mic", e);
       this.disposeGraph(this.graph);
       this.graph = undefined;
-      this.processedTrack = opts.track;
+      this.processedTrack = undefined;
     }
   }
 

src/lotus/lotusQuality.test.ts

@@ -0,0 +1,93 @@
+/*
+Copyright 2026 Lotus Guild
+
+SPDX-License-Identifier: AGPL-3.0-only OR LicenseRef-Element-Commercial
+Please see LICENSE in the repository root for full details.
+*/
+
+import { describe, expect, test, vi } from "vitest";
+
+import { buildPatch, patchSender } from "./lotusQuality";
+
+function makeSender(
+  initialEncodings: RTCRtpEncodingParameters[] = [{}],
+): RTCRtpSender {
+  let encodings = initialEncodings;
+  return {
+    getParameters: vi.fn(() => ({ encodings })),
+    setParameters: vi.fn(async (params: RTCRtpSendParameters) => {
+      await Promise.resolve();
+      encodings = params.encodings ?? [];
+    }),
+  } as unknown as RTCRtpSender;
+}
+
+describe("lotusQuality set_quality -> clear (#11)", () => {
+  test("clearing a previously-set cap actively unsets it on the sender", async () => {
+    const sender = makeSender();
+    const writtenKeys = new WeakMap<
+      RTCRtpSender,
+      Set<keyof RTCRtpEncodingParameters>
+    >();
+
+    // Set: audioMaxBitrate = 64000.
+    const setPatch = buildPatch(sender, { maxBitrate: 64_000 }, writtenKeys);
+    expect(setPatch).toEqual({ maxBitrate: 64_000 });
+    await patchSender(sender, setPatch, writtenKeys);
+
+    expect(sender.getParameters().encodings[0].maxBitrate).toBe(64_000);
+    expect(writtenKeys.get(sender)).toEqual(new Set(["maxBitrate"]));
+
+    // Clear: host sends `null`, so the caller now wants an empty desired
+    // patch. buildPatch must still emit an explicit `undefined` for the key
+    // it previously wrote, instead of an empty patch that leaves the stale
+    // cap on the sender.
+    const clearPatch = buildPatch(sender, {}, writtenKeys);
+    expect(clearPatch).toEqual({ maxBitrate: undefined });
+    await patchSender(sender, clearPatch, writtenKeys);
+
+    const finalEncoding = sender.getParameters().encodings[0];
+    expect(finalEncoding.maxBitrate).toBeUndefined();
+    expect("maxBitrate" in finalEncoding).toBe(true); // explicitly cleared, not merely absent
+    expect(writtenKeys.has(sender)).toBe(false);
+  });
+
+  test("clearing one of several caps leaves the others (and their tracking) intact", async () => {
+    const sender = makeSender();
+    const writtenKeys = new WeakMap<
+      RTCRtpSender,
+      Set<keyof RTCRtpEncodingParameters>
+    >();
+
+    const setPatch = buildPatch(
+      sender,
+      { maxBitrate: 500_000, maxFramerate: 24 },
+      writtenKeys,
+    );
+    await patchSender(sender, setPatch, writtenKeys);
+    expect(writtenKeys.get(sender)).toEqual(
+      new Set(["maxBitrate", "maxFramerate"]),
+    );
+
+    // Only maxFramerate is still desired; maxBitrate should be actively
+    // cleared.
+    const clearPatch = buildPatch(sender, { maxFramerate: 24 }, writtenKeys);
+    expect(clearPatch).toEqual({ maxFramerate: 24, maxBitrate: undefined });
+    await patchSender(sender, clearPatch, writtenKeys);
+
+    const finalEncoding = sender.getParameters().encodings[0];
+    expect(finalEncoding.maxBitrate).toBeUndefined();
+    expect(finalEncoding.maxFramerate).toBe(24);
+    expect(writtenKeys.get(sender)).toEqual(new Set(["maxFramerate"]));
+  });
+
+  test("no sender means an empty patch and no call", () => {
+    const writtenKeys = new WeakMap<
+      RTCRtpSender,
+      Set<keyof RTCRtpEncodingParameters>
+    >();
+    expect(buildPatch(undefined, { maxBitrate: 1000 }, writtenKeys)).toEqual(
+      {},
+    );
+  });
+});

src/lotus/lotusQuality.ts

@@ -45,6 +45,14 @@ export function startLotusQuality(vm: CallViewModel): () => void {
   if (!w) return () => undefined;
 
   const settings: QualitySettings = {};
+  // [lotus] Tracks which RTCRtpEncodingParameters keys this module has
+  // actively written on each sender, so a later `null` (clear) can write
+  // `undefined` into those same keys instead of just dropping the sticky
+  // setting and leaving the stale cap live on the sender (#11).
+  const writtenKeys = new WeakMap<
+    RTCRtpSender,
+    Set<keyof RTCRtpEncodingParameters>
+  >();
   // Per-room LocalTrackPublished listeners, so sticky settings re-apply on
   // every (re)publish.
   const roomListeners = new Map<LivekitRoom, () => void>();
@@ -57,24 +65,27 @@ export function startLotusQuality(vm: CallViewModel): () => void {
   const applyToRoom = (room: LivekitRoom): void => {
     const lp = room.localParticipant;
 
-    if (settings.audioMaxBitrate !== undefined) {
     const mic = lp.getTrackPublication(Track.Source.Microphone)?.track as
       | LocalTrack
       | undefined;
-      void patchSender(mic?.sender, { maxBitrate: settings.audioMaxBitrate });
-    }
+    const micDesired: Partial<RTCRtpEncodingParameters> = {};
+    if (settings.audioMaxBitrate !== undefined)
+      micDesired.maxBitrate = settings.audioMaxBitrate;
+    const micPatch = buildPatch(mic?.sender, micDesired, writtenKeys);
+    if (Object.keys(micPatch).length > 0)
+      void patchSender(mic?.sender, micPatch, writtenKeys);
 
-    const ssPatch: Partial<RTCRtpEncodingParameters> = {};
+    const ssDesired: Partial<RTCRtpEncodingParameters> = {};
     if (settings.screenshareMaxBitrate !== undefined)
-      ssPatch.maxBitrate = settings.screenshareMaxBitrate;
+      ssDesired.maxBitrate = settings.screenshareMaxBitrate;
     if (settings.screenshareMaxFramerate !== undefined)
-      ssPatch.maxFramerate = settings.screenshareMaxFramerate;
-    if (Object.keys(ssPatch).length > 0) {
+      ssDesired.maxFramerate = settings.screenshareMaxFramerate;
     const ss = lp.getTrackPublication(Track.Source.ScreenShare)?.track as
       | LocalTrack
       | undefined;
-      void patchSender(ss?.sender, ssPatch);
-    }
+    const ssPatch = buildPatch(ss?.sender, ssDesired, writtenKeys);
+    if (Object.keys(ssPatch).length > 0)
+      void patchSender(ss?.sender, ssPatch, writtenKeys);
   };
 
   const applyToAll = (): void => rooms.forEach(applyToRoom);
@@ -164,9 +175,35 @@ export function startLotusQuality(vm: CallViewModel): () => void {
   };
 }
 
-async function patchSender(
+// [lotus] Build the patch to actually send to a sender: the desired caps,
+// plus an explicit `undefined` for any key this module previously wrote to
+// this sender but no longer wants (see writtenKeys / #11) — otherwise a
+// cleared sticky setting would simply be skipped here and the stale
+// maxBitrate/maxFramerate would stay live on the RTCRtpSender.
+// Exported for unit testing only (see lotusQuality.test.ts) — not part of
+// the module's public surface used by callers.
+export function buildPatch(
+  sender: RTCRtpSender | undefined,
+  desired: Partial<RTCRtpEncodingParameters>,
+  writtenKeys: WeakMap<RTCRtpSender, Set<keyof RTCRtpEncodingParameters>>,
+): Partial<RTCRtpEncodingParameters> {
+  if (!sender) return {};
+  const patch: Partial<RTCRtpEncodingParameters> = { ...desired };
+  const prev = writtenKeys.get(sender);
+  if (prev) {
+    for (const key of prev) {
+      if (!(key in patch)) patch[key] = undefined;
+    }
+  }
+  return patch;
+}
+
+// Exported for unit testing only (see lotusQuality.test.ts) — not part of
+// the module's public surface used by callers.
+export async function patchSender(
   sender: RTCRtpSender | undefined,
   patch: Partial<RTCRtpEncodingParameters>,
+  writtenKeys: WeakMap<RTCRtpSender, Set<keyof RTCRtpEncodingParameters>>,
 ): Promise<void> {
   if (!sender) return;
   try {
@@ -178,6 +215,15 @@ async function patchSender(
     // full-resolution layer — the real bandwidth hog — is a later encoding.
     for (const enc of params.encodings) Object.assign(enc, patch);
     await sender.setParameters(params);
+    // Remember only the caps that are still active (defined) after this
+    // write, so a later clear knows exactly which keys to unset.
+    const active = new Set<keyof RTCRtpEncodingParameters>();
+    for (const [key, value] of Object.entries(patch)) {
+      if (value !== undefined)
+        active.add(key as keyof RTCRtpEncodingParameters);
+    }
+    if (active.size > 0) writtenKeys.set(sender, active);
+    else writtenKeys.delete(sender);
   } catch (e) {
     logger.warn("[lotus] set_quality: setParameters failed", e);
   }

src/widget.ts

@@ -94,6 +94,16 @@ export const initializeWidget = (
       const parentOrigin = new URL(parentUrl).origin;
       logger.info("Widget API is available");
       const api = new WidgetApi(widgetId, parentOrigin);
+      // [lotus] matrix-widget-api's PostmessageTransport defaults
+      // strictOriginCheck to false, which would let any frame holding a
+      // handle to our window post toWidget actions (including the
+      // io.lotus.* actions below). The Lotus deployment serves EC
+      // same-origin with the host (cinny loads /public/element-call/index.html),
+      // so globalThis.origin === parentOrigin and this check passes safely.
+      // A cross-origin deployment would need to compare ev.origin to
+      // parentOrigin instead, since strictOriginCheck compares against
+      // globalThis.origin.
+      api.transport.strictOriginCheck = true;
       api.requestCapability(MatrixCapabilities.AlwaysOnScreen);
       api.requestCapability(MatrixCapabilities.MSC4039DownloadFile);