12 Commits
Author SHA1 Message Date
Lotus CIandClaude Opus 5 fcb7f8456d chore(lotus): 0.25.0-lotus.9
CI / Build embedded bundle (push) Successful in 1m34s
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Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 22:45:23 -04:00
Lotus CIandClaude Opus 5 c2267800be feat(lotus): report speakingWhileMuted for the local participant in io.lotus.call_state (#37)
While the mic is muted LiveKit keeps the capture alive and merely disables the
published MediaStreamTrack, so VAD says nothing. lotusMutedSpeech taps a CLONE
of the published track (post-denoise when the in-source processor is active),
samples RMS at 10 Hz through an AnalyserNode and runs a hysteresis gate (300 ms
of voice on, 800 ms of quiet off, threshold ≈ −36 dBFS). The tap only exists
while a muted mic publication exists; the flag is set on the local entry of
io.lotus.call_state only and is never sent to other participants. Gate is
unit-tested; verified end-to-end: muted with a tone mic → true within 2.5 s,
silence → false, unmute → false.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 22:45:23 -04:00
Lotus CIandClaude Opus 5 5421d545c4 chore(lotus): 0.25.0-lotus.8
CI / Build embedded bundle (push) Successful in 2m34s
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Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 14:35:59 -04:00
Lotus CIandClaude Opus 5 2fafa3cf36 test(lotus): oxlint-clean mock in the LocalTransport refusal test
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 14:35:59 -04:00
Lotus CIandClaude Opus 5 667230f6e3 chore(lotus): 0.25.0-lotus.7
CI / Build embedded bundle (push) Failing after 50s
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Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 14:35:03 -04:00
Lotus CIandClaude Opus 5 746917a4c6 test(lotus): satisfy oxlint require-await in the LocalTransport refusal test
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 14:35:03 -04:00
Lotus CIandClaude Opus 5 d881833491 chore(lotus): 0.25.0-lotus.6
CI / Build embedded bundle (push) Failing after 56s
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Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 14:04:45 -04:00
Lotus CIandClaude Opus 5 021b1881e5 fix(lotus): LocalTransport must not re-wrap SFUTokenRefusedError
mapAuthErrorToUserFriendlyError turned every non-whitelisted error back into
FailToGetOpenIdToken, so lotus.5's refusal reason still surfaced as the
generic page (caught end-to-end with a routed 403 on /sfu/get). Pass it
through like the other user-facing auth errors. Unit-tested.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 14:04:45 -04:00
Lotus CIandClaude Opus 5 1b609d997b chore(lotus): 0.25.0-lotus.5
CI / Build embedded bundle (push) Successful in 1m47s
CI / Publish to Gitea npm registry (push) Successful in 1m2s
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 13:42:22 -04:00
Lotus CIandClaude Opus 5 d9ac9a0fa4 fix(lotus): show the SFU token service's refusal reason instead of "Something went wrong"
When the voice-limit guard (or any JWT service) answers 403 with a reason —
"This voice channel is full.", "You don't have permission to …" — EC wrapped it
in FailToGetOpenIdToken and the user saw the generic error page with
OPEN_ID_ERROR. A 403 MatrixError with a non-empty `error` now throws
SFUTokenRefusedError ("Can't join this call" + the server's sentence), so the
reason is the description. Other failures are unchanged. Unit-tested.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-19 13:42:22 -04:00
Lotus CIandClaude Opus 5 66bfead7f1 chore(lotus): 0.25.0-lotus.4
CI / Build embedded bundle (push) Successful in 1m44s
CI / Publish to Gitea npm registry (push) Successful in 54s
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-18 23:54:25 -04:00
Lotus CIandClaude Opus 5 ea579cb998 fix(lotus): screenshare-audio mute survives a re-share — mute via the renderer, not setVolume
"Mute Screenshare Audio" (io.lotus.set_deafen screenshareAudioMuted) used
RemoteParticipant.setVolume(0, ScreenShareAudio). EC's own createVolumeControls
writes volume 1 through the same setter the moment a new screenshare media item
resolves, so when the sharer stopped and re-shared (or a late joiner shared)
the audio came back at full volume while the host button still said
"Unmute Screenshare Audio". Reproduced on the local calls stack with two
headless clients: after a re-share the screen_share_audio element read vol=1.

Now the flag is a global behavior (muteScreenshareAudio$) that
LivekitRoomAudioRenderer turns into the `muted` prop of every
Track.Source.ScreenShareAudio element — the exact mechanism deafen already uses
(pub.setEnabled(false): the server stops sending). Verified via the
RemoteTrackPublication behind each <audio>: the re-published track (new sid)
mounts with enabled=false while muted and re-enables on unmute; deafen +
undeafen leaves it muted; teardown resets the flag so the next call starts
clean. Unit tests updated; renderer test asserts only ScreenShareAudio
elements get muted by the new prop.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
2026-09-18 23:54:25 -04:00
17 changed files with 479 additions and 150 deletions
+1 -1
View File
@@ -1,6 +1,6 @@
{
"name": "@lotusguild/element-call-embedded",
"version": "0.25.0-lotus.3",
"version": "0.25.0-lotus.9",
"files": [
"README.md",
"LICENSE-AGPL-3.0",
+1
View File
@@ -116,6 +116,7 @@
"peer_connection_timeout_description": "Connection to the media server timed out. Try switching to a different network or disabling your VPN. If the problem persists, see our <0>troubleshooting guide</0> or contact your server administrator.",
"room_creation_restricted": "Failed to create call",
"room_creation_restricted_description": "Call creation might be restricted to authorized users only. Try again later, or contact your server admin if the problem persists.",
"sfu_token_refused": "Can't join this call",
"sticky_events_required": "Homeserver does not support Matrix 2.0 calls",
"sticky_events_required_description": "This deployment is configured to use Matrix 2.0 call mode, but the homeserver does not advertise support for sticky events (MSC4354). Ask your server admin to upgrade, or switch the deployment to a compatible mode.",
"unexpected_ec_error": "An unexpected error occurred (<0>Error Code:</0> <1>{{ errorCode }}</1>). Please contact your server admin."
+63 -2
View File
@@ -52,9 +52,16 @@ afterEach(() => {
vi.mock("@livekit/components-react", async (importOriginal) => {
return {
...(await importOriginal()),
AudioTrack: (props: { trackRef: TrackReference }): ReactNode => {
AudioTrack: (props: {
trackRef: TrackReference;
muted?: boolean;
}): ReactNode => {
return (
<audio data-testid={"audio"}>
<audio
data-testid={"audio"}
data-source={props.trackRef.publication.source}
data-muted={String(!!props.muted)}
>
{getTrackReferenceId(props.trackRef)}
</audio>
);
@@ -83,6 +90,7 @@ function renderTestComponent(
kind: Track.Kind;
source: Track.Source;
}[],
props: { muted?: boolean; screenshareAudioMuted?: boolean } = {},
): RenderResult {
const liveKitParticipants = livekitParticipantIdentities.map((identity) =>
mockRemoteParticipant({ identity }),
@@ -117,6 +125,7 @@ function renderTestComponent(
validIdentities={participants.map((p) => p.identity)}
livekitRoom={livekitRoom}
url={""}
{...props}
/>
</MediaDevicesProvider>,
);
@@ -286,3 +295,55 @@ it("should setup audioContext gain and pan", () => {
expect(testAudioContext.gain.gain.value).toEqual(0.1);
expect(testAudioContext.pan.pan.value).toEqual(1);
});
// [lotus] The host's "Mute Screenshare Audio" mutes ONLY screenshare-audio
// elements, via the same `muted` prop path as deafen (so it survives re-renders
// and later-published shares); deafen still mutes everything.
it("screenshareAudioMuted mutes only ScreenShareAudio tracks", () => {
const explicitTracks = [
{
participantId: "@alice:DEV0",
kind: Track.Kind.Audio,
source: Track.Source.Microphone,
},
{
participantId: "@alice:DEV0",
kind: Track.Kind.Audio,
source: Track.Source.ScreenShareAudio,
},
];
const mutedBySource = (r: RenderResult): Record<string, string> =>
Object.fromEntries(
r
.queryAllByTestId("audio")
.map((el) => [el.dataset.source, el.dataset.muted]),
);
expect(
mutedBySource(
renderTestComponent(
[{ userId: "@alice", deviceId: "DEV0" }],
["@alice:DEV0"],
explicitTracks,
{ screenshareAudioMuted: true },
),
),
).toEqual({
[Track.Source.Microphone]: "false",
[Track.Source.ScreenShareAudio]: "true",
});
expect(
mutedBySource(
renderTestComponent(
[{ userId: "@alice", deviceId: "DEV0" }],
["@alice:DEV0"],
explicitTracks,
{ muted: true },
),
),
).toEqual({
[Track.Source.Microphone]: "true",
[Track.Source.ScreenShareAudio]: "true",
});
});
+12 -1
View File
@@ -38,6 +38,12 @@ export interface MatrixAudioRendererProps {
* If set to `true`, the server will stop sending audio track data to the client.
*/
muted?: boolean;
/**
* [lotus] If set to `true`, mutes only the `Track.Source.ScreenShareAudio`
* tracks (the host's "Mute Screenshare Audio" control) — same mechanism as
* `muted`, so it holds across re-renders and later-published shares.
*/
screenshareAudioMuted?: boolean;
}
/**
@@ -58,6 +64,7 @@ export function LivekitRoomAudioRenderer({
livekitRoom,
validIdentities,
muted,
screenshareAudioMuted,
}: MatrixAudioRendererProps): ReactNode {
const logger = rootLogger.getChild("[MatrixAudioRenderer]");
const tracks = useTracks(
@@ -143,7 +150,11 @@ export function LivekitRoomAudioRenderer({
<AudioTrackWithAudioNodes
key={getTrackReferenceId(trackRef)}
trackRef={trackRef}
muted={muted}
muted={
muted ||
(screenshareAudioMuted &&
trackRef.publication.source === Track.Source.ScreenShareAudio)
}
audioContext={shouldUseAudioContext ? audioContext : undefined}
audioNodes={audioNodes}
/>
+26 -1
View File
@@ -20,7 +20,7 @@ import { MatrixError } from "matrix-js-sdk";
import { getSFUConfigWithOpenID, type OpenIDClientParts } from "./openIDSFU";
import { testJWTToken } from "../utils/test-fixtures";
import { ownMemberMock } from "../utils/test";
import { FailToGetOpenIdToken } from "../utils/errors";
import { FailToGetOpenIdToken, SFUTokenRefusedError } from "../utils/errors";
const sfuUrl = "https://sfu.example.org";
@@ -91,6 +91,31 @@ describe("getSFUConfigWithOpenID", () => {
expect.fail("Expected test to throw;");
});
it("[lotus] surfaces a 403 refusal's reason instead of the generic error", async () => {
fetchMock.post("https://sfu.example.org/sfu/get", () => {
return {
status: 403,
body: { errcode: "M_FORBIDDEN", error: "This voice channel is full." },
};
});
try {
await getSFUConfigWithOpenID(
matrixClient,
ownMemberMock,
"https://sfu.example.org",
"!example_room_id",
);
} catch (ex: unknown) {
expect(ex).toBeInstanceOf(SFUTokenRefusedError);
expect((ex as SFUTokenRefusedError).localisedMessage).toEqual(
"This voice channel is full.",
);
void (await fetchMock.flush());
return;
}
expect.fail("Expected test to throw;");
});
it("should retry without delay params if the JWT service legacy endpoint returns M_BAD_JSON 400", async () => {
let callCount = 0;
+10
View File
@@ -8,6 +8,7 @@ Please see LICENSE in the repository root for full details.
import {
type IOpenIDToken,
type MatrixClient,
MatrixError,
parseErrorResponse,
} from "matrix-js-sdk";
import { type CallMembershipIdentityParts } from "matrix-js-sdk/lib/matrixrtc/EncryptionManager";
@@ -16,6 +17,7 @@ import { type Logger } from "matrix-js-sdk/lib/logger";
import {
FailToGetOpenIdToken,
NoMatrix2AuthorizationService,
SFUTokenRefusedError,
} from "../utils/errors";
import { doNetworkOperationWithRetry } from "../utils/matrix";
import { Config } from "../config/Config";
@@ -165,6 +167,14 @@ export async function getSFUConfigWithOpenID(
logger?.info(`Got JWT from call's active focus URL.`);
return extractFullConfigFromToken(sfuConfig);
} catch (ex) {
// [lotus] A 403 from the token service carries the reason the user needs
// ("This voice channel is full.") — surface it instead of the generic error.
if (ex instanceof MatrixError && ex.httpStatus === 403) {
const reason = (ex.data as { error?: unknown } | undefined)?.error;
if (typeof reason === "string" && reason.trim()) {
throw new SFUTokenRefusedError(reason, ex);
}
}
throw new FailToGetOpenIdToken(
ex instanceof Error ? ex : new Error(`Unknown error ${ex}`),
);
+5 -1
View File
@@ -53,7 +53,11 @@ function mockMember(id: string, userId: string): Member {
}
function mockVm(members: Member[]): CallViewModel {
return { userMedia$: of(members) } as unknown as CallViewModel;
return {
userMedia$: of(members),
// no livekit connections → the muted-speech tap never starts
allConnections$: of({ getConnections: () => [] }),
} as unknown as CallViewModel;
}
function participantsOf(call: number): unknown[] {
+41 -28
View File
@@ -5,7 +5,7 @@ SPDX-License-Identifier: AGPL-3.0-only OR LicenseRef-Element-Commercial
Please see LICENSE in the repository root for full details.
*/
import { combineLatest, of, type Subscription } from "rxjs";
import { combineLatest, of, startWith, type Subscription } from "rxjs";
import {
distinctUntilChanged,
map,
@@ -16,6 +16,7 @@ import {
import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
import { widget } from "../widget";
import { LotusWidgetActions, lotusFlag, lotusSendToHost } from "./lotusWidget";
import { observeSpeakingWhileMuted$ } from "./lotusMutedSpeech";
interface ParticipantState {
/** EC media id (`${userId}:${deviceId}`), stable per participant device. */
@@ -25,6 +26,11 @@ interface ParticipantState {
speaking: boolean;
audioEnabled: boolean;
videoEnabled: boolean;
/**
* [lotus #37] LOCAL participant only: voice detected on the mic while it is
* muted (see lotusMutedSpeech.ts). Absent for remote participants.
*/
speakingWhileMuted?: boolean;
}
/**
@@ -45,7 +51,8 @@ function participantsEqual(
p.userId === b[i].userId &&
p.speaking === b[i].speaking &&
p.audioEnabled === b[i].audioEnabled &&
p.videoEnabled === b[i].videoEnabled,
p.videoEnabled === b[i].videoEnabled &&
p.speakingWhileMuted === b[i].speakingWhileMuted,
)
);
}
@@ -66,34 +73,40 @@ export function startLotusCallState(vm: CallViewModel): () => void {
// lotusDecorations.ts.
if (!widget) return () => undefined;
const sub: Subscription = vm.userMedia$
.pipe(
switchMap((members) =>
members.length === 0
? of([] as ParticipantState[])
: combineLatest(
members.map((m) =>
combineLatest([
m.speaking$,
m.audioEnabled$,
m.videoEnabled$,
]).pipe(
map(
([
speaking,
audioEnabled,
videoEnabled,
]): ParticipantState => ({
id: m.id,
userId: m.userId,
speaking,
audioEnabled,
videoEnabled,
}),
),
),
const participants$ = vm.userMedia$.pipe(
switchMap((members) =>
members.length === 0
? of([] as (ParticipantState & { local: boolean })[])
: combineLatest(
members.map((m) =>
combineLatest([
m.speaking$,
m.audioEnabled$,
m.videoEnabled$,
]).pipe(
map(([speaking, audioEnabled, videoEnabled]) => ({
id: m.id,
userId: m.userId,
speaking,
audioEnabled,
videoEnabled,
local: m.local === true,
})),
),
),
),
),
);
const sub: Subscription = combineLatest([
participants$,
observeSpeakingWhileMuted$(vm).pipe(startWith(false)),
])
.pipe(
map(([members, speakingWhileMuted]): ParticipantState[] =>
members.map(({ local, ...p }) =>
local ? { ...p, speakingWhileMuted } : p,
),
),
// `speaking` flips rapidly; drop no-op repeats BEFORE throttling so
// the throttle window isn't spent re-emitting an unchanged value.
+25 -43
View File
@@ -7,13 +7,11 @@ 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";
import { setScreenshareAudioMuted$ } from "./lotusScreenshareAudio";
const lazyActions = new EventEmitter();
@@ -27,18 +25,6 @@ vi.mock("../widget", () => ({
},
}));
/** 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$: of({ getConnections: () => [{ livekitRoom }] }),
} as unknown as CallViewModel;
}
function send(data: unknown): void {
lazyActions.emit(LotusWidgetActions.SetDeafen, {
detail: { data },
@@ -60,7 +46,7 @@ afterEach(() => {
});
test("deafen mutes, and undeafen unmutes, EC's global audio output", () => {
const stop = startLotusDeafen(mockVm());
const stop = startLotusDeafen();
send({ deafened: true, screenshareAudioMuted: false });
expect(emissions).toEqual([false]);
@@ -73,7 +59,7 @@ test("deafen mutes, and undeafen unmutes, EC's global audio output", () => {
});
test("re-sending the same state is idempotent (host resendForkState)", () => {
const stop = startLotusDeafen(mockVm());
const stop = startLotusDeafen();
send({ deafened: true, screenshareAudioMuted: false });
send({ deafened: true, screenshareAudioMuted: false });
@@ -86,7 +72,7 @@ test("re-sending the same state is idempotent (host resendForkState)", () => {
});
test("undeafen does not re-enable audio the user had muted themselves", () => {
const stop = startLotusDeafen(mockVm());
const stop = startLotusDeafen();
// The user mutes all audio through EC's own control first.
setAudioEnabled$.next(false);
@@ -101,48 +87,44 @@ test("undeafen does not re-enable audio the user had muted themselves", () => {
expect(emissions).toEqual([false]);
});
test("deafen never touches per-participant volume", () => {
const participant = { setVolume: vi.fn() };
const stop = startLotusDeafen(mockVm([participant]));
test("deafen never touches the screenshare-audio flag", () => {
const stop = startLotusDeafen();
send({ deafened: true, screenshareAudioMuted: false });
send({ deafened: false, screenshareAudioMuted: false });
expect(participant.setVolume).not.toHaveBeenCalled();
expect(setScreenshareAudioMuted$.value).toBe(false);
stop();
});
test("screenshare-audio mute is applied per source and only undone for participants we muted", () => {
const participant = { setVolume: vi.fn() };
const stop = startLotusDeafen(mockVm([participant]));
test("screenshare-audio mute is a global flag the renderer reads, reset on teardown", () => {
const seen: boolean[] = [];
const flagSub = setScreenshareAudioMuted$.subscribe((v) => seen.push(v));
const stop = startLotusDeafen();
// Not muted yet: no volume writes at all.
// Not muted yet: nothing pushed beyond the BehaviorSubject's initial value.
send({ deafened: false, screenshareAudioMuted: false });
expect(participant.setVolume).not.toHaveBeenCalled();
expect(seen).toEqual([false]);
send({ deafened: false, screenshareAudioMuted: true });
expect(participant.setVolume).toHaveBeenCalledWith(
0,
Track.Source.ScreenShareAudio,
);
expect(setScreenshareAudioMuted$.value).toBe(true);
// Re-applying the same state (host resend after reconnect) is a no-op.
send({ deafened: false, screenshareAudioMuted: true });
expect(seen).toEqual([false, true]);
participant.setVolume.mockClear();
send({ deafened: false, screenshareAudioMuted: false });
expect(participant.setVolume).toHaveBeenCalledWith(
1,
Track.Source.ScreenShareAudio,
);
// Releasing again writes nothing: we no longer own that participant.
participant.setVolume.mockClear();
send({ deafened: false, screenshareAudioMuted: false });
expect(participant.setVolume).not.toHaveBeenCalled();
// Deafen + undeafen while screenshare audio is muted leaves it muted.
send({ deafened: true, screenshareAudioMuted: true });
send({ deafened: false, screenshareAudioMuted: true });
expect(setScreenshareAudioMuted$.value).toBe(true);
// Leaving the call clears it so the next call starts clean.
stop();
expect(setScreenshareAudioMuted$.value).toBe(false);
flagSub.unsubscribe();
});
test("a partial payload only moves the flag it names", () => {
const stop = startLotusDeafen(mockVm());
const stop = startLotusDeafen();
send({ deafened: true });
expect(emissions).toEqual([false]);
+17 -71
View File
@@ -5,19 +5,13 @@ SPDX-License-Identifier: AGPL-3.0-only OR LicenseRef-Element-Commercial
Please see LICENSE in the repository root for full details.
*/
import {
type RemoteParticipant,
type Room as LivekitRoom,
RoomEvent,
Track,
} from "livekit-client";
import { logger } from "matrix-js-sdk/lib/logger";
import { type IWidgetApiRequest } from "matrix-widget-api";
import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
import { setAudioEnabled$ } from "../controls";
import { widget } from "../widget";
import { LotusWidgetActions } from "./lotusActions";
import { setScreenshareAudioMuted$ } from "./lotusScreenshareAudio";
/**
* Handle the host's `io.lotus.set_deafen` toWidget action, replacing cinny's
@@ -41,22 +35,23 @@ import { LotusWidgetActions } from "./lotusActions";
* 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.
* tab/game audio while still hearing voices). It goes through the same
* mechanism as deafen — a global behavior (`muteScreenshareAudio$`) that
* `LivekitRoomAudioRenderer` turns into the `muted` prop of every
* `Track.Source.ScreenShareAudio` element — rather than
* `RemoteParticipant.setVolume(0, ScreenShareAudio)`: EC's own
* `createVolumeControls` writes volume 1 through that very setter whenever a
* new screenshare media item resolves, so a sharer who stopped and re-shared
* (or a late joiner's share) came back at full volume while the host's button
* still said "Unmute Screenshare Audio".
*
* Undeafen restores the user's OWN output-enabled state as it was before the
* deafen (and never touches the `mute-all-audio` setting), so a user who had
* already muted all audio themselves stays muted.
*
* State is closure-scoped (per invocation, matching the sibling lotus modules).
* State is closure-scoped (per invocation, matching the sibling lotus modules);
* the screenshare-audio flag additionally lives in `setScreenshareAudioMuted$`
* so the renderer can read it, and is reset on teardown.
* 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
@@ -64,7 +59,7 @@ import { LotusWidgetActions } from "./lotusActions";
*
* No effect unless the host sends the action. Returns a teardown function.
*/
export function startLotusDeafen(vm: CallViewModel): () => void {
export function startLotusDeafen(): () => void {
const w = widget;
if (!w) return () => undefined;
@@ -107,53 +102,6 @@ export function startLotusDeafen(vm: CallViewModel): () => void {
}
});
// 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 => {
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 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.allConnections$.subscribe((data) => {
const next = data.getConnections().map((c) => c.livekitRoom);
rooms = next;
// Detach listeners for rooms that went away.
for (const [room, off] of roomListeners) {
if (!next.includes(room)) {
off();
roomListeners.delete(room);
}
}
// Attach to new rooms + apply the current state to their participants.
for (const room of next) {
if (!roomListeners.has(room)) {
room.on(RoomEvent.ParticipantConnected, applyToParticipant);
roomListeners.set(room, () =>
room.off(RoomEvent.ParticipantConnected, applyToParticipant),
);
applyToRoom(room);
}
}
});
const handler = (ev: CustomEvent<IWidgetApiRequest>): void => {
w.api.transport.reply(ev.detail, {});
const data = ev.detail.data as
@@ -168,20 +116,18 @@ export function startLotusDeafen(vm: CallViewModel): () => void {
`[lotus] set_deafen: deafened=${deafened} screenshareAudioMuted=${screenshareAudioMuted}`,
);
applyGlobalMute();
rooms.forEach(applyToRoom);
if (setScreenshareAudioMuted$.value !== screenshareAudioMuted)
setScreenshareAudioMuted$.next(screenshareAudioMuted);
};
w.lazyActions.on(LotusWidgetActions.SetDeafen, handler);
return () => {
sub.unsubscribe();
for (const off of roomListeners.values()) off();
roomListeners.clear();
// Leave the user's own output state as they had it before deafen.
if (deafened) {
deafened = false;
applyGlobalMute();
}
screenshareMuted.clear();
if (setScreenshareAudioMuted$.value) setScreenshareAudioMuted$.next(false);
audioSub.unsubscribe();
w.lazyActions.off(LotusWidgetActions.SetDeafen, handler);
};
+27
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@@ -0,0 +1,27 @@
/*
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 { expect, test } from "vitest";
import { MutedSpeechGate } from "./lotusMutedSpeech";
test("needs 300 ms of voice to switch on and 800 ms of quiet to switch off", () => {
const g = new MutedSpeechGate(0.015);
expect(g.push(0.1)).toBe(false);
expect(g.push(0.1)).toBe(false);
expect(g.push(0.1)).toBe(true);
// a short dip does not drop it
for (let i = 0; i < 7; i++) expect(g.push(0.0)).toBe(true);
expect(g.push(0.0)).toBe(false);
// one loud sample after quiet does not re-trigger
expect(g.push(0.2)).toBe(false);
});
test("keyboard-level noise below the threshold never triggers", () => {
const g = new MutedSpeechGate(0.015);
for (let i = 0; i < 50; i++) expect(g.push(0.01)).toBe(false);
});
+154
View File
@@ -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 {
type LocalTrackPublication,
type Room as LivekitRoom,
RoomEvent,
Track,
} from "livekit-client";
import { Observable, distinctUntilChanged, switchMap } from "rxjs";
import { type CallViewModel } from "../state/CallViewModel/CallViewModel";
/**
* [lotus #37] "Talking while muted" detection for the LOCAL participant.
*
* While the mic is muted LiveKit keeps the capture alive and merely disables
* the published MediaStreamTrack, so VAD/`speaking` is false and nothing
* tells the user they are talking into a muted mic. This taps a CLONE of the
* published track (the post-processor track when the in-source denoiser is
* active, so keyboard noise doesn't count), samples RMS at ~10 Hz and emits a
* debounced boolean. Zero cost when unmuted (tap torn down), local-only —
* the flag rides `io.lotus.call_state` to the host and never reaches other
* participants.
*/
export const MUTED_SPEECH_RMS = 0.015; // ≈ 36 dBFS; normal speech into a headset is 0.050.3
const SAMPLE_MS = 100;
const ON_SAMPLES = 3; // 300 ms of voice before we say "talking"
const OFF_SAMPLES = 8; // 800 ms of quiet before we drop it
/** Pure hysteresis gate over successive RMS samples (unit-tested). */
export class MutedSpeechGate {
private above = 0;
private below = 0;
private on = false;
public constructor(private readonly threshold = MUTED_SPEECH_RMS) {}
public push(rms: number): boolean {
if (rms >= this.threshold) {
this.above += 1;
this.below = 0;
if (!this.on && this.above >= ON_SAMPLES) this.on = true;
} else {
this.below += 1;
this.above = 0;
if (this.on && this.below >= OFF_SAMPLES) this.on = false;
}
return this.on;
}
public get value(): boolean {
return this.on;
}
}
const mutedMicTrack = (room: LivekitRoom): MediaStreamTrack | null => {
const pub: LocalTrackPublication | undefined =
room.localParticipant.getTrackPublication(Track.Source.Microphone);
const track = pub?.track?.mediaStreamTrack;
return pub?.isMuted && track && track.readyState === "live" ? track : null;
};
/**
* Emits while the local mic is muted and voice is detected on it. Emits
* `false` whenever the mic is unmuted, unpublished or the connection changes.
*/
export function observeSpeakingWhileMuted$(
vm: CallViewModel,
): Observable<boolean> {
return vm.allConnections$.pipe(
switchMap(
(data) =>
new Observable<boolean>((subscriber) => {
const rooms = data.getConnections().map((c) => c.livekitRoom);
let ctx: AudioContext | null = null;
let clone: MediaStreamTrack | null = null;
let timer: ReturnType<typeof setInterval> | undefined;
let tapped: MediaStreamTrack | null = null;
const stopTap = (): void => {
if (timer !== undefined) clearInterval(timer);
timer = undefined;
clone?.stop();
clone = null;
void ctx?.close().catch(() => undefined);
ctx = null;
tapped = null;
subscriber.next(false);
};
const startTap = (source: MediaStreamTrack): void => {
try {
clone = source.clone();
clone.enabled = true; // the source is disabled by the mute — the clone must not be
ctx = new AudioContext();
const analyser = ctx.createAnalyser();
analyser.fftSize = 1024;
ctx
.createMediaStreamSource(new MediaStream([clone]))
.connect(analyser);
const buf = new Float32Array(analyser.fftSize);
const gate = new MutedSpeechGate();
tapped = source;
timer = setInterval(() => {
analyser.getFloatTimeDomainData(buf);
let sum = 0;
for (let i = 0; i < buf.length; i += 1) sum += buf[i] * buf[i];
subscriber.next(gate.push(Math.sqrt(sum / buf.length)));
}, SAMPLE_MS);
} catch {
stopTap();
}
};
const reconcile = (): void => {
const track =
rooms.map(mutedMicTrack).find((t) => t !== null) ?? null;
if (track === tapped) return;
if (tapped) stopTap();
if (track) startTap(track);
};
const events = [
RoomEvent.TrackMuted,
RoomEvent.TrackUnmuted,
RoomEvent.LocalTrackPublished,
RoomEvent.LocalTrackUnpublished,
RoomEvent.Reconnected,
RoomEvent.Disconnected,
] as const;
rooms.forEach((room) =>
events.forEach((ev) => room.on(ev, reconcile)),
);
subscriber.next(false);
reconcile();
return () => {
rooms.forEach((room) =>
events.forEach((ev) => room.off(ev, reconcile)),
);
if (tapped) stopTap();
};
}),
),
distinctUntilChanged(),
);
}
+27
View File
@@ -0,0 +1,27 @@
/*
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 { BehaviorSubject } from "rxjs";
import { globalScope } from "../state/ObservableScope";
/**
* Whether the host has muted remote screenshare AUDIO (`io.lotus.set_deafen`
* `screenshareAudioMuted`). Consumed by `LivekitRoomAudioRenderer` exactly the
* way `muteAllAudio$` is: it becomes the `muted` prop of every
* `Track.Source.ScreenShareAudio` element, so it survives re-renders, applies
* to tracks that are published LATER (a sharer stopping and re-sharing, a late
* joiner) and never fights EC's per-tile volume controls — which was the
* failure mode of the old `RemoteParticipant.setVolume(0, ScreenShareAudio)`
* approach: `createVolumeControls` writes its own volume (1) through the same
* sink the moment a new screenshare media item appears, un-muting it.
*/
export const setScreenshareAudioMuted$ = new BehaviorSubject(false);
export const muteScreenshareAudio$ = globalScope.behavior(
setScreenshareAudioMuted$,
);
+5 -1
View File
@@ -73,6 +73,7 @@ import { matrixRTCMode as matrixRTCModeSetting } from "../settings/settings";
import { ReactionsReader } from "../reactions/ReactionsReader";
import { LivekitRoomAudioRenderer } from "../livekit/MatrixAudioRenderer.tsx";
import { muteAllAudio$ } from "../state/MuteAllAudioModel.ts";
import { muteScreenshareAudio$ } from "../lotus/lotusScreenshareAudio";
import { useMediaDevices } from "../MediaDevicesContext.ts";
import { EarpieceOverlay } from "./EarpieceOverlay.tsx";
import {
@@ -261,6 +262,8 @@ export const InCallView: FC<InCallViewProps> = ({
const { showControls, header: headerStyle } = useUrlParams();
const muteAllAudio = useBehavior(muteAllAudio$);
// [lotus] host-driven "Mute Screenshare Audio" (io.lotus.set_deafen).
const muteScreenshareAudio = useBehavior(muteScreenshareAudio$);
const toggleAudio = useBehavior(muteStates.audio.toggle$);
const toggleVideo = useBehavior(muteStates.video.toggle$);
const setAudioEnabled = useBehavior(muteStates.audio.setEnabled$);
@@ -309,7 +312,7 @@ export const InCallView: FC<InCallViewProps> = ({
// [lotus] Handle the host's io.lotus.set_deafen action to silence remote
// audio (and optionally screenshare audio) at the LiveKit source. No-op
// unless the host sends the action.
useEffect(() => startLotusDeafen(vm), [vm]);
useEffect(() => startLotusDeafen(), []);
const fatalCallError = useBehavior(vm.fatalError$);
// Stop the rendering and throw for the error boundary
@@ -657,6 +660,7 @@ export const InCallView: FC<InCallViewProps> = ({
livekitRoom={livekitRoom}
validIdentities={participants}
muted={muteAllAudio}
screenshareAudioMuted={muteScreenshareAudio}
/>
))}
{renderContent()}
@@ -37,6 +37,7 @@ import { Epoch, ObservableScope } from "../../ObservableScope";
import {
MatrixRTCTransportMissingError,
FailToGetOpenIdToken,
SFUTokenRefusedError,
} from "../../../utils/errors";
import * as openIDSFU from "../../../livekit/openIDSFU";
import { customLivekitUrl } from "../../../settings/settings";
@@ -125,6 +126,46 @@ describe("LocalTransport", () => {
expect(() => active$.value).toThrow(expectedError);
});
it("[lotus] passes SFUTokenRefusedError through untouched", async () => {
const scope = new ObservableScope();
mockConfig({
livekit: { livekit_service_url: "https://lk.example.org" },
});
const refused = new SFUTokenRefusedError("This voice channel is full.");
vi.spyOn(openIDSFU, "getSFUConfigWithOpenID").mockImplementation(
async () => {
await Promise.resolve();
throw refused;
},
);
const errors: Error[] = [];
const { active$ } = createLocalTransport$({
scope,
roomId: "!example_room_id",
memberships$: constant(new Epoch<CallMembership[]>([])),
client: {
baseUrl: "https://example.org",
getDomain: () => "example.org",
// eslint-disable-next-line @typescript-eslint/naming-convention
_unstable_getRTCTransports: async () => Promise.resolve([]),
getOpenIdToken: vi.fn(),
getDeviceId: vi.fn(),
},
ownMembershipIdentity: ownMemberMock,
forceJwtEndpoint: JwtEndpointVersion.Legacy,
delayId$: constant("delay_id_mock"),
});
active$.subscribe(
() => undefined,
(e) => errors.push(e),
);
await flushPromises();
expect(errors).toStrictEqual([refused]);
expect((errors[0] as SFUTokenRefusedError).localisedMessage).toBe(
"This voice channel is full.",
);
});
it("emits preferred transport after OpenID resolves", async () => {
// Use config so transport discovery succeeds, but delay OpenID JWT fetch
mockConfig({
@@ -26,6 +26,7 @@ import { type Epoch, type ObservableScope } from "../../ObservableScope.ts";
import { Config } from "../../../config/Config.ts";
import {
FailToGetOpenIdToken,
SFUTokenRefusedError,
MatrixRTCTransportMissingError,
NoMatrix2AuthorizationService,
} from "../../../utils/errors.ts";
@@ -261,7 +262,9 @@ async function doOpenIdAndJWTFromUrl(
function mapAuthErrorToUserFriendlyError(e: unknown): Error {
if (
e instanceof FailToGetOpenIdToken ||
e instanceof NoMatrix2AuthorizationService
e instanceof NoMatrix2AuthorizationService ||
// [lotus] carries the token service's own refusal reason — keep it.
e instanceof SFUTokenRefusedError
) {
// rethrow as is
return e;
+20
View File
@@ -23,6 +23,8 @@ export enum ErrorCode {
E2EE_NOT_SUPPORTED = "E2EE_NOT_SUPPORTED",
STICKY_EVENTS_NOT_SUPPORTED = "STICKY_EVENTS_NOT_SUPPORTED",
OPEN_ID_ERROR = "OPEN_ID_ERROR",
/** [lotus] The SFU token service refused us with a human-readable reason (e.g. the voice-limit guard: channel full / no permission). */
SFU_TOKEN_REFUSED = "SFU_TOKEN_REFUSED",
NO_MATRIX_2_AUTHORIZATION_SERVICE = "NO_MATRIX_2_0_AUTHORIZATION_SERVICE",
SFU_ERROR = "SFU_ERROR",
UNKNOWN_ERROR = "UNKNOWN_ERROR",
@@ -234,6 +236,24 @@ export class FailToStartLivekitConnection extends ElementCallError {
/**
* Error indicating that a LiveKit's server has hit its track limits.
*/
/**
* [lotus] The SFU token service answered 403 with a reason we can show verbatim
* — the voice-limit guard says things like "This voice channel is full." or
* "You don't have permission to share your screen here." Without this the user
* only ever saw "Something went wrong (OPEN_ID_ERROR)".
*/
export class SFUTokenRefusedError extends ElementCallError {
public constructor(reason: string, cause?: Error) {
super(
t("error.sfu_token_refused"),
ErrorCode.SFU_TOKEN_REFUSED,
ErrorCategory.CONFIGURATION_ISSUE,
reason,
cause,
);
}
}
export class InsufficientCapacityError extends ElementCallError {
public constructor() {
super(