element-call/src/lotus/lotusQuality.ts

/*
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 LocalTrack,
  ParticipantEvent,
  type Room as LivekitRoom,
  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 { widget } from "../widget";
import { LotusWidgetActions } from "./lotusActions";

interface QualitySettings {
  /** Max audio (mic) bitrate in bits/sec. */
  audioMaxBitrate?: number;
  /** Max screenshare video bitrate in bits/sec. */
  screenshareMaxBitrate?: number;
  /** Max screenshare framerate in fps. */
  screenshareMaxFramerate?: number;
}

/**
 * Handle the host's `io.lotus.set_quality` toWidget action (#7): apply
 * audio/screenshare encoding limits (bitrate, framerate) to the local
 * published tracks via `RTCRtpSender.setParameters` — no republish needed.
 * These controls live in EC's module scope and were unreachable from the host
 * against the prebuilt bundle.
 *
 * Settings are sticky and re-applied whenever a matching local track is
 * (re)published, so they survive mute/unmute and reconnects. The server-side
 * voice-limit-guard remains the enforcement backstop.
 *
 * No effect unless the host sends the action. Returns a teardown function.
 */
export function startLotusQuality(vm: CallViewModel): () => void {
  const w = widget;
  if (!w) return () => undefined;

  const settings: QualitySettings = {};
  // Per-room LocalTrackPublished listeners, so sticky settings re-apply on
  // every (re)publish.
  const roomListeners = new Map<LivekitRoom, () => void>();
  // Per-room settle re-apply timers, so we can cancel a pending 500ms re-apply
  // when a room is removed or on teardown — otherwise it would fire against a
  // torn-down room.
  const settleTimers = new Map<LivekitRoom, ReturnType<typeof setTimeout>>();
  let rooms: LivekitRoom[] = [];

  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 ssPatch: Partial<RTCRtpEncodingParameters> = {};
    if (settings.screenshareMaxBitrate !== undefined)
      ssPatch.maxBitrate = settings.screenshareMaxBitrate;
    if (settings.screenshareMaxFramerate !== undefined)
      ssPatch.maxFramerate = settings.screenshareMaxFramerate;
    if (Object.keys(ssPatch).length > 0) {
      const ss = lp.getTrackPublication(Track.Source.ScreenShare)?.track as
        | LocalTrack
        | undefined;
      void patchSender(ss?.sender, ssPatch);
    }
  };

  const applyToAll = (): void => rooms.forEach(applyToRoom);

  // Keep the LocalTrackPublished listeners in sync with the connected rooms.
  // Drive off the LOCAL participant's connection(s), not `livekitRoomItems$` —
  // that stream omits rooms with no remote members (returns null for isLocal),
  // so caps wouldn't apply to the local senders while you're alone. Map the
  // connections to their livekit rooms exactly like `lotusDenoise.ts` does.
  const sub = vm.allConnections$.subscribe((data) => {
    const next = data.getConnections().map((c) => c.livekitRoom);
    rooms = next;
    // Remove listeners for rooms that went away.
    for (const [room, off] of roomListeners) {
      if (!next.includes(room)) {
        off();
        roomListeners.delete(room);
      }
    }
    // Add listeners for new rooms + apply current settings to them.
    for (const room of next) {
      if (!roomListeners.has(room)) {
        // Re-apply on (re)publish AND unmute/track-restart: LiveKit's
        // refreshSenderEncodings() overwrites maxBitrate/maxFramerate from the
        // publish presets on replaceTrack (device/source switch, processor
        // toggle, restart-on-unmute), and those paths don't emit
        // LocalTrackPublished. The settle re-apply lands after LiveKit's async
        // recompute so our cap wins.
        const reapply = (): void => {
          applyToRoom(room);
          // Store the settle timer per room and cancel any pending one, so it
          // can be cleared on removal/teardown and never fires against a
          // torn-down room.
          const prev = settleTimers.get(room);
          if (prev !== undefined) clearTimeout(prev);
          settleTimers.set(
            room,
            setTimeout(() => {
              settleTimers.delete(room);
              applyToRoom(room);
            }, 500),
          );
        };
        const events = [
          ParticipantEvent.LocalTrackPublished,
          ParticipantEvent.TrackUnmuted,
        ] as const;
        for (const e of events) room.localParticipant.on(e, reapply);
        roomListeners.set(room, () => {
          for (const e of events) room.localParticipant.off(e, reapply);
          const t = settleTimers.get(room);
          if (t !== undefined) clearTimeout(t);
          settleTimers.delete(room);
        });
        applyToRoom(room);
      }
    }
  });

  const handler = (ev: CustomEvent<IWidgetApiRequest>): void => {
    void w.api.transport.reply(ev.detail, {});
    const data = ev.detail.data as Record<string, unknown> | undefined;
    if (!data) return;
    // Clamp to sane ranges so a typo can't brick the encoder (e.g. a 1 bps mic).
    const ranges: Record<keyof QualitySettings, [number, number]> = {
      audioMaxBitrate: [6_000, 510_000],
      screenshareMaxBitrate: [50_000, 20_000_000],
      screenshareMaxFramerate: [1, 60],
    };
    for (const key of Object.keys(ranges) as (keyof QualitySettings)[]) {
      const v = data[key];
      if (v === null) settings[key] = undefined;
      else if (typeof v === "number" && Number.isFinite(v)) {
        const [lo, hi] = ranges[key];
        settings[key] = Math.min(hi, Math.max(lo, v));
      }
    }
    applyToAll();
  };

  w.lazyActions.on(LotusWidgetActions.SetQuality, handler);
  return () => {
    sub.unsubscribe();
    for (const off of roomListeners.values()) off();
    roomListeners.clear();
    w.lazyActions.off(LotusWidgetActions.SetQuality, handler);
  };
}

async function patchSender(
  sender: RTCRtpSender | undefined,
  patch: Partial<RTCRtpEncodingParameters>,
): Promise<void> {
  if (!sender) return;
  try {
    const params = sender.getParameters();
    if (!params.encodings || params.encodings.length === 0)
      params.encodings = [{}];
    // Apply to EVERY encoding, not just encodings[0]: screenshare publishes
    // with simulcast (VP8), so encodings[0] is the small layer and the
    // 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);
  } catch (e) {
    logger.warn("[lotus] set_quality: setParameters failed", e);
  }
}