lotus(#1): fix restart-silence (A7), 48kHz ctx; protocol + CI hardening

Denoise deep-review (CRITICAL): restart() read opts.audioContext, which
LiveKit does NOT pass on restart — so reconnect (the A7 scenario) and mic
device-switch threw after stopping the old track, leaving the mic SILENT
(A7 reintroduced). Fix:
- Processor owns a dedicated 48kHz AudioContext (sapphi worklets require
  48kHz; H1), reused across restart, closed on destroy.
- restart() never throws and never leaves a stopped track on the sender:
  builds the new graph first, then disposes the old; on failure degrades
  to RAW mic audio rather than silence.
- Cache wasm per URL (no re-fetch each reconnect); gate threshold default
  -45 and accept an explicit 0 (M2); document the cross-repo asset contract.

Protocol audit:
- Non-silent warning when an unsupported denoise model (dtln/deepfilternet)
  is requested instead of silent rnnoise fallback (F3).
- Correct the call_state enum comment (immediate error-reply, not 10s) (F2).

Build/CI audit:
- Stamp VITE_APP_VERSION in CI; document the vX.Y.Z-lotus.N version scheme.

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>

.gitea/workflows/ci.yml

@@ -21,6 +21,8 @@ on:
 env:
   # element-call's build:full sets 16384 already; keep parity for safety.
   NODE_OPTIONS: '--max-old-space-size=16384'
+  # Stamp the build so analytics/rageshakes aren't labelled "dev".
+  VITE_APP_VERSION: ${{ github.ref_name }}
 
 jobs:
   build:
@@ -106,8 +108,13 @@ jobs:
 
       - name: Set published version from tag
         working-directory: embedded/web
+        # Versioning scheme: reserve bare vX.Y.Z for upstream-parity points only
+        # (e.g. v0.20.1 == upstream 0.20.1). For Lotus-only iterations on top of
+        # an upstream base, tag a semver prerelease — v0.20.1-lotus.1, -lotus.2,
+        # … — so we never collide with an already-published upstream-parity
+        # version on the registry.
         run: |
-          TAG="${GITHUB_REF_NAME#v}"   # v0.20.1 -> 0.20.1
+          TAG="${GITHUB_REF_NAME#v}"   # v0.20.1-lotus.1 -> 0.20.1-lotus.1
           npm version "$TAG" --no-git-tag-version --allow-same-version
 
       - name: Publish

src/lotus/lotusActions.ts

@@ -14,9 +14,11 @@ export enum LotusWidgetActions {
   /**
    * fromWidget: in-call per-participant speaking / mute state.
    * NOTE: matrix-widget-api `transport.send` is request/response — the host
-   * MUST reply/ack each one (cinny's `listenAction` does, replying `{}`),
-   * otherwise every send sits pending for the 10s transport timeout and then
-   * rejects, producing continuous churn + log noise for the whole call.
+   * MUST register a handler that replies/acks each one (cinny's `listenAction`
+   * does, replying `{}`). If the host has no handler, ClientWidgetApi
+   * immediately error-replies "unsupported from-widget action", so the data is
+   * silently dropped and each throttled send rejects (caught) — functional miss
+   * + log churn, not a hang.
    */
   CallState = "io.lotus.call_state",
   /** toWidget: pin/spotlight (or clear, with userId=null) a participant. */

src/lotus/lotusDenoise.ts

@@ -53,11 +53,28 @@ function safeAssetBase(raw: string | null): string {
 export function startLotusDenoise(vm: CallViewModel): () => void {
   if (lotusParam("lotusDenoise") !== "ml") return () => undefined;
 
+  const requestedModel = lotusParam("lotusModel");
+  // Only RNNoise/Speex are implemented in-source so far. Don't silently
+  // degrade an unsupported model (e.g. the host's dtln/deepfilternet) to
+  // rnnoise — log it so the mismatch is visible.
+  let model: LotusDenoiseConfig["model"] = "rnnoise";
+  if (requestedModel === "speex") model = "speex";
+  else if (requestedModel && requestedModel !== "rnnoise")
+    logger.warn(
+      `[lotus] denoise model "${requestedModel}" not implemented in-source; using rnnoise`,
+    );
+
+  const rawThreshold = lotusParam("lotusGateThreshold");
   const config: LotusDenoiseConfig = {
-    model: lotusParam("lotusModel") === "speex" ? "speex" : "rnnoise",
+    model,
     assetBase: safeAssetBase(lotusParam("lotusDenoiseBase")),
     gate: lotusFlag("lotusGate"),
-    gateThreshold: Number(lotusParam("lotusGateThreshold")) || -50,
+    // Default -45 (matches the reference shim); accept an explicit 0 (don't
+    // coerce it away via `|| default`).
+    gateThreshold:
+      rawThreshold !== null && Number.isFinite(Number(rawThreshold))
+        ? Number(rawThreshold)
+        : -45,
   };
 
   const micOf = (room: LivekitRoom): LocalAudioTrack | undefined =>

src/lotus/lotusDenoiseProcessor.ts

@@ -25,6 +25,14 @@ export interface LotusDenoiseConfig {
 // Flat sapphi worklets: each registers a processor under these names when its
 // script module is added. Same assets the Lotus host already ships under
 // public/element-call/denoise/, so no new EC dependency is needed.
+//
+// ⚠️ CONTRACT: this is an undeclared, cross-repo asset dependency. The embedded
+// fork build does NOT bundle these — they are copied in by cinny's
+// vite.config.js `lotusDenoise()` plugin from `@sapphi-red/web-noise-suppressor`.
+// The worklet/wasm version must match what this processor expects (sapphi
+// rnnoise/speex, 48kHz). If the fork ever wants to own them, add
+// `@sapphi-red/web-noise-suppressor` as a dep + a copy step here. Until then,
+// an integration smoke-check should assert GET .../denoise/rnnoise.wasm == 200.
 const PROCESSORS: Record<
   LotusDenoiseModel,
   { name: string; script: string; wasm: string }
@@ -43,15 +51,46 @@ const PROCESSORS: Record<
 const GATE_NAME = "@sapphi-red/web-noise-suppressor/noiseGate";
 const GATE_SCRIPT = "noiseGateWorklet.js";
 
+// The sapphi rnnoise/speex worklets are 48kHz-fullband and don't resample, so
+// the graph MUST run at 48kHz regardless of the hardware default.
+const SAMPLE_RATE = 48_000;
+
+// Cache fetched wasm per URL so a reconnect/device-switch (which rebuilds the
+// graph) doesn't re-download it.
+const wasmCache = new Map<string, Promise<ArrayBuffer>>();
+function loadWasm(url: string): Promise<ArrayBuffer> {
+  let p = wasmCache.get(url);
+  if (!p) {
+    p = fetch(url).then((r) => {
+      if (!r.ok) throw new Error(`denoise wasm ${r.status}`);
+      return r.arrayBuffer();
+    });
+    wasmCache.set(url, p);
+  }
+  return p;
+}
+
+interface Graph {
+  source: MediaStreamAudioSourceNode;
+  nodes: AudioWorkletNode[];
+  track: MediaStreamTrack;
+}
+
 /**
  * A LiveKit audio TrackProcessor that runs Lotus ML noise suppression
  * (RNNoise/Speex) on the local microphone track, as a first-class stage in
  * Element Call's publish pipeline.
  *
  * This replaces the host's `getUserMedia` monkeypatch: because it's a real
- * LiveKit processor, EC re-applies it on every (re)publish, so denoise
- * survives EC's own mid-call reconnect — the root cause of the A7
- * "mic dead after reconnect" bug.
+ * LiveKit processor, EC re-applies it on every (re)publish/restart, so denoise
+ * survives EC's own mid-call reconnect — the root cause of the A7 "mic dead
+ * after reconnect" bug.
+ *
+ * It owns a dedicated 48kHz AudioContext (the worklets require it, and LiveKit
+ * does NOT pass an audioContext to restart()), reused across restarts and
+ * closed on destroy. restart() never throws and never leaves a stopped track
+ * on the sender: on failure it degrades to the raw (un-denoised) track rather
+ * than silence.
  */
 export class LotusDenoiseProcessor
   implements TrackProcessor<Track.Kind.Audio, AudioProcessorOptions>
@@ -59,41 +98,58 @@ export class LotusDenoiseProcessor
   public readonly name = "lotus-denoise";
   public processedTrack?: MediaStreamTrack;
 
-  private source?: MediaStreamAudioSourceNode;
-  private nodes: AudioWorkletNode[] = [];
+  private ctx?: AudioContext;
+  private graph?: Graph;
 
   public constructor(private readonly config: LotusDenoiseConfig) {}
 
-  public async init(opts: AudioProcessorOptions): Promise<void> {
-    await this.build(opts.audioContext, opts.track);
+  public async init(_opts: AudioProcessorOptions): Promise<void> {
+    // Own context at the required rate — do NOT use opts.audioContext (it's the
+    // room's hardware-rate context, and is undefined on restart()).
+    this.ctx = new AudioContext({ sampleRate: SAMPLE_RATE });
+    const graph = await this.buildGraph(_opts.track);
+    this.graph = graph;
+    this.processedTrack = graph.track;
   }
 
   public async restart(opts: AudioProcessorOptions): Promise<void> {
-    this.teardownGraph();
-    await this.build(opts.audioContext, opts.track);
+    if (!this.ctx || this.ctx.state === "closed")
+      this.ctx = new AudioContext({ sampleRate: SAMPLE_RATE });
+    try {
+      const next = await this.buildGraph(opts.track);
+      this.disposeGraph(this.graph);
+      this.graph = next;
+      this.processedTrack = next.track;
+    } catch (e) {
+      // Never go silent on the A7/device-switch path: fall back to raw audio.
+      logger.warn("[lotus] denoise restart failed; using raw mic", e);
+      this.disposeGraph(this.graph);
+      this.graph = undefined;
+      this.processedTrack = opts.track;
+    }
   }
 
   public async destroy(): Promise<void> {
-    this.teardownGraph();
+    this.disposeGraph(this.graph);
+    this.graph = undefined;
+    this.processedTrack = undefined;
+    if (this.ctx && this.ctx.state !== "closed")
+      await this.ctx.close().catch(() => undefined);
+    this.ctx = undefined;
   }
 
-  private async build(
-    ctx: AudioContext,
-    track: MediaStreamTrack,
-  ): Promise<void> {
+  private async buildGraph(track: MediaStreamTrack): Promise<Graph> {
+    const ctx = this.ctx!;
     const base = this.config.assetBase;
     const proc = PROCESSORS[this.config.model];
 
-    // Register worklet modules (idempotent) and load the wasm binary.
     await ctx.audioWorklet.addModule(base + proc.script);
     if (this.config.gate) await ctx.audioWorklet.addModule(base + GATE_SCRIPT);
-    const wasmBinary = await fetch(base + proc.wasm).then((r) => {
-      if (!r.ok) throw new Error(`denoise wasm ${r.status}`);
-      return r.arrayBuffer();
-    });
+    const wasmBinary = await loadWasm(base + proc.wasm);
 
     const source = ctx.createMediaStreamSource(new MediaStream([track]));
     const dest = ctx.createMediaStreamDestination();
+    const nodes: AudioWorkletNode[] = [];
     let head: AudioNode = source;
 
     if (this.config.gate) {
@@ -107,26 +163,26 @@ export class LotusDenoiseProcessor
       });
       head.connect(gate);
       head = gate;
-      this.nodes.push(gate);
+      nodes.push(gate);
     }
 
     const ml = new AudioWorkletNode(ctx, proc.name, {
       channelCount: 1,
       numberOfInputs: 1,
       numberOfOutputs: 1,
-      processorOptions: { maxChannels: 1, wasmBinary },
+      processorOptions: { maxChannels: 1, wasmBinary: wasmBinary.slice(0) },
     });
     head.connect(ml);
     ml.connect(dest);
-    this.nodes.push(ml);
+    nodes.push(ml);
 
-    this.source = source;
-    this.processedTrack = dest.stream.getAudioTracks()[0];
     logger.info(`[lotus] denoise processor active (${this.config.model})`);
+    return { source, nodes, track: dest.stream.getAudioTracks()[0] };
   }
 
-  private teardownGraph(): void {
-    for (const node of this.nodes) {
+  private disposeGraph(graph: Graph | undefined): void {
+    if (!graph) return;
+    for (const node of graph.nodes) {
       try {
         node.port.postMessage("destroy");
       } catch {
@@ -138,14 +194,11 @@ export class LotusDenoiseProcessor
         /* ignore */
       }
     }
-    this.nodes = [];
     try {
-      this.source?.disconnect();
+      graph.source.disconnect();
     } catch {
       /* ignore */
     }
-    this.source = undefined;
-    this.processedTrack?.stop();
-    this.processedTrack = undefined;
+    graph.track.stop();
   }
 }