refactor(core): tighten WebAudio rate fix per review

- Hoist duplicated test mock helpers (createMockAudioContext / setupTransport /
  mockBuffer / mockEl) from the two describe blocks to module scope.
- Drop redundant math-derivation comments in schedulePlayback; the dedicated
  rate-aware tests are the canonical proof.
- Tighten setRate JSDoc.
- Add no-op guard in setRate when the new rate equals the current rate, so a
  duplicate set-playback-rate postMessage doesn't re-anchor or walk active
  sources for nothing.
- Add a regression test for the no-op guard, and strengthen the clamp test
  to schedule at rate=2 first so the clamp-to-1 assertion is non-vacuous.

packages/core/src/runtime/webAudioTransport.test.ts

@@ -1,6 +1,49 @@
 import { describe, it, expect, vi } from "vitest";
 import { WebAudioTransport } from "./webAudioTransport";
 
+function createMockAudioContext(currentTime = 100) {
+  const startFn = vi.fn();
+  const sourceNode = {
+    buffer: null as AudioBuffer | null,
+    playbackRate: { value: 1 },
+    start: startFn,
+    stop: vi.fn(),
+    disconnect: vi.fn(),
+    connect: vi.fn(),
+  };
+  const gainNode = {
+    gain: { value: 1 },
+    connect: vi.fn(),
+    disconnect: vi.fn(),
+  };
+  const masterGain = {
+    gain: { value: 1 },
+    connect: vi.fn(),
+  };
+  const ctx = {
+    currentTime,
+    state: "running",
+    resume: vi.fn(),
+    createBufferSource: vi.fn(() => sourceNode),
+    createGain: vi.fn(() => gainNode),
+    destination: {},
+    close: vi.fn(),
+  };
+  return { ctx, sourceNode, gainNode, masterGain, startFn };
+}
+
+function setupTransport(currentTime = 100) {
+  const transport = new WebAudioTransport();
+  const mock = createMockAudioContext(currentTime);
+  (transport as unknown as { _ctx: unknown })._ctx = mock.ctx;
+  (transport as unknown as { _masterGain: unknown })._masterGain = mock.masterGain;
+  const gen = transport.startGeneration();
+  return { transport, mock, gen };
+}
+
+const mockBuffer = {} as AudioBuffer;
+const mockEl = { muted: false } as HTMLMediaElement;
+
 describe("WebAudioTransport", () => {
   it("tracks play generation for async race prevention", () => {
     const transport = new WebAudioTransport();
@@ -82,49 +125,6 @@ describe("WebAudioTransport", () => {
   });
 
   describe("schedulePlayback timing", () => {
-    function createMockAudioContext(currentTime = 100) {
-      const startFn = vi.fn();
-      const sourceNode = {
-        buffer: null as AudioBuffer | null,
-        playbackRate: { value: 1 },
-        start: startFn,
-        stop: vi.fn(),
-        disconnect: vi.fn(),
-        connect: vi.fn(),
-      };
-      const gainNode = {
-        gain: { value: 1 },
-        connect: vi.fn(),
-        disconnect: vi.fn(),
-      };
-      const masterGain = {
-        gain: { value: 1 },
-        connect: vi.fn(),
-      };
-      const ctx = {
-        currentTime,
-        state: "running",
-        resume: vi.fn(),
-        createBufferSource: vi.fn(() => sourceNode),
-        createGain: vi.fn(() => gainNode),
-        destination: {},
-        close: vi.fn(),
-      };
-      return { ctx, sourceNode, gainNode, masterGain, startFn };
-    }
-
-    function setupTransport(currentTime = 100) {
-      const transport = new WebAudioTransport();
-      const mock = createMockAudioContext(currentTime);
-      (transport as unknown as { _ctx: unknown })._ctx = mock.ctx;
-      (transport as unknown as { _masterGain: unknown })._masterGain = mock.masterGain;
-      const gen = transport.startGeneration();
-      return { transport, mock, gen };
-    }
-
-    const mockBuffer = {} as AudioBuffer;
-    const mockEl = { muted: false } as HTMLMediaElement;
-
     it("starts in-progress clips immediately with correct buffer offset", async () => {
       const { transport, mock, gen } = setupTransport(100);
 
@@ -167,49 +167,6 @@ describe("WebAudioTransport", () => {
   });
 
   describe("playback rate", () => {
-    function createMockAudioContext(currentTime = 100) {
-      const startFn = vi.fn();
-      const sourceNode = {
-        buffer: null as AudioBuffer | null,
-        playbackRate: { value: 1 },
-        start: startFn,
-        stop: vi.fn(),
-        disconnect: vi.fn(),
-        connect: vi.fn(),
-      };
-      const gainNode = {
-        gain: { value: 1 },
-        connect: vi.fn(),
-        disconnect: vi.fn(),
-      };
-      const masterGain = {
-        gain: { value: 1 },
-        connect: vi.fn(),
-      };
-      const ctx = {
-        currentTime,
-        state: "running",
-        resume: vi.fn(),
-        createBufferSource: vi.fn(() => sourceNode),
-        createGain: vi.fn(() => gainNode),
-        destination: {},
-        close: vi.fn(),
-      };
-      return { ctx, sourceNode, gainNode, masterGain, startFn };
-    }
-
-    function setupTransport(currentTime = 100) {
-      const transport = new WebAudioTransport();
-      const mock = createMockAudioContext(currentTime);
-      (transport as unknown as { _ctx: unknown })._ctx = mock.ctx;
-      (transport as unknown as { _masterGain: unknown })._masterGain = mock.masterGain;
-      const gen = transport.startGeneration();
-      return { transport, mock, gen };
-    }
-
-    const mockBuffer = {} as AudioBuffer;
-    const mockEl = { muted: false } as HTMLMediaElement;
-
     it("sets sourceNode.playbackRate.value when rate is provided", async () => {
       const { transport, mock, gen } = setupTransport(100);
 
@@ -236,10 +193,6 @@ describe("WebAudioTransport", () => {
     });
 
     it("keeps in-progress buffer offset at elapsed + mediaStart regardless of rate", async () => {
-      // Audio that has been playing for `elapsed` composition seconds at rate r:
-      //   wallclock elapsed = elapsed / r
-      //   buffer advanced  = (elapsed / r) × r = elapsed
-      // so the buffer offset is `elapsed + mediaStart` for any rate.
       const { transport, mock, gen } = setupTransport(100);
 
       await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 2);
@@ -263,16 +216,34 @@ describe("WebAudioTransport", () => {
       expect(() => transport.setRate(2)).not.toThrow();
     });
 
+    it("setRate is a no-op when the rate is unchanged", async () => {
+      const { transport, mock, gen } = setupTransport(100);
+      await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 2);
+
+      mock.ctx.currentTime = 100.5;
+      const timeBefore = transport.getTime();
+      transport.setRate(2);
+      const timeAfter = transport.getTime();
+
+      expect(timeAfter).toBe(timeBefore);
+      // No re-anchor, so the next 0.5s of wallclock still maps to 1s of comp time.
+      mock.ctx.currentTime = 101;
+      expect(transport.getTime()).toBeCloseTo(10, 10);
+    });
+
     it("setRate clamps non-finite or non-positive values to 1", async () => {
       const { transport, mock, gen } = setupTransport(100);
-      await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 1);
+      await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 2);
+      expect(mock.sourceNode.playbackRate.value).toBe(2);
 
       transport.setRate(Number.NaN);
       expect(mock.sourceNode.playbackRate.value).toBe(1);
 
+      transport.setRate(2);
       transport.setRate(0);
       expect(mock.sourceNode.playbackRate.value).toBe(1);
 
+      transport.setRate(2);
       transport.setRate(-1);
       expect(mock.sourceNode.playbackRate.value).toBe(1);
     });

packages/core/src/runtime/webAudioTransport.ts

@@ -112,14 +112,8 @@ export class WebAudioTransport {
       this._rateAnchorComp = compositionTime;
 
       if (elapsed >= 0) {
-        // Audio that has already been "playing" for `elapsed` composition
-        // seconds at rate r has advanced exactly `elapsed` buffer seconds —
-        // wallclock elapsed is `elapsed / r` and the source plays at rate r,
-        // so buffer advance = (elapsed / r) × r = elapsed.
         sourceNode.start(0, elapsed + mediaStart);
       } else {
-        // Future clip: composition time will reach `compositionStart` after
-        // `-elapsed` composition seconds, which is `-elapsed / r` wallclock.
         const delay = -elapsed / safeRate;
         sourceNode.start(scheduledAt + delay, mediaStart);
       }
@@ -146,18 +140,14 @@ export class WebAudioTransport {
   }
 
   /**
-   * Change the playback rate of every currently-active source in place.
+   * Rebases the composition-time reference frame before swapping rate so
-   *
+   * `getTime()` stays continuous across the change. Sources scheduled to
-   * Also rebases the composition-time reference frame so `getTime()` keeps
+   * start in the future keep their original wallclock start time — callers
-   * returning the same value across the rate change (i.e. the clock doesn't
+   * that need rate-correct future starts should `stopAll()` and reschedule.
-   * jump, it just advances at a different slope from here on).
-   *
-   * Note: sources scheduled to start in the future via `sourceNode.start(when)`
-   * keep their original wallclock start time. Callers that need rate-correct
-   * future-start times should `stopAll()` and reschedule.
    */
   setRate(rate: number): void {
     const safeRate = normalizeRate(rate);
+    if (safeRate === this._rate) return;
     if (this._ctx && !this._paused) {
       this._rateAnchorComp = this.getTime();
       this._rateAnchorCtx = this._ctx.currentTime;