fix(core): thread playback rate into WebAudio audio sources

WebAudioTransport scheduled AudioBufferSourceNodes with the implicit
default playbackRate of 1, so non-1x transport rates desynced visuals
from audio: GSAP timelines, the transport clock, and native <video>
all sped up while WebAudio-routed <audio> clips kept playing at 1x.

- schedulePlayback now accepts a rate, sets sourceNode.playbackRate,
  and scales the future-clip start delay by the rate (the in-progress
  buffer offset stays elapsed + mediaStart, which is rate-independent).
- New setRate() updates active sources in place and rebases the
  getTime() reference frame so the audio-master clock stays continuous
  across mid-playback rate changes.
- Runtime onSetPlaybackRate now forwards into webAudio.setRate, and
  player.play() schedules each clip with state.playbackRate.

Fixes #713
This commit is contained in:
James
2026-05-11 13:59:19 +00:00
parent d4ba9080f3
commit 89ee1e36d7
3 changed files with 204 additions and 4 deletions
@@ -86,12 +86,16 @@ describe("WebAudioTransport", () => {
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 },
@@ -161,4 +165,149 @@ describe("WebAudioTransport", () => {
expect(mock.startFn).toHaveBeenCalledWith(0, 0);
});
});
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);
await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 2);
expect(mock.sourceNode.playbackRate.value).toBe(2);
});
it("defaults rate to 1 when not provided", async () => {
const { transport, mock, gen } = setupTransport(100);
await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen);
expect(mock.sourceNode.playbackRate.value).toBe(1);
});
it("scales delay by rate for future clips so they fire at the right wallclock", async () => {
const { transport, mock, gen } = setupTransport(100);
// compStart=10, compositionTime=2, rate=2 → 8s of comp time = 4s wallclock
await transport.schedulePlayback(mockEl, mockBuffer, 10, 0, 2, 1, gen, 2);
expect(mock.startFn).toHaveBeenCalledWith(104, 0);
});
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);
expect(mock.startFn).toHaveBeenCalledWith(0, 3);
});
it("setRate updates active sources in place", async () => {
const { transport, mock, gen } = setupTransport(100);
await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 1);
expect(mock.sourceNode.playbackRate.value).toBe(1);
transport.setRate(2);
expect(mock.sourceNode.playbackRate.value).toBe(2);
});
it("setRate before any sources are scheduled does not throw", () => {
const transport = new WebAudioTransport();
expect(() => transport.setRate(2)).not.toThrow();
});
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);
transport.setRate(Number.NaN);
expect(mock.sourceNode.playbackRate.value).toBe(1);
transport.setRate(0);
expect(mock.sourceNode.playbackRate.value).toBe(1);
transport.setRate(-1);
expect(mock.sourceNode.playbackRate.value).toBe(1);
});
it("getTime advances at the configured rate", async () => {
const { transport, mock, gen } = setupTransport(100);
await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 2);
// At schedule time, ctx.currentTime=100, compositionTime=8.
expect(transport.getTime()).toBeCloseTo(8, 10);
// Advance the audio-context clock by 0.5 wallclock seconds; at rate=2,
// composition time should have advanced 1s.
mock.ctx.currentTime = 100.5;
expect(transport.getTime()).toBeCloseTo(9, 10);
});
it("getTime tracks composition time after a mid-playback setRate", async () => {
const { transport, mock, gen } = setupTransport(100);
await transport.schedulePlayback(mockEl, mockBuffer, 5, 0, 8, 1, gen, 1);
expect(transport.getTime()).toBeCloseTo(8, 10);
// 0.5s passes at rate=1 → composition time = 8.5
mock.ctx.currentTime = 100.5;
expect(transport.getTime()).toBeCloseTo(8.5, 10);
// Bump rate to 2 — composition time should NOT jump.
transport.setRate(2);
expect(transport.getTime()).toBeCloseTo(8.5, 10);
// Another 0.5s wallclock at rate=2 → composition time = 9.5
mock.ctx.currentTime = 101;
expect(transport.getTime()).toBeCloseTo(9.5, 10);
});
});
});