@xmaxai/web-sdk 1.0.8 → 1.0.9

package/README.md

@@ -1,5 +1,5 @@
 <p align="center">
-  <img src="./docs/images/brand/xmax-sdk.png" alt="XmaxSDK — Realtime Interactive Video Generation" width="880">
+  <img src="./materials/images/brand/xmax-sdk.png" alt="XmaxSDK — Realtime Interactive Video Generation" width="880">
 </p>
 
 <p align="center">
@@ -10,7 +10,7 @@
 
 We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video generation via Xmax models. XmaxSDK implements an end-to-end pipeline covering media acquisition, video streaming, frame-by-frame generation, and on-device rendering, enabling developers to seamlessly integrate low-latency, high-fidelity video transformations into creative applications at a much lower cost than alternative solutions.
 
-<p align="center"><img src="./docs/images/xlab/generation-demo.gif" alt="X-Lab realtime generation demo" width="33%" /><img src="./docs/images/xlab/index-demo.gif" alt="X-Lab index demo" width="33%" /><img src="./docs/images/xlab/storage-demo.gif" alt="X-Lab storage demo" width="33%" /></p>
+<p align="center"><img src="./materials/images/xlab/generation-demo.gif" alt="X-Lab realtime generation demo" width="33%" /><img src="./materials/images/xlab/index-demo.gif" alt="X-Lab index demo" width="33%" /><img src="./materials/images/xlab/storage-demo.gif" alt="X-Lab storage demo" width="33%" /></p>
 
 <br>
 
@@ -30,8 +30,8 @@ We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video
       Replace anyone in your live feed with a designated avatar in real-time.
     </td>
     <td rowspan="2" width="16%" align="center" valign="middle">
-      <a href="./docs/videos/use-cases/character-swapping.mp4">
-        <img src="./docs/images/use-cases/character-swapping-poster.png" alt="Play the Character Swapping demo" width="120">
+      <a href="./materials/videos/use-cases/character-swapping.mp4">
+        <img src="./materials/images/use-cases/character-swapping-poster.png" alt="Play the Character Swapping demo" width="120">
         <br>
         <sub>▶ Play demo</sub>
       </a>
@@ -53,8 +53,8 @@ We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video
       authentic fit.
     </td>
     <td rowspan="2" width="16%" align="center" valign="middle">
-      <a href="./docs/videos/use-cases/virtual-try-on.mp4">
-        <img src="./docs/images/use-cases/virtual-try-on-poster.png" alt="Play the Virtual Try-On demo" width="120">
+      <a href="./materials/videos/use-cases/virtual-try-on.mp4">
+        <img src="./materials/images/use-cases/virtual-try-on-poster.png" alt="Play the Virtual Try-On demo" width="120">
         <br>
         <sub>▶ Play demo</sub>
       </a>
@@ -75,8 +75,8 @@ We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video
       Reimagine your world in any style with an immersive visual experience.
     </td>
     <td rowspan="2" width="16%" align="center" valign="middle">
-      <a href="./docs/videos/use-cases/video-restyling.mp4">
-        <img src="./docs/images/use-cases/video-restyling-poster.png" alt="Play the Video Restyling demo" width="120">
+      <a href="./materials/videos/use-cases/video-restyling.mp4">
+        <img src="./materials/images/use-cases/video-restyling-poster.png" alt="Play the Video Restyling demo" width="120">
         <br>
         <sub>▶ Play demo</sub>
       </a>
@@ -98,8 +98,8 @@ We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video
       through gestures.
     </td>
     <td rowspan="2" width="16%" align="center" valign="middle">
-      <a href="./docs/videos/use-cases/ai-companions.mp4">
-        <img src="./docs/images/use-cases/ai-companions-poster.png" alt="Play the AI Companions demo" width="120">
+      <a href="./materials/videos/use-cases/ai-companions.mp4">
+        <img src="./materials/images/use-cases/ai-companions-poster.png" alt="Play the AI Companions demo" width="120">
         <br>
         <sub>▶ Play demo</sub>
       </a>
@@ -121,8 +121,8 @@ We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video
       trajectories.
     </td>
     <td rowspan="2" width="16%" align="center" valign="middle">
-      <a href="./docs/videos/use-cases/live-photo.mp4">
-        <img src="./docs/images/use-cases/live-photo-poster.png" alt="Play the Live Photo demo" width="120">
+      <a href="./materials/videos/use-cases/live-photo.mp4">
+        <img src="./materials/images/use-cases/live-photo-poster.png" alt="Play the Live Photo demo" width="120">
         <br>
         <sub>▶ Play demo</sub>
       </a>
@@ -145,21 +145,21 @@ We introduce XmaxSDK, a JavaScript SDK designed for real-time interactive video
   <thead>
     <tr>
       <th height="104" align="center" valign="middle">
-        <img src="./docs/images/why/low-latency.svg" alt="Low latency" width="36" height="36"><br>Low latency
+        <img src="./materials/images/why/low-latency.svg" alt="Low latency" width="36" height="36"><br>Low latency
       </th>
       <th height="104" align="center" valign="middle">
-        <img src="./docs/images/why/low-cost.svg" alt="Cost efficiency" width="36" height="36"><br>Cost efficiency
+        <img src="./materials/images/why/low-cost.svg" alt="Cost efficiency" width="36" height="36"><br>Cost efficiency
       </th>
       <th height="104" align="center" valign="middle">
-        <img src="./docs/images/why/high-fidelity.svg" alt="High fidelity" width="36" height="36"><br>High fidelity
+        <img src="./materials/images/why/high-fidelity.svg" alt="High fidelity" width="36" height="36"><br>High fidelity
       </th>
     </tr>
   </thead>
   <tbody>
     <tr>
-      <td>End-to-end latency is measured in <img src="./docs/images/why/latency-highlight.svg" alt="hundreds of milliseconds" width="192" height="20" align="absmiddle">, ensuring that updates to generation conditions and interaction controls are reflected instantly.</td>
-      <td>Run on a <img src="./docs/images/why/gpu-highlight.svg" alt="single RTX 5090" width="126" height="20" align="absmiddle">, reducing inference costs by orders of magnitude versus datacenter GPUs like H100.</td>
-      <td>Our models support real-time generation at up to <img src="./docs/images/why/resolution-highlight.svg" alt="1080p" width="48" height="20" align="absmiddle">, delivering production-ready, high-quality video output.</td>
+      <td>End-to-end latency is measured in <img src="./materials/images/why/latency-highlight.svg" alt="hundreds of milliseconds" width="192" height="20" align="absmiddle">, ensuring that updates to generation conditions and interaction controls are reflected instantly.</td>
+      <td>Run on a <img src="./materials/images/why/gpu-highlight.svg" alt="single RTX 5090" width="126" height="20" align="absmiddle">, reducing inference costs by orders of magnitude versus datacenter GPUs like H100.</td>
+      <td>Our models support real-time generation at up to <img src="./materials/images/why/resolution-highlight.svg" alt="1080p" width="48" height="20" align="absmiddle">, delivering production-ready, high-quality video output.</td>
     </tr>
   </tbody>
 </table>
@@ -282,6 +282,88 @@ await realtime.updateUplinkVideoFormat(new RealtimeVideoFormat({
 
 <br>
 
+### Generate from a local image
+
+Use `createLocalImageStream` instead of `createLocalCameraStream` to use a local
+image as the video input. It accepts a browser-decodable `File` or `Blob`, fits the
+image into the model's input dimensions without cropping, and continuously sends
+the same frame without audio. No image upload or camera permission is required.
+
+```javascript
+const localStream = await realtime.createLocalImageStream({
+  file: imageFile, // File from an <input type="file" accept="image/*">.
+  videoFormat: new RealtimeVideoFormat({ width: 1024, height: 1920, fps: 30 }),
+});
+
+const remoteStream = await realtime.startGeneration({
+  localStream,
+  context: new RealtimeContext({ prompt: "Make the scene come alive" }),
+});
+```
+
+Bind the returned tracks to the same video views used for camera input.
+TRTC, Agora, and VeRTC use the existing external-video publishing path.
+`disconnect()` retains the image for reconnection; `close()` releases its resources.
+Animated images are captured as a single still frame.
+
+<br>
+
+### Record a generated clip
+
+Recording is off by default. Set `record: true` when starting a new generation
+to record approximately 20 seconds of the original remote video and audio.
+Register the listener before starting:
+
+```javascript
+import { RealtimeRecordingEventType } from "@xmaxai/web-sdk";
+
+realtime.setRecordingListener(event => {
+  if (event.type === RealtimeRecordingEventType.failed) {
+    console.error("Recording failed", event.taskID, event.error);
+    return;
+  }
+
+  const { blob, hasAudio, reason } = event.result;
+  // Save, upload, or preview blob here. Use blob.type to identify its format.
+  // completed: reached the time limit; interrupted: finalized an earlier end.
+  console.log(event.taskID, blob, hasAudio, reason);
+});
+
+await realtime.startGeneration({
+  localStream,
+  context: new RealtimeContext({ prompt: "Make the scene come alive" }),
+  record: true,
+});
+```
+
+- The timer starts when a remote video frame is available, independently of the
+  attached view. Frame interpolation, interaction effects, and overlays are not recorded.
+- Recording errors do not stop generation. `startGeneration` does not wait for the file.
+- Before delivery, the SDK shortens unusually long video-frame intervals and cuts
+  the same intervals from audio. Cuts align to complete audio-packet boundaries;
+  neither track is re-encoded, and all video frames are preserved. The resulting
+  clip can be shorter than the recording period. Speech during removed intervals
+  is lost, and audio splices can be audible. This does not reconstruct missing
+  motion or smooth visual jumps.
+- Gap removal runs after recording and does not delay RTC cleanup. It requires
+  a supported MP4 or WebM recording; if processing fails or times out, the SDK
+  logs a warning and returns the original clip instead.
+- `record` is ignored on condition updates; it cannot stop or restart a recording.
+- Restarting ends the old recording. `restart: true` with `record: true` reports a
+  recording failure for the new task; generation still restarts normally.
+- An early end returns `interrupted` only if the recorder finalizes a nonempty file
+  without an encoding error. Empty or failed recordings return `failed`.
+- Audio arriving later is included after its arrival. `hasAudio` indicates that an
+  enabled, unmuted remote audio track was observed during recording, not audible speech.
+- Formats depend on browser support; MP4 is not guaranteed. Background scheduling
+  can delay the stop timer, so files are not cropped to exactly 20.000 seconds.
+- Audio processing must be allowed by the browser. TRTC recording on mobile Safari
+  is currently unsupported because playback muting also disables audio reception.
+- Recording buffers are limited to 64 MiB. The SDK does not upload recordings or
+  retain a copy after delivery. Clear the listener with `setRecordingListener(undefined)`.
+
+<br>
+
 ### Using React
 
 Use `XmaxRealtimeVideo` as your primary React view. Store the local and remote
@@ -302,6 +384,23 @@ import { VideoContentMode } from "@xmaxai/web-sdk";
 See the [React component](./packages/xmax-sdk/src/react/XmaxRealtimeVideo.tsx) for state binding and the
 [example project](#example-project) for a complete implementation.
 
+### Interact with the generated video
+
+`XmaxVideo` and `XmaxRealtimeVideo` support dragging on the remote output with a
+mouse, pen, or multiple touch points. Interaction becomes available after generation
+starts. The SDK maps the displayed coordinates to the model input and sends the
+trajectory automatically; local previews do not send interaction events.
+
+Set `isInteractionEnabled={false}` to disable interaction without stopping playback.
+The framework-independent views expose the same `isInteractionEnabled` property.
+
+Use `trajectoryRenderer` to supply a `TrajectoryEffectRendering` implementation for
+custom visuals, or leave it undefined to use the default glowing trail. A renderer
+only controls visuals, not sampling or signaling; use a separate instance per view
+and release its animation resources in `reset()`.
+
+The generated response to dragging depends on the model and generation prompt.
+
 <br>
 
 ### Listen for events

package/dist/RecordingProcessor-4TP5NICO.cjs

@@ -0,0 +1,211 @@
+"use strict";Object.defineProperty(exports, "__esModule", {value: true}); function _nullishCoalesce(lhs, rhsFn) { if (lhs != null) { return lhs; } else { return rhsFn(); } } function _optionalChain(ops) { let lastAccessLHS = undefined; let value = ops[0]; let i = 1; while (i < ops.length) { const op = ops[i]; const fn = ops[i + 1]; i += 2; if ((op === 'optionalAccess' || op === 'optionalCall') && value == null) { return undefined; } if (op === 'access' || op === 'optionalAccess') { lastAccessLHS = value; value = fn(value); } else if (op === 'call' || op === 'optionalCall') { value = fn((...args) => value.call(lastAccessLHS, ...args)); lastAccessLHS = undefined; } } return value; }
+
+var _chunk4DE2IREAcjs = require('./chunk-4DE2IREA.cjs');
+
+// src/Media/Recording/RecordingProcessor.ts
+
+
+
+
+
+
+
+
+
+
+
+
+var _mediabunny = require('mediabunny');
+
+// src/Media/Recording/RecordingTimeline.ts
+var _RecordingTimeline = class _RecordingTimeline {
+  /**
+   * 从显示时间戳估计正常帧间隔，只裁掉显著超长的等待部分。
+   */
+  constructor(timestamps, duration, audioPackets) {
+    _chunk4DE2IREAcjs.__publicField.call(void 0, this, "duration", duration);
+    /**
+     * 原始时长、正常帧时长与待删除区间
+     */
+    _chunk4DE2IREAcjs.__publicField.call(void 0, this, "frameDuration");
+    _chunk4DE2IREAcjs.__publicField.call(void 0, this, "cuts");
+    if (!Number.isFinite(duration) || duration <= 0 || timestamps.length === 0 || timestamps.some((value) => !Number.isFinite(value) || value < 0 || value > duration)) {
+      throw new Error("Invalid recording timeline");
+    }
+    const sorted = [...timestamps].sort((a, b) => a - b);
+    const intervals = sorted.slice(1).map((time, index) => time - sorted[index]).filter((value) => value > 0).sort((a, b) => a - b);
+    this.frameDuration = _nullishCoalesce(intervals[Math.floor(intervals.length / 2)], () => ( _RecordingTimeline.fallbackFrameDuration));
+    const threshold = Math.max(_RecordingTimeline.minimumGap, this.frameDuration * _RecordingTimeline.gapMultiplier);
+    const cuts = sorted.flatMap((start, index) => {
+      const end = _nullishCoalesce(sorted[index + 1], () => ( duration));
+      return end - start > threshold + 1e-6 ? [{ start: start + this.frameDuration, end }] : [];
+    });
+    this.cuts = audioPackets ? cuts.flatMap((cut) => {
+      const contained = audioPackets.filter((packet) => packet.end > packet.start && packet.start >= cut.start && packet.end <= cut.end);
+      if (contained.length === 0) {
+        return [];
+      }
+      return [{
+        start: Math.min(...contained.map((packet) => packet.start)),
+        end: Math.max(...contained.map((packet) => packet.end))
+      }];
+    }) : cuts;
+  }
+  /**
+   * 将原始时间映射到裁剪后的时间；删除区间内的时刻映射到同一接缝。
+   */
+  map(time) {
+    let removed = 0;
+    for (const cut of this.cuts) {
+      if (time <= cut.start) {
+        break;
+      }
+      removed += Math.min(time, cut.end) - cut.start;
+    }
+    return time - removed;
+  }
+  /**
+   * 判断一个音频包是否位于删除区间；区间已对齐包边界，不会切断压缩数据。
+   */
+  removes(start, end) {
+    return this.cuts.some((cut) => start >= cut.start && end <= cut.end);
+  }
+};
+/**
+ * 长间隔判定与默认帧时长
+ */
+_chunk4DE2IREAcjs.__publicField.call(void 0, _RecordingTimeline, "minimumGap", 0.12);
+_chunk4DE2IREAcjs.__publicField.call(void 0, _RecordingTimeline, "gapMultiplier", 3);
+_chunk4DE2IREAcjs.__publicField.call(void 0, _RecordingTimeline, "fallbackFrameDuration", 1 / 30);
+var RecordingTimeline = _RecordingTimeline;
+
+// src/Media/Recording/RecordingProcessor.ts
+var _RecordingProcessor = class _RecordingProcessor {
+  /**
+   * 裁剪成功返回新文件，无需裁剪时返回原文件；失败交由调用方保留原始录像。
+   */
+  async process(blob) {
+    if (blob.size > _RecordingProcessor.maximumBytes) {
+      throw new Error("Recording is too large to process");
+    }
+    const input = new (0, _mediabunny.Input)({ source: new (0, _mediabunny.BlobSource)(blob), formats: [_mediabunny.MP4, _mediabunny.WEBM] });
+    const controller = new AbortController();
+    let output;
+    let timer;
+    const timeout = new Promise((_, reject) => {
+      timer = setTimeout(() => {
+        controller.abort();
+        reject(new Error("Recording processing timed out"));
+      }, _RecordingProcessor.timeoutMs);
+    });
+    try {
+      return await Promise.race([
+        this.rewrite(input, blob, controller.signal, (value) => {
+          output = value;
+        }),
+        timeout
+      ]);
+    } finally {
+      controller.abort();
+      clearTimeout(timer);
+      input.dispose();
+      if (output && output.state !== "finalized" && output.state !== "canceled") {
+        void output.cancel().catch(() => {
+        });
+      }
+    }
+  }
+  /**
+   * 读取有限数量的编码包，维持原始解码顺序。
+   */
+  async readPackets(track, signal) {
+    const packets = [];
+    for await (const packet of new (0, _mediabunny.EncodedPacketSink)(track).packets()) {
+      signal.throwIfAborted();
+      packets.push(packet);
+      if (packets.length > _RecordingProcessor.maximumPackets) {
+        throw new Error("Recording has too many media packets");
+      }
+    }
+    return packets;
+  }
+  /**
+   * 保留所有视频帧，删除等待区间内的完整音频包，并重新封装两条时间线。
+   */
+  async rewrite(input, original, signal, registerOutput) {
+    const video = await input.getPrimaryVideoTrack();
+    if (!video) {
+      throw new Error("Recording has no video track");
+    }
+    const videoPackets = await this.readPackets(video, signal);
+    const audio = await input.getPrimaryAudioTrack();
+    const audioPackets = audio ? await this.readPackets(audio, signal) : [];
+    const duration = await input.computeDuration();
+    const audioRanges = audioPackets.map((packet, index) => ({
+      start: packet.timestamp,
+      end: packet.duration > 0 ? packet.timestamp + packet.duration : _nullishCoalesce(_optionalChain([audioPackets, 'access', _2 => _2[index + 1], 'optionalAccess', _3 => _3.timestamp]), () => ( duration))
+    }));
+    const timeline = new RecordingTimeline(videoPackets.map((packet) => packet.timestamp), duration, audio ? audioRanges : void 0);
+    if (timeline.cuts.length === 0) {
+      return original;
+    }
+    const videoCodec = await video.getCodec();
+    const videoConfig = await video.getDecoderConfig();
+    const audioCodec = await _optionalChain([audio, 'optionalAccess', _4 => _4.getCodec, 'call', _5 => _5()]);
+    const audioConfig = await _optionalChain([audio, 'optionalAccess', _6 => _6.getDecoderConfig, 'call', _7 => _7()]);
+    if (!videoCodec || !videoConfig || audio && (!audioCodec || !audioConfig)) {
+      throw new Error("Unsupported recording codec");
+    }
+    const isWebM = (await input.getFormat()).name === _mediabunny.WEBM.name;
+    const rotation = await video.getRotation();
+    const flip = await video.getFlip();
+    signal.throwIfAborted();
+    const output = new (0, _mediabunny.Output)({
+      format: isWebM ? new (0, _mediabunny.WebMOutputFormat)() : new (0, _mediabunny.Mp4OutputFormat)(),
+      target: new (0, _mediabunny.BufferTarget)()
+    });
+    registerOutput(output);
+    const videoSource = new (0, _mediabunny.EncodedVideoPacketSource)(videoCodec);
+    output.addVideoTrack(videoSource, { rotation, flip });
+    const audioSource = audioCodec ? new (0, _mediabunny.EncodedAudioPacketSource)(audioCodec) : void 0;
+    if (audioSource) {
+      output.addAudioTrack(audioSource);
+    }
+    await output.start();
+    for (const packet of videoPackets) {
+      signal.throwIfAborted();
+      const timestamp = timeline.map(packet.timestamp);
+      const duration2 = timeline.map(packet.timestamp + packet.duration) - timestamp;
+      await videoSource.add(packet.clone({ timestamp, duration: duration2 }), { decoderConfig: videoConfig });
+    }
+    videoSource.close();
+    if (audioSource && audioConfig) {
+      for (const [index, packet] of audioPackets.entries()) {
+        signal.throwIfAborted();
+        const range = audioRanges[index];
+        if (timeline.removes(range.start, range.end)) {
+          continue;
+        }
+        await audioSource.add(packet.clone({ timestamp: timeline.map(packet.timestamp) }), { decoderConfig: audioConfig });
+      }
+      audioSource.close();
+    }
+    signal.throwIfAborted();
+    await output.finalize();
+    if (!output.target.buffer || output.target.buffer.byteLength > _RecordingProcessor.maximumBytes) {
+      throw new Error("Processed recording exceeds the memory limit");
+    }
+    return new Blob([output.target.buffer], { type: output.format.mimeType });
+  }
+};
+/**
+ * 后处理资源上限，与实时连接的清理超时分离
+ */
+_chunk4DE2IREAcjs.__publicField.call(void 0, _RecordingProcessor, "timeoutMs", 1e4);
+_chunk4DE2IREAcjs.__publicField.call(void 0, _RecordingProcessor, "maximumBytes", 64 * 1024 * 1024);
+_chunk4DE2IREAcjs.__publicField.call(void 0, _RecordingProcessor, "maximumPackets", 1e4);
+var RecordingProcessor = _RecordingProcessor;
+
+
+exports.RecordingProcessor = RecordingProcessor;
+//# sourceMappingURL=RecordingProcessor-4TP5NICO.cjs.map

package/dist/RecordingProcessor-4TP5NICO.cjs.map

@@ -0,0 +1 @@
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{\n  BlobSource, BufferTarget, EncodedAudioPacketSource, EncodedPacketSink,\n  EncodedVideoPacketSource, Input, MP4, WEBM, Mp4OutputFormat, Output, WebMOutputFormat,\n  type EncodedPacket, type InputTrack,\n} from \"mediabunny\";\nimport { RecordingTimeline } from \"./RecordingTimeline\";\n\n/**\n * 录制完成后裁掉长帧等待；音视频共用按音频包边界对齐的时间映射，不重新编码。\n */\nexport class RecordingProcessor {\n  /**\n   * 后处理资源上限，与实时连接的清理超时分离\n   */\n  private static readonly timeoutMs = 10_000;\n  private static readonly maximumBytes = 64 * 1024 * 1024;\n  private static readonly maximumPackets = 10_000;\n\n  /**\n   * 裁剪成功返回新文件，无需裁剪时返回原文件；失败交由调用方保留原始录像。\n   */\n  async process(blob: Blob): Promise<Blob> {\n    if (blob.size > RecordingProcessor.maximumBytes) {\n      throw new Error(\"Recording is too large to process\");\n    }\n\n    const input = new Input({ source: new BlobSource(blob), formats: [MP4, WEBM] });\n    const controller = new AbortController();\n    let output: Output | undefined;\n    let timer: ReturnType<typeof setTimeout> | undefined;\n    const timeout = new Promise<never>((_, reject) => {\n      timer = setTimeout(() => {\n        controller.abort();\n        reject(new Error(\"Recording processing timed out\"));\n      }, RecordingProcessor.timeoutMs);\n    });\n\n    try {\n      return await Promise.race([\n        this.rewrite(input, blob, controller.signal, value => { output = value; }),\n        timeout,\n      ]);\n    } finally {\n      controller.abort();\n      clearTimeout(timer);\n      input.dispose();\n      if (output && output.state !== \"finalized\" && output.state !== \"canceled\") {\n        void output.cancel().catch(() => {});\n      }\n    }\n  }\n\n  /**\n   * 读取有限数量的编码包，维持原始解码顺序。\n   */\n  private async readPackets(track: InputTrack, signal: AbortSignal): Promise<EncodedPacket[]> {\n    const packets: EncodedPacket[] = [];\n    for await (const packet of new EncodedPacketSink(track).packets()) {\n      signal.throwIfAborted();\n      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1]?.timestamp ?? duration,\n    }));\n    const timeline = new RecordingTimeline(videoPackets.map(packet => packet.timestamp), duration, audio ? audioRanges : undefined);\n    if (timeline.cuts.length === 0) {\n      return original;\n    }\n\n    const videoCodec = await video.getCodec();\n    const videoConfig = await video.getDecoderConfig();\n    const audioCodec = await audio?.getCodec();\n    const audioConfig = await audio?.getDecoderConfig();\n    if (!videoCodec || !videoConfig || (audio && (!audioCodec || !audioConfig))) {\n      throw new Error(\"Unsupported recording codec\");\n    }\n    const isWebM = (await input.getFormat()).name === WEBM.name;\n    const rotation = await video.getRotation();\n    const flip = await video.getFlip();\n    signal.throwIfAborted();\n\n    const output = new Output({\n      format: isWebM ? new WebMOutputFormat() : new Mp4OutputFormat(),\n      target: new BufferTarget(),\n    });\n    registerOutput(output);\n    const videoSource = new EncodedVideoPacketSource(videoCodec);\n    output.addVideoTrack(videoSource, { rotation, flip });\n    const audioSource = audioCodec ? new EncodedAudioPacketSource(audioCodec) : undefined;\n    if (audioSource) {\n      output.addAudioTrack(audioSource);\n    }\n    await output.start();\n\n    // 不指定固定帧率，避免容器把原本不同的显示时间戳量化到同一帧。\n    for (const packet of videoPackets) {\n      signal.throwIfAborted();\n      const timestamp = timeline.map(packet.timestamp);\n      const duration = timeline.map(packet.timestamp + packet.duration) - timestamp;\n      await videoSource.add(packet.clone({ timestamp, duration }), { decoderConfig: videoConfig });\n    }\n    videoSource.close();\n\n    if (audioSource && audioConfig) {\n      for (const [index, packet] of audioPackets.entries()) {\n        signal.throwIfAborted();\n        const range = audioRanges[index]!;\n        if (timeline.removes(range.start, range.end)) {\n          continue;\n        }\n        await audioSource.add(packet.clone({ timestamp: timeline.map(packet.timestamp) }), { decoderConfig: audioConfig });\n      }\n      audioSource.close();\n    }\n    signal.throwIfAborted();\n    await output.finalize();\n    if (!output.target.buffer || output.target.buffer.byteLength > RecordingProcessor.maximumBytes) {\n      throw new Error(\"Processed recording exceeds the memory limit\");\n    }\n    return new Blob([output.target.buffer], { type: output.format.mimeType });\n  }\n}\n","/**\n * 原始录制时间线上的区间，单位为秒。\n */\nexport interface RecordingTimeRange {\n  start: number;\n  end: number;\n}\n\n/**\n * 视频长帧裁剪计划。音视频共用同一映射，避免分别累计时间造成漂移。\n */\nexport class RecordingTimeline {\n  /**\n   * 长间隔判定与默认帧时长\n   */\n  private static readonly minimumGap = 0.12;\n  private static readonly gapMultiplier = 3;\n  private static readonly fallbackFrameDuration = 1 / 30;\n\n  /**\n   * 原始时长、正常帧时长与待删除区间\n   */\n  readonly frameDuration: number;\n  readonly cuts: readonly RecordingTimeRange[];\n\n  /**\n   * 从显示时间戳估计正常帧间隔，只裁掉显著超长的等待部分。\n   */\n  constructor(timestamps: readonly number[], readonly duration: number, audioPackets?: readonly RecordingTimeRange[]) {\n    if (!Number.isFinite(duration) || duration <= 0 || timestamps.length === 0 ||\n        timestamps.some(value => !Number.isFinite(value) || value < 0 || value > duration)) {\n      throw new Error(\"Invalid recording timeline\");\n    }\n\n    const sorted = [...timestamps].sort((a, b) => a - b);\n    const intervals = sorted.slice(1).map((time, index) => time - sorted[index]!)\n      .filter(value => value > 0).sort((a, b) => a - b);\n    this.frameDuration = intervals[Math.floor(intervals.length / 2)] ?? RecordingTimeline.fallbackFrameDuration;\n    const threshold = Math.max(RecordingTimeline.minimumGap, this.frameDuration * RecordingTimeline.gapMultiplier);\n\n    const cuts = sorted.flatMap((start, index) => {\n      const end = sorted[index + 1] ?? duration;\n      return end - start > threshold + 1e-6 ? [{ start: start + this.frameDuration, end }] : [];\n    });\n\n    // 只删除完全落在等待区间内的音频包；同一裁剪边界同时用于视频，避免重编码延迟。\n    this.cuts = audioPackets ? cuts.flatMap(cut => {\n      const contained = audioPackets.filter(packet => packet.end > packet.start &&\n        packet.start >= cut.start && packet.end <= cut.end);\n      if (contained.length === 0) {\n        return [];\n      }\n      return [{\n        start: Math.min(...contained.map(packet => packet.start)),\n        end: Math.max(...contained.map(packet => packet.end)),\n      }];\n    }) : cuts;\n  }\n\n  /**\n   * 将原始时间映射到裁剪后的时间；删除区间内的时刻映射到同一接缝。\n   */\n  map(time: number): number {\n    let removed = 0;\n    for (const cut of this.cuts) {\n      if (time <= cut.start) {\n        break;\n      }\n      removed += Math.min(time, cut.end) - cut.start;\n    }\n    return time - removed;\n  }\n\n  /**\n   * 判断一个音频包是否位于删除区间；区间已对齐包边界，不会切断压缩数据。\n   */\n  removes(start: number, end: number): boolean {\n    return this.cuts.some(cut => start >= cut.start && end <= cut.end);\n  }\n}\n"]}