@xmaxai/web-sdk 1.0.10 → 1.1.0

package/dist/RecordingProcessor-4TP5NICO.cjs

@@ -1,211 +0,0 @@
-"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
-
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-
-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

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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({ 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 */\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"]}

package/dist/RecordingProcessor-ZSBJFF4J.js

@@ -1,211 +0,0 @@
-import {
-  __publicField
-} from "./chunk-V6TY7KAL.js";
-
-// src/Media/Recording/RecordingProcessor.ts
-import {
-  BlobSource,
-  BufferTarget,
-  EncodedAudioPacketSource,
-  EncodedPacketSink,
-  EncodedVideoPacketSource,
-  Input,
-  MP4,
-  WEBM,
-  Mp4OutputFormat,
-  Output,
-  WebMOutputFormat
-} from "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) {
-    __publicField(this, "duration", duration);
-    /**
-     * 原始时长、正常帧时长与待删除区间
-     */
-    __publicField(this, "frameDuration");
-    __publicField(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 = 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 = 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
- */
-__publicField(_RecordingTimeline, "minimumGap", 0.12);
-__publicField(_RecordingTimeline, "gapMultiplier", 3);
-__publicField(_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 Input({ source: new BlobSource(blob), formats: [MP4, 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 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 : audioPackets[index + 1]?.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 audio?.getCodec();
-    const audioConfig = await audio?.getDecoderConfig();
-    if (!videoCodec || !videoConfig || audio && (!audioCodec || !audioConfig)) {
-      throw new Error("Unsupported recording codec");
-    }
-    const isWebM = (await input.getFormat()).name === WEBM.name;
-    const rotation = await video.getRotation();
-    const flip = await video.getFlip();
-    signal.throwIfAborted();
-    const output = new Output({
-      format: isWebM ? new WebMOutputFormat() : new Mp4OutputFormat(),
-      target: new BufferTarget()
-    });
-    registerOutput(output);
-    const videoSource = new EncodedVideoPacketSource(videoCodec);
-    output.addVideoTrack(videoSource, { rotation, flip });
-    const audioSource = audioCodec ? new 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
- */
-__publicField(_RecordingProcessor, "timeoutMs", 1e4);
-__publicField(_RecordingProcessor, "maximumBytes", 64 * 1024 * 1024);
-__publicField(_RecordingProcessor, "maximumPackets", 1e4);
-var RecordingProcessor = _RecordingProcessor;
-export {
-  RecordingProcessor
-};
-//# sourceMappingURL=RecordingProcessor-ZSBJFF4J.js.map

package/dist/RecordingProcessor-ZSBJFF4J.js.map

@@ -1 +0,0 @@
-{"version":3,"sources":["../src/Media/Recording/RecordingProcessor.ts","../src/Media/Recording/RecordingTimeline.ts"],"sourcesContent":["import {\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      packets.push(packet);\n      if (packets.length > RecordingProcessor.maximumPackets) {\n        throw new Error(\"Recording has too many media packets\");\n      }\n    }\n    return packets;\n  }\n\n  /**\n   * 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。\n   */\n  private async rewrite(\n    input: Input,\n    original: Blob,\n    signal: AbortSignal,\n    registerOutput: (output: Output) => void,\n  ): Promise<Blob> {\n    const video = await input.getPrimaryVideoTrack();\n    if (!video) {\n      throw new Error(\"Recording has no video track\");\n    }\n    const videoPackets = await this.readPackets(video, signal);\n    const audio = await input.getPrimaryAudioTrack();\n    const audioPackets = audio ? await this.readPackets(audio, signal) : [];\n    const duration = await input.computeDuration();\n    const audioRanges = audioPackets.map((packet, index) => ({\n      start: packet.timestamp,\n      end: packet.duration > 0 ? packet.timestamp + packet.duration : audioPackets[index + 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  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package/dist/chunk-4DE2IREA.cjs

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-
-
-
-exports.__publicField = __publicField;
-//# sourceMappingURL=chunk-4DE2IREA.cjs.map

package/dist/chunk-4DE2IREA.cjs.map

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