@xmaxai/web-sdk 1.0.9 → 1.0.10

package/README.md

@@ -232,7 +232,7 @@ const client = new XmaxClient(
 );
 
 const realtime = client.createRealtimeManager(
-  new RealtimeConfiguration({ model: RealtimeModel.x2_1_preview })
+  new RealtimeConfiguration({ model: RealtimeModel.x2_1_1_preview_cn })
 );
 
 const localStream = await realtime.createLocalCameraStream({
@@ -433,6 +433,13 @@ For camera input, bind the returned video track to a preview view. The SDK enter
 `ready` after it has received a valid frame and the preview view is bound; observe
 this through `setStateListener`.
 
+`startGeneration()` returns a remote stream for mounting; it does not wait for
+the video view to finish preparing. Bind its track to an SDK video component.
+The state remains `RealtimeConnectionState.starting` until the first frame and
+the 1000 ms render warmup are complete, then becomes
+`RealtimeConnectionState.generating` as the video fades in. Use this state to
+dismiss your loading UI. Without an SDK video view, the state stays `starting`.
+
 At generation start, the SDK sends `params.mirror: true` for the front camera
 and `false` for the rear camera, local files, and network videos. The backend
 mirrors the input; generated results are not automatically mirrored in the

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

@@ -1 +1 @@
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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  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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  }\n}\n"]}

package/dist/{TrajectoryEffectRendering-C7E2KJbV.d.cts → TrajectoryEffectRendering-BKKPO7Uk.d.cts}

@@ -60,7 +60,7 @@ declare class RealtimeVideoFormat {
      * 上传编码配置
      */
     /**
-     * 最低上传码率，单位为 kbps；未指定时由 SDK 计算，0 表示不设最低码率。
+     * 最低上传码率，单位为 kbps；缺省为 100，0 表示不设最低码率；仅支持该参数的 RTC 链路生效。
      */
     readonly minimumBitrate?: number;
     /**
@@ -77,7 +77,7 @@ declare class RealtimeVideoFormat {
      * @param init.width 视频宽度，单位为像素。
      * @param init.height 视频高度，单位为像素。
      * @param init.fps 视频帧率，必须大于 0。
-     * @param init.minimumBitrate 最低上传码率，单位为 kbps；缺省按最终上传尺寸和帧率计算。
+     * @param init.minimumBitrate 最低上传码率，单位为 kbps；缺省为 100，仅支持该参数的 RTC 链路生效。
      * @param init.maximumBitrate 最高上传码率，单位为 kbps；缺省按最终上传尺寸和帧率计算。
      * @param init.encoderPreference 上传编码策略偏好，默认值为 maintainFramerate。
      */

package/dist/{TrajectoryEffectRendering-C7E2KJbV.d.ts → TrajectoryEffectRendering-BKKPO7Uk.d.ts}

@@ -60,7 +60,7 @@ declare class RealtimeVideoFormat {
      * 上传编码配置
      */
     /**
-     * 最低上传码率，单位为 kbps；未指定时由 SDK 计算，0 表示不设最低码率。
+     * 最低上传码率，单位为 kbps；缺省为 100，0 表示不设最低码率；仅支持该参数的 RTC 链路生效。
      */
     readonly minimumBitrate?: number;
     /**
@@ -77,7 +77,7 @@ declare class RealtimeVideoFormat {
      * @param init.width 视频宽度，单位为像素。
      * @param init.height 视频高度，单位为像素。
      * @param init.fps 视频帧率，必须大于 0。
-     * @param init.minimumBitrate 最低上传码率，单位为 kbps；缺省按最终上传尺寸和帧率计算。
+     * @param init.minimumBitrate 最低上传码率，单位为 kbps；缺省为 100，仅支持该参数的 RTC 链路生效。
      * @param init.maximumBitrate 最高上传码率，单位为 kbps；缺省按最终上传尺寸和帧率计算。
      * @param init.encoderPreference 上传编码策略偏好，默认值为 maintainFramerate。
      */

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