@xmov/avatar 2.1.0 → 2.1.2

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@xmov/avatar",
-  "version": "2.1.0",
+  "version": "2.1.2",
   "description": "魔珐数字人 JS-SDK",
   "type": "module",
   "source": "src/index.ts",
@@ -10,12 +10,19 @@
       "default": "./dist/index.modern.js"
     },
     "./agent": {
-      "types": "./src/agent/index.ts",
+      "types": "./dist/agent/index.d.ts",
       "require": "./dist/agent/index.cjs",
       "default": "./dist/agent/avatar.modern.js"
     },
     "./agent/umd": "./dist/agent/index.umd.js"
   },
+  "typesVersions": {
+    "*": {
+      "agent": [
+        "dist/agent/index.d.ts"
+      ]
+    }
+  },
   "amdName": "XmovAvatar",
   "main": "./dist/index.cjs",
   "types": "./dist/index.d.ts",
@@ -23,12 +30,22 @@
   "unpkg": "./dist/index.umd.js",
   "files": [
     "dist/",
-    "src/"
+    "src/",
+    "!src/agent/",
+    "!dist/agent/**/*.map",
+    "!dist/agent/__tests__/",
+    "!dist/agent/audio-debug.d.ts",
+    "!dist/agent/audio-uplink.d.ts",
+    "!dist/agent/cdn-entry.d.ts",
+    "!dist/agent/e2e-client.d.ts",
+    "!dist/agent/fixed-audio-track.d.ts",
+    "!dist/agent/internal-types.d.ts",
+    "!dist/agent/microphone.d.ts"
   ],
   "scripts": {
     "build": "pnpm run build:core && pnpm run build:agent && pnpm run build:e2e:cdn && pnpm run test:e2e:cdn && node upload-sdk.js build",
     "build:core": "microbundle --workers --define ENV=production,VERSION=$npm_package_version --external none --tsconfig tsconfig.json",
-    "build:agent": "microbundle --entry src/agent/index.ts --output dist/agent --format modern,esm,cjs,umd --external @xmov/avatar --alias ../index=@xmov/avatar --globals @xmov/avatar=XmovAvatar --name XingyunAvatarAgentModule --tsconfig tsconfig.agent.json && node scripts/build-agent-entry.mjs",
+    "build:agent": "microbundle --entry src/agent/index.ts --output dist/agent --format modern,esm,cjs,umd --external @xmov/avatar --alias ../index=@xmov/avatar --globals @xmov/avatar=XmovAvatar --name XingyunAvatarAgentModule --tsconfig tsconfig.agent.json --no-sourcemap && node scripts/build-agent-entry.mjs",
     "build:e2e:cdn": "microbundle --entry src/agent/cdn-entry.ts --output cdn-dist/xmovAvatar_e2e.js --format iife --external none --workers --define ENV=production,VERSION=$npm_package_version --tsconfig tsconfig.agent.json --no-pkg-main --no-sourcemap",
     "sdk": "pnpm --filter @xmov/avatar run --parallel --no-bail \"/^sdk:.+$/\"",
     "sdk:core": "microbundle watch --workers --define ENV=development,VERSION=$npm_package_version --external none --tsconfig tsconfig.json",

package/src/control/RenderScheduler.ts

@@ -636,7 +636,6 @@ export default class RenderScheduler {
 
     // 1. 暂停动画帧循环（保留当前帧索引）
     this.frameAnimationController?.pause();
-    this.decoder.abort();
 
     // 2. 停止并清空音频播放（丢弃所有剩余的音频数据）
     this.audioRenderer.stop(-1); // 传入 -1 清空所有音频数据
@@ -674,7 +673,7 @@ export default class RenderScheduler {
     // 4. 恢复后立即进入渲染中状态
     this.renderState = RenderState.resumed;
     this.onRenderChange(this.renderState);
-    (window as any).avatarSDKLogger?.log(this.TAG, ' 渲染已恢复，状态: this.currentSpeechId===');
+    (window as any).avatarSDKLogger?.log(this.TAG, ' 渲染已恢复，状态: this.currentSpeechId===' + this.currentSpeechId);
   }
 
   /**
@@ -897,16 +896,6 @@ export default class RenderScheduler {
     this.audioRenderer.resume();
   }
 
-  /**
-   * app 切后台时暂停音频并清空对齐状态（复用断网离线时的 pause()）
-   * 不能保留缓存续播：Android 后台视频帧号持续递增而音频停留在暂停位置，
-   * 恢复后音画基准脱节；且后台期间可能下发新 speak 数据，旧缓存与新 sf 冲突。
-   * 恢复后依赖新下发音频的 sf 作为 firstFrameIndex 重新对齐
-   */
-  pauseForBackground() {
-    this.audioRenderer.pause();
-  }
-
   /**
    * 设置数字人canvas的显隐状态
    * @param visible 是否可见

package/src/encoding/webcodec-opus-encoder.ts

@@ -25,7 +25,7 @@ interface PendingEncode {
   chunks: EncodedAudioChunk[];
   resolve(frames: EncodedOpusFrame[]): void;
   reject(error: Error): void;
-  timeoutId: number;
+  timeoutId: any;
 }
 
 function buildAudioEncoderConfig(config: OpusEncoderConfig) {

package/src/index.ts

@@ -1051,28 +1051,19 @@ export default class XmovAvatar {
 
   visibilitychange() {
     if (document.hidden) {
-      // 进入后台：暂停渲染循环和解码器（省电）、清空音频与表情数据
-      // 仅渲染中暂停（resumeRender 要求 paused 状态才能恢复），
-      // 其他状态（init/stopped/隐身 paused）只停止音频，避免强制暂停后无法恢复
-      const rs = this.renderScheduler?.getRenderState();
-      if (rs === RenderState.rendering || rs === RenderState.resumed) {
-        this.renderScheduler?.pauseRender();
-      } else {
-        this.renderScheduler?.pauseForBackground();
-      }
+        this.setVolume(0);
     } else {
-      // 回到前台：若处于暂停态则恢复渲染（内部已 forceSyncDecoder 对齐最新帧）
-      if (this.renderScheduler?.getRenderState() === RenderState.paused) {
-        this.renderScheduler?.resumeRender();
-      } else if (this.ttsa?.getStatus()) {
-        // 未暂停但连接正常：兜底同步解帧队列
+      if (this.ttsa?.getStatus()) {
+        // 连接正常：兜底同步解帧队列
         this.renderScheduler?.forceSyncDecoder();
+        this.setVolume(1);
       }
       // WebSocket 在后台断开且已进入离线模式，网络可用则主动触发重连，
       // 否则用户切回前台续播时无响应且无任何报错
       if (!this.ttsa?.getStatus() && this.status === AvatarStatus.offline && NetworkMonitor.ONLINE) {
         (window as any).avatarSDKLogger.warn(this.TAG, "切回前台发现连接已断开，主动触发重连");
         this.onlineMode();
+        this.setVolume(1);
       }
       // tab 切换回来时，恢复音频捕获的 AudioContext
       if (this.captureAudioCtx && this.captureAudioCtx.state === 'suspended') {

package/dist/agent/__tests__/agent.test.d.ts

@@ -1 +0,0 @@
-export {};

package/dist/agent/audio-debug.d.ts

@@ -1,3 +0,0 @@
-type AudioChunkDebugMetadata = Record<string, unknown>;
-export declare function logAudioChunkContent(label: string, chunk: Blob | ArrayBuffer | ArrayBufferView, metadata?: AudioChunkDebugMetadata, previewBytes?: number): Promise<void>;
-export {};

package/dist/agent/audio-uplink.d.ts

@@ -1,62 +0,0 @@
-export declare const AU_STREAM_NEAREND: 1;
-export declare const AU_STREAM_FAREND: 2;
-export type AuStreamId = typeof AU_STREAM_NEAREND | typeof AU_STREAM_FAREND;
-export type AuPayload = Blob | ArrayBuffer | ArrayBufferView;
-export interface AgentAudioBlock {
-    data: AuPayload;
-    /** Epoch timestamp for the beginning of this 100 ms media block. */
-    timestamp: number;
-    /** Whether this block starts a new WebM container stream. */
-    isFirstChunk?: boolean;
-}
-interface AgentAudioUplinkOptions {
-    sendBinary(frame: ArrayBuffer): void;
-    onError?(error: Error): void;
-}
-export declare function packAuAudioFrame(streamId: AuStreamId, tsMs: number, payload: ArrayBuffer | ArrayBufferView): ArrayBuffer;
-/**
- * Owns the connection-level AU timeline and serializes nearend/farend writes.
- * Nearend/farend are matched by 100 ms bucket. When independent capture clocks
- * straddle a bucket boundary, an aged nearend may consume farend from
- * the immediately preceding bucket. This keeps farend-before-nearend wire
- * order without uploading farend-only buckets or dropping WebM continuations.
- */
-export declare class AgentAudioUplink {
-    private readonly options;
-    private connectionStartedAt;
-    private connectionGeneration;
-    private sendChain;
-    private failed;
-    private readonly pendingFarend;
-    private readonly pendingNearend;
-    private readonly closedNearendBuckets;
-    private readonly lastTimestampByStream;
-    private readonly lastSentTimestampByStream;
-    private readonly pendingWebmPrefixByStream;
-    private farendStreamState;
-    constructor(options: AgentAudioUplinkOptions);
-    reset(connectionStartedAt?: number | null): void;
-    startInput(): void;
-    clearFarend(): void;
-    offerFarend(block: AgentAudioBlock): void;
-    offerNearend(block: AgentAudioBlock): void;
-    drain(): Promise<void>;
-    private normalizeBlock;
-    private enqueue;
-    /**
-     * Chromium may emit the first WebM byte (0x1a) as its own MediaRecorder
-     * Blob and put the remaining EBML magic in the next Blob. A one-byte AU
-     * payload is not a valid timed audio block, so keep an incomplete WebM
-     * magic prefix and prepend it to the next block from the same stream.
-     */
-    private coalesceSplitWebmMagic;
-    private flushOlderNearend;
-    private flushExpiredNearend;
-    private flushBucket;
-    private takeNearend;
-    private takeFarendThrough;
-    private abandonFarendStream;
-    private prune;
-}
-export declare const AU_AUDIO_BLOCK_MS = 100;
-export {};