@h-ai/ai 0.1.0-alpha9 → 1.0.0-beta.1

package/dist/browser.d.ts

@@ -1,9 +1,12 @@
-export { A as A2AConfig, a as A2AConfigSchema, b as A2ASkillConfigSchema, c as AIConfig, d as AIConfigInput, e as AIConfigSchema, C as CompressConfig, f as CompressConfigSchema, E as EmbeddingConfig, g as EmbeddingConfigSchema, h as EntityType, i as EntityTypeSchema, F as FileConfig, j as FileConfigSchema, K as KnowledgeConfig, k as KnowledgeConfigSchema, L as LLMConfig, l as LLMConfigSchema, M as MCPConfig, m as MCPConfigSchema, n as MCPServerCapabilities, o as MCPServerCapabilitiesSchema, p as MCPServerConfig, q as MCPServerConfigSchema, r as MemoryConfig, s as MemoryConfigSchema, t as MemoryType, u as MemoryTypeSchema, v as ModelEntry, w as ModelEntrySchema, x as ModelScenario, y as ModelScenarioSchema, R as ResolveRequiredModelEntryOptions, z as ResolvedModelConfig, B as RetrievalConfig, D as RetrievalConfigSchema, G as RetrievalSourceConfig, H as RetrievalSourceSchema, S as SummaryConfig, I as SummaryConfigSchema, T as TokenConfig, J as TokenConfigSchema, N as resolveModelEntry } from './ai-reasoning-types-CGUgkLF7.js';
-export { A as AIFunctions, a as AIInitOptions, C as CompressionStrategy, b as CompressionStrategySchema, H as HaiAIError } from './ai-types-CRCjluXI.js';
-export { A2AClientOperations, AIApiAdapter, AIClient, AIClientConfig, StreamOptions, StreamProgress, collectStreamContent, createA2AClient, createAIClient, parseSSE } from './client/index.js';
-import '@a2a-js/sdk/server';
+export { A as A2AAgentCardConfig, a as A2AApiKeySecurity, b as A2AAuthenticator, c as A2ACallOptions, d as A2ACallResult, e as A2ACallerIdentity, f as A2AConfig, g as A2AConfigSchema, h as A2AContextInfo, i as A2AHandleResult, j as A2AMessageRecord, k as A2AOperations, l as A2ASecurityConfig, m as A2ASkillConfigSchema, n as A2ATaskFilter, o as AIConfig, p as AIConfigInput, q as AIConfigSchema, r as AIFunctions, s as AIInitOptions, t as AIRelStore, u as AIRelStoreCompareAndSwap, v as AIRelStoreOptions, w as AIStoreProvider, x as AIVectorBackend, y as AIVectorStore, z as AIVersionedRecord, B as AUDIO_WS_PATH, C as ApiType, D as ApiTypeSchema, E as AskOptions, F as AssistantMessage, G as AudioConfig, H as AudioConfigSchema, I as AudioFormatSchema, J as AudioModelEntry, K as AudioModelEntrySchema, L as AudioOperation, M as AudioOperationSchema, N as AudioOperationsSchema, O as AudioProviderName, P as AudioProviderSchema, Q as AudioWsClientMessage, R as AudioWsClientMessageSchema, S as AudioWsDoneMessage, T as AudioWsDoneMessageSchema, U as AudioWsEndMessage, V as AudioWsErrorMessage, W as AudioWsSegmentDoneMessage, X as AudioWsSegmentStartedMessage, Y as AudioWsServerMessage, Z as AudioWsSpeechMessage, _ as AudioWsStartMessage, $ as AudioWsStartMessageSchema, a0 as AudioWsTextMessage, a1 as AudioWsTextMessageSchema, a2 as AudioWsTranscriptMessage, a3 as ChatCompletionChoice, a4 as ChatCompletionChunk, a5 as ChatCompletionDelta, a6 as ChatCompletionRequest, a7 as ChatCompletionResponse, a8 as ChatHistoryOptions, a9 as ChatMessage, aa as ChatRecord, ab as Citation, ac as CommitTurnInput, ad as CompressConfig, ae as CompressConfigSchema, af as CompressOperations, ag as CompressOptions, ah as CompressResult, ai as CompressionStrategy, aj as CompressionStrategySchema, ak as ConsolidateOptions, al as ConsolidateResult, am as ContextChatOptions, an as ContextChatResult, ao as ContextManager, ap as ContextManagerOptions, aq as ContextOperations, ar as ContextResetOptions, as as ContextStreamEvent, at as ConversationTurn, au as ConversationTurnStatus, av as DefineToolOptions, aw as DeveloperMessage, ax as EmbeddingConfig, ay as EmbeddingConfigSchema, az as EmbeddingItem, aA as EmbeddingOperations, aB as EmbeddingRequest, aC as EmbeddingResponse, aD as EntityDocumentRelation, aE as EntityDocumentResult, aF as EntityListOptions, aG as EntityQueryOptions, aH as EntityType, aI as EntityTypeSchema, aJ as FileConfig, aK as FileConfigSchema, aL as FileOperations, aM as FileParseMethod, aN as FileParseOptions, aO as FileParseRequest, aP as FileParseResult, aQ as GenerateImageRequest, aR as GenerateImageResult, aS as GenerateObjectRequest, aT as GeneratedImage, aU as HaiAIError, aV as ImageConfig, aW as ImageConfigSchema, aX as ImageContent, aY as ImageModelEntry, aZ as ImageModelEntrySchema, a_ as ImageOperations, a$ as ImageProviderName, b0 as ImageProviderSchema, b1 as ImageSize, b2 as InteractionScope, b3 as KnowledgeAskOptions, b4 as KnowledgeAskResult, b5 as KnowledgeConfig, b6 as KnowledgeConfigSchema, b7 as KnowledgeDocumentInfo, b8 as KnowledgeDocumentListOptions, b9 as KnowledgeDocumentMetadata, ba as KnowledgeDocumentRemoveOptions, bb as KnowledgeDocumentRevision, bc as KnowledgeEntity, bd as KnowledgeIngestBatchProgress, be as KnowledgeIngestBatchResult, bf as KnowledgeIngestFileInput, bg as KnowledgeIngestInput, bh as KnowledgeIngestResult, bi as KnowledgeOperations, bj as KnowledgeRepairOptions, bk as KnowledgeRepairProgress, bl as KnowledgeRepairResult, bm as KnowledgeRetrieveItem, bn as KnowledgeRetrieveOptions, bo as KnowledgeRetrieveResult, bp as KnowledgeSetupOptions, bq as KnowledgeStore, br as LLMConfig, bs as LLMConfigSchema, bt as LLMOperations, bu as MCPConfig, bv as MCPConfigSchema, bw as MCPContext, bx as MCPOperations, by as MCPPrompt, bz as MCPPromptArgument, bA as MCPPromptContent, bB as MCPPromptMessage, bC as MCPResource, bD as MCPResourceContent, bE as MCPServerCapabilities, bF as MCPServerCapabilitiesSchema, bG as MCPServerConfig, bH as MCPServerConfigSchema, bI as MCPToolDefinition, bJ as MCPToolHandler, bK as McpServerOptions, bL as MemoryAccessScope, bM as MemoryAdminOperations, bN as MemoryClearAllOptions, bO as MemoryClearOptions, bP as MemoryConfig, bQ as MemoryConfigSchema, bR as MemoryCoreOperations, bS as MemoryEntry, bT as MemoryEntryInput, bU as MemoryExtractOptions, bV as MemoryInjectionOptions, bW as MemoryListOptions, bX as MemoryListPageOptions, bY as MemoryOperations, bZ as MemoryRecallOptions, b_ as MemoryType, b$ as MemoryTypeSchema, c0 as MemoryUpdateInput, c1 as MessageContent, c2 as MessageRole, c3 as ModelEntry, c4 as ModelEntrySchema, c5 as ModelScenario, c6 as ModelScenarioSchema, c7 as ObjectRef, c8 as OptionalSecretSchema, c9 as OutputFormat, ca as PersonaOperations, cb as PersonaProfile, cc as PersonaProfileInput, cd as PersonaProfileUpdate, ce as PersonaScopeOptions, cf as RagContextItem, cg as RagOperations, ch as RagOptions, ci as RagResult, cj as RagStreamEvent, ck as ReasoningOperations, cl as ReasoningOptions, cm as ReasoningResult, cn as ReasoningStep, co as ReasoningStepType, cp as ReasoningStrategy, cq as ReasoningStreamEvent, cr as ReferenceImage, cs as RerankDocument, ct as RerankItem, cu as RerankOperations, cv as RerankRequest, cw as RerankResponse, cx as ResolveRequiredModelEntryOptions, cy as ResolvedAudioModel, cz as ResolvedImageModel, cA as ResolvedModelConfig, cB as RetrievalConfig, cC as RetrievalConfigSchema, cD as RetrievalOperations, cE as RetrievalRequest, cF as RetrievalResult, cG as RetrievalResultItem, cH as RetrievalSource, cI as RetrievalSourceConfig, cJ as RetrievalSourceSchema, cK as SSEDecoder, cL as SSEEvent, cM as ScopedMemoryBinding, cN as ScopedMemoryOperations, cO as SessionInfo, cP as StoreFilter, cQ as StorePage, cR as StoreScope, cS as StreamOperations, cT as StreamProcessor, cU as StreamResult, cV as SummaryConfig, cW as SummaryConfigSchema, cX as SummaryOperations, cY as SummaryOptions, cZ as SummaryResult, c_ as SystemMessage, c$ as TempModelConfig, d0 as TextContent, d1 as TokenConfig, d2 as TokenConfigSchema, d3 as TokenOperations, d4 as TokenUsage, d5 as Tool, d6 as ToolAuthorizationRequest, d7 as ToolAuthorizer, d8 as ToolCall, d9 as ToolDefinition, da as ToolErrorType, db as ToolExecutionContext, dc as ToolExecutionOptions, dd as ToolMessage, de as ToolRegistryOperations, df as ToolRegistryOptions, dg as ToolsOperations, dh as UserMessage, di as WhereClause, dj as WhereOperator, dk as WhereValue, dl as ensureAudioCredential, dm as resolveAudioModel, dn as resolveImageModel, dp as resolveModelApi, dq as resolveModelEntry, dr as serializePlayableAudio } from './ai-audio-ws-protocol-nfhO0tOo.js';
+export { A as AudioCapabilitiesRequest, a as AudioContent, b as AudioFormat, c as AudioInputStream, d as AudioModelCapabilities, e as AudioOperations, f as AudioReference, P as PlayableAudio, S as SynthesisEvent, g as SynthesisOptions, h as SynthesisRequest, i as SynthesisResult, j as SynthesisStreamRequest, k as SynthesisTextSegment, T as TranscriptionEvent, l as TranscriptionOptions, m as TranscriptionRequest, n as TranscriptionResult, o as TranscriptionSegment, p as TranscriptionStreamRequest, q as TranscriptionTimestampGranularity, r as TranscriptionWord } from './ai-audio-types-4R5yUAVl.js';
+export { AudioClientConfig, AudioClientOperations, AudioTicketRequest, createAudioClient } from './client/index.js';
+export { AgentCapabilities, AgentCard, AgentProvider, AgentSkill, Artifact, DataPart, FilePart, FileWithBytes, FileWithUri, Message, MessageSendParams, Part, PushNotificationConfig, Task, TaskArtifactUpdateEvent, TaskIdParams, TaskQueryParams, TaskState, TaskStatus, TaskStatusUpdateEvent, TextPart } from '@a2a-js/sdk';
+export { AgentExecutor, ExecutionEventBus } from '@a2a-js/sdk/server';
 import '@h-ai/core';
-import '@h-ai/datapipe';
 import 'zod';
 import 'openai';
 import 'node:buffer';
+import '@h-ai/datapipe';
+import '@modelcontextprotocol/sdk/types.js';
+import 'zod/v4/core';

package/dist/browser.js

@@ -1,4 +1,5 @@
-export { A2AConfigSchema, A2ASkillConfigSchema, AIConfigSchema, CompressConfigSchema, CompressionStrategySchema, EmbeddingConfigSchema, EntityTypeSchema, FileConfigSchema, HaiAIError, KnowledgeConfigSchema, LLMConfigSchema, MCPConfigSchema, MCPServerCapabilitiesSchema, MCPServerConfigSchema, MemoryConfigSchema, MemoryTypeSchema, ModelEntrySchema, ModelScenarioSchema, RetrievalConfigSchema, RetrievalSourceSchema, SummaryConfigSchema, TokenConfigSchema, resolveModelEntry } from './chunk-6PPXWQ7I.js';
-export { collectStreamContent, createA2AClient, createAIClient, parseSSE } from './chunk-ABP7RKIP.js';
+export { A2AConfigSchema, A2ASkillConfigSchema, AIConfigSchema, AUDIO_WS_PATH, ApiTypeSchema, AudioConfigSchema, AudioFormatSchema, AudioModelEntrySchema, AudioOperationSchema, AudioOperationsSchema, AudioProviderSchema, AudioWsClientMessageSchema, AudioWsDoneMessageSchema, AudioWsStartMessageSchema, AudioWsTextMessageSchema, CompressConfigSchema, CompressionStrategySchema, EmbeddingConfigSchema, EntityTypeSchema, FileConfigSchema, ImageConfigSchema, ImageModelEntrySchema, ImageProviderSchema, KnowledgeConfigSchema, LLMConfigSchema, MCPConfigSchema, MCPServerCapabilitiesSchema, MCPServerConfigSchema, MemoryConfigSchema, MemoryTypeSchema, ModelEntrySchema, ModelScenarioSchema, OptionalSecretSchema, RetrievalConfigSchema, RetrievalSourceSchema, SummaryConfigSchema, TokenConfigSchema, ensureAudioCredential, resolveAudioModel, resolveImageModel, resolveModelApi, resolveModelEntry, serializePlayableAudio } from './chunk-HEVXR252.js';
+export { createAudioClient } from './chunk-GMQFIWND.js';
+export { HaiAIError } from './chunk-HYMP57AC.js';
 //# sourceMappingURL=browser.js.map
 //# sourceMappingURL=browser.js.map

package/dist/chunk-GMQFIWND.js

@@ -0,0 +1,254 @@
+import { HaiAIError } from './chunk-HYMP57AC.js';
+
+// src/client/ai-audio-client.ts
+function audioClientError(code, message) {
+  const error = new Error(message);
+  error.code = code;
+  return error;
+}
+function openBrowserWs(url, signal) {
+  return new Promise((resolve, reject) => {
+    const ws = new WebSocket(url);
+    ws.binaryType = "arraybuffer";
+    const queue = [];
+    let waiting = null;
+    let closed = false;
+    let opened = false;
+    let aborted = false;
+    let failure = null;
+    function settleClosed() {
+      if (closed)
+        return;
+      closed = true;
+      signal?.removeEventListener("abort", onAbort);
+      if (waiting) {
+        const w = waiting;
+        waiting = null;
+        w({ value: void 0, done: true });
+      }
+    }
+    function onAbort() {
+      aborted = true;
+      if (!opened) {
+        settleClosed();
+        reject(audioClientError(HaiAIError.AUDIO_CANCELLED.code, "Audio operation cancelled"));
+      }
+      ws.close();
+    }
+    function terminalError() {
+      if (aborted)
+        return audioClientError(HaiAIError.AUDIO_CANCELLED.code, "Audio operation cancelled");
+      if (failure)
+        return failure;
+      return audioClientError(HaiAIError.AUDIO_CONNECTION_FAILED.code, "Audio WebSocket closed before completion");
+    }
+    function push(message) {
+      if (waiting) {
+        const w = waiting;
+        waiting = null;
+        w({ value: message, done: false });
+      } else {
+        queue.push(message);
+      }
+    }
+    ws.onmessage = (event) => {
+      if (typeof event.data === "string")
+        push({ isBinary: false, text: event.data });
+      else
+        push({ isBinary: true, binary: new Uint8Array(event.data) });
+    };
+    ws.onerror = () => {
+      failure = audioClientError(HaiAIError.AUDIO_CONNECTION_FAILED.code, "Audio WebSocket connection failed");
+      if (!opened) {
+        settleClosed();
+        reject(failure);
+      }
+    };
+    ws.onclose = () => settleClosed();
+    if (signal) {
+      if (signal.aborted)
+        onAbort();
+      else
+        signal.addEventListener("abort", onAbort, { once: true });
+    }
+    ws.onopen = () => {
+      opened = true;
+      resolve({
+        // DOM WebSocket.send 的 BufferSource 泛型限定为 ArrayBuffer，Uint8Array 需显式收窄
+        send: (data) => ws.send(data),
+        async *messages() {
+          while (true) {
+            if (queue.length > 0) {
+              yield queue.shift();
+              continue;
+            }
+            if (closed)
+              throw terminalError();
+            const next = await new Promise((res) => {
+              waiting = res;
+            });
+            if (next.done)
+              throw terminalError();
+            yield next.value;
+          }
+        },
+        close: () => ws.close()
+      });
+    };
+  });
+}
+function createAudioClient(config) {
+  async function buildUrl(request) {
+    const ticket = await config.getTicket(request);
+    if (!ticket)
+      throw new Error("Audio WebSocket ticket is required");
+    const separator = config.url.includes("?") ? "&" : "?";
+    return `${config.url}${separator}ticket=${encodeURIComponent(ticket)}`;
+  }
+  function startMessage(operation, extra) {
+    return JSON.stringify({ type: "start", operation, ...extra });
+  }
+  async function* transcribeStream(request) {
+    const conn = await openBrowserWs(await buildUrl({ operation: "transcribe", model: request.model }), request.signal);
+    try {
+      const format = request.audio.format;
+      const sampleRate = request.audio.sampleRate;
+      conn.send(startMessage("transcribe", { stream: true, model: request.model, language: request.language, contextHints: request.contextHints, format, sampleRate, channels: request.audio.channels }));
+      const sendAudio = (async () => {
+        if ("chunks" in request.audio) {
+          for await (const part of request.audio.chunks)
+            conn.send(part);
+        } else {
+          conn.send(request.audio.data);
+        }
+        conn.send(JSON.stringify({ type: "done" }));
+      })();
+      let sendError;
+      sendAudio.catch((e) => {
+        sendError = e;
+      });
+      for await (const message of conn.messages()) {
+        if (message.isBinary || !message.text)
+          continue;
+        const event = JSON.parse(message.text);
+        if (event.type === "transcript")
+          yield { type: "transcript", text: event.text, final: event.final };
+        else if (event.type === "speech_started")
+          yield { type: "speech_started" };
+        else if (event.type === "speech_stopped")
+          yield { type: "speech_stopped" };
+        else if (event.type === "error")
+          throw audioClientError(event.code, event.message);
+        else if (event.type === "end")
+          break;
+      }
+      if (sendError)
+        throw sendError;
+    } finally {
+      conn.close();
+    }
+  }
+  async function transcribe(request) {
+    const conn = await openBrowserWs(await buildUrl({ operation: "transcribe", model: request.model }), request.signal);
+    try {
+      conn.send(startMessage("transcribe", { stream: false, model: request.model, language: request.language, contextHints: request.contextHints, format: request.audio.format, sampleRate: request.audio.sampleRate, channels: request.audio.channels }));
+      conn.send(request.audio.data);
+      conn.send(JSON.stringify({ type: "done" }));
+      let text = "";
+      for await (const message of conn.messages()) {
+        if (message.isBinary || !message.text)
+          continue;
+        const event = JSON.parse(message.text);
+        if (event.type === "transcript")
+          text = event.text;
+        else if (event.type === "error")
+          throw audioClientError(event.code, event.message);
+        else if (event.type === "end")
+          break;
+      }
+      return { text };
+    } finally {
+      conn.close();
+    }
+  }
+  async function* synthesizeStream(request) {
+    const conn = await openBrowserWs(await buildUrl({ operation: "synthesize", model: request.model }), request.signal);
+    try {
+      conn.send(startMessage("synthesize", { model: request.model, voice: request.voice, instruction: request.instruction, format: request.format, sampleRate: request.sampleRate }));
+      const sendText = (async () => {
+        const segments = isSynthesisTextSegment(request.text) ? singleSegment(request.text) : request.text;
+        for await (const segment of segments)
+          conn.send(JSON.stringify({ type: "text", segmentId: segment.id, text: segment.text }));
+        conn.send(JSON.stringify({ type: "done" }));
+      })();
+      let sendError;
+      sendText.catch((e) => {
+        sendError = e;
+      });
+      let activeSegmentId;
+      for await (const message of conn.messages()) {
+        if (message.isBinary && message.binary) {
+          if (!activeSegmentId)
+            throw new Error("Audio frame received outside a synthesis segment");
+          yield { type: "audio", segmentId: activeSegmentId, data: message.binary };
+          continue;
+        }
+        if (!message.text)
+          continue;
+        const event = JSON.parse(message.text);
+        if (event.type === "segment_started") {
+          if (activeSegmentId)
+            throw new Error("Synthesis segment started before the previous segment completed");
+          activeSegmentId = event.segmentId;
+          yield event;
+        } else if (event.type === "segment_done") {
+          if (activeSegmentId !== event.segmentId)
+            throw new Error("Synthesis segment completion does not match the active segment");
+          yield event;
+          activeSegmentId = void 0;
+        } else if (event.type === "error") {
+          throw audioClientError(event.code, event.message);
+        } else if (event.type === "end") {
+          if (activeSegmentId)
+            throw audioClientError(HaiAIError.AUDIO_PROTOCOL_ERROR.code, "Synthesis ended before the active segment completed");
+          break;
+        }
+      }
+      if (sendError)
+        throw sendError;
+    } finally {
+      conn.close();
+    }
+  }
+  async function synthesize(request) {
+    const chunks = [];
+    let meta;
+    for await (const event of synthesizeStream({ ...request, text: { id: "synthesis", text: request.text } })) {
+      if (event.type === "segment_started")
+        meta = { format: event.format, sampleRate: event.sampleRate, channels: event.channels };
+      else if (event.type === "audio")
+        chunks.push(event.data);
+    }
+    if (!meta)
+      throw audioClientError(HaiAIError.AUDIO_PROTOCOL_ERROR.code, "Synthesis completed without segment metadata");
+    const total = chunks.reduce((sum, c) => sum + c.length, 0);
+    const data = new Uint8Array(total);
+    let offset = 0;
+    for (const chunk of chunks) {
+      data.set(chunk, offset);
+      offset += chunk.length;
+    }
+    return { data, format: meta.format, sampleRate: meta.sampleRate, channels: meta.channels ?? 1 };
+  }
+  return { transcribe, transcribeStream, synthesize, synthesizeStream };
+}
+function isSynthesisTextSegment(value) {
+  return typeof value === "object" && value !== null && "id" in value && "text" in value;
+}
+async function* singleSegment(segment) {
+  yield segment;
+}
+
+export { createAudioClient };
+//# sourceMappingURL=chunk-GMQFIWND.js.map
+//# sourceMappingURL=chunk-GMQFIWND.js.map

package/dist/chunk-GMQFIWND.js.map

@@ -0,0 +1 @@
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* @h-ai/ai/client — 浏览器语音客户端\n *\n * 通过统一 WebSocket 连接 `@h-ai/serv` 的语音入口，向浏览器暴露与 Node 端一致的\n * `ai.audio.*` 业务 API。WebSocket 建立、消息编解码等传输细节在此内部完成，\n * 调用方不接触任何厂商原生事件。\n * @module client/ai-audio-client\n */\n\nimport type {\n  SynthesisEvent,\n  SynthesisRequest,\n  SynthesisResult,\n  SynthesisStreamRequest,\n  SynthesisTextSegment,\n  TranscriptionEvent,\n  TranscriptionRequest,\n  TranscriptionResult,\n  TranscriptionStreamRequest,\n} from '../audio/ai-audio-types.js'\nimport type { AudioWsServerMessage, AudioWsStartMessage } from '../audio/ai-audio-ws-protocol.js'\nimport { HaiAIError } from '../ai-types.js'\n\n/**\n * 构造带领域错误码的浏览器客户端错误\n *\n * 浏览器 Client 约定出错抛异常；此处在 `Error.code` 上保留领域错误码\n * （取消 / 连接失败 / 服务端领域错误），供调用方区分正常结束、取消与异常断连。\n */\nfunction audioClientError(code: string | number, message: string): Error {\n  const error = new Error(message)\n  ;(error as { code?: string | number }).code = code\n  return error\n}\n\n// ─── 客户端配置 ───\n\n/**\n * getTicket 收到的当前 Audio 请求摘要\n *\n * 供应用签发与本次操作严格绑定的能力票据（如 `iam.ticket.issue({ grant })`），\n * 使票据只能用于本次操作 / 模型 / 会话，降低票据被重放的风险。\n *\n * @example\n * ```ts\n * import type { AudioTicketRequest } from '@h-ai/ai/client'\n * function grant(request: AudioTicketRequest) {\n *   return { operation: request.operation, model: request.model } // 经已登录 HTTP 接口签发并在服务端原子消费。\n * }\n * ```\n */\nexport interface AudioTicketRequest {\n  /** 本次操作类型 */\n  operation: 'transcribe' | 'synthesize'\n  /** 本次使用的模型 ID（如有指定） */\n  model?: string\n}\n\n/**\n * 浏览器语音客户端配置\n *\n * @example\n * ```ts\n * import { createAudioClient } from '@h-ai/ai/client'\n * import type { AudioTicketRequest } from '@h-ai/ai/client'\n * function connect(issueTicket: (request: AudioTicketRequest) => Promise<string>) {\n *   return createAudioClient({ url: 'wss://example.com/api/v1/ai/audio', getTicket: issueTicket })\n * }\n * ```\n */\nexport interface AudioClientConfig {\n  /** 语音 WebSocket 完整 URL（如 `wss://host/api/v1/ai/audio`） */\n  url: string\n  /**\n   * 获取短期、一次性的 Audio WebSocket ticket\n   *\n   * ticket 应由已登录 HTTP 请求签发，并在服务端首次校验时原子消费。\n   * 收到当前请求摘要（operation / model），可签发与本次操作严格绑定的票据。\n   */\n  getTicket: (request: AudioTicketRequest) => string | Promise<string>\n}\n\n/**\n * 浏览器语音操作接口\n *\n * 与 Node 端 `AudioOperations` 业务形态一致；出错时抛出异常（浏览器 Client 约定）。\n *\n * @example\n * ```ts\n * import type { AudioClientOperations } from '@h-ai/ai/client'\n * async function speak(audio: AudioClientOperations) {\n *   // 浏览器 AudioClient 普通方法也显式抛错，由应用交互边界处理。\n *   const result = await audio.synthesize({ text: '你好', format: 'mp3' })\n *   return result.data\n * }\n * ```\n */\nexport interface AudioClientOperations {\n  /** 将完整音频识别为完整文本 */\n  transcribe: (request: TranscriptionRequest) => Promise<TranscriptionResult>\n  /** 持续输入音频或增量返回识别文本（含语音起止事件） */\n  transcribeStream: (request: TranscriptionStreamRequest) => AsyncIterable<TranscriptionEvent>\n  /** 将完整文本合成为完整音频 */\n  synthesize: (request: SynthesisRequest) => Promise<SynthesisResult>\n  /** 持续输入文本段并按段输出结构化音频事件 */\n  synthesizeStream: (request: SynthesisStreamRequest) => AsyncIterable<SynthesisEvent>\n}\n\n// ─── 浏览器 WebSocket 辅助 ───\n\n/**\n * WebSocket 收到的消息\n *\n * @example\n * ```ts\n * // 模块内部上下文：示例复用当前声明所在模块的实际绑定。\n * const frame: BrowserWsMessage = { isBinary: true, binary: new Uint8Array([0, 0]) }\n * const bytes = frame.isBinary ? frame.binary : undefined\n * ```\n */\ninterface BrowserWsMessage {\n  text?: string\n  binary?: Uint8Array\n  isBinary: boolean\n}\n\n/**\n * 浏览器 WebSocket 连接封装\n *\n * @example\n * ```ts\n * // 模块内部上下文：示例复用当前声明所在模块的实际绑定。\n * async function consume(connection: BrowserWsConnection) {\n *   try {\n *     connection.send(JSON.stringify({ type: 'done' }))\n *     for await (const frame of connection.messages()) {\n *       if (!frame.isBinary && frame.text) return JSON.parse(frame.text)\n *     }\n *     return undefined\n *   }\n *   finally { connection.close() }\n * }\n * ```\n */\ninterface BrowserWsConnection {\n  send: (data: string | Uint8Array) => void\n  messages: () => AsyncIterableIterator<BrowserWsMessage>\n  close: () => void\n}\n\n/** 建立浏览器 WebSocket 连接（内部消息队列 → 异步迭代） */\nfunction openBrowserWs(url: string, signal?: AbortSignal): Promise<BrowserWsConnection> {\n  return new Promise((resolve, reject) => {\n    const ws = new WebSocket(url)\n    ws.binaryType = 'arraybuffer'\n\n    const queue: BrowserWsMessage[] = []\n    let waiting: ((result: IteratorResult<BrowserWsMessage>) => void) | null = null\n    let closed = false\n    let opened = false\n    let aborted = false\n    let failure: Error | null = null\n\n    function settleClosed(): void {\n      if (closed)\n        return\n      closed = true\n      signal?.removeEventListener('abort', onAbort)\n      if (waiting) {\n        const w = waiting\n        waiting = null\n        w({ value: undefined, done: true })\n      }\n    }\n\n    function onAbort(): void {\n      aborted = true\n      if (!opened) {\n        // 连接建立前取消：直接以取消错误拒绝建连\n        settleClosed()\n        reject(audioClientError(HaiAIError.AUDIO_CANCELLED.code, 'Audio operation cancelled'))\n      }\n      // 连接建立后取消：关闭连接，由消息迭代器在终止时抛出 AUDIO_CANCELLED\n      ws.close()\n    }\n\n    /**\n     * socket 关闭且未按协议正常收到 `end` 时的终止错误。\n     *\n     * 消费方在收到服务端 `end` 时会主动 break，因而正常结束不会走到此分支；\n     * 走到此分支即意味着连接在完成前关闭，须区分取消 / 网络异常抛出对应领域错误。\n     */\n    function terminalError(): Error {\n      if (aborted)\n        return audioClientError(HaiAIError.AUDIO_CANCELLED.code, 'Audio operation cancelled')\n      if (failure)\n        return failure\n      return audioClientError(HaiAIError.AUDIO_CONNECTION_FAILED.code, 'Audio WebSocket closed before completion')\n    }\n\n    function push(message: BrowserWsMessage): void {\n      if (waiting) {\n        const w = waiting\n        waiting = null\n        w({ value: message, done: false })\n      }\n      else {\n        queue.push(message)\n      }\n    }\n\n    ws.onmessage = (event: MessageEvent): void => {\n      if (typeof event.data === 'string')\n        push({ isBinary: false, text: event.data })\n      else\n        push({ isBinary: true, binary: new Uint8Array(event.data as ArrayBuffer) })\n    }\n    // 连接建立前后的错误都记录：建立前直接拒绝，建立后由迭代器在关闭时抛出 AUDIO_CONNECTION_FAILED\n    ws.onerror = (): void => {\n      failure = audioClientError(HaiAIError.AUDIO_CONNECTION_FAILED.code, 'Audio WebSocket connection failed')\n      if (!opened) {\n        settleClosed()\n        reject(failure)\n      }\n    }\n    ws.onclose = (): void => settleClosed()\n\n    if (signal) {\n      if (signal.aborted)\n        onAbort()\n      else\n        signal.addEventListener('abort', onAbort, { once: true })\n    }\n\n    ws.onopen = (): void => {\n      opened = true\n      resolve({\n        // DOM WebSocket.send 的 BufferSource 泛型限定为 ArrayBuffer，Uint8Array 需显式收窄\n        send: data => ws.send(data as string | Uint8Array<ArrayBuffer>),\n        async* messages() {\n          while (true) {\n            if (queue.length > 0) {\n              yield queue.shift() as BrowserWsMessage\n              continue\n            }\n            if (closed)\n              throw terminalError()\n            const next = await new Promise<IteratorResult<BrowserWsMessage>>((res) => {\n              waiting = res\n            })\n            if (next.done)\n              throw terminalError()\n            yield next.value\n          }\n        },\n        close: () => ws.close(),\n      })\n    }\n  })\n}\n\n// ─── 客户端实现 ───\n\n/**\n * 创建浏览器语音客户端\n *\n * @param config - 语音客户端配置\n * @returns 浏览器语音操作接口\n *\n * @example\n * ```ts\n * import { createAudioClient } from '@h-ai/ai/client'\n * import type { AudioTicketRequest } from '@h-ai/ai/client'\n * function connect(issueTicket: (request: AudioTicketRequest) => Promise<string>) {\n *   return createAudioClient({ url: 'wss://example.com/api/v1/ai/audio', getTicket: issueTicket })\n * }\n * ```\n */\nexport function createAudioClient(config: AudioClientConfig): AudioClientOperations {\n  async function buildUrl(request: AudioTicketRequest): Promise<string> {\n    const ticket = await config.getTicket(request)\n    if (!ticket)\n      throw new Error('Audio WebSocket ticket is required')\n    const separator = config.url.includes('?') ? '&' : '?'\n    return `${config.url}${separator}ticket=${encodeURIComponent(ticket)}`\n  }\n\n  function startMessage(operation: 'transcribe' | 'synthesize', extra: Partial<AudioWsStartMessage>): string {\n    return JSON.stringify({ type: 'start', operation, ...extra } satisfies AudioWsStartMessage)\n  }\n\n  async function* transcribeStream(request: TranscriptionStreamRequest): AsyncIterable<TranscriptionEvent> {\n    const conn = await openBrowserWs(await buildUrl({ operation: 'transcribe', model: request.model }), request.signal)\n    try {\n      const format = request.audio.format\n      const sampleRate = request.audio.sampleRate\n      conn.send(startMessage('transcribe', { stream: true, model: request.model, language: request.language, contextHints: request.contextHints, format, sampleRate, channels: request.audio.channels }))\n\n      const sendAudio = (async () => {\n        if ('chunks' in request.audio) {\n          for await (const part of request.audio.chunks)\n            conn.send(part)\n        }\n        else {\n          conn.send(request.audio.data)\n        }\n        conn.send(JSON.stringify({ type: 'done' }))\n      })()\n      let sendError: unknown\n      sendAudio.catch((e: unknown) => {\n        sendError = e\n      })\n\n      for await (const message of conn.messages()) {\n        if (message.isBinary || !message.text)\n          continue\n        const event = JSON.parse(message.text) as AudioWsServerMessage\n        if (event.type === 'transcript')\n          yield { type: 'transcript', text: event.text, final: event.final }\n        else if (event.type === 'speech_started')\n          yield { type: 'speech_started' }\n        else if (event.type === 'speech_stopped')\n          yield { type: 'speech_stopped' }\n        else if (event.type === 'error')\n          throw audioClientError(event.code, event.message)\n        else if (event.type === 'end')\n          break\n      }\n      if (sendError)\n        throw sendError\n    }\n    finally {\n      conn.close()\n    }\n  }\n\n  async function transcribe(request: TranscriptionRequest): Promise<TranscriptionResult> {\n    // 完整识别：服务端缓冲全部音频后返回单条最终结果\n    const conn = await openBrowserWs(await buildUrl({ operation: 'transcribe', model: request.model }), request.signal)\n    try {\n      conn.send(startMessage('transcribe', { stream: false, model: request.model, language: request.language, contextHints: request.contextHints, format: request.audio.format, sampleRate: request.audio.sampleRate, channels: request.audio.channels }))\n      conn.send(request.audio.data)\n      conn.send(JSON.stringify({ type: 'done' }))\n\n      let text = ''\n      for await (const message of conn.messages()) {\n        if (message.isBinary || !message.text)\n          continue\n        const event = JSON.parse(message.text) as AudioWsServerMessage\n        if (event.type === 'transcript')\n          text = event.text\n        else if (event.type === 'error')\n          throw audioClientError(event.code, event.message)\n        else if (event.type === 'end')\n          break\n      }\n      return { text }\n    }\n    finally {\n      conn.close()\n    }\n  }\n\n  async function* synthesizeStream(request: SynthesisStreamRequest): AsyncIterable<SynthesisEvent> {\n    const conn = await openBrowserWs(await buildUrl({ operation: 'synthesize', model: request.model }), request.signal)\n    try {\n      conn.send(startMessage('synthesize', { model: request.model, voice: request.voice, instruction: request.instruction, format: request.format, sampleRate: request.sampleRate }))\n\n      const sendText = (async () => {\n        const segments = isSynthesisTextSegment(request.text) ? singleSegment(request.text) : request.text\n        for await (const segment of segments)\n          conn.send(JSON.stringify({ type: 'text', segmentId: segment.id, text: segment.text }))\n        conn.send(JSON.stringify({ type: 'done' }))\n      })()\n      let sendError: unknown\n      sendText.catch((e: unknown) => {\n        sendError = e\n      })\n\n      let activeSegmentId: string | undefined\n      for await (const message of conn.messages()) {\n        if (message.isBinary && message.binary) {\n          if (!activeSegmentId)\n            throw new Error('Audio frame received outside a synthesis segment')\n          yield { type: 'audio', segmentId: activeSegmentId, data: message.binary }\n          continue\n        }\n        if (!message.text)\n          continue\n        const event = JSON.parse(message.text) as AudioWsServerMessage\n        if (event.type === 'segment_started') {\n          if (activeSegmentId)\n            throw new Error('Synthesis segment started before the previous segment completed')\n          activeSegmentId = event.segmentId\n          yield event\n        }\n        else if (event.type === 'segment_done') {\n          if (activeSegmentId !== event.segmentId)\n            throw new Error('Synthesis segment completion does not match the active segment')\n          yield event\n          activeSegmentId = undefined\n        }\n        else if (event.type === 'error') {\n          throw audioClientError(event.code, event.message)\n        }\n        else if (event.type === 'end') {\n          // 收到 end 但仍有未完成的合成段：视为本次合成失败，不得把部分音频当作成功结果\n          if (activeSegmentId)\n            throw audioClientError(HaiAIError.AUDIO_PROTOCOL_ERROR.code, 'Synthesis ended before the active segment completed')\n          break\n        }\n      }\n      if (sendError)\n        throw sendError\n    }\n    finally {\n      conn.close()\n    }\n  }\n\n  async function synthesize(request: SynthesisRequest): Promise<SynthesisResult> {\n    const chunks: Uint8Array[] = []\n    let meta: { format: SynthesisResult['format'], sampleRate?: number, channels?: 1 | 2 } | undefined\n    for await (const event of synthesizeStream({ ...request, text: { id: 'synthesis', text: request.text } })) {\n      if (event.type === 'segment_started')\n        meta = { format: event.format, sampleRate: event.sampleRate, channels: event.channels }\n      else if (event.type === 'audio')\n        chunks.push(event.data)\n    }\n    // 音频格式来自服务端解析后的 Provider 真实输出（随 segment_started 下发），不由请求参数猜测\n    if (!meta)\n      throw audioClientError(HaiAIError.AUDIO_PROTOCOL_ERROR.code, 'Synthesis completed without segment metadata')\n    const total = chunks.reduce((sum, c) => sum + c.length, 0)\n    const data = new Uint8Array(total)\n    let offset = 0\n    for (const chunk of chunks) {\n      data.set(chunk, offset)\n      offset += chunk.length\n    }\n    return { data, format: meta.format, sampleRate: meta.sampleRate, channels: meta.channels ?? 1 }\n  }\n\n  return { transcribe, transcribeStream, synthesize, synthesizeStream }\n}\n\nfunction isSynthesisTextSegment(value: SynthesisStreamRequest['text']): value is SynthesisTextSegment {\n  return typeof value === 'object' && value !== null && 'id' in value && 'text' in value\n}\n\nasync function* singleSegment(segment: SynthesisTextSegment): AsyncIterable<SynthesisTextSegment> {\n  yield segment\n}\n"]}