@roboflow/inference-sdk 0.1.8 → 0.1.10

package/dist/index.es.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.es.js","sources":["../src/inference-api.ts","../src/streams.ts","../src/video-upload.ts","../src/webrtc.ts"],"sourcesContent":["/**\n * Base URL for the Roboflow API (used for TURN server configuration)\n * Can be overridden via environment variable in Node.js environments\n */\nconst RF_API_BASE_URL = typeof process !== \"undefined\" && process.env?.RF_API_BASE_URL\n  ? process.env.RF_API_BASE_URL\n  : \"https://api.roboflow.com\";\n\n/**\n * List of known Roboflow serverless API URLs where auto TURN config applies\n */\nconst ROBOFLOW_SERVERLESS_URLS = [\n  \"https://serverless.roboflow.com\"\n];\n\nexport interface WebRTCWorkerConfig {\n  imageInputName?: string;\n  streamOutputNames?: string[];\n  dataOutputNames?: string[];\n  threadPoolWorkers?: number;\n  /**\n   * Workflow parameters to pass to the workflow execution\n   */\n  workflowsParameters?: Record<string, any>;\n  /**\n   * ICE servers for WebRTC connections (used for both client and server)\n   */\n  iceServers?: RTCIceServerConfig[];\n  /**\n   * Processing timeout in seconds (serverless only)\n   * @default 600\n   */\n  processingTimeout?: number;\n  /**\n   * Requested compute plan (serverless only)\n   * @example \"webrtc-gpu-small\"\n   */\n  requestedPlan?: string;\n  /**\n   * Requested region for processing (serverless only)\n   * @example \"us\"\n   */\n  requestedRegion?: string;\n  /**\n   * Set to false for file upload mode (batch processing).\n   * When false, server processes all frames sequentially instead of dropping frames.\n   * @default true\n   */\n  realtimeProcessing?: boolean;\n}\n\n/**\n * ICE server configuration for WebRTC connections\n *\n * Use this to configure custom STUN/TURN servers for users behind\n * symmetric NAT or restrictive firewalls.\n */\nexport interface RTCIceServerConfig {\n  urls: string[];\n  username?: string;\n  credential?: string;\n}\n\nexport interface WebRTCOffer {\n  sdp: string;\n  type: string;\n}\n\nexport type WorkflowSpec = Record<string, any>;\n\nexport interface WebRTCWorkerResponse {\n  status?: string;\n  sdp: string;\n  type: string;\n  context?: {\n    request_id: string | null;\n    pipeline_id: string | null;\n  };\n}\n\nexport interface WebRTCParams {\n  workflowSpec?: WorkflowSpec;\n  workspaceName?: string;\n  workflowId?: string;\n  imageInputName?: string;\n  streamOutputNames?: string[];\n  dataOutputNames?: string[];\n  threadPoolWorkers?: number;\n  /**\n   * Workflow parameters to pass to the workflow execution\n   */\n  workflowsParameters?: Record<string, any>;\n  /**\n   * ICE servers for WebRTC connections (used for both client and server)\n   *\n   * Use this to specify custom STUN/TURN servers for users behind\n   * symmetric NAT or restrictive firewalls. The same configuration is\n   * used for both the client-side RTCPeerConnection and sent to the\n   * server via webrtc_config.\n   *\n   * @example\n   * ```typescript\n   * iceServers: [\n   *   { urls: [\"stun:stun.l.google.com:19302\"] },\n   *   { urls: [\"turn:turn.example.com:3478\"], username: \"user\", credential: \"pass\" }\n   * ]\n   * ```\n   */\n  iceServers?: RTCIceServerConfig[];\n  /**\n   * Processing timeout in seconds (serverless only)\n   * @default 600\n   */\n  processingTimeout?: number;\n  /**\n   * Requested compute plan (serverless only)\n   * @example \"webrtc-gpu-small\"\n   */\n  requestedPlan?: string;\n  /**\n   * Requested region for processing (serverless only)\n   * @example \"us\"\n   */\n  requestedRegion?: string;\n  /**\n   * Set to false for file upload mode (batch processing).\n   * When false, server processes all frames sequentially instead of dropping frames.\n   * @default true\n   */\n  realtimeProcessing?: boolean;\n}\n\nexport interface Connector {\n  connectWrtc(offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse>;\n  /**\n   * Fetch ICE servers (TURN configuration) for WebRTC connections\n   * This should be called BEFORE creating the RTCPeerConnection to ensure\n   * proper NAT traversal configuration.\n   *\n   * @returns Promise resolving to ICE server configuration, or null/undefined if not available\n   */\n  getIceServers?(): Promise<RTCIceServerConfig[] | null>;\n  _apiKey?: string;\n  _serverUrl?: string;\n}\n\nexport class InferenceHTTPClient {\n  private apiKey: string;\n  private serverUrl: string;\n\n  /**\n   * @private\n   * Use InferenceHTTPClient.init() instead\n   */\n  private constructor(apiKey: string, serverUrl: string = \"https://serverless.roboflow.com\") {\n    this.apiKey = apiKey;\n    this.serverUrl = serverUrl;\n  }\n\n  static init({ apiKey, serverUrl }: { apiKey: string; serverUrl?: string }): InferenceHTTPClient {\n    if (!apiKey) {\n      throw new Error(\"apiKey is required\");\n    }\n    return new InferenceHTTPClient(apiKey, serverUrl);\n  }\n\n  /**\n   * Initialize a WebRTC worker pipeline\n   *\n   * @param params - Pipeline parameters\n   * @param params.offer - WebRTC offer { sdp, type }\n   * @param params.workflowSpec - Workflow specification\n   * @param params.config - Additional configuration\n   * @param params.config.imageInputName - Input image name (default: \"image\")\n   * @param params.config.streamOutputNames - Output stream names for video (default: [])\n   * @param params.config.dataOutputNames - Output data names (default: [\"string\"])\n   * @param params.config.threadPoolWorkers - Thread pool workers (default: 4)\n   * @returns Promise resolving to answer with SDP and pipeline ID\n   *\n   * @example\n   * ```typescript\n   * const answer = await client.initializeWebrtcWorker({\n   *   offer: { sdp, type },\n   *   workflowSpec: { ... },\n   *   config: {\n   *     imageInputName: \"image\",\n   *     streamOutputNames: [\"output_image\"]\n   *   }\n   * });\n   * ```\n   */\n  async initializeWebrtcWorker({\n    offer,\n    workflowSpec,\n    workspaceName,\n    workflowId,\n    config = {}\n  }: {\n    offer: WebRTCOffer;\n    workflowSpec?: WorkflowSpec;\n    workspaceName?: string;\n    workflowId?: string;\n    config?: WebRTCWorkerConfig;\n  }): Promise<WebRTCWorkerResponse> {\n    if (!offer || !offer.sdp || !offer.type) {\n      throw new Error(\"offer with sdp and type is required\");\n    }\n\n    // Validate that either workflowSpec OR (workspaceName + workflowId) is provided\n    const hasWorkflowSpec = !!workflowSpec;\n    const hasWorkspaceIdentifier = !!(workspaceName && workflowId);\n\n    if (!hasWorkflowSpec && !hasWorkspaceIdentifier) {\n      throw new Error(\"Either workflowSpec OR (workspaceName + workflowId) is required\");\n    }\n    if (hasWorkflowSpec && hasWorkspaceIdentifier) {\n      throw new Error(\"Provide either workflowSpec OR (workspaceName + workflowId), not both\");\n    }\n\n    const {\n      imageInputName = \"image\",\n      streamOutputNames = [],\n      dataOutputNames = [\"string\"],\n      threadPoolWorkers = 4,\n      workflowsParameters = {},\n      iceServers,\n      processingTimeout,\n      requestedPlan,\n      requestedRegion,\n      realtimeProcessing = true\n    } = config as any;\n\n    // Build workflow_configuration based on what's provided\n    const workflowConfiguration: any = {\n      type: \"WorkflowConfiguration\",\n      image_input_name: imageInputName,\n      workflows_parameters: workflowsParameters,\n      workflows_thread_pool_workers: threadPoolWorkers,\n      cancel_thread_pool_tasks_on_exit: true,\n      video_metadata_input_name: \"video_metadata\"\n    };\n\n    if (hasWorkflowSpec) {\n      workflowConfiguration.workflow_specification = workflowSpec;\n    } else {\n      workflowConfiguration.workspace_name = workspaceName;\n      workflowConfiguration.workflow_id = workflowId;\n    }\n\n    const payload: Record<string, any> = {\n      workflow_configuration: workflowConfiguration,\n      api_key: this.apiKey,\n      webrtc_realtime_processing: realtimeProcessing,\n      webrtc_offer: {\n        sdp: offer.sdp,\n        type: offer.type\n      },\n      webrtc_config: iceServers ? { iceServers } : null,\n      stream_output: streamOutputNames,\n      data_output: dataOutputNames\n    };\n\n    // Add serverless-specific fields if provided\n    if (processingTimeout !== undefined) {\n      payload.processing_timeout = processingTimeout;\n    }\n    if (requestedPlan !== undefined) {\n      payload.requested_plan = requestedPlan;\n    }\n    if (requestedRegion !== undefined) {\n      payload.requested_region = requestedRegion;\n    }\n    const response = await fetch(`${this.serverUrl}/initialise_webrtc_worker`, {\n      method: \"POST\",\n      headers: { \"Content-Type\": \"application/json\" },\n      body: JSON.stringify(payload)\n    });\n\n    if (!response.ok) {\n      const errorText = await response.text().catch(() => \"\");\n      throw new Error(`initialise_webrtc_worker failed (${response.status}): ${errorText}`);\n    }\n\n    const result = await response.json();\n\n    return result;\n  }\n\n  async terminatePipeline({ pipelineId }: { pipelineId: string }): Promise<void> {\n    if (!pipelineId) {\n      throw new Error(\"pipelineId is required\");\n    }\n\n    await fetch(\n      `${this.serverUrl}/inference_pipelines/${pipelineId}/terminate?api_key=${this.apiKey}`,\n      {\n        method: \"POST\",\n        headers: { \"Content-Type\": \"application/json\" }\n      }\n    );\n  }\n\n  /**\n   * Fetch TURN server configuration from Roboflow API\n   *\n   * This automatically fetches TURN server credentials for improved WebRTC\n   * connectivity through firewalls and NAT. Only applicable when using\n   * Roboflow serverless infrastructure.\n   *\n   * @returns Promise resolving to ICE server configuration, or null if not applicable\n   *\n   * @example\n   * ```typescript\n   * const client = InferenceHTTPClient.init({ apiKey: \"your-api-key\" });\n   * const iceServers = await client.fetchTurnConfig();\n   * // Returns: [{ urls: [\"turn:...\"], username: \"...\", credential: \"...\" }]\n   * ```\n   */\n  async fetchTurnConfig(): Promise<RTCIceServerConfig[] | null> {\n    // // Only fetch TURN config for Roboflow serverless URLs\n    if (!ROBOFLOW_SERVERLESS_URLS.includes(this.serverUrl)) {\n      return null;\n    }\n    try {\n      const response = await fetch(\n        `${RF_API_BASE_URL}/webrtc_turn_config?api_key=${this.apiKey}`,\n        {\n          method: \"GET\",\n          headers: { \"Content-Type\": \"application/json\" }\n        }\n      );\n\n      if (!response.ok) {\n        console.warn(`[RFWebRTC] Failed to fetch TURN config (${response.status}), using defaults`);\n        return null;\n      }\n\n      const turnConfig = await response.json();\n\n      // Handle 3 formats:\n      // 1. Single server object: { urls, username, credential }\n      // 2. Array of servers: [{ urls, username, credential }, ...]\n      // 3. Object with iceServers: { iceServers: [...] }\n      let iceServersRaw: any[];\n\n      if (Array.isArray(turnConfig)) {\n        // Format 2: array of servers\n        iceServersRaw = turnConfig;\n      } else if (turnConfig.iceServers && Array.isArray(turnConfig.iceServers)) {\n        // Format 3: object with iceServers array\n        iceServersRaw = turnConfig.iceServers;\n      } else if (turnConfig.urls) {\n        // Format 1: single server object - wrap in array\n        iceServersRaw = [turnConfig];\n      } else {\n        console.warn(\"[RFWebRTC] Invalid TURN config format, using defaults\");\n        return null;\n      }\n\n      // Normalize the ICE servers format\n      const iceServers: RTCIceServerConfig[] = iceServersRaw.map((server: any) => ({\n        urls: Array.isArray(server.urls) ? server.urls : [server.urls],\n        username: server.username,\n        credential: server.credential\n      }));\n\n      return iceServers;\n    } catch (err) {\n      console.warn(\"[RFWebRTC] Error fetching TURN config:\", err);\n      return null;\n    }\n  }\n}\n\n/**\n * Connectors for establishing WebRTC connections to Roboflow\n */\nexport const connectors = {\n  /**\n   * Create a connector that uses API key directly\n   *\n   * **WARNING**: If you use this in the frontend, it will expose your API key. \n   * Use only for demos/testing.\n   * For production, use withProxyUrl() with a backend proxy.\n   *\n   * @param apiKey - Roboflow API key\n   * @param options - Additional options\n   * @param options.serverUrl - Custom Roboflow server URL\n   * @returns Connector with connectWrtc method\n   *\n   * @example\n   * ```typescript\n   * const connector = connectors.withApiKey(\"your-api-key\");\n   * const answer = await connector.connectWrtc(offer, wrtcParams);\n   * ```\n   */\n  withApiKey(apiKey: string, options: { serverUrl?: string } = {}): Connector {\n    const { serverUrl } = options;\n\n    // Warn if running in browser context\n    if (typeof window !== 'undefined') {\n      console.warn(\n        '[Security Warning] Using API key directly in browser will expose it. ' +\n        'Use connectors.withProxyUrl() for production. ' +\n        'See: https://docs.roboflow.com/api-reference/authentication#securing-your-api-key'\n      );\n    }\n\n    const client = InferenceHTTPClient.init({ apiKey, serverUrl });\n\n    return {\n      connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n        console.debug(\"wrtcParams\", wrtcParams);\n        const answer = await client.initializeWebrtcWorker({\n          offer,\n          workflowSpec: wrtcParams.workflowSpec,\n          workspaceName: wrtcParams.workspaceName,\n          workflowId: wrtcParams.workflowId,\n          config: {\n            imageInputName: wrtcParams.imageInputName,\n            streamOutputNames: wrtcParams.streamOutputNames,\n            dataOutputNames: wrtcParams.dataOutputNames,\n            threadPoolWorkers: wrtcParams.threadPoolWorkers,\n            workflowsParameters: wrtcParams.workflowsParameters,\n            iceServers: wrtcParams.iceServers,\n            processingTimeout: wrtcParams.processingTimeout,\n            requestedPlan: wrtcParams.requestedPlan,\n            requestedRegion: wrtcParams.requestedRegion,\n            realtimeProcessing: wrtcParams.realtimeProcessing\n          }\n        });\n\n        return answer;\n      },\n\n      /**\n       * Fetch TURN server configuration for improved WebRTC connectivity\n       */\n      getIceServers: async (): Promise<RTCIceServerConfig[] | null> => {\n        return await client.fetchTurnConfig();\n      },\n\n      // Store apiKey for cleanup\n      _apiKey: apiKey,\n      _serverUrl: serverUrl\n    };\n  },\n\n  /**\n   * Create a connector that uses a backend proxy (recommended for production)\n   *\n   * Your backend receives the offer and wrtcParams, adds the secret API key,\n   * and forwards to Roboflow. This keeps your API key secure.\n   *\n   * For improved WebRTC connectivity through firewalls, implement a separate\n   * endpoint for TURN server configuration that calls `fetchTurnConfig()`.\n   *\n   * @param proxyUrl - Backend proxy endpoint URL for WebRTC initialization\n   * @param options - Additional options\n   * @param options.turnConfigUrl - Optional URL for fetching TURN server configuration\n   * @returns Connector with connectWrtc and optional getIceServers methods\n   *\n   * @example\n   * ```typescript\n   * // Frontend: Create connector with TURN config endpoint\n   * const connector = connectors.withProxyUrl('/api/init-webrtc', {\n   *   turnConfigUrl: '/api/turn-config'\n   * });\n   * ```\n   *\n   * @example\n   * Backend implementation (Express) with TURN server support:\n   * ```typescript\n   * // Endpoint for TURN configuration (called first by SDK)\n   * app.get('/api/turn-config', async (req, res) => {\n   *   const client = InferenceHTTPClient.init({\n   *     apiKey: process.env.ROBOFLOW_API_KEY\n   *   });\n   *   const iceServers = await client.fetchTurnConfig();\n   *   res.json({ iceServers });\n   * });\n   *\n   * // Endpoint for WebRTC initialization\n   * app.post('/api/init-webrtc', async (req, res) => {\n   *   const { offer, wrtcParams } = req.body;\n   *   const client = InferenceHTTPClient.init({\n   *     apiKey: process.env.ROBOFLOW_API_KEY\n   *   });\n   *\n   *   const answer = await client.initializeWebrtcWorker({\n   *     offer,\n   *     workflowSpec: wrtcParams.workflowSpec,\n   *     workspaceName: wrtcParams.workspaceName,\n   *     workflowId: wrtcParams.workflowId,\n   *     config: {\n   *       imageInputName: wrtcParams.imageInputName,\n   *       streamOutputNames: wrtcParams.streamOutputNames,\n   *       dataOutputNames: wrtcParams.dataOutputNames,\n   *       threadPoolWorkers: wrtcParams.threadPoolWorkers,\n   *       workflowsParameters: wrtcParams.workflowsParameters,\n   *       iceServers: wrtcParams.iceServers,\n   *       processingTimeout: wrtcParams.processingTimeout,\n   *       requestedPlan: wrtcParams.requestedPlan,\n   *       requestedRegion: wrtcParams.requestedRegion\n   *     }\n   *   });\n   *\n   *   res.json(answer);\n   * });\n   * ```\n   */\n  withProxyUrl(proxyUrl: string, options: { turnConfigUrl?: string } = {}): Connector {\n    const { turnConfigUrl } = options;\n\n    return {\n      connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n        const response = await fetch(proxyUrl, {\n          method: \"POST\",\n          headers: { \"Content-Type\": \"application/json\" },\n          body: JSON.stringify({\n            offer,\n            wrtcParams\n          })\n        });\n\n        if (!response.ok) {\n          const errorText = await response.text().catch(() => \"\");\n          throw new Error(`Proxy request failed (${response.status}): ${errorText}`);\n        }\n\n        return await response.json();\n      },\n\n      /**\n       * Fetch TURN server configuration from the proxy backend\n       * Only available if turnConfigUrl was provided\n       */\n      getIceServers: turnConfigUrl\n        ? async (): Promise<RTCIceServerConfig[] | null> => {\n            try {\n              const response = await fetch(turnConfigUrl, {\n                method: \"GET\",\n                headers: { \"Content-Type\": \"application/json\" }\n              });\n\n              if (!response.ok) {\n                console.warn(`[RFWebRTC] Failed to fetch TURN config from proxy (${response.status})`);\n                return null;\n              }\n\n              const data = await response.json();\n              return data.iceServers || null;\n            } catch (err) {\n              console.warn(\"[RFWebRTC] Error fetching TURN config from proxy:\", err);\n              return null;\n            }\n          }\n        : undefined\n    };\n  }\n};\n","/**\n * Get a camera stream with the given constraints.\n *\n * @param constraints - MediaStreamConstraints for getUserMedia\n * @returns Promise that resolves to MediaStream\n *\n * @example\n * ```typescript\n * const stream = await useCamera({\n *   video: {\n *     facingMode: { ideal: \"user\" },\n *     width: { ideal: 1280 },\n *     height: { ideal: 720 },\n *     frameRate: { ideal: 30 }\n *   },\n *   audio: false\n * });\n * ```\n */\nexport async function useCamera(constraints: MediaStreamConstraints = { video: true }): Promise<MediaStream> {\n  try {\n    console.log(\"[RFStreams] requesting with\", constraints);\n    const stream = await navigator.mediaDevices.getUserMedia(constraints);\n    console.log(\"[RFStreams] got stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n    return stream;\n  } catch (err) {\n    console.warn(\"[RFStreams] failed, falling back\", err);\n    const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: false });\n    console.log(\"[RFStreams] fallback stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n    return stream;\n  }\n}\n  \nexport function stopStream(stream: MediaStream | null | undefined): void {\n  if (stream) {\n    stream.getTracks().forEach(track => track.stop());\n    console.log(\"[RFStreams] Stream stopped\");\n  }\n}\n","/**\n * Video file upload via WebRTC datachannel\n *\n * This module provides the FileUploader class for chunked file uploads\n * through WebRTC datachannels with backpressure handling.\n */\n\n/**\n * Configuration constants for file upload (matching Python SDK)\n */\nconst CHUNK_SIZE = 49152;       // 49KB - safe for WebRTC\nconst BUFFER_LIMIT = 262144;    // 256KB - backpressure threshold\nconst POLL_INTERVAL = 10;       // 10ms buffer check interval\n\n/**\n * Helper to sleep for a given number of milliseconds\n */\nfunction sleep(ms: number): Promise<void> {\n  return new Promise(resolve => setTimeout(resolve, ms));\n}\n\n/**\n * FileUploader handles chunked file upload with backpressure\n *\n * Uploads files through a WebRTC datachannel in 49KB chunks with\n * intelligent backpressure handling to prevent overwhelming the network.\n */\nexport class FileUploader {\n  private file: File;\n  private channel: RTCDataChannel;\n  private totalChunks: number;\n  private cancelled: boolean = false;\n\n  constructor(file: File, channel: RTCDataChannel) {\n    this.file = file;\n    this.channel = channel;\n    this.totalChunks = Math.ceil(file.size / CHUNK_SIZE);\n  }\n\n  /**\n   * Cancel the upload\n   */\n  cancel(): void {\n    this.cancelled = true;\n  }\n\n  /**\n   * Upload the file in chunks with backpressure handling\n   *\n   * @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n   */\n  async upload(onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n    const totalBytes = this.file.size;\n\n    for (let chunkIndex = 0; chunkIndex < this.totalChunks; chunkIndex++) {\n      // Check for cancellation\n      if (this.cancelled) {\n        throw new Error(\"Upload cancelled\");\n      }\n\n      // Check channel state\n      if (this.channel.readyState !== \"open\") {\n        throw new Error(\"Video upload interrupted\");\n      }\n\n      // Read chunk from file\n      const start = chunkIndex * CHUNK_SIZE;\n      const end = Math.min(start + CHUNK_SIZE, totalBytes);\n      const chunkBlob = this.file.slice(start, end);\n      const chunkData = new Uint8Array(await chunkBlob.arrayBuffer());\n\n      // Create message with 8-byte header (chunkIndex + totalChunks as uint32 LE)\n      const message = new ArrayBuffer(8 + chunkData.length);\n      const view = new DataView(message);\n      view.setUint32(0, chunkIndex, true);      // little-endian\n      view.setUint32(4, this.totalChunks, true); // little-endian\n      new Uint8Array(message, 8).set(chunkData);\n\n      // Backpressure: wait for buffer to drain\n      while (this.channel.bufferedAmount > BUFFER_LIMIT) {\n        if (this.channel.readyState !== \"open\") {\n          throw new Error(\"Video upload interrupted\");\n        }\n        await sleep(POLL_INTERVAL);\n      }\n\n      // Send chunk\n      this.channel.send(message);\n\n      // Report progress\n      if (onProgress) {\n        onProgress(end, totalBytes);\n      }\n    }\n  }\n}\n","\nimport { InferenceHTTPClient, Connector, WebRTCParams, RTCIceServerConfig } from \"./inference-api\";\nimport { stopStream } from \"./streams\";\nimport { WebRTCOutputData } from \"./webrtc-types\";\nimport { FileUploader } from \"./video-upload\";\n\n// Re-export shared types\nexport type { WebRTCVideoMetadata, WebRTCOutputData } from \"./webrtc-types\";\n\n// Re-export FileUploader from video-upload\nexport { FileUploader } from \"./video-upload\";\n\n/**\n * Binary protocol header size (frame_id + chunk_index + total_chunks)\n * Each field is 4 bytes uint32 little-endian\n */\nconst HEADER_SIZE = 12;\n\n/**\n * Reassembles chunked binary messages from the datachannel\n */\nexport class ChunkReassembler {\n  private pendingFrames: Map<number, {\n    chunks: Map<number, Uint8Array>;\n    totalChunks: number;\n  }> = new Map();\n\n  /**\n   * Process an incoming chunk and return the complete message if all chunks received\n   */\n  processChunk(frameId: number, chunkIndex: number, totalChunks: number, payload: Uint8Array): Uint8Array | null {\n    // Single chunk message - return immediately\n    if (totalChunks === 1) {\n      return payload;\n    }\n\n    // Multi-chunk message - accumulate\n    if (!this.pendingFrames.has(frameId)) {\n      this.pendingFrames.set(frameId, {\n        chunks: new Map(),\n        totalChunks\n      });\n    }\n\n    const frame = this.pendingFrames.get(frameId)!;\n    frame.chunks.set(chunkIndex, payload);\n\n    // Check if all chunks received\n    if (frame.chunks.size === totalChunks) {\n      // Reassemble in order\n      const totalLength = Array.from(frame.chunks.values()).reduce((sum, chunk) => sum + chunk.length, 0);\n      const result = new Uint8Array(totalLength);\n      let offset = 0;\n\n      for (let i = 0; i < totalChunks; i++) {\n        const chunk = frame.chunks.get(i)!;\n        result.set(chunk, offset);\n        offset += chunk.length;\n      }\n\n      this.pendingFrames.delete(frameId);\n      return result;\n    }\n\n    return null;\n  }\n\n  /**\n   * Clear all pending frames (for cleanup)\n   */\n  clear(): void {\n    this.pendingFrames.clear();\n  }\n}\n\n/**\n * Parse the binary header from a datachannel message\n */\nexport function parseBinaryHeader(buffer: ArrayBuffer): { frameId: number; chunkIndex: number; totalChunks: number; payload: Uint8Array } {\n  const view = new DataView(buffer);\n  const frameId = view.getUint32(0, true);      // little-endian\n  const chunkIndex = view.getUint32(4, true);   // little-endian\n  const totalChunks = view.getUint32(8, true);  // little-endian\n  const payload = new Uint8Array(buffer, HEADER_SIZE);\n\n  return { frameId, chunkIndex, totalChunks, payload };\n}\n\nexport interface UseStreamOptions {\n  disableInputStreamDownscaling?: boolean;\n}\n\nexport interface UseStreamParams {\n  source: MediaStream;\n  connector: Connector;\n  wrtcParams: WebRTCParams;\n  onData?: (data: WebRTCOutputData) => void;\n  options?: UseStreamOptions;\n}\n\nasync function waitForIceGathering(pc: RTCPeerConnection, timeoutMs = 6000): Promise<void> {\n  if (pc.iceGatheringState === \"complete\") return;\n\n  let hasSrflx = false;\n\n  // Track if we get a good candidate (srflx = public IP via STUN)\n  const candidateHandler = (event: RTCPeerConnectionIceEvent) => {\n    if (event.candidate && event.candidate.type === \"srflx\") {\n      hasSrflx = true;\n    }\n  };\n  pc.addEventListener(\"icecandidate\", candidateHandler);\n\n  try {\n    await Promise.race([\n      new Promise<void>(resolve => {\n        const check = () => {\n          if (pc.iceGatheringState === \"complete\") {\n            pc.removeEventListener(\"icegatheringstatechange\", check);\n            resolve();\n          }\n        };\n        pc.addEventListener(\"icegatheringstatechange\", check);\n      }),\n      new Promise<void>((resolve, reject) => {\n        setTimeout(() => {\n          if (!hasSrflx) {\n            console.error(\"[ICE] timeout with NO srflx candidate! Connection may fail.\");\n            reject(new Error(\"ICE gathering timeout without srflx candidate\"));\n          } else {\n            resolve();\n          }\n        }, timeoutMs);\n      })\n    ]);\n  } finally {\n    pc.removeEventListener(\"icecandidate\", candidateHandler);\n  }\n}\n\nfunction setupRemoteStreamListener(pc: RTCPeerConnection): Promise<MediaStream> {\n  return new Promise((resolve) => {\n    pc.addEventListener(\"track\", (event: RTCTrackEvent) => {\n      if (event.streams && event.streams[0]) {\n        resolve(event.streams[0]);\n      }\n    });\n  });\n}\n\nconst DEFAULT_ICE_SERVERS: RTCIceServerConfig[] = [\n  { urls: [\"stun:stun.l.google.com:19302\"] }\n];\n\nasync function preparePeerConnection(\n  localStream?: MediaStream,\n  file?: File,\n  customIceServers?: RTCIceServerConfig[]\n): Promise<{\n  pc: RTCPeerConnection;\n  offer: RTCSessionDescriptionInit;\n  remoteStreamPromise: Promise<MediaStream>;\n  dataChannel: RTCDataChannel;\n  uploadChannel?: RTCDataChannel;\n}> {\n  if (!localStream && !file || (localStream && file)) {\n    throw new Error(\"Either localStream or file must be provided, but not both\");\n  }\n  const iceServers = customIceServers ?? DEFAULT_ICE_SERVERS;\n\n  const pc = new RTCPeerConnection({\n    iceServers: iceServers as RTCIceServer[]\n  });\n\n  // Add transceiver for receiving remote video (BEFORE adding tracks - order matters!)\n  try {\n    pc.addTransceiver(\"video\", { direction: \"recvonly\" });\n  } catch (err) {\n    console.warn(\"[RFWebRTC] Could not add transceiver:\", err);\n  }\n\n  if (localStream) {\n    // Add local tracks\n    localStream.getVideoTracks().forEach(track => {\n      try {\n        // @ts-ignore - contentHint is not in all TypeScript definitions\n        track.contentHint = \"detail\";\n      } catch (e) {\n        // Ignore if contentHint not supported\n      }\n      pc.addTrack(track, localStream);\n    });\n  }\n\n  // Setup remote stream listener\n  const remoteStreamPromise = setupRemoteStreamListener(pc);\n\n  // Create control datachannel (named \"inference\" to match Python SDK)\n  const dataChannel = pc.createDataChannel(\"inference\", {\n    ordered: true\n  });\n\n  // Create upload datachannel for file uploads\n  let uploadChannel: RTCDataChannel | undefined;\n  if (file) {\n    uploadChannel = pc.createDataChannel(\"video_upload\");\n  }\n\n  // Create offer\n  const offer = await pc.createOffer();\n  await pc.setLocalDescription(offer);\n\n  // Wait for ICE gathering\n  await waitForIceGathering(pc);\n\n  return {\n    pc,\n    offer: pc.localDescription!,\n    remoteStreamPromise,\n    dataChannel,\n    uploadChannel\n  };\n}\n\n/**\n * Disable input stream downscaling\n * @private\n */\nasync function disableInputStreamDownscaling(pc: RTCPeerConnection): Promise<void> {\n  const sender = pc.getSenders().find(s => s.track && s.track.kind === \"video\");\n  if (!sender) return;\n\n  const params = sender.getParameters();\n  params.encodings = params.encodings || [{}];\n  params.encodings[0].scaleResolutionDownBy = 1;\n\n  try {\n    await sender.setParameters(params);\n  } catch (err) {\n    console.warn(\"[RFWebRTC] Failed to set encoding parameters:\", err);\n  }\n}\n\n/**\n * Helper to wait for datachannel to open\n */\nfunction waitForChannelOpen(channel: RTCDataChannel, timeoutMs = 30000): Promise<void> {\n  return new Promise((resolve, reject) => {\n    if (channel.readyState === \"open\") {\n      resolve();\n      return;\n    }\n\n    const openHandler = () => {\n      channel.removeEventListener(\"open\", openHandler);\n      channel.removeEventListener(\"error\", errorHandler);\n      clearTimeout(timeout);\n      resolve();\n    };\n\n    const errorHandler = () => {\n      channel.removeEventListener(\"open\", openHandler);\n      channel.removeEventListener(\"error\", errorHandler);\n      clearTimeout(timeout);\n      reject(new Error(\"Datachannel error\"));\n    };\n\n    const timeout = setTimeout(() => {\n      channel.removeEventListener(\"open\", openHandler);\n      channel.removeEventListener(\"error\", errorHandler);\n      reject(new Error(\"Datachannel open timeout\"));\n    }, timeoutMs);\n\n    channel.addEventListener(\"open\", openHandler);\n    channel.addEventListener(\"error\", errorHandler);\n  });\n}\n\n/**\n * WebRTC Connection object\n *\n * Represents an active WebRTC connection to Roboflow for streaming inference\n * or file-based batch processing.\n */\nexport class RFWebRTCConnection {\n  private pc: RTCPeerConnection;\n  private _localStream?: MediaStream;\n  private remoteStreamPromise: Promise<MediaStream>;\n  private pipelineId: string | null;\n  private apiKey: string | null;\n  private dataChannel: RTCDataChannel;\n  private reassembler: ChunkReassembler;\n  private uploadChannel?: RTCDataChannel;\n  private uploader?: FileUploader;\n  private onComplete?: () => void;\n\n  /** @private */\n  constructor(\n    pc: RTCPeerConnection,\n    remoteStreamPromise: Promise<MediaStream>,\n    pipelineId: string | null,\n    apiKey: string | null,\n    dataChannel: RTCDataChannel,\n    options?: {\n      localStream?: MediaStream;\n      uploadChannel?: RTCDataChannel;\n      onData?: (data: any) => void;\n      onComplete?: () => void;\n    }\n  ) {\n    this.pc = pc;\n    this._localStream = options?.localStream;\n    this.remoteStreamPromise = remoteStreamPromise;\n    this.pipelineId = pipelineId;\n    this.apiKey = apiKey;\n    this.dataChannel = dataChannel;\n    this.reassembler = new ChunkReassembler();\n    this.uploadChannel = options?.uploadChannel;\n    this.onComplete = options?.onComplete;\n\n    // Set binary mode for datachannel\n    this.dataChannel.binaryType = \"arraybuffer\";\n\n    const onData = options?.onData;\n\n    // Setup data channel event listeners\n    if (onData) {\n      this.dataChannel.addEventListener(\"message\", (messageEvent: MessageEvent) => {\n        try {\n          // Handle binary protocol with chunking\n          if (messageEvent.data instanceof ArrayBuffer) {\n            const { frameId, chunkIndex, totalChunks, payload } = parseBinaryHeader(messageEvent.data);\n            const completePayload = this.reassembler.processChunk(frameId, chunkIndex, totalChunks, payload);\n\n            if (completePayload) {\n              // Decode UTF-8 JSON payload\n              const decoder = new TextDecoder(\"utf-8\");\n              const jsonString = decoder.decode(completePayload);\n              const data = JSON.parse(jsonString);\n              onData(data);\n            }\n          } else {\n            // Fallback for string messages (shouldn't happen with new protocol)\n            const data = JSON.parse(messageEvent.data);\n            onData(data);\n          }\n        } catch (err) {\n          console.error(\"[RFWebRTC] Failed to parse data channel message:\", err);\n        }\n      });\n\n      this.dataChannel.addEventListener(\"error\", (error) => {\n        console.error(\"[RFWebRTC] Data channel error:\", error);\n      });\n    }\n\n    // Handle channel close - call onComplete when processing finishes\n    this.dataChannel.addEventListener(\"close\", () => {\n      this.reassembler.clear();\n      if (this.onComplete) {\n        this.onComplete();\n      }\n    });\n  }\n\n  /**\n   * Get the remote stream (processed video from Roboflow)\n   *\n   * @returns Promise resolving to the remote MediaStream\n   *\n   * @example\n   * ```typescript\n   * const conn = await useStream({ ... });\n   * const remoteStream = await conn.remoteStream();\n   * videoElement.srcObject = remoteStream;\n   * ```\n   */\n  async remoteStream(): Promise<MediaStream> {\n    return await this.remoteStreamPromise;\n  }\n\n  /**\n   * Get the local stream (original camera)\n   *\n   * @returns The local MediaStream, or undefined if using file upload mode\n   *\n   * @example\n   * ```typescript\n   * const conn = await useStream({ ... });\n   * const localStream = conn.localStream();\n   * if (localStream) {\n   *   videoElement.srcObject = localStream;\n   * }\n   * ```\n   */\n  localStream(): MediaStream | undefined {\n    return this._localStream;\n  }\n\n  /**\n   * Cleanup and close connection\n   *\n   * Terminates the pipeline on Roboflow, closes the peer connection,\n   * and stops the local media stream (if applicable).\n   *\n   * @returns Promise that resolves when cleanup is complete\n   *\n   * @example\n   * ```typescript\n   * const conn = await useStream({ ... });\n   * // ... use connection ...\n   * await conn.cleanup(); // Clean up when done\n   * ```\n   */\n  async cleanup(): Promise<void> {\n    // Cancel any ongoing upload\n    if (this.uploader) {\n      this.uploader.cancel();\n    }\n\n    // Clear pending chunks\n    this.reassembler.clear();\n\n    // Terminate pipeline\n    if (this.pipelineId && this.apiKey) {\n      try {\n        const client = InferenceHTTPClient.init({ apiKey: this.apiKey });\n        await client.terminatePipeline({ pipelineId: this.pipelineId });\n      } catch (err) {\n        console.warn(\"[RFWebRTC] Failed to terminate pipeline:\", err);\n      }\n    }\n\n    // Close peer connection\n    if (this.pc && this.pc.connectionState !== \"closed\") {\n      this.pc.close();\n    }\n\n    // Stop local stream if present\n    if (this._localStream) {\n      stopStream(this._localStream);\n    }\n  }\n\n  /**\n   * Start uploading a file through the connection\n   *\n   * @param file - The file to upload\n   * @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n   * @returns Promise that resolves when upload is complete\n   * @throws Error if no upload channel is available\n   *\n   * @example\n   * ```typescript\n   * await connection.startUpload(videoFile, (uploaded, total) => {\n   *   console.log(`Upload progress: ${(uploaded / total * 100).toFixed(1)}%`);\n   * });\n   * ```\n   */\n  async startUpload(file: File, onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n    if (!this.uploadChannel) {\n      throw new Error(\"No upload channel available. This connection was not created for file uploads.\");\n    }\n\n    // Wait for upload channel to open\n    await waitForChannelOpen(this.uploadChannel);\n\n    this.uploader = new FileUploader(file, this.uploadChannel);\n    await this.uploader.upload(onProgress);\n  }\n\n  /**\n   * Cancel any ongoing file upload\n   */\n  cancelUpload(): void {\n    if (this.uploader) {\n      this.uploader.cancel();\n    }\n  }\n\n  /**\n   * Reconfigure pipeline outputs at runtime\n   *\n   * Dynamically change stream and data outputs without restarting the connection.\n   * Set a field to `null` to leave it unchanged, or to `null` value to enable all outputs,\n   * or to `[]` to disable/auto-detect.\n   *\n   * @param config - Output configuration\n   * @param config.streamOutput - Stream output names (null = unchanged, [] = auto-detect, [\"name\"] = specific output)\n   * @param config.dataOutput - Data output names (null = unchanged, [] = disable, [\"name\"] = specific outputs, null value = all outputs)\n   *\n   * @example\n   * ```typescript\n   * // Change to different stream output\n   * connection.reconfigureOutputs({\n   *   streamOutput: [\"annotated_image\"],\n   *   dataOutput: null  // unchanged\n   * });\n   *\n   * // Enable all data outputs\n   * connection.reconfigureOutputs({\n   *   streamOutput: null,  // unchanged\n   *   dataOutput: null     // null value = all outputs\n   * });\n   *\n   * // Disable all data outputs\n   * connection.reconfigureOutputs({\n   *   streamOutput: null,  // unchanged\n   *   dataOutput: []       // empty array = disable\n   * });\n   * ```\n   */\n  reconfigureOutputs(config: { streamOutput?: string[] | null; dataOutput?: string[] | null }): void {\n    const message: any = {};\n\n    if (config.streamOutput !== undefined) {\n      message.stream_output = config.streamOutput;\n    }\n\n    if (config.dataOutput !== undefined) {\n      message.data_output = config.dataOutput;\n    }\n\n    this.sendData(message);\n  }\n\n  /**\n   * Send data through the data channel\n   * @private\n   */\n  private sendData(data: any): void {\n    if (this.dataChannel.readyState !== \"open\") {\n      console.warn(\"[RFWebRTC] Data channel is not open. Current state:\", this.dataChannel.readyState);\n      return;\n    }\n\n    try {\n      const message = typeof data === \"string\" ? data : JSON.stringify(data);\n      this.dataChannel.send(message);\n    } catch (err) {\n      console.error(\"[RFWebRTC] Failed to send data:\", err);\n    }\n  }\n}\n\n/**\n * Internal base function for establishing WebRTC connection\n * Used by both useStream and useVideoFile\n * @private\n */\ninterface BaseUseStreamParams {\n  source: MediaStream | File;\n  connector: Connector;\n  wrtcParams: WebRTCParams;\n  onData?: (data: WebRTCOutputData) => void;\n  onComplete?: () => void;\n  onFileUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n  options?: UseStreamOptions;\n}\n\nasync function baseUseStream({\n  source,\n  connector,\n  wrtcParams,\n  onData,\n  onComplete,\n  onFileUploadProgress,\n  options = {}\n}: BaseUseStreamParams): Promise<RFWebRTCConnection> {\n  // Validate connector\n  if (!connector || typeof connector.connectWrtc !== \"function\") {\n    throw new Error(\"connector must have a connectWrtc method\");\n  }\n\n  const isFile = source instanceof File;\n  const localStream = isFile ? undefined : source;\n  const file = isFile ? source : undefined;\n\n  // Step 1: Determine ICE servers to use\n  // Priority: 1) User-provided in wrtcParams, 2) From connector.getIceServers(), 3) Defaults\n  let iceServers = wrtcParams.iceServers;\n  if ((!iceServers || iceServers.length === 0) && connector.getIceServers) {\n    try {\n      const turnConfig = await connector.getIceServers();\n      if (turnConfig && turnConfig.length > 0) {\n        iceServers = turnConfig;\n        console.log(\"[RFWebRTC] Using TURN servers from connector\");\n      }\n    } catch (err) {\n      console.warn(\"[RFWebRTC] Failed to fetch TURN config, using defaults:\", err);\n    }\n  }\n\n  // Step 2: Prepare peer connection and create offer\n  const { pc, offer, remoteStreamPromise, dataChannel, uploadChannel } = await preparePeerConnection(\n    localStream,\n    file,\n    iceServers\n  );\n\n  // Update wrtcParams with resolved iceServers so server also uses them\n  // For file uploads, default to batch mode (realtimeProcessing: false)\n  const resolvedWrtcParams = {\n    ...wrtcParams,\n    iceServers: iceServers,\n    realtimeProcessing: wrtcParams.realtimeProcessing ?? !isFile\n  };\n\n  // Step 3: Call connector.connectWrtc to exchange SDP and get answer\n  const answer = await connector.connectWrtc(\n    { sdp: offer.sdp!, type: offer.type! },\n    resolvedWrtcParams\n  );\n\n  // API returns sdp and type at root level\n  const sdpAnswer = { sdp: answer.sdp, type: answer.type } as RTCSessionDescriptionInit;\n\n  if (!sdpAnswer?.sdp || !sdpAnswer?.type) {\n    console.error(\"[RFWebRTC] Invalid answer from server:\", answer);\n    throw new Error(\"connector.connectWrtc must return answer with sdp and type\");\n  }\n\n  const pipelineId = answer?.context?.pipeline_id || null;\n\n  // Step 4: Set remote description\n  await pc.setRemoteDescription(sdpAnswer);\n\n  // Step 5: Wait for connection to establish\n  await new Promise<void>((resolve, reject) => {\n    const checkState = () => {\n      if (pc.connectionState === \"connected\") {\n        pc.removeEventListener(\"connectionstatechange\", checkState);\n        resolve();\n      } else if (pc.connectionState === \"failed\") {\n        pc.removeEventListener(\"connectionstatechange\", checkState);\n        reject(new Error(\"WebRTC connection failed\"));\n      }\n    };\n\n    pc.addEventListener(\"connectionstatechange\", checkState);\n    checkState(); // Check immediately in case already connected\n\n    // Timeout after 30 seconds\n    setTimeout(() => {\n      pc.removeEventListener(\"connectionstatechange\", checkState);\n      reject(new Error(\"WebRTC connection timeout after 30s\"));\n    }, 30000);\n  });\n\n  // Step 6: Optimize quality for MediaStream (disable downsampling by default)\n  if (localStream) {\n    const shouldDisableDownscaling = options.disableInputStreamDownscaling !== false;\n    if (shouldDisableDownscaling) {\n      await disableInputStreamDownscaling(pc);\n    }\n  }\n\n  // Get apiKey from connector if available (for cleanup)\n  const apiKey = connector._apiKey || null;\n\n  // Step 7: Create connection object\n  const connection = new RFWebRTCConnection(\n    pc,\n    remoteStreamPromise,\n    pipelineId,\n    apiKey,\n    dataChannel,\n    {\n      localStream,\n      uploadChannel,\n      onData,\n      onComplete\n    }\n  );\n\n  // Step 8: Start file upload if applicable (runs in background)\n  if (file && uploadChannel) {\n    connection.startUpload(file, onFileUploadProgress).catch(err => {\n      console.error(\"[RFWebRTC] Upload error:\", err);\n    });\n  }\n\n  return connection;\n}\n\n/**\n * Main function to establish WebRTC streaming connection\n *\n * Creates a WebRTC connection to Roboflow for real-time inference on video streams.\n *\n * @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n *\n * @example\n * ```typescript\n * import { useStream } from 'inferencejs/webrtc';\n * import { connectors } from 'inferencejs/api';\n * import { useCamera } from 'inferencejs/streams';\n *\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const stream = await useCamera({ video: { facingMode: { ideal: \"environment\" } } });\n * const conn = await useStream({\n *   source: stream,\n *   connector,\n *   wrtcParams: {\n *     workflowSpec: {\n *       // Your workflow specification\n *     },\n *     imageInputName: \"image\",\n *     streamOutputNames: [\"output\"],\n *     dataOutputNames: [\"predictions\"]\n *   },\n *   onData: (data) => {\n *     console.log(\"Inference results:\", data);\n *   }\n * });\n *\n * const remoteStream = await conn.remoteStream();\n * videoElement.srcObject = remoteStream;\n * ```\n */\nexport async function useStream({\n  source,\n  connector,\n  wrtcParams,\n  onData,\n  options = {}\n}: UseStreamParams): Promise<RFWebRTCConnection> {\n  if (source instanceof File) {\n    throw new Error(\"useStream requires a MediaStream. Use useVideoFile for File uploads.\");\n  }\n\n  return baseUseStream({\n    source,\n    connector,\n    wrtcParams,\n    onData,\n    options\n  });\n}\n\n/**\n * Parameters for useVideoFile function\n */\nexport interface UseVideoFileParams {\n  /** The video file to upload */\n  file: File;\n  /** Connector for WebRTC signaling */\n  connector: Connector;\n  /** WebRTC parameters for the workflow */\n  wrtcParams: WebRTCParams;\n  /** Callback for inference results */\n  onData?: (data: WebRTCOutputData) => void;\n  /** Callback for upload progress */\n  onUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n  /** Callback when processing completes (datachannel closes) */\n  onComplete?: () => void;\n}\n\n/**\n * Upload a video file for batch inference processing\n *\n * Creates a WebRTC connection to Roboflow for uploading a video file\n * and receiving inference results. The file is uploaded via datachannel\n * with intelligent backpressure handling.\n *\n * @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n *\n * @example\n * ```typescript\n * import { connectors, webrtc } from '@roboflow/inference-sdk';\n *\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const connection = await webrtc.useVideoFile({\n *   file: videoFile,\n *   connector,\n *   wrtcParams: {\n *     workflowSpec: { ... },\n *     imageInputName: \"image\",\n *     dataOutputNames: [\"predictions\"]\n *   },\n *   onData: (data) => {\n *     console.log(\"Inference results:\", data);\n *     if (data.processing_complete) {\n *       console.log(\"Processing complete!\");\n *     }\n *   },\n *   onUploadProgress: (uploaded, total) => {\n *     console.log(`Upload: ${(uploaded / total * 100).toFixed(1)}%`);\n *   }\n * });\n *\n * // When done\n * await connection.cleanup();\n * ```\n */\nexport async function useVideoFile({\n  file,\n  connector,\n  wrtcParams,\n  onData,\n  onUploadProgress,\n  onComplete\n}: UseVideoFileParams): Promise<RFWebRTCConnection> {\n  return baseUseStream({\n    source: file,\n    connector,\n    wrtcParams: {\n      ...wrtcParams,\n      realtimeProcessing: wrtcParams.realtimeProcessing ?? true\n    },\n    onData,\n    onComplete,\n    onFileUploadProgress: onUploadProgress\n  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Supports credentials in URL format: rtsp://user:pass@host/stream\n   * @example \"rtsp://camera.local/stream\"\n   */\n  rtspUrl?: string;\n}\n\n/**\n * ICE server configuration for WebRTC connections\n *\n * Use this to configure custom STUN/TURN servers for users behind\n * symmetric NAT or restrictive firewalls.\n */\nexport interface RTCIceServerConfig {\n  urls: string[];\n  username?: string;\n  credential?: string;\n}\n\nexport interface WebRTCOffer {\n  sdp: string;\n  type: string;\n}\n\nexport type WorkflowSpec = Record<string, any>;\n\nexport interface WebRTCWorkerResponse {\n  status?: string;\n  sdp: string;\n  type: string;\n  context?: {\n    request_id: string | null;\n    pipeline_id: string | null;\n  };\n}\n\nexport interface WebRTCParams {\n  workflowSpec?: WorkflowSpec;\n  workspaceName?: string;\n  workflowId?: string;\n  imageInputName?: string;\n  streamOutputNames?: string[];\n  dataOutputNames?: string[];\n  threadPoolWorkers?: number;\n  /**\n   * Workflow parameters to pass to the workflow execution\n   */\n  workflowsParameters?: Record<string, any>;\n  /**\n   * ICE servers for WebRTC connections (used for both client and server)\n   *\n   * Use this to specify custom STUN/TURN servers for users behind\n   * symmetric NAT or restrictive firewalls. The same configuration is\n   * used for both the client-side RTCPeerConnection and sent to the\n   * server via webrtc_config.\n   *\n   * @example\n   * ```typescript\n   * iceServers: [\n   *   { urls: [\"stun:stun.l.google.com:19302\"] },\n   *   { urls: [\"turn:turn.example.com:3478\"], username: \"user\", credential: \"pass\" }\n   * ]\n   * ```\n   */\n  iceServers?: RTCIceServerConfig[];\n  /**\n   * Processing timeout in seconds (serverless only)\n   * @default 600\n   */\n  processingTimeout?: number;\n  /**\n   * Requested compute plan (serverless only)\n   * @example \"webrtc-gpu-small\"\n   */\n  requestedPlan?: string;\n  /**\n   * Requested region for processing (serverless only)\n   * @example \"us\"\n   */\n  requestedRegion?: string;\n  /**\n   * Set to false for file upload mode (batch processing).\n   * When false, server processes all frames sequentially instead of dropping frames.\n   * @default true\n   */\n  realtimeProcessing?: boolean;\n  /**\n   * RTSP URL for server-side video capture.\n   * When provided, the server captures video from this RTSP stream instead of receiving\n   * video from the client. Supports credentials in URL format: rtsp://user:pass@host/stream\n   * @example \"rtsp://camera.local/stream\"\n   */\n  rtspUrl?: string;\n}\n\nexport interface Connector {\n  connectWrtc(offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse>;\n  /**\n   * Fetch ICE servers (TURN configuration) for WebRTC connections\n   * This should be called BEFORE creating the RTCPeerConnection to ensure\n   * proper NAT traversal configuration.\n   *\n   * @returns Promise resolving to ICE server configuration, or null/undefined if not available\n   */\n  getIceServers?(): Promise<RTCIceServerConfig[] | null>;\n  _apiKey?: string;\n  _serverUrl?: string;\n}\n\nexport class InferenceHTTPClient {\n  private apiKey: string;\n  private serverUrl: string;\n\n  /**\n   * @private\n   * Use InferenceHTTPClient.init() instead\n   */\n  private constructor(apiKey: string, serverUrl: string = \"https://serverless.roboflow.com\") {\n    this.apiKey = apiKey;\n    this.serverUrl = serverUrl;\n  }\n\n  static init({ apiKey, serverUrl }: { apiKey: string; serverUrl?: string }): InferenceHTTPClient {\n    if (!apiKey) {\n      throw new Error(\"apiKey is required\");\n    }\n    return new InferenceHTTPClient(apiKey, serverUrl);\n  }\n\n  /**\n   * Initialize a WebRTC worker pipeline\n   *\n   * @param params - Pipeline parameters\n   * @param params.offer - WebRTC offer { sdp, type }\n   * @param params.workflowSpec - Workflow specification\n   * @param params.config - Additional configuration\n   * @param params.config.imageInputName - Input image name (default: \"image\")\n   * @param params.config.streamOutputNames - Output stream names for video (default: [])\n   * @param params.config.dataOutputNames - Output data names (default: [\"string\"])\n   * @param params.config.threadPoolWorkers - Thread pool workers (default: 4)\n   * @returns Promise resolving to answer with SDP and pipeline ID\n   *\n   * @example\n   * ```typescript\n   * const answer = await client.initializeWebrtcWorker({\n   *   offer: { sdp, type },\n   *   workflowSpec: { ... },\n   *   config: {\n   *     imageInputName: \"image\",\n   *     streamOutputNames: [\"output_image\"]\n   *   }\n   * });\n   * ```\n   */\n  async initializeWebrtcWorker({\n    offer,\n    workflowSpec,\n    workspaceName,\n    workflowId,\n    config = {}\n  }: {\n    offer: WebRTCOffer;\n    workflowSpec?: WorkflowSpec;\n    workspaceName?: string;\n    workflowId?: string;\n    config?: WebRTCWorkerConfig;\n  }): Promise<WebRTCWorkerResponse> {\n    if (!offer || !offer.sdp || !offer.type) {\n      throw new Error(\"offer with sdp and type is required\");\n    }\n\n    // Validate that either workflowSpec OR (workspaceName + workflowId) is provided\n    const hasWorkflowSpec = !!workflowSpec;\n    const hasWorkspaceIdentifier = !!(workspaceName && workflowId);\n\n    if (!hasWorkflowSpec && !hasWorkspaceIdentifier) {\n      throw new Error(\"Either workflowSpec OR (workspaceName + workflowId) is required\");\n    }\n    if (hasWorkflowSpec && hasWorkspaceIdentifier) {\n      throw new Error(\"Provide either workflowSpec OR (workspaceName + workflowId), not both\");\n    }\n\n    const {\n      imageInputName = \"image\",\n      streamOutputNames = [],\n      dataOutputNames = [],\n      threadPoolWorkers = 4,\n      workflowsParameters = {},\n      iceServers,\n      processingTimeout,\n      requestedPlan,\n      requestedRegion,\n      realtimeProcessing = true,\n      rtspUrl\n    } = config as any;\n\n    // Build workflow_configuration based on what's provided\n    const workflowConfiguration: any = {\n      type: \"WorkflowConfiguration\",\n      image_input_name: imageInputName,\n      workflows_parameters: workflowsParameters,\n      workflows_thread_pool_workers: threadPoolWorkers,\n      cancel_thread_pool_tasks_on_exit: true,\n      video_metadata_input_name: \"video_metadata\"\n    };\n\n    if (hasWorkflowSpec) {\n      workflowConfiguration.workflow_specification = workflowSpec;\n    } else {\n      workflowConfiguration.workspace_name = workspaceName;\n      workflowConfiguration.workflow_id = workflowId;\n    }\n\n    const payload: Record<string, any> = {\n      workflow_configuration: workflowConfiguration,\n      api_key: this.apiKey,\n      webrtc_realtime_processing: realtimeProcessing,\n      webrtc_offer: {\n        sdp: offer.sdp,\n        type: offer.type\n      },\n      webrtc_config: iceServers ? { iceServers } : null,\n      stream_output: streamOutputNames,\n      data_output: dataOutputNames\n    };\n\n    // Add serverless-specific fields if provided\n    if (processingTimeout !== undefined) {\n      payload.processing_timeout = processingTimeout;\n    }\n    if (requestedPlan !== undefined) {\n      payload.requested_plan = requestedPlan;\n    }\n    if (requestedRegion !== undefined) {\n      payload.requested_region = requestedRegion;\n    }\n    // Add RTSP URL for server-side video capture\n    if (rtspUrl) {\n      payload.rtsp_url = rtspUrl;\n    }\n    const response = await fetch(`${this.serverUrl}/initialise_webrtc_worker`, {\n      method: \"POST\",\n      headers: { \"Content-Type\": \"application/json\" },\n      body: JSON.stringify(payload)\n    });\n\n    if (!response.ok) {\n      const errorText = await response.text().catch(() => \"\");\n      throw new Error(`initialise_webrtc_worker failed (${response.status}): ${errorText}`);\n    }\n\n    const result = await response.json();\n\n    return result;\n  }\n\n  async terminatePipeline({ pipelineId }: { pipelineId: string }): Promise<void> {\n    if (!pipelineId) {\n      throw new Error(\"pipelineId is required\");\n    }\n\n    await fetch(\n      `${this.serverUrl}/inference_pipelines/${pipelineId}/terminate?api_key=${this.apiKey}`,\n      {\n        method: \"POST\",\n        headers: { \"Content-Type\": \"application/json\" }\n      }\n    );\n  }\n\n  /**\n   * Fetch TURN server configuration from Roboflow API\n   *\n   * This automatically fetches TURN server credentials for improved WebRTC\n   * connectivity through firewalls and NAT. Only applicable when using\n   * Roboflow serverless infrastructure.\n   *\n   * @returns Promise resolving to ICE server configuration, or null if not applicable\n   *\n   * @example\n   * ```typescript\n   * const client = InferenceHTTPClient.init({ apiKey: \"your-api-key\" });\n   * const iceServers = await client.fetchTurnConfig();\n   * // Returns: [{ urls: [\"turn:...\"], username: \"...\", credential: \"...\" }]\n   * ```\n   */\n  async fetchTurnConfig(): Promise<RTCIceServerConfig[] | null> {\n    // // Only fetch TURN config for Roboflow serverless URLs\n    if (!ROBOFLOW_SERVERLESS_URLS.includes(this.serverUrl)) {\n      return null;\n    }\n    try {\n      const response = await fetch(\n        `${RF_API_BASE_URL}/webrtc_turn_config?api_key=${this.apiKey}`,\n        {\n          method: \"GET\",\n          headers: { \"Content-Type\": \"application/json\" }\n        }\n      );\n\n      if (!response.ok) {\n        console.warn(`[RFWebRTC] Failed to fetch TURN config (${response.status}), using defaults`);\n        return null;\n      }\n\n      const turnConfig = await response.json();\n\n      // Handle 3 formats:\n      // 1. Single server object: { urls, username, credential }\n      // 2. Array of servers: [{ urls, username, credential }, ...]\n      // 3. Object with iceServers: { iceServers: [...] }\n      let iceServersRaw: any[];\n\n      if (Array.isArray(turnConfig)) {\n        // Format 2: array of servers\n        iceServersRaw = turnConfig;\n      } else if (turnConfig.iceServers && Array.isArray(turnConfig.iceServers)) {\n        // Format 3: object with iceServers array\n        iceServersRaw = turnConfig.iceServers;\n      } else if (turnConfig.urls) {\n        // Format 1: single server object - wrap in array\n        iceServersRaw = [turnConfig];\n      } else {\n        console.warn(\"[RFWebRTC] Invalid TURN config format, using defaults\");\n        return null;\n      }\n\n      // Normalize the ICE servers format\n      const iceServers: RTCIceServerConfig[] = iceServersRaw.map((server: any) => ({\n        urls: Array.isArray(server.urls) ? server.urls : [server.urls],\n        username: server.username,\n        credential: server.credential\n      }));\n\n      return iceServers;\n    } catch (err) {\n      console.warn(\"[RFWebRTC] Error fetching TURN config:\", err);\n      return null;\n    }\n  }\n}\n\n/**\n * Connectors for establishing WebRTC connections to Roboflow\n */\nexport const connectors = {\n  /**\n   * Create a connector that uses API key directly\n   *\n   * **WARNING**: If you use this in the frontend, it will expose your API key. \n   * Use only for demos/testing.\n   * For production, use withProxyUrl() with a backend proxy.\n   *\n   * @param apiKey - Roboflow API key\n   * @param options - Additional options\n   * @param options.serverUrl - Custom Roboflow server URL\n   * @returns Connector with connectWrtc method\n   *\n   * @example\n   * ```typescript\n   * const connector = connectors.withApiKey(\"your-api-key\");\n   * const answer = await connector.connectWrtc(offer, wrtcParams);\n   * ```\n   */\n  withApiKey(apiKey: string, options: { serverUrl?: string } = {}): Connector {\n    const { serverUrl } = options;\n\n    // Warn if running in browser context\n    if (typeof window !== 'undefined') {\n      console.warn(\n        '[Security Warning] Using API key directly in browser will expose it. ' +\n        'Use connectors.withProxyUrl() for production. ' +\n        'See: https://docs.roboflow.com/api-reference/authentication#securing-your-api-key'\n      );\n    }\n\n    const client = InferenceHTTPClient.init({ apiKey, serverUrl });\n\n    return {\n      connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n        console.debug(\"wrtcParams\", wrtcParams);\n        const answer = await client.initializeWebrtcWorker({\n          offer,\n          workflowSpec: wrtcParams.workflowSpec,\n          workspaceName: wrtcParams.workspaceName,\n          workflowId: wrtcParams.workflowId,\n          config: {\n            imageInputName: wrtcParams.imageInputName,\n            streamOutputNames: wrtcParams.streamOutputNames,\n            dataOutputNames: wrtcParams.dataOutputNames,\n            threadPoolWorkers: wrtcParams.threadPoolWorkers,\n            workflowsParameters: wrtcParams.workflowsParameters,\n            iceServers: wrtcParams.iceServers,\n            processingTimeout: wrtcParams.processingTimeout,\n            requestedPlan: wrtcParams.requestedPlan,\n            requestedRegion: wrtcParams.requestedRegion,\n            realtimeProcessing: wrtcParams.realtimeProcessing,\n            rtspUrl: wrtcParams.rtspUrl\n          }\n        });\n\n        return answer;\n      },\n\n      /**\n       * Fetch TURN server configuration for improved WebRTC connectivity\n       */\n      getIceServers: async (): Promise<RTCIceServerConfig[] | null> => {\n        return await client.fetchTurnConfig();\n      },\n\n      // Store apiKey for cleanup\n      _apiKey: apiKey,\n      _serverUrl: serverUrl\n    };\n  },\n\n  /**\n   * Create a connector that uses a backend proxy (recommended for production)\n   *\n   * Your backend receives the offer and wrtcParams, adds the secret API key,\n   * and forwards to Roboflow. This keeps your API key secure.\n   *\n   * For improved WebRTC connectivity through firewalls, implement a separate\n   * endpoint for TURN server configuration that calls `fetchTurnConfig()`.\n   *\n   * @param proxyUrl - Backend proxy endpoint URL for WebRTC initialization\n   * @param options - Additional options\n   * @param options.turnConfigUrl - Optional URL for fetching TURN server configuration\n   * @returns Connector with connectWrtc and optional getIceServers methods\n   *\n   * @example\n   * ```typescript\n   * // Frontend: Create connector with TURN config endpoint\n   * const connector = connectors.withProxyUrl('/api/init-webrtc', {\n   *   turnConfigUrl: '/api/turn-config'\n   * });\n   * ```\n   *\n   * @example\n   * Backend implementation (Express) with TURN server support:\n   * ```typescript\n   * // Endpoint for TURN configuration (called first by SDK)\n   * app.get('/api/turn-config', async (req, res) => {\n   *   const client = InferenceHTTPClient.init({\n   *     apiKey: process.env.ROBOFLOW_API_KEY\n   *   });\n   *   const iceServers = await client.fetchTurnConfig();\n   *   res.json({ iceServers });\n   * });\n   *\n   * // Endpoint for WebRTC initialization\n   * app.post('/api/init-webrtc', async (req, res) => {\n   *   const { offer, wrtcParams } = req.body;\n   *   const client = InferenceHTTPClient.init({\n   *     apiKey: process.env.ROBOFLOW_API_KEY\n   *   });\n   *\n   *   const answer = await client.initializeWebrtcWorker({\n   *     offer,\n   *     workflowSpec: wrtcParams.workflowSpec,\n   *     workspaceName: wrtcParams.workspaceName,\n   *     workflowId: wrtcParams.workflowId,\n   *     config: {\n   *       imageInputName: wrtcParams.imageInputName,\n   *       streamOutputNames: wrtcParams.streamOutputNames,\n   *       dataOutputNames: wrtcParams.dataOutputNames,\n   *       threadPoolWorkers: wrtcParams.threadPoolWorkers,\n   *       workflowsParameters: wrtcParams.workflowsParameters,\n   *       iceServers: wrtcParams.iceServers,\n   *       processingTimeout: wrtcParams.processingTimeout,\n   *       requestedPlan: wrtcParams.requestedPlan,\n   *       requestedRegion: wrtcParams.requestedRegion\n   *     }\n   *   });\n   *\n   *   res.json(answer);\n   * });\n   * ```\n   */\n  withProxyUrl(proxyUrl: string, options: { turnConfigUrl?: string } = {}): Connector {\n    const { turnConfigUrl } = options;\n\n    return {\n      connectWrtc: async (offer: WebRTCOffer, wrtcParams: WebRTCParams): Promise<WebRTCWorkerResponse> => {\n        const response = await fetch(proxyUrl, {\n          method: \"POST\",\n          headers: { \"Content-Type\": \"application/json\" },\n          body: JSON.stringify({\n            offer,\n            wrtcParams\n          })\n        });\n\n        if (!response.ok) {\n          const errorText = await response.text().catch(() => \"\");\n          throw new Error(`Proxy request failed (${response.status}): ${errorText}`);\n        }\n\n        return await response.json();\n      },\n\n      /**\n       * Fetch TURN server configuration from the proxy backend\n       * Only available if turnConfigUrl was provided\n       */\n      getIceServers: turnConfigUrl\n        ? async (): Promise<RTCIceServerConfig[] | null> => {\n            try {\n              const response = await fetch(turnConfigUrl, {\n                method: \"GET\",\n                headers: { \"Content-Type\": \"application/json\" }\n              });\n\n              if (!response.ok) {\n                console.warn(`[RFWebRTC] Failed to fetch TURN config from proxy (${response.status})`);\n                return null;\n              }\n\n              const data = await response.json();\n              return data.iceServers || null;\n            } catch (err) {\n              console.warn(\"[RFWebRTC] Error fetching TURN config from proxy:\", err);\n              return null;\n            }\n          }\n        : undefined\n    };\n  }\n};\n","/**\n * Get a camera stream with the given constraints.\n *\n * @param constraints - MediaStreamConstraints for getUserMedia\n * @returns Promise that resolves to MediaStream\n *\n * @example\n * ```typescript\n * const stream = await useCamera({\n *   video: {\n *     facingMode: { ideal: \"user\" },\n *     width: { ideal: 1280 },\n *     height: { ideal: 720 },\n *     frameRate: { ideal: 30 }\n *   },\n *   audio: false\n * });\n * ```\n */\nexport async function useCamera(constraints: MediaStreamConstraints = { video: true }): Promise<MediaStream> {\n  try {\n    console.log(\"[RFStreams] requesting with\", constraints);\n    const stream = await navigator.mediaDevices.getUserMedia(constraints);\n    console.log(\"[RFStreams] got stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n    return stream;\n  } catch (err) {\n    console.warn(\"[RFStreams] failed, falling back\", err);\n    const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: false });\n    console.log(\"[RFStreams] fallback stream\", stream.getVideoTracks().map(t => ({ id: t.id, label: t.label })));\n    return stream;\n  }\n}\n  \nexport function stopStream(stream: MediaStream | null | undefined): void {\n  if (stream) {\n    stream.getTracks().forEach(track => track.stop());\n    console.log(\"[RFStreams] Stream stopped\");\n  }\n}\n","/**\n * Video file upload via WebRTC datachannel\n *\n * This module provides the FileUploader class for chunked file uploads\n * through WebRTC datachannels with backpressure handling.\n */\n\n/**\n * Configuration constants for file upload (matching Python SDK)\n */\nconst CHUNK_SIZE = 49152;       // 49KB - safe for WebRTC\nconst BUFFER_LIMIT = 262144;    // 256KB - backpressure threshold\nconst POLL_INTERVAL = 10;       // 10ms buffer check interval\n\n/**\n * Helper to sleep for a given number of milliseconds\n */\nfunction sleep(ms: number): Promise<void> {\n  return new Promise(resolve => setTimeout(resolve, ms));\n}\n\n/**\n * FileUploader handles chunked file upload with backpressure\n *\n * Uploads files through a WebRTC datachannel in 49KB chunks with\n * intelligent backpressure handling to prevent overwhelming the network.\n */\nexport class FileUploader {\n  private file: File;\n  private channel: RTCDataChannel;\n  private totalChunks: number;\n  private cancelled: boolean = false;\n\n  constructor(file: File, channel: RTCDataChannel) {\n    this.file = file;\n    this.channel = channel;\n    this.totalChunks = Math.ceil(file.size / CHUNK_SIZE);\n  }\n\n  /**\n   * Cancel the upload\n   */\n  cancel(): void {\n    this.cancelled = true;\n  }\n\n  /**\n   * Upload the file in chunks with backpressure handling\n   *\n   * @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n   */\n  async upload(onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n    const totalBytes = this.file.size;\n\n    for (let chunkIndex = 0; chunkIndex < this.totalChunks; chunkIndex++) {\n      // Check for cancellation\n      if (this.cancelled) {\n        throw new Error(\"Upload cancelled\");\n      }\n\n      // Check channel state\n      if (this.channel.readyState !== \"open\") {\n        throw new Error(\"Video upload interrupted\");\n      }\n\n      // Read chunk from file\n      const start = chunkIndex * CHUNK_SIZE;\n      const end = Math.min(start + CHUNK_SIZE, totalBytes);\n      const chunkBlob = this.file.slice(start, end);\n      const chunkData = new Uint8Array(await chunkBlob.arrayBuffer());\n\n      // Create message with 8-byte header (chunkIndex + totalChunks as uint32 LE)\n      const message = new ArrayBuffer(8 + chunkData.length);\n      const view = new DataView(message);\n      view.setUint32(0, chunkIndex, true);      // little-endian\n      view.setUint32(4, this.totalChunks, true); // little-endian\n      new Uint8Array(message, 8).set(chunkData);\n\n      // Backpressure: wait for buffer to drain\n      while (this.channel.bufferedAmount > BUFFER_LIMIT) {\n        if (this.channel.readyState !== \"open\") {\n          throw new Error(\"Video upload interrupted\");\n        }\n        await sleep(POLL_INTERVAL);\n      }\n\n      // Send chunk\n      this.channel.send(message);\n\n      // Report progress\n      if (onProgress) {\n        onProgress(end, totalBytes);\n      }\n    }\n  }\n}\n","\nimport { InferenceHTTPClient, Connector, WebRTCParams, RTCIceServerConfig } from \"./inference-api\";\nimport { stopStream } from \"./streams\";\nimport { WebRTCOutputData, WebRTCHooks } from \"./webrtc-types\";\nimport { FileUploader } from \"./video-upload\";\n\n// Re-export shared types\nexport type { WebRTCVideoMetadata, WebRTCOutputData, WebRTCHooks } from \"./webrtc-types\";\n\n// Re-export FileUploader from video-upload\nexport { FileUploader } from \"./video-upload\";\n\n/**\n * Binary protocol header size (frame_id + chunk_index + total_chunks)\n * Each field is 4 bytes uint32 little-endian\n */\nconst HEADER_SIZE = 12;\n\n/**\n * Reassembles chunked binary messages from the datachannel\n */\nexport class ChunkReassembler {\n  private pendingFrames: Map<number, {\n    chunks: Map<number, Uint8Array>;\n    totalChunks: number;\n  }> = new Map();\n\n  /**\n   * Process an incoming chunk and return the complete message if all chunks received\n   */\n  processChunk(frameId: number, chunkIndex: number, totalChunks: number, payload: Uint8Array): Uint8Array | null {\n    // Single chunk message - return immediately\n    if (totalChunks === 1) {\n      return payload;\n    }\n\n    // Multi-chunk message - accumulate\n    if (!this.pendingFrames.has(frameId)) {\n      this.pendingFrames.set(frameId, {\n        chunks: new Map(),\n        totalChunks\n      });\n    }\n\n    const frame = this.pendingFrames.get(frameId)!;\n    frame.chunks.set(chunkIndex, payload);\n\n    // Check if all chunks received\n    if (frame.chunks.size === totalChunks) {\n      // Reassemble in order\n      const totalLength = Array.from(frame.chunks.values()).reduce((sum, chunk) => sum + chunk.length, 0);\n      const result = new Uint8Array(totalLength);\n      let offset = 0;\n\n      for (let i = 0; i < totalChunks; i++) {\n        const chunk = frame.chunks.get(i)!;\n        result.set(chunk, offset);\n        offset += chunk.length;\n      }\n\n      this.pendingFrames.delete(frameId);\n      return result;\n    }\n\n    return null;\n  }\n\n  /**\n   * Clear all pending frames (for cleanup)\n   */\n  clear(): void {\n    this.pendingFrames.clear();\n  }\n}\n\n/**\n * Parse the binary header from a datachannel message\n */\nexport function parseBinaryHeader(buffer: ArrayBuffer): { frameId: number; chunkIndex: number; totalChunks: number; payload: Uint8Array } {\n  const view = new DataView(buffer);\n  const frameId = view.getUint32(0, true);      // little-endian\n  const chunkIndex = view.getUint32(4, true);   // little-endian\n  const totalChunks = view.getUint32(8, true);  // little-endian\n  const payload = new Uint8Array(buffer, HEADER_SIZE);\n\n  return { frameId, chunkIndex, totalChunks, payload };\n}\n\nexport interface UseStreamOptions {\n  disableInputStreamDownscaling?: boolean;\n}\n\nexport interface UseStreamParams {\n  source: MediaStream;\n  connector: Connector;\n  wrtcParams: WebRTCParams;\n  onData?: (data: WebRTCOutputData) => void;\n  options?: UseStreamOptions;\n  /** Lifecycle hooks for customizing WebRTC behavior */\n  hooks?: WebRTCHooks;\n}\n\nasync function waitForIceGathering(pc: RTCPeerConnection, timeoutMs = 6000): Promise<void> {\n  if (pc.iceGatheringState === \"complete\") return;\n\n  let hasSrflx = false;\n\n  // Track if we get a good candidate (srflx = public IP via STUN)\n  const candidateHandler = (event: RTCPeerConnectionIceEvent) => {\n    if (event.candidate && event.candidate.type === \"srflx\") {\n      hasSrflx = true;\n    }\n  };\n  pc.addEventListener(\"icecandidate\", candidateHandler);\n\n  try {\n    await Promise.race([\n      new Promise<void>(resolve => {\n        const check = () => {\n          if (pc.iceGatheringState === \"complete\") {\n            pc.removeEventListener(\"icegatheringstatechange\", check);\n            resolve();\n          }\n        };\n        pc.addEventListener(\"icegatheringstatechange\", check);\n      }),\n      new Promise<void>((resolve, reject) => {\n        setTimeout(() => {\n          if (!hasSrflx) {\n            console.error(\"[ICE] timeout with NO srflx candidate! Connection may fail.\");\n            reject(new Error(\"ICE gathering timeout without srflx candidate\"));\n          } else {\n            resolve();\n          }\n        }, timeoutMs);\n      })\n    ]);\n  } finally {\n    pc.removeEventListener(\"icecandidate\", candidateHandler);\n  }\n}\n\nfunction setupRemoteStreamListener(pc: RTCPeerConnection): Promise<MediaStream> {\n  return new Promise((resolve) => {\n    pc.addEventListener(\"track\", (event: RTCTrackEvent) => {\n      if (event.streams && event.streams[0]) {\n        resolve(event.streams[0]);\n      }\n    });\n  });\n}\n\nconst DEFAULT_ICE_SERVERS: RTCIceServerConfig[] = [\n  { urls: [\"stun:stun.l.google.com:19302\"] }\n];\n\nasync function preparePeerConnection(\n  localStream?: MediaStream,\n  file?: File,\n  customIceServers?: RTCIceServerConfig[],\n  rtspUrl?: string,\n  hooks?: WebRTCHooks\n): Promise<{\n  pc: RTCPeerConnection;\n  offer: RTCSessionDescriptionInit;\n  remoteStreamPromise: Promise<MediaStream>;\n  dataChannel: RTCDataChannel;\n  uploadChannel?: RTCDataChannel;\n}> {\n  // Validate: exactly one source type must be provided\n  const hasLocalStream = !!localStream;\n  const hasFile = !!file;\n  const hasRtspUrl = !!rtspUrl;\n  const sourceCount = [hasLocalStream, hasFile, hasRtspUrl].filter(Boolean).length;\n\n  if (sourceCount !== 1) {\n    throw new Error(\"Exactly one of localStream, file, or rtspUrl must be provided\");\n  }\n\n  const iceServers = customIceServers ?? DEFAULT_ICE_SERVERS;\n\n  const pc = new RTCPeerConnection({\n    iceServers: iceServers as RTCIceServer[]\n  });\n\n  // Call onPeerConnectionCreated hook\n  if (hooks?.onPeerConnectionCreated) {\n    await hooks.onPeerConnectionCreated(pc);\n  }\n\n  // Add transceiver for receiving remote video (BEFORE adding tracks - order matters!)\n  try {\n    pc.addTransceiver(\"video\", { direction: \"recvonly\" });\n  } catch (err) {\n    console.warn(\"[RFWebRTC] Could not add transceiver:\", err);\n  }\n\n  if (localStream) {\n    // Add local tracks\n    for (const track of localStream.getVideoTracks()) {\n      const sender = pc.addTrack(track, localStream);\n\n      // Call onTrackAdded hook\n      if (hooks?.onTrackAdded) {\n        await hooks.onTrackAdded(track, sender, pc);\n      }\n    }\n  }\n  // Note: For RTSP, no local tracks are added (receive-only mode)\n\n  // Setup remote stream listener\n  const remoteStreamPromise = setupRemoteStreamListener(pc);\n\n  // Create control datachannel (named \"inference\" to match Python SDK)\n  const dataChannel = pc.createDataChannel(\"inference\", {\n    ordered: true\n  });\n\n  // Create upload datachannel for file uploads (not needed for RTSP)\n  let uploadChannel: RTCDataChannel | undefined;\n  if (file) {\n    uploadChannel = pc.createDataChannel(\"video_upload\");\n  }\n\n  // Create offer\n  let offer = await pc.createOffer();\n\n  // Call onOfferCreated hook to allow SDP modification\n  if (hooks?.onOfferCreated) {\n    const modifiedOffer = await hooks.onOfferCreated(offer);\n    if (modifiedOffer) {\n      offer = modifiedOffer;\n    }\n  }\n\n  await pc.setLocalDescription(offer);\n\n  // Wait for ICE gathering\n  await waitForIceGathering(pc);\n\n  return {\n    pc,\n    offer: pc.localDescription!,\n    remoteStreamPromise,\n    dataChannel,\n    uploadChannel\n  };\n}\n\n/**\n * Disable input stream downscaling\n * @private\n */\nasync function disableInputStreamDownscaling(pc: RTCPeerConnection): Promise<void> {\n  const sender = pc.getSenders().find(s => s.track && s.track.kind === \"video\");\n  if (!sender) return;\n\n  const params = sender.getParameters();\n  params.encodings = params.encodings || [{}];\n  params.encodings[0].scaleResolutionDownBy = 1;\n\n  try {\n    await sender.setParameters(params);\n  } catch (err) {\n    console.warn(\"[RFWebRTC] Failed to set encoding parameters:\", err);\n  }\n}\n\n/**\n * Helper to wait for datachannel to open\n */\nfunction waitForChannelOpen(channel: RTCDataChannel, timeoutMs = 30000): Promise<void> {\n  return new Promise((resolve, reject) => {\n    if (channel.readyState === \"open\") {\n      resolve();\n      return;\n    }\n\n    const openHandler = () => {\n      channel.removeEventListener(\"open\", openHandler);\n      channel.removeEventListener(\"error\", errorHandler);\n      clearTimeout(timeout);\n      resolve();\n    };\n\n    const errorHandler = () => {\n      channel.removeEventListener(\"open\", openHandler);\n      channel.removeEventListener(\"error\", errorHandler);\n      clearTimeout(timeout);\n      reject(new Error(\"Datachannel error\"));\n    };\n\n    const timeout = setTimeout(() => {\n      channel.removeEventListener(\"open\", openHandler);\n      channel.removeEventListener(\"error\", errorHandler);\n      reject(new Error(\"Datachannel open timeout\"));\n    }, timeoutMs);\n\n    channel.addEventListener(\"open\", openHandler);\n    channel.addEventListener(\"error\", errorHandler);\n  });\n}\n\n/**\n * WebRTC Connection object\n *\n * Represents an active WebRTC connection to Roboflow for streaming inference\n * or file-based batch processing.\n */\nexport class RFWebRTCConnection {\n  /**\n   * The underlying RTCPeerConnection.\n   * Exposed for advanced use cases like getting stats or accessing senders.\n   */\n  public readonly peerConnection: RTCPeerConnection;\n  private _localStream?: MediaStream;\n  private remoteStreamPromise: Promise<MediaStream>;\n  private pipelineId: string | null;\n  private apiKey: string | null;\n  /**\n   * The data channel used for receiving inference results.\n   * Exposed for advanced use cases.\n   */\n  public readonly dataChannel: RTCDataChannel;\n  private reassembler: ChunkReassembler;\n  /**\n   * The data channel used for uploading video files (only available in file upload mode).\n   * Exposed for advanced use cases.\n   */\n  public readonly uploadChannel?: RTCDataChannel;\n  private uploader?: FileUploader;\n  private onComplete?: () => void;\n\n  /** @private */\n  constructor(\n    pc: RTCPeerConnection,\n    remoteStreamPromise: Promise<MediaStream>,\n    pipelineId: string | null,\n    apiKey: string | null,\n    dataChannel: RTCDataChannel,\n    options?: {\n      localStream?: MediaStream;\n      uploadChannel?: RTCDataChannel;\n      onData?: (data: any) => void;\n      onComplete?: () => void;\n    }\n  ) {\n    this.peerConnection = pc;\n    this._localStream = options?.localStream;\n    this.remoteStreamPromise = remoteStreamPromise;\n    this.pipelineId = pipelineId;\n    this.apiKey = apiKey;\n    this.dataChannel = dataChannel;\n    this.reassembler = new ChunkReassembler();\n    this.uploadChannel = options?.uploadChannel;\n    this.onComplete = options?.onComplete;\n\n    // Set binary mode for datachannel\n    this.dataChannel.binaryType = \"arraybuffer\";\n\n    const onData = options?.onData;\n\n    // Setup data channel event listeners\n    if (onData) {\n      this.dataChannel.addEventListener(\"message\", (messageEvent: MessageEvent) => {\n        try {\n          // Handle binary protocol with chunking\n          if (messageEvent.data instanceof ArrayBuffer) {\n            const { frameId, chunkIndex, totalChunks, payload } = parseBinaryHeader(messageEvent.data);\n            const completePayload = this.reassembler.processChunk(frameId, chunkIndex, totalChunks, payload);\n\n            if (completePayload) {\n              // Decode UTF-8 JSON payload\n              const decoder = new TextDecoder(\"utf-8\");\n              const jsonString = decoder.decode(completePayload);\n              const data = JSON.parse(jsonString);\n              onData(data);\n            }\n          } else {\n            // Fallback for string messages (shouldn't happen with new protocol)\n            const data = JSON.parse(messageEvent.data);\n            onData(data);\n          }\n        } catch (err) {\n          console.error(\"[RFWebRTC] Failed to parse data channel message:\", err);\n        }\n      });\n\n      this.dataChannel.addEventListener(\"error\", (error) => {\n        console.error(\"[RFWebRTC] Data channel error:\", error);\n      });\n    }\n\n    // Handle channel close - call onComplete when processing finishes\n    this.dataChannel.addEventListener(\"close\", () => {\n      this.reassembler.clear();\n      if (this.onComplete) {\n        this.onComplete();\n      }\n    });\n  }\n\n  /**\n   * Get the remote stream (processed video from Roboflow)\n   *\n   * @returns Promise resolving to the remote MediaStream\n   *\n   * @example\n   * ```typescript\n   * const conn = await useStream({ ... });\n   * const remoteStream = await conn.remoteStream();\n   * videoElement.srcObject = remoteStream;\n   * ```\n   */\n  async remoteStream(): Promise<MediaStream> {\n    return await this.remoteStreamPromise;\n  }\n\n  /**\n   * Get the local stream (original camera)\n   *\n   * @returns The local MediaStream, or undefined if using file upload mode\n   *\n   * @example\n   * ```typescript\n   * const conn = await useStream({ ... });\n   * const localStream = conn.localStream();\n   * if (localStream) {\n   *   videoElement.srcObject = localStream;\n   * }\n   * ```\n   */\n  localStream(): MediaStream | undefined {\n    return this._localStream;\n  }\n\n  /**\n   * Cleanup and close connection\n   *\n   * Terminates the pipeline on Roboflow, closes the peer connection,\n   * and stops the local media stream (if applicable).\n   *\n   * @returns Promise that resolves when cleanup is complete\n   *\n   * @example\n   * ```typescript\n   * const conn = await useStream({ ... });\n   * // ... use connection ...\n   * await conn.cleanup(); // Clean up when done\n   * ```\n   */\n  async cleanup(): Promise<void> {\n    // Cancel any ongoing upload\n    if (this.uploader) {\n      this.uploader.cancel();\n    }\n\n    // Clear pending chunks\n    this.reassembler.clear();\n\n    // Terminate pipeline\n    if (this.pipelineId && this.apiKey) {\n      try {\n        const client = InferenceHTTPClient.init({ apiKey: this.apiKey });\n        await client.terminatePipeline({ pipelineId: this.pipelineId });\n      } catch (err) {\n        console.warn(\"[RFWebRTC] Failed to terminate pipeline:\", err);\n      }\n    }\n\n    // Close peer connection\n    if (this.peerConnection && this.peerConnection.connectionState !== \"closed\") {\n      this.peerConnection.close();\n    }\n\n    // Stop local stream if present\n    if (this._localStream) {\n      stopStream(this._localStream);\n    }\n  }\n\n  /**\n   * Start uploading a file through the connection\n   *\n   * @param file - The file to upload\n   * @param onProgress - Optional callback for progress updates (bytesUploaded, totalBytes)\n   * @returns Promise that resolves when upload is complete\n   * @throws Error if no upload channel is available\n   *\n   * @example\n   * ```typescript\n   * await connection.startUpload(videoFile, (uploaded, total) => {\n   *   console.log(`Upload progress: ${(uploaded / total * 100).toFixed(1)}%`);\n   * });\n   * ```\n   */\n  async startUpload(file: File, onProgress?: (bytesUploaded: number, totalBytes: number) => void): Promise<void> {\n    if (!this.uploadChannel) {\n      throw new Error(\"No upload channel available. This connection was not created for file uploads.\");\n    }\n\n    // Wait for upload channel to open\n    await waitForChannelOpen(this.uploadChannel);\n\n    this.uploader = new FileUploader(file, this.uploadChannel);\n    await this.uploader.upload(onProgress);\n  }\n\n  /**\n   * Cancel any ongoing file upload\n   */\n  cancelUpload(): void {\n    if (this.uploader) {\n      this.uploader.cancel();\n    }\n  }\n\n  /**\n   * Reconfigure pipeline outputs at runtime\n   *\n   * Dynamically change stream and data outputs without restarting the connection.\n   * Set a field to `null` to leave it unchanged, or to `null` value to enable all outputs,\n   * or to `[]` to disable/auto-detect.\n   *\n   * @param config - Output configuration\n   * @param config.streamOutput - Stream output names (null = unchanged, [] = auto-detect, [\"name\"] = specific output)\n   * @param config.dataOutput - Data output names (null = unchanged, [] = disable, [\"name\"] = specific outputs, null value = all outputs)\n   *\n   * @example\n   * ```typescript\n   * // Change to different stream output\n   * connection.reconfigureOutputs({\n   *   streamOutput: [\"annotated_image\"],\n   *   dataOutput: null  // unchanged\n   * });\n   *\n   * // Enable all data outputs\n   * connection.reconfigureOutputs({\n   *   streamOutput: null,  // unchanged\n   *   dataOutput: null     // null value = all outputs\n   * });\n   *\n   * // Disable all data outputs\n   * connection.reconfigureOutputs({\n   *   streamOutput: null,  // unchanged\n   *   dataOutput: []       // empty array = disable\n   * });\n   * ```\n   */\n  reconfigureOutputs(config: { streamOutput?: string[] | null; dataOutput?: string[] | null }): void {\n    const message: any = {};\n\n    if (config.streamOutput !== undefined) {\n      message.stream_output = config.streamOutput;\n    }\n\n    if (config.dataOutput !== undefined) {\n      message.data_output = config.dataOutput;\n    }\n\n    this.sendData(message);\n  }\n\n  /**\n   * Send data through the data channel\n   * @private\n   */\n  private sendData(data: any): void {\n    if (this.dataChannel.readyState !== \"open\") {\n      console.warn(\"[RFWebRTC] Data channel is not open. Current state:\", this.dataChannel.readyState);\n      return;\n    }\n\n    try {\n      const message = typeof data === \"string\" ? data : JSON.stringify(data);\n      this.dataChannel.send(message);\n    } catch (err) {\n      console.error(\"[RFWebRTC] Failed to send data:\", err);\n    }\n  }\n}\n\n/**\n * Internal base function for establishing WebRTC connection\n * Used by both useStream and useVideoFile\n * @private\n */\ninterface BaseUseStreamParams {\n  source?: MediaStream | File;\n  rtspUrl?: string;\n  connector: Connector;\n  wrtcParams: WebRTCParams;\n  onData?: (data: WebRTCOutputData) => void;\n  onComplete?: () => void;\n  onFileUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n  options?: UseStreamOptions;\n  hooks?: WebRTCHooks;\n}\n\nasync function baseUseStream({\n  source,\n  rtspUrl,\n  connector,\n  wrtcParams,\n  onData,\n  onComplete,\n  onFileUploadProgress,\n  options = {},\n  hooks\n}: BaseUseStreamParams): Promise<RFWebRTCConnection> {\n  // Validate connector\n  if (!connector || typeof connector.connectWrtc !== \"function\") {\n    throw new Error(\"connector must have a connectWrtc method\");\n  }\n\n  // Determine source type\n  const isRtsp = !!rtspUrl;\n  const isFile = !isRtsp && source instanceof File;\n  const localStream = !isRtsp && !isFile && source ? (source as MediaStream) : undefined;\n  const file = isFile ? (source as File) : undefined;\n\n  // Step 1: Determine ICE servers to use\n  // Priority: 1) User-provided in wrtcParams, 2) From connector.getIceServers(), 3) Defaults\n  let iceServers = wrtcParams.iceServers;\n  if ((!iceServers || iceServers.length === 0) && connector.getIceServers) {\n    try {\n      const turnConfig = await connector.getIceServers();\n      if (turnConfig && turnConfig.length > 0) {\n        iceServers = turnConfig;\n        console.log(\"[RFWebRTC] Using TURN servers from connector\");\n      }\n    } catch (err) {\n      console.warn(\"[RFWebRTC] Failed to fetch TURN config, using defaults:\", err);\n    }\n  }\n\n  // Step 2: Prepare peer connection and create offer\n  const { pc, offer, remoteStreamPromise, dataChannel, uploadChannel } = await preparePeerConnection(\n    localStream,\n    file,\n    iceServers,\n    rtspUrl,\n    hooks\n  );\n\n  // Update wrtcParams with resolved iceServers so server also uses them\n  // For file uploads, default to batch mode (realtimeProcessing: false)\n  // For RTSP, default to realtime processing\n  const resolvedWrtcParams = {\n    ...wrtcParams,\n    iceServers: iceServers,\n    realtimeProcessing: wrtcParams.realtimeProcessing ?? !isFile,\n    rtspUrl: rtspUrl\n  };\n\n  // Step 3: Call connector.connectWrtc to exchange SDP and get answer\n  const answer = await connector.connectWrtc(\n    { sdp: offer.sdp!, type: offer.type! },\n    resolvedWrtcParams\n  );\n\n  // API returns sdp and type at root level\n  const sdpAnswer = { sdp: answer.sdp, type: answer.type } as RTCSessionDescriptionInit;\n\n  if (!sdpAnswer?.sdp || !sdpAnswer?.type) {\n    console.error(\"[RFWebRTC] Invalid answer from server:\", answer);\n    throw new Error(\"connector.connectWrtc must return answer with sdp and type\");\n  }\n\n  const pipelineId = answer?.context?.pipeline_id || null;\n\n  // Step 4: Set remote description\n  await pc.setRemoteDescription(sdpAnswer);\n\n  // Step 5: Wait for connection to establish\n  await new Promise<void>((resolve, reject) => {\n    const checkState = () => {\n      if (pc.connectionState === \"connected\") {\n        pc.removeEventListener(\"connectionstatechange\", checkState);\n        resolve();\n      } else if (pc.connectionState === \"failed\") {\n        pc.removeEventListener(\"connectionstatechange\", checkState);\n        reject(new Error(\"WebRTC connection failed\"));\n      }\n    };\n\n    pc.addEventListener(\"connectionstatechange\", checkState);\n    checkState(); // Check immediately in case already connected\n\n    // Timeout after 30 seconds\n    setTimeout(() => {\n      pc.removeEventListener(\"connectionstatechange\", checkState);\n      reject(new Error(\"WebRTC connection timeout after 30s\"));\n    }, 30000);\n  });\n\n  // Step 6: Optimize quality for MediaStream (disable downsampling by default)\n  if (localStream) {\n    const shouldDisableDownscaling = options.disableInputStreamDownscaling !== false;\n    if (shouldDisableDownscaling) {\n      await disableInputStreamDownscaling(pc);\n    }\n  }\n\n  // Get apiKey from connector if available (for cleanup)\n  const apiKey = connector._apiKey || null;\n\n  // Step 7: Create connection object\n  const connection = new RFWebRTCConnection(\n    pc,\n    remoteStreamPromise,\n    pipelineId,\n    apiKey,\n    dataChannel,\n    {\n      localStream,\n      uploadChannel,\n      onData,\n      onComplete\n    }\n  );\n\n  // Step 8: Start file upload if applicable (runs in background)\n  if (file && uploadChannel) {\n    connection.startUpload(file, onFileUploadProgress).catch(err => {\n      console.error(\"[RFWebRTC] Upload error:\", err);\n    });\n  }\n\n  return connection;\n}\n\n/**\n * Main function to establish WebRTC streaming connection\n *\n * Creates a WebRTC connection to Roboflow for real-time inference on video streams.\n *\n * @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n *\n * @example\n * ```typescript\n * import { useStream } from 'inferencejs/webrtc';\n * import { connectors } from 'inferencejs/api';\n * import { useCamera } from 'inferencejs/streams';\n *\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const stream = await useCamera({ video: { facingMode: { ideal: \"environment\" } } });\n * const conn = await useStream({\n *   source: stream,\n *   connector,\n *   wrtcParams: {\n *     workflowSpec: {\n *       // Your workflow specification\n *     },\n *     imageInputName: \"image\",\n *     streamOutputNames: [\"output\"],\n *     dataOutputNames: [\"predictions\"]\n *   },\n *   onData: (data) => {\n *     console.log(\"Inference results:\", data);\n *   }\n * });\n *\n * const remoteStream = await conn.remoteStream();\n * videoElement.srcObject = remoteStream;\n * ```\n */\nexport async function useStream({\n  source,\n  connector,\n  wrtcParams,\n  onData,\n  options = {},\n  hooks\n}: UseStreamParams): Promise<RFWebRTCConnection> {\n  if (source instanceof File) {\n    throw new Error(\"useStream requires a MediaStream. Use useVideoFile for File uploads.\");\n  }\n\n  return baseUseStream({\n    source,\n    connector,\n    wrtcParams,\n    onData,\n    options,\n    hooks\n  });\n}\n\n/**\n * Parameters for useVideoFile function\n */\nexport interface UseVideoFileParams {\n  /** The video file to upload */\n  file: File;\n  /** Connector for WebRTC signaling */\n  connector: Connector;\n  /** WebRTC parameters for the workflow */\n  wrtcParams: WebRTCParams;\n  /** Callback for inference results */\n  onData?: (data: WebRTCOutputData) => void;\n  /** Callback for upload progress */\n  onUploadProgress?: (bytesUploaded: number, totalBytes: number) => void;\n  /** Callback when processing completes (datachannel closes) */\n  onComplete?: () => void;\n  /** Lifecycle hooks for customizing WebRTC behavior */\n  hooks?: WebRTCHooks;\n}\n\n/**\n * Upload a video file for batch inference processing\n *\n * Creates a WebRTC connection to Roboflow for uploading a video file\n * and receiving inference results. The file is uploaded via datachannel\n * with intelligent backpressure handling.\n *\n * @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n *\n * @example\n * ```typescript\n * import { connectors, webrtc } from '@roboflow/inference-sdk';\n *\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const connection = await webrtc.useVideoFile({\n *   file: videoFile,\n *   connector,\n *   wrtcParams: {\n *     workflowSpec: { ... },\n *     imageInputName: \"image\",\n *     dataOutputNames: [\"predictions\"]\n *   },\n *   onData: (data) => {\n *     console.log(\"Inference results:\", data);\n *     if (data.processing_complete) {\n *       console.log(\"Processing complete!\");\n *     }\n *   },\n *   onUploadProgress: (uploaded, total) => {\n *     console.log(`Upload: ${(uploaded / total * 100).toFixed(1)}%`);\n *   }\n * });\n *\n * // When done\n * await connection.cleanup();\n * ```\n */\nexport async function useVideoFile({\n  file,\n  connector,\n  wrtcParams,\n  onData,\n  onUploadProgress,\n  onComplete,\n  hooks\n}: UseVideoFileParams): Promise<RFWebRTCConnection> {\n  return baseUseStream({\n    source: file,\n    connector,\n    wrtcParams: {\n      ...wrtcParams,\n      realtimeProcessing: wrtcParams.realtimeProcessing ?? true\n    },\n    onData,\n    onComplete,\n    onFileUploadProgress: onUploadProgress,\n    hooks\n  });\n}\n\n/**\n * Parameters for useRtspStream function\n */\nexport interface UseRtspStreamParams {\n  /** RTSP URL for server-side video capture (e.g., \"rtsp://camera.local/stream\") */\n  rtspUrl: string;\n  /** Connector for WebRTC signaling */\n  connector: Connector;\n  /** WebRTC parameters for the workflow */\n  wrtcParams: WebRTCParams;\n  /** Callback for inference results */\n  onData?: (data: WebRTCOutputData) => void;\n  /** Lifecycle hooks for customizing WebRTC behavior */\n  hooks?: WebRTCHooks;\n}\n\n/**\n * Connect to an RTSP stream for inference processing\n *\n * Creates a WebRTC connection where the server captures video from an RTSP URL\n * and sends processed video back to the client. This is a receive-only mode -\n * no video is sent from the browser to the server.\n *\n * @param params - Connection parameters\n * @returns Promise resolving to RFWebRTCConnection\n *\n * @example\n * ```typescript\n * import { connectors, webrtc } from '@roboflow/inference-sdk';\n *\n * const connector = connectors.withApiKey(\"your-api-key\");\n * const connection = await webrtc.useRtspStream({\n *   rtspUrl: \"rtsp://camera.local/stream\",\n *   connector,\n *   wrtcParams: {\n *     workflowSpec: { ... },\n *     imageInputName: \"image\",\n *     dataOutputNames: [\"predictions\"]\n *   },\n *   onData: (data) => {\n *     console.log(\"Inference results:\", data);\n *   }\n * });\n *\n * // Get processed video stream from server\n * const remoteStream = await connection.remoteStream();\n * videoElement.srcObject = remoteStream;\n *\n * // When done\n * await connection.cleanup();\n * ```\n */\nexport async function useRtspStream({\n  rtspUrl,\n  connector,\n  wrtcParams,\n  onData,\n  hooks\n}: UseRtspStreamParams): Promise<RFWebRTCConnection> {\n  // Validate RTSP URL format\n  if (!rtspUrl.startsWith(\"rtsp://\") && !rtspUrl.startsWith(\"rtsps://\")) {\n    throw new Error(\"Invalid RTSP URL: must start with rtsp:// or rtsps://\");\n  }\n\n  return baseUseStream({\n    rtspUrl,\n    connector,\n    wrtcParams,\n    onData,\n    hooks\n  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package/dist/index.js

@@ -1,2 +1,2 @@
-(function(u,p){typeof exports=="object"&&typeof module<"u"?p(exports):typeof define=="function"&&define.amd?define(["exports"],p):(u=typeof globalThis<"u"?globalThis:u||self,p(u.RoboflowClient={}))})(this,function(u){"use strict";var te=Object.defineProperty;var re=(u,p,v)=>p in u?te(u,p,{enumerable:!0,configurable:!0,writable:!0,value:v}):u[p]=v;var d=(u,p,v)=>re(u,typeof p!="symbol"?p+"":p,v);var A;const p=typeof process<"u"&&((A=process.env)!=null&&A.RF_API_BASE_URL)?process.env.RF_API_BASE_URL:"https://api.roboflow.com",v=["https://serverless.roboflow.com"];class k{constructor(e,t="https://serverless.roboflow.com"){d(this,"apiKey");d(this,"serverUrl");this.apiKey=e,this.serverUrl=t}static init({apiKey:e,serverUrl:t}){if(!e)throw new Error("apiKey is required");return new k(e,t)}async initializeWebrtcWorker({offer:e,workflowSpec:t,workspaceName:a,workflowId:o,config:r={}}){if(!e||!e.sdp||!e.type)throw new Error("offer with sdp and type is required");const i=!!t,l=!!(a&&o);if(!i&&!l)throw new Error("Either workflowSpec OR (workspaceName + workflowId) is required");if(i&&l)throw new Error("Provide either workflowSpec OR (workspaceName + workflowId), not both");const{imageInputName:c="image",streamOutputNames:s=[],dataOutputNames:h=["string"],threadPoolWorkers:f=4,workflowsParameters:C={},iceServers:y,processingTimeout:T,requestedPlan:S,requestedRegion:R,realtimeProcessing:g=!0}=r,m={type:"WorkflowConfiguration",image_input_name:c,workflows_parameters:C,workflows_thread_pool_workers:f,cancel_thread_pool_tasks_on_exit:!0,video_metadata_input_name:"video_metadata"};i?m.workflow_specification=t:(m.workspace_name=a,m.workflow_id=o);const _={workflow_configuration:m,api_key:this.apiKey,webrtc_realtime_processing:g,webrtc_offer:{sdp:e.sdp,type:e.type},webrtc_config:y?{iceServers:y}:null,stream_output:s,data_output:h};T!==void 0&&(_.processing_timeout=T),S!==void 0&&(_.requested_plan=S),R!==void 0&&(_.requested_region=R);const b=await fetch(`${this.serverUrl}/initialise_webrtc_worker`,{method:"POST",headers:{"Content-Type":"application/json"},body:JSON.stringify(_)});if(!b.ok){const F=await b.text().catch(()=>"");throw new Error(`initialise_webrtc_worker failed (${b.status}): ${F}`)}return await b.json()}async terminatePipeline({pipelineId:e}){if(!e)throw new Error("pipelineId is required");await fetch(`${this.serverUrl}/inference_pipelines/${e}/terminate?api_key=${this.apiKey}`,{method:"POST",headers:{"Content-Type":"application/json"}})}async fetchTurnConfig(){if(!v.includes(this.serverUrl))return null;try{const e=await fetch(`${p}/webrtc_turn_config?api_key=${this.apiKey}`,{method:"GET",headers:{"Content-Type":"application/json"}});if(!e.ok)return console.warn(`[RFWebRTC] Failed to fetch TURN config (${e.status}), using defaults`),null;const t=await e.json();let a;if(Array.isArray(t))a=t;else if(t.iceServers&&Array.isArray(t.iceServers))a=t.iceServers;else if(t.urls)a=[t];else return console.warn("[RFWebRTC] Invalid TURN config format, using defaults"),null;return a.map(r=>({urls:Array.isArray(r.urls)?r.urls:[r.urls],username:r.username,credential:r.credential}))}catch(e){return console.warn("[RFWebRTC] Error fetching TURN config:",e),null}}}const q={withApiKey(n,e={}){const{serverUrl:t}=e;typeof window<"u"&&console.warn("[Security Warning] Using API key directly in browser will expose it. Use connectors.withProxyUrl() for production. See: https://docs.roboflow.com/api-reference/authentication#securing-your-api-key");const a=k.init({apiKey:n,serverUrl:t});return{connectWrtc:async(o,r)=>(console.debug("wrtcParams",r),await a.initializeWebrtcWorker({offer:o,workflowSpec:r.workflowSpec,workspaceName:r.workspaceName,workflowId:r.workflowId,config:{imageInputName:r.imageInputName,streamOutputNames:r.streamOutputNames,dataOutputNames:r.dataOutputNames,threadPoolWorkers:r.threadPoolWorkers,workflowsParameters:r.workflowsParameters,iceServers:r.iceServers,processingTimeout:r.processingTimeout,requestedPlan:r.requestedPlan,requestedRegion:r.requestedRegion,realtimeProcessing:r.realtimeProcessing}})),getIceServers:async()=>await a.fetchTurnConfig(),_apiKey:n,_serverUrl:t}},withProxyUrl(n,e={}){const{turnConfigUrl:t}=e;return{connectWrtc:async(a,o)=>{const r=await fetch(n,{method:"POST",headers:{"Content-Type":"application/json"},body:JSON.stringify({offer:a,wrtcParams:o})});if(!r.ok){const i=await r.text().catch(()=>"");throw new Error(`Proxy request failed (${r.status}): ${i}`)}return await r.json()},getIceServers:t?async()=>{try{const a=await fetch(t,{method:"GET",headers:{"Content-Type":"application/json"}});return a.ok?(await a.json()).iceServers||null:(console.warn(`[RFWebRTC] Failed to fetch TURN config from proxy (${a.status})`),null)}catch(a){return console.warn("[RFWebRTC] Error fetching TURN config from proxy:",a),null}}:void 0}}};async function j(n={video:!0}){try{console.log("[RFStreams] requesting with",n);const e=await navigator.mediaDevices.getUserMedia(n);return console.log("[RFStreams] got stream",e.getVideoTracks().map(t=>({id:t.id,label:t.label}))),e}catch(e){console.warn("[RFStreams] failed, falling back",e);const t=await navigator.mediaDevices.getUserMedia({video:!0,audio:!1});return console.log("[RFStreams] fallback stream",t.getVideoTracks().map(a=>({id:a.id,label:a.label}))),t}}function W(n){n&&(n.getTracks().forEach(e=>e.stop()),console.log("[RFStreams] Stream stopped"))}const K=Object.freeze(Object.defineProperty({__proto__:null,stopStream:W,useCamera:j},Symbol.toStringTag,{value:"Module"})),U=49152,B=262144,V=10;function $(n){return new Promise(e=>setTimeout(e,n))}class I{constructor(e,t){d(this,"file");d(this,"channel");d(this,"totalChunks");d(this,"cancelled",!1);this.file=e,this.channel=t,this.totalChunks=Math.ceil(e.size/U)}cancel(){this.cancelled=!0}async upload(e){const t=this.file.size;for(let a=0;a<this.totalChunks;a++){if(this.cancelled)throw new Error("Upload cancelled");if(this.channel.readyState!=="open")throw new Error("Video upload interrupted");const o=a*U,r=Math.min(o+U,t),i=this.file.slice(o,r),l=new Uint8Array(await i.arrayBuffer()),c=new ArrayBuffer(8+l.length),s=new DataView(c);for(s.setUint32(0,a,!0),s.setUint32(4,this.totalChunks,!0),new Uint8Array(c,8).set(l);this.channel.bufferedAmount>B;){if(this.channel.readyState!=="open")throw new Error("Video upload interrupted");await $(V)}this.channel.send(c),e&&e(r,t)}}}const M=12;class L{constructor(){d(this,"pendingFrames",new Map)}processChunk(e,t,a,o){if(a===1)return o;this.pendingFrames.has(e)||this.pendingFrames.set(e,{chunks:new Map,totalChunks:a});const r=this.pendingFrames.get(e);if(r.chunks.set(t,o),r.chunks.size===a){const i=Array.from(r.chunks.values()).reduce((s,h)=>s+h.length,0),l=new Uint8Array(i);let c=0;for(let s=0;s<a;s++){const h=r.chunks.get(s);l.set(h,c),c+=h.length}return this.pendingFrames.delete(e),l}return null}clear(){this.pendingFrames.clear()}}function O(n){const e=new DataView(n),t=e.getUint32(0,!0),a=e.getUint32(4,!0),o=e.getUint32(8,!0),r=new Uint8Array(n,M);return{frameId:t,chunkIndex:a,totalChunks:o,payload:r}}async function H(n,e=6e3){if(n.iceGatheringState==="complete")return;let t=!1;const a=o=>{o.candidate&&o.candidate.type==="srflx"&&(t=!0)};n.addEventListener("icecandidate",a);try{await Promise.race([new Promise(o=>{const r=()=>{n.iceGatheringState==="complete"&&(n.removeEventListener("icegatheringstatechange",r),o())};n.addEventListener("icegatheringstatechange",r)}),new Promise((o,r)=>{setTimeout(()=>{t?o():(console.error("[ICE] timeout with NO srflx candidate! Connection may fail."),r(new Error("ICE gathering timeout without srflx candidate")))},e)})])}finally{n.removeEventListener("icecandidate",a)}}function z(n){return new Promise(e=>{n.addEventListener("track",t=>{t.streams&&t.streams[0]&&e(t.streams[0])})})}const G=[{urls:["stun:stun.l.google.com:19302"]}];async function J(n,e,t){if(!n&&!e||n&&e)throw new Error("Either localStream or file must be provided, but not both");const a=t??G,o=new RTCPeerConnection({iceServers:a});try{o.addTransceiver("video",{direction:"recvonly"})}catch(s){console.warn("[RFWebRTC] Could not add transceiver:",s)}n&&n.getVideoTracks().forEach(s=>{try{s.contentHint="detail"}catch{}o.addTrack(s,n)});const r=z(o),i=o.createDataChannel("inference",{ordered:!0});let l;e&&(l=o.createDataChannel("video_upload"));const c=await o.createOffer();return await o.setLocalDescription(c),await H(o),{pc:o,offer:o.localDescription,remoteStreamPromise:r,dataChannel:i,uploadChannel:l}}async function Z(n){const e=n.getSenders().find(a=>a.track&&a.track.kind==="video");if(!e)return;const t=e.getParameters();t.encodings=t.encodings||[{}],t.encodings[0].scaleResolutionDownBy=1;try{await e.setParameters(t)}catch(a){console.warn("[RFWebRTC] Failed to set encoding parameters:",a)}}function Q(n,e=3e4){return new Promise((t,a)=>{if(n.readyState==="open"){t();return}const o=()=>{n.removeEventListener("open",o),n.removeEventListener("error",r),clearTimeout(i),t()},r=()=>{n.removeEventListener("open",o),n.removeEventListener("error",r),clearTimeout(i),a(new Error("Datachannel error"))},i=setTimeout(()=>{n.removeEventListener("open",o),n.removeEventListener("error",r),a(new Error("Datachannel open timeout"))},e);n.addEventListener("open",o),n.addEventListener("error",r)})}class D{constructor(e,t,a,o,r,i){d(this,"pc");d(this,"_localStream");d(this,"remoteStreamPromise");d(this,"pipelineId");d(this,"apiKey");d(this,"dataChannel");d(this,"reassembler");d(this,"uploadChannel");d(this,"uploader");d(this,"onComplete");this.pc=e,this._localStream=i==null?void 0:i.localStream,this.remoteStreamPromise=t,this.pipelineId=a,this.apiKey=o,this.dataChannel=r,this.reassembler=new L,this.uploadChannel=i==null?void 0:i.uploadChannel,this.onComplete=i==null?void 0:i.onComplete,this.dataChannel.binaryType="arraybuffer";const l=i==null?void 0:i.onData;l&&(this.dataChannel.addEventListener("message",c=>{try{if(c.data instanceof ArrayBuffer){const{frameId:s,chunkIndex:h,totalChunks:f,payload:C}=O(c.data),y=this.reassembler.processChunk(s,h,f,C);if(y){const S=new TextDecoder("utf-8").decode(y),R=JSON.parse(S);l(R)}}else{const s=JSON.parse(c.data);l(s)}}catch(s){console.error("[RFWebRTC] Failed to parse data channel message:",s)}}),this.dataChannel.addEventListener("error",c=>{console.error("[RFWebRTC] Data channel error:",c)})),this.dataChannel.addEventListener("close",()=>{this.reassembler.clear(),this.onComplete&&this.onComplete()})}async remoteStream(){return await this.remoteStreamPromise}localStream(){return this._localStream}async cleanup(){if(this.uploader&&this.uploader.cancel(),this.reassembler.clear(),this.pipelineId&&this.apiKey)try{await k.init({apiKey:this.apiKey}).terminatePipeline({pipelineId:this.pipelineId})}catch(e){console.warn("[RFWebRTC] Failed to terminate pipeline:",e)}this.pc&&this.pc.connectionState!=="closed"&&this.pc.close(),this._localStream&&W(this._localStream)}async startUpload(e,t){if(!this.uploadChannel)throw new Error("No upload channel available. This connection was not created for file uploads.");await Q(this.uploadChannel),this.uploader=new I(e,this.uploadChannel),await this.uploader.upload(t)}cancelUpload(){this.uploader&&this.uploader.cancel()}reconfigureOutputs(e){const t={};e.streamOutput!==void 0&&(t.stream_output=e.streamOutput),e.dataOutput!==void 0&&(t.data_output=e.dataOutput),this.sendData(t)}sendData(e){if(this.dataChannel.readyState!=="open"){console.warn("[RFWebRTC] Data channel is not open. Current state:",this.dataChannel.readyState);return}try{const t=typeof e=="string"?e:JSON.stringify(e);this.dataChannel.send(t)}catch(t){console.error("[RFWebRTC] Failed to send data:",t)}}}async function N({source:n,connector:e,wrtcParams:t,onData:a,onComplete:o,onFileUploadProgress:r,options:i={}}){var F;if(!e||typeof e.connectWrtc!="function")throw new Error("connector must have a connectWrtc method");const l=n instanceof File,c=l?void 0:n,s=l?n:void 0;let h=t.iceServers;if((!h||h.length===0)&&e.getIceServers)try{const w=await e.getIceServers();w&&w.length>0&&(h=w,console.log("[RFWebRTC] Using TURN servers from connector"))}catch(w){console.warn("[RFWebRTC] Failed to fetch TURN config, using defaults:",w)}const{pc:f,offer:C,remoteStreamPromise:y,dataChannel:T,uploadChannel:S}=await J(c,s,h),R={...t,iceServers:h,realtimeProcessing:t.realtimeProcessing??!l},g=await e.connectWrtc({sdp:C.sdp,type:C.type},R),m={sdp:g.sdp,type:g.type};if(!(m!=null&&m.sdp)||!(m!=null&&m.type))throw console.error("[RFWebRTC] Invalid answer from server:",g),new Error("connector.connectWrtc must return answer with sdp and type");const _=((F=g==null?void 0:g.context)==null?void 0:F.pipeline_id)||null;await f.setRemoteDescription(m),await new Promise((w,x)=>{const E=()=>{f.connectionState==="connected"?(f.removeEventListener("connectionstatechange",E),w()):f.connectionState==="failed"&&(f.removeEventListener("connectionstatechange",E),x(new Error("WebRTC connection failed")))};f.addEventListener("connectionstatechange",E),E(),setTimeout(()=>{f.removeEventListener("connectionstatechange",E),x(new Error("WebRTC connection timeout after 30s"))},3e4)}),c&&i.disableInputStreamDownscaling!==!1&&await Z(f);const b=e._apiKey||null,P=new D(f,y,_,b,T,{localStream:c,uploadChannel:S,onData:a,onComplete:o});return s&&S&&P.startUpload(s,r).catch(w=>{console.error("[RFWebRTC] Upload error:",w)}),P}async function X({source:n,connector:e,wrtcParams:t,onData:a,options:o={}}){if(n instanceof File)throw new Error("useStream requires a MediaStream. Use useVideoFile for File uploads.");return N({source:n,connector:e,wrtcParams:t,onData:a,options:o})}async function Y({file:n,connector:e,wrtcParams:t,onData:a,onUploadProgress:o,onComplete:r}){return N({source:n,connector:e,wrtcParams:{...t,realtimeProcessing:t.realtimeProcessing??!0},onData:a,onComplete:r,onFileUploadProgress:o})}const ee=Object.freeze(Object.defineProperty({__proto__:null,ChunkReassembler:L,FileUploader:I,RFWebRTCConnection:D,parseBinaryHeader:O,useStream:X,useVideoFile:Y},Symbol.toStringTag,{value:"Module"}));u.InferenceHTTPClient=k,u.connectors=q,u.streams=K,u.webrtc=ee,Object.defineProperty(u,Symbol.toStringTag,{value:"Module"})});
+(function(f,w){typeof exports=="object"&&typeof module<"u"?w(exports):typeof define=="function"&&define.amd?define(["exports"],w):(f=typeof globalThis<"u"?globalThis:f||self,w(f.RoboflowClient={}))})(this,function(f){"use strict";var oe=Object.defineProperty;var ie=(f,w,_)=>w in f?oe(f,w,{enumerable:!0,configurable:!0,writable:!0,value:_}):f[w]=_;var l=(f,w,_)=>ie(f,typeof w!="symbol"?w+"":w,_);var A;const w=typeof process<"u"&&((A=process.env)!=null&&A.RF_API_BASE_URL)?process.env.RF_API_BASE_URL:"https://api.roboflow.com",_=["https://serverless.roboflow.com"];class b{constructor(e,t="https://serverless.roboflow.com"){l(this,"apiKey");l(this,"serverUrl");this.apiKey=e,this.serverUrl=t}static init({apiKey:e,serverUrl:t}){if(!e)throw new Error("apiKey is required");return new b(e,t)}async initializeWebrtcWorker({offer:e,workflowSpec:t,workspaceName:n,workflowId:o,config:a={}}){if(!e||!e.sdp||!e.type)throw new Error("offer with sdp and type is required");const i=!!t,p=!!(n&&o);if(!i&&!p)throw new Error("Either workflowSpec OR (workspaceName + workflowId) is required");if(i&&p)throw new Error("Provide either workflowSpec OR (workspaceName + workflowId), not both");const{imageInputName:d="image",streamOutputNames:c=[],dataOutputNames:s=[],threadPoolWorkers:C=4,workflowsParameters:v={},iceServers:m,processingTimeout:u,requestedPlan:h,requestedRegion:R,realtimeProcessing:P=!0,rtspUrl:E}=a,T={type:"WorkflowConfiguration",image_input_name:d,workflows_parameters:v,workflows_thread_pool_workers:C,cancel_thread_pool_tasks_on_exit:!0,video_metadata_input_name:"video_metadata"};i?T.workflow_specification=t:(T.workspace_name=n,T.workflow_id=o);const g={workflow_configuration:T,api_key:this.apiKey,webrtc_realtime_processing:P,webrtc_offer:{sdp:e.sdp,type:e.type},webrtc_config:m?{iceServers:m}:null,stream_output:c,data_output:s};u!==void 0&&(g.processing_timeout=u),h!==void 0&&(g.requested_plan=h),R!==void 0&&(g.requested_region=R),E&&(g.rtsp_url=E);const y=await fetch(`${this.serverUrl}/initialise_webrtc_worker`,{method:"POST",headers:{"Content-Type":"application/json"},body:JSON.stringify(g)});if(!y.ok){const W=await y.text().catch(()=>"");throw new Error(`initialise_webrtc_worker failed (${y.status}): ${W}`)}return await y.json()}async terminatePipeline({pipelineId:e}){if(!e)throw new Error("pipelineId is required");await fetch(`${this.serverUrl}/inference_pipelines/${e}/terminate?api_key=${this.apiKey}`,{method:"POST",headers:{"Content-Type":"application/json"}})}async fetchTurnConfig(){if(!_.includes(this.serverUrl))return null;try{const e=await fetch(`${w}/webrtc_turn_config?api_key=${this.apiKey}`,{method:"GET",headers:{"Content-Type":"application/json"}});if(!e.ok)return console.warn(`[RFWebRTC] Failed to fetch TURN config (${e.status}), using defaults`),null;const t=await e.json();let n;if(Array.isArray(t))n=t;else if(t.iceServers&&Array.isArray(t.iceServers))n=t.iceServers;else if(t.urls)n=[t];else return console.warn("[RFWebRTC] Invalid TURN config format, using defaults"),null;return n.map(a=>({urls:Array.isArray(a.urls)?a.urls:[a.urls],username:a.username,credential:a.credential}))}catch(e){return console.warn("[RFWebRTC] Error fetching TURN config:",e),null}}}const B={withApiKey(r,e={}){const{serverUrl:t}=e;typeof window<"u"&&console.warn("[Security Warning] Using API key directly in browser will expose it. Use connectors.withProxyUrl() for production. See: https://docs.roboflow.com/api-reference/authentication#securing-your-api-key");const n=b.init({apiKey:r,serverUrl:t});return{connectWrtc:async(o,a)=>(console.debug("wrtcParams",a),await n.initializeWebrtcWorker({offer:o,workflowSpec:a.workflowSpec,workspaceName:a.workspaceName,workflowId:a.workflowId,config:{imageInputName:a.imageInputName,streamOutputNames:a.streamOutputNames,dataOutputNames:a.dataOutputNames,threadPoolWorkers:a.threadPoolWorkers,workflowsParameters:a.workflowsParameters,iceServers:a.iceServers,processingTimeout:a.processingTimeout,requestedPlan:a.requestedPlan,requestedRegion:a.requestedRegion,realtimeProcessing:a.realtimeProcessing,rtspUrl:a.rtspUrl}})),getIceServers:async()=>await n.fetchTurnConfig(),_apiKey:r,_serverUrl:t}},withProxyUrl(r,e={}){const{turnConfigUrl:t}=e;return{connectWrtc:async(n,o)=>{const a=await fetch(r,{method:"POST",headers:{"Content-Type":"application/json"},body:JSON.stringify({offer:n,wrtcParams:o})});if(!a.ok){const i=await a.text().catch(()=>"");throw new Error(`Proxy request failed (${a.status}): ${i}`)}return await a.json()},getIceServers:t?async()=>{try{const n=await fetch(t,{method:"GET",headers:{"Content-Type":"application/json"}});return n.ok?(await n.json()).iceServers||null:(console.warn(`[RFWebRTC] Failed to fetch TURN config from proxy (${n.status})`),null)}catch(n){return console.warn("[RFWebRTC] Error fetching TURN config from proxy:",n),null}}:void 0}}};async function V(r={video:!0}){try{console.log("[RFStreams] requesting with",r);const e=await navigator.mediaDevices.getUserMedia(r);return console.log("[RFStreams] got stream",e.getVideoTracks().map(t=>({id:t.id,label:t.label}))),e}catch(e){console.warn("[RFStreams] failed, falling back",e);const t=await navigator.mediaDevices.getUserMedia({video:!0,audio:!1});return console.log("[RFStreams] fallback stream",t.getVideoTracks().map(n=>({id:n.id,label:n.label}))),t}}function I(r){r&&(r.getTracks().forEach(e=>e.stop()),console.log("[RFStreams] Stream stopped"))}const $=Object.freeze(Object.defineProperty({__proto__:null,stopStream:I,useCamera:V},Symbol.toStringTag,{value:"Module"})),F=49152,M=262144,z=10;function H(r){return new Promise(e=>setTimeout(e,r))}class L{constructor(e,t){l(this,"file");l(this,"channel");l(this,"totalChunks");l(this,"cancelled",!1);this.file=e,this.channel=t,this.totalChunks=Math.ceil(e.size/F)}cancel(){this.cancelled=!0}async upload(e){const t=this.file.size;for(let n=0;n<this.totalChunks;n++){if(this.cancelled)throw new Error("Upload cancelled");if(this.channel.readyState!=="open")throw new Error("Video upload interrupted");const o=n*F,a=Math.min(o+F,t),i=this.file.slice(o,a),p=new Uint8Array(await i.arrayBuffer()),d=new ArrayBuffer(8+p.length),c=new DataView(d);for(c.setUint32(0,n,!0),c.setUint32(4,this.totalChunks,!0),new Uint8Array(d,8).set(p);this.channel.bufferedAmount>M;){if(this.channel.readyState!=="open")throw new Error("Video upload interrupted");await H(z)}this.channel.send(d),e&&e(a,t)}}}const G=12;class O{constructor(){l(this,"pendingFrames",new Map)}processChunk(e,t,n,o){if(n===1)return o;this.pendingFrames.has(e)||this.pendingFrames.set(e,{chunks:new Map,totalChunks:n});const a=this.pendingFrames.get(e);if(a.chunks.set(t,o),a.chunks.size===n){const i=Array.from(a.chunks.values()).reduce((c,s)=>c+s.length,0),p=new Uint8Array(i);let d=0;for(let c=0;c<n;c++){const s=a.chunks.get(c);p.set(s,d),d+=s.length}return this.pendingFrames.delete(e),p}return null}clear(){this.pendingFrames.clear()}}function D(r){const e=new DataView(r),t=e.getUint32(0,!0),n=e.getUint32(4,!0),o=e.getUint32(8,!0),a=new Uint8Array(r,G);return{frameId:t,chunkIndex:n,totalChunks:o,payload:a}}async function J(r,e=6e3){if(r.iceGatheringState==="complete")return;let t=!1;const n=o=>{o.candidate&&o.candidate.type==="srflx"&&(t=!0)};r.addEventListener("icecandidate",n);try{await Promise.race([new Promise(o=>{const a=()=>{r.iceGatheringState==="complete"&&(r.removeEventListener("icegatheringstatechange",a),o())};r.addEventListener("icegatheringstatechange",a)}),new Promise((o,a)=>{setTimeout(()=>{t?o():(console.error("[ICE] timeout with NO srflx candidate! Connection may fail."),a(new Error("ICE gathering timeout without srflx candidate")))},e)})])}finally{r.removeEventListener("icecandidate",n)}}function Z(r){return new Promise(e=>{r.addEventListener("track",t=>{t.streams&&t.streams[0]&&e(t.streams[0])})})}const Q=[{urls:["stun:stun.l.google.com:19302"]}];async function X(r,e,t,n,o){if([!!r,!!e,!!n].filter(Boolean).length!==1)throw new Error("Exactly one of localStream, file, or rtspUrl must be provided");const c=t??Q,s=new RTCPeerConnection({iceServers:c});o!=null&&o.onPeerConnectionCreated&&await o.onPeerConnectionCreated(s);try{s.addTransceiver("video",{direction:"recvonly"})}catch(h){console.warn("[RFWebRTC] Could not add transceiver:",h)}if(r)for(const h of r.getVideoTracks()){const R=s.addTrack(h,r);o!=null&&o.onTrackAdded&&await o.onTrackAdded(h,R,s)}const C=Z(s),v=s.createDataChannel("inference",{ordered:!0});let m;e&&(m=s.createDataChannel("video_upload"));let u=await s.createOffer();if(o!=null&&o.onOfferCreated){const h=await o.onOfferCreated(u);h&&(u=h)}return await s.setLocalDescription(u),await J(s),{pc:s,offer:s.localDescription,remoteStreamPromise:C,dataChannel:v,uploadChannel:m}}async function Y(r){const e=r.getSenders().find(n=>n.track&&n.track.kind==="video");if(!e)return;const t=e.getParameters();t.encodings=t.encodings||[{}],t.encodings[0].scaleResolutionDownBy=1;try{await e.setParameters(t)}catch(n){console.warn("[RFWebRTC] Failed to set encoding parameters:",n)}}function ee(r,e=3e4){return new Promise((t,n)=>{if(r.readyState==="open"){t();return}const o=()=>{r.removeEventListener("open",o),r.removeEventListener("error",a),clearTimeout(i),t()},a=()=>{r.removeEventListener("open",o),r.removeEventListener("error",a),clearTimeout(i),n(new Error("Datachannel error"))},i=setTimeout(()=>{r.removeEventListener("open",o),r.removeEventListener("error",a),n(new Error("Datachannel open timeout"))},e);r.addEventListener("open",o),r.addEventListener("error",a)})}class N{constructor(e,t,n,o,a,i){l(this,"peerConnection");l(this,"_localStream");l(this,"remoteStreamPromise");l(this,"pipelineId");l(this,"apiKey");l(this,"dataChannel");l(this,"reassembler");l(this,"uploadChannel");l(this,"uploader");l(this,"onComplete");this.peerConnection=e,this._localStream=i==null?void 0:i.localStream,this.remoteStreamPromise=t,this.pipelineId=n,this.apiKey=o,this.dataChannel=a,this.reassembler=new O,this.uploadChannel=i==null?void 0:i.uploadChannel,this.onComplete=i==null?void 0:i.onComplete,this.dataChannel.binaryType="arraybuffer";const p=i==null?void 0:i.onData;p&&(this.dataChannel.addEventListener("message",d=>{try{if(d.data instanceof ArrayBuffer){const{frameId:c,chunkIndex:s,totalChunks:C,payload:v}=D(d.data),m=this.reassembler.processChunk(c,s,C,v);if(m){const h=new TextDecoder("utf-8").decode(m),R=JSON.parse(h);p(R)}}else{const c=JSON.parse(d.data);p(c)}}catch(c){console.error("[RFWebRTC] Failed to parse data channel message:",c)}}),this.dataChannel.addEventListener("error",d=>{console.error("[RFWebRTC] Data channel error:",d)})),this.dataChannel.addEventListener("close",()=>{this.reassembler.clear(),this.onComplete&&this.onComplete()})}async remoteStream(){return await this.remoteStreamPromise}localStream(){return this._localStream}async cleanup(){if(this.uploader&&this.uploader.cancel(),this.reassembler.clear(),this.pipelineId&&this.apiKey)try{await b.init({apiKey:this.apiKey}).terminatePipeline({pipelineId:this.pipelineId})}catch(e){console.warn("[RFWebRTC] Failed to terminate pipeline:",e)}this.peerConnection&&this.peerConnection.connectionState!=="closed"&&this.peerConnection.close(),this._localStream&&I(this._localStream)}async startUpload(e,t){if(!this.uploadChannel)throw new Error("No upload channel available. This connection was not created for file uploads.");await ee(this.uploadChannel),this.uploader=new L(e,this.uploadChannel),await this.uploader.upload(t)}cancelUpload(){this.uploader&&this.uploader.cancel()}reconfigureOutputs(e){const t={};e.streamOutput!==void 0&&(t.stream_output=e.streamOutput),e.dataOutput!==void 0&&(t.data_output=e.dataOutput),this.sendData(t)}sendData(e){if(this.dataChannel.readyState!=="open"){console.warn("[RFWebRTC] Data channel is not open. Current state:",this.dataChannel.readyState);return}try{const t=typeof e=="string"?e:JSON.stringify(e);this.dataChannel.send(t)}catch(t){console.error("[RFWebRTC] Failed to send data:",t)}}}async function U({source:r,rtspUrl:e,connector:t,wrtcParams:n,onData:o,onComplete:a,onFileUploadProgress:i,options:p={},hooks:d}){var j;if(!t||typeof t.connectWrtc!="function")throw new Error("connector must have a connectWrtc method");const c=!!e,s=!c&&r instanceof File,C=!c&&!s&&r?r:void 0,v=s?r:void 0;let m=n.iceServers;if((!m||m.length===0)&&t.getIceServers)try{const S=await t.getIceServers();S&&S.length>0&&(m=S,console.log("[RFWebRTC] Using TURN servers from connector"))}catch(S){console.warn("[RFWebRTC] Failed to fetch TURN config, using defaults:",S)}const{pc:u,offer:h,remoteStreamPromise:R,dataChannel:P,uploadChannel:E}=await X(C,v,m,e,d),T={...n,iceServers:m,realtimeProcessing:n.realtimeProcessing??!s,rtspUrl:e},g=await t.connectWrtc({sdp:h.sdp,type:h.type},T),y={sdp:g.sdp,type:g.type};if(!(y!=null&&y.sdp)||!(y!=null&&y.type))throw console.error("[RFWebRTC] Invalid answer from server:",g),new Error("connector.connectWrtc must return answer with sdp and type");const x=((j=g==null?void 0:g.context)==null?void 0:j.pipeline_id)||null;await u.setRemoteDescription(y),await new Promise((S,K)=>{const k=()=>{u.connectionState==="connected"?(u.removeEventListener("connectionstatechange",k),S()):u.connectionState==="failed"&&(u.removeEventListener("connectionstatechange",k),K(new Error("WebRTC connection failed")))};u.addEventListener("connectionstatechange",k),k(),setTimeout(()=>{u.removeEventListener("connectionstatechange",k),K(new Error("WebRTC connection timeout after 30s"))},3e4)}),C&&p.disableInputStreamDownscaling!==!1&&await Y(u);const W=t._apiKey||null,q=new N(u,R,x,W,P,{localStream:C,uploadChannel:E,onData:o,onComplete:a});return v&&E&&q.startUpload(v,i).catch(S=>{console.error("[RFWebRTC] Upload error:",S)}),q}async function te({source:r,connector:e,wrtcParams:t,onData:n,options:o={},hooks:a}){if(r instanceof File)throw new Error("useStream requires a MediaStream. Use useVideoFile for File uploads.");return U({source:r,connector:e,wrtcParams:t,onData:n,options:o,hooks:a})}async function re({file:r,connector:e,wrtcParams:t,onData:n,onUploadProgress:o,onComplete:a,hooks:i}){return U({source:r,connector:e,wrtcParams:{...t,realtimeProcessing:t.realtimeProcessing??!0},onData:n,onComplete:a,onFileUploadProgress:o,hooks:i})}async function ne({rtspUrl:r,connector:e,wrtcParams:t,onData:n,hooks:o}){if(!r.startsWith("rtsp://")&&!r.startsWith("rtsps://"))throw new Error("Invalid RTSP URL: must start with rtsp:// or rtsps://");return U({rtspUrl:r,connector:e,wrtcParams:t,onData:n,hooks:o})}const ae=Object.freeze(Object.defineProperty({__proto__:null,ChunkReassembler:O,FileUploader:L,RFWebRTCConnection:N,parseBinaryHeader:D,useRtspStream:ne,useStream:te,useVideoFile:re},Symbol.toStringTag,{value:"Module"}));f.InferenceHTTPClient=b,f.connectors=B,f.streams=$,f.webrtc=ae,Object.defineProperty(f,Symbol.toStringTag,{value:"Module"})});
 //# sourceMappingURL=index.js.map