@framers/agentos 0.1.112 → 0.1.114

package/dist/voice-pipeline/WebSocketStreamTransport.js

@@ -21,11 +21,11 @@
  * not crash the transport, allowing the session to continue.
  *
  * ### Binary frames (outbound)
- * {@link sendAudio} sends the raw `Buffer` from an {@link EncodedAudioChunk}
+ * `sendAudio()` sends the raw `Buffer` from an {@link EncodedAudioChunk}
  * (or converts a `Float32Array` to a `Buffer` for raw {@link AudioFrame}s).
  *
  * ### Text frames (outbound)
- * {@link sendControl} JSON-stringifies a {@link ServerTextMessage} and sends
+ * `sendControl()` JSON-stringifies a {@link ServerTextMessage} and sends
  * it as a text frame.
  *
  * ## Reconnection behaviour
@@ -59,8 +59,8 @@ import { randomUUID } from 'node:crypto';
  *
  * | Method          | Behaviour                                              |
  * |-----------------|--------------------------------------------------------|
- * | {@link sendAudio}   | Serialises audio to a binary `Buffer` and calls `ws.send()`. |
- * | {@link sendControl} | JSON-stringifies the message and calls `ws.send()`.    |
+ * | `sendAudio()`   | Serialises audio to a binary `Buffer` and calls `ws.send()`. |
+ * | `sendControl()` | JSON-stringifies the message and calls `ws.send()`.    |
  *
  * ## Lifecycle events (re-emitted on `this`)
  *
@@ -97,7 +97,7 @@ export class WebSocketStreamTransport extends EventEmitter {
      * Create a new transport wrapping an existing WebSocket connection.
      *
      * The constructor inspects `ws.readyState` to determine the initial
-     * {@link state}: if the socket is already OPEN (readyState === 1) the state
+     * `state`: if the socket is already OPEN (readyState === 1) the state
      * is set to `'open'`; otherwise it is set to `'connecting'` and will
      * transition to `'open'` when the `'open'` event fires.
      *

package/dist/voice-pipeline/index.d.ts

@@ -42,6 +42,7 @@ export { SoftFadeBargeinHandler } from './SoftFadeBargeinHandler.js';
 export { HeuristicEndpointDetector } from './HeuristicEndpointDetector.js';
 export { AcousticEndpointDetector } from './AcousticEndpointDetector.js';
 export { WebSocketStreamTransport } from './WebSocketStreamTransport.js';
+export { WebRTCStreamTransport, createWebRTCTransport } from './WebRTCStreamTransport.js';
 export { VoicePipelineOrchestrator } from './VoicePipelineOrchestrator.js';
 export { VoiceInterruptError } from './VoiceInterruptError.js';
 //# sourceMappingURL=index.d.ts.map

package/dist/voice-pipeline/index.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../../src/voice-pipeline/index.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GAqCG;AAIH,cAAc,YAAY,CAAC;AAG3B,OAAO,EAAE,qBAAqB,EAAE,MAAM,4BAA4B,CAAC;AACnE,OAAO,EAAE,sBAAsB,EAAE,MAAM,6BAA6B,CAAC;AAGrE,OAAO,EAAE,yBAAyB,EAAE,MAAM,gCAAgC,CAAC;AAC3E,OAAO,EAAE,wBAAwB,EAAE,MAAM,+BAA+B,CAAC;AAGzE,OAAO,EAAE,wBAAwB,EAAE,MAAM,+BAA+B,CAAC;AAGzE,OAAO,EAAE,qBAAqB,EAAE,qBAAqB,EAAE,MAAM,4BAA4B,CAAC;AAG1F,OAAO,EAAE,yBAAyB,EAAE,MAAM,gCAAgC,CAAC;AAG3E,OAAO,EAAE,mBAAmB,EAAE,MAAM,0BAA0B,CAAC"}

package/dist/voice-pipeline/index.js

@@ -47,6 +47,8 @@ export { HeuristicEndpointDetector } from './HeuristicEndpointDetector.js';
 export { AcousticEndpointDetector } from './AcousticEndpointDetector.js';
 // WebSocket-based transport implementation
 export { WebSocketStreamTransport } from './WebSocketStreamTransport.js';
+// WebRTC DataChannel-based transport implementation
+export { WebRTCStreamTransport, createWebRTCTransport } from './WebRTCStreamTransport.js';
 // Central pipeline orchestrator (state machine)
 export { VoicePipelineOrchestrator } from './VoicePipelineOrchestrator.js';
 // Typed error for barge-in interruptions

package/dist/voice-pipeline/index.js.map

@@ -1 +1 @@
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+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/voice-pipeline/index.ts"],"names":[],"mappings":"AAAA;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;GAqCG;AAEH,wDAAwD;AACxD,8EAA8E;AAC9E,cAAc,YAAY,CAAC;AAE3B,4CAA4C;AAC5C,OAAO,EAAE,qBAAqB,EAAE,MAAM,4BAA4B,CAAC;AACnE,OAAO,EAAE,sBAAsB,EAAE,MAAM,6BAA6B,CAAC;AAErE,6CAA6C;AAC7C,OAAO,EAAE,yBAAyB,EAAE,MAAM,gCAAgC,CAAC;AAC3E,OAAO,EAAE,wBAAwB,EAAE,MAAM,+BAA+B,CAAC;AAEzE,2CAA2C;AAC3C,OAAO,EAAE,wBAAwB,EAAE,MAAM,+BAA+B,CAAC;AAEzE,oDAAoD;AACpD,OAAO,EAAE,qBAAqB,EAAE,qBAAqB,EAAE,MAAM,4BAA4B,CAAC;AAE1F,gDAAgD;AAChD,OAAO,EAAE,yBAAyB,EAAE,MAAM,gCAAgC,CAAC;AAE3E,yCAAyC;AACzC,OAAO,EAAE,mBAAmB,EAAE,MAAM,0BAA0B,CAAC"}

package/dist/voice-pipeline/types.d.ts

@@ -79,7 +79,7 @@ export interface AudioFrame {
      * (e.g. a hardware device label or a WebRTC peer ID). Used by the
      * diarization engine when native speaker IDs are unavailable.
      *
-     * @see {@link DiarizedSegment.speakerId} for the post-diarization label.
+     * See `DiarizedSegment.speakerId` for the post-diarization label.
      */
     speakerHint?: string;
 }
@@ -104,11 +104,11 @@ export interface AudioFrame {
  */
 export interface EncodedAudioChunk {
     /**
-     * Raw encoded bytes in the format specified by {@link format}.
+     * Raw encoded bytes in the format specified by `format`.
      */
     audio: Buffer;
     /**
-     * Codec/container format of {@link audio}.
+     * Codec/container format of `audio`.
      *
      * - `'pcm'` -- raw signed 16-bit LE samples (lowest latency, highest bandwidth).
      * - `'mp3'` -- MPEG Layer 3 (wide browser support, moderate latency).
@@ -122,12 +122,12 @@ export interface EncodedAudioChunk {
     /**
      * Playback duration of this chunk in milliseconds.
      * Used by the orchestrator to track cumulative played time for
-     * barge-in context ({@link BargeinContext.playedDurationMs}).
+     * barge-in context (`BargeinContext.playedDurationMs`).
      */
     durationMs: number;
     /**
      * The text fragment that was synthesised into this chunk. Preserved so
-     * barge-in handlers can report {@link VoiceTurnMetadata.interruptedRemainder}
+     * barge-in handlers can report `VoiceTurnMetadata.interruptedRemainder`
      * accurately when playback is cut short.
      */
     text: string;
@@ -136,7 +136,7 @@ export interface EncodedAudioChunk {
  * Discriminated union of control messages sent from the pipeline to the
  * underlying stream transport (e.g. a WebSocket or WebRTC data-channel).
  *
- * @see {@link IStreamTransport.sendControl} which accepts these messages.
+ * See `IStreamTransport.sendControl()` for the transport method that accepts these messages.
  * @see {@link ServerTextMessage} for the full server-to-client protocol.
  *
  * @example
@@ -223,10 +223,10 @@ export interface IStreamTransport extends EventEmitter {
     close(code?: number, reason?: string): void;
 }
 /**
- * Configuration passed to {@link IStreamingSTT.startSession} when opening a new
+ * Configuration passed to `IStreamingSTT.startSession()` when opening a new
  * speech recognition stream.
  *
- * @see {@link VoicePipelineConfig.sttOptions} for provider-level overrides.
+ * See `VoicePipelineConfig.sttOptions` for provider-level overrides.
  *
  * @example
  * ```typescript
@@ -273,7 +273,7 @@ export interface StreamingSTTConfig {
  * A single word within a {@link TranscriptEvent}, augmented with timing and
  * optional speaker attribution.
  *
- * @see {@link TranscriptEvent.words} which contains an array of these.
+ * See `TranscriptEvent.words` for the array that contains these entries.
  */
 export interface TranscriptWord {
     /**
@@ -304,7 +304,7 @@ export interface TranscriptWord {
  * Emitted by a {@link StreamingSTTSession} each time the provider produces a
  * recognition hypothesis.
  *
- * @see {@link IEndpointDetector.pushTranscript} which consumes these events.
+ * See `IEndpointDetector.pushTranscript()` for the endpoint detector hook that consumes these events.
  *
  * @example
  * ```typescript
@@ -321,7 +321,7 @@ export interface TranscriptEvent {
      */
     text: string;
     /**
-     * Aggregate confidence score for {@link text} in the range [0, 1].
+     * Aggregate confidence score for `text` in the range [0, 1].
      */
     confidence: number;
     /**
@@ -355,7 +355,7 @@ export interface TranscriptEvent {
  * | `'error'`      | `Error`               | Unrecoverable provider error.      |
  * | `'close'`      | *(none)*              | Session has been fully terminated. |
  *
- * @see {@link IStreamingSTT.startSession} which creates these sessions.
+ * See `IStreamingSTT.startSession()` for the factory method that creates these sessions.
  */
 export interface StreamingSTTSession extends EventEmitter {
     /**
@@ -382,7 +382,7 @@ export interface StreamingSTTSession extends EventEmitter {
  * Implementations are registered via the `EXTENSION_KIND_STREAMING_STT`
  * extension kind and resolved by the voice pipeline at session creation time.
  *
- * @see {@link StreamingSTTSession} for the session interface returned by {@link startSession}.
+ * See {@link StreamingSTTSession} for the session interface returned by `startSession()`.
  */
 export interface IStreamingSTT {
     /**
@@ -406,7 +406,7 @@ export interface IStreamingSTT {
  * A VAD (Voice Activity Detection) or STT-derived event describing speech
  * energy transitions over time.
  *
- * @see {@link IEndpointDetector.pushVadEvent} which consumes these.
+ * See `IEndpointDetector.pushVadEvent()` for the endpoint-detector hook that consumes these.
  *
  * @example
  * ```typescript
@@ -460,8 +460,8 @@ export interface VadEvent {
  * | `manual`           | Explicitly triggered by a ClientTextMessage control     |
  * | `timeout`          | Hard maximum turn duration elapsed                     |
  *
- * @see {@link TurnCompleteEvent.reason} which carries this value.
- * @see {@link VoiceTurnMetadata.endpointReason} where it is forwarded to the agent.
+ * See `TurnCompleteEvent.reason` for the field that carries this value.
+ * See `VoiceTurnMetadata.endpointReason` for where it is forwarded to the agent.
  */
 export type EndpointReason = 'silence_timeout' | 'punctuation' | 'syntax_complete' | 'semantic_model' | 'manual' | 'timeout';
 /**
@@ -547,7 +547,7 @@ export interface IEndpointDetector extends EventEmitter {
  * Configuration for a diarization session. Controls expected speaker count and
  * chunking behaviour for providers that require buffered audio.
  *
- * @see {@link IDiarizationEngine.startSession} which accepts this config.
+ * See `IDiarizationEngine.startSession()` for the factory method that accepts this config.
  */
 export interface DiarizationConfig {
     /**
@@ -615,7 +615,7 @@ export interface DiarizedSegment extends TranscriptSegment {
      */
     speakerId: string;
     /**
-     * Confidence that this segment belongs to {@link speakerId}, in the range [0, 1].
+     * Confidence that this segment belongs to `speakerId`, in the range [0, 1].
      */
     speakerConfidence: number;
 }
@@ -632,7 +632,7 @@ export interface DiarizedSegment extends TranscriptSegment {
  * | `'error'`          | `Error`                                  | Unrecoverable engine error.    |
  * | `'close'`          | *(none)*                                 | Session terminated.            |
  *
- * @see {@link IDiarizationEngine.startSession} which creates these sessions.
+ * See `IDiarizationEngine.startSession()` for the factory method that creates these sessions.
  */
 export interface DiarizationSession extends EventEmitter {
     /**
@@ -668,7 +668,7 @@ export interface DiarizationSession extends EventEmitter {
  *
  * Registered via `EXTENSION_KIND_DIARIZATION`.
  *
- * @see {@link DiarizationSession} for the session interface returned by {@link startSession}.
+ * See {@link DiarizationSession} for the session interface returned by `startSession()`.
  */
 export interface IDiarizationEngine {
     /**
@@ -680,10 +680,10 @@ export interface IDiarizationEngine {
     startSession(config?: DiarizationConfig): Promise<DiarizationSession>;
 }
 /**
- * Configuration passed to {@link IStreamingTTS.startSession} when opening a new
+ * Configuration passed to `IStreamingTTS.startSession()` when opening a new
  * text-to-speech synthesis stream.
  *
- * @see {@link VoicePipelineConfig.ttsOptions} for provider-level overrides.
+ * See `VoicePipelineConfig.ttsOptions` for provider-level overrides.
  *
  * @example
  * ```typescript
@@ -707,7 +707,7 @@ export interface StreamingTTSConfig {
      */
     format?: 'pcm' | 'mp3' | 'opus';
     /**
-     * Output sample rate in Hz. Must be supported by the chosen {@link format}.
+     * Output sample rate in Hz. Must be supported by the chosen `format`.
      * @defaultValue 24000
      */
     sampleRate?: number;
@@ -722,7 +722,7 @@ export interface StreamingTTSConfig {
     chunkingMode?: 'sentence' | 'word' | 'paragraph';
     /**
      * Maximum number of milliseconds of audio to buffer before forcing a flush,
-     * regardless of {@link chunkingMode}. Prevents unbounded memory growth for very
+     * regardless of `chunkingMode`. Prevents unbounded memory growth for very
      * long utterances.
      * @defaultValue 3000
      */
@@ -745,12 +745,12 @@ export interface StreamingTTSConfig {
  * | `'error'`          | `Error`                   | Unrecoverable synthesis error.       |
  * | `'close'`          | *(none)*                  | Session terminated.                  |
  *
- * @see {@link IStreamingTTS.startSession} which creates these sessions.
+ * See `IStreamingTTS.startSession()` for the factory method that creates these sessions.
  */
 export interface StreamingTTSSession extends EventEmitter {
     /**
      * Push one or more LLM output tokens into the synthesis buffer.
-     * The session will chunk and synthesise them according to {@link StreamingTTSConfig.chunkingMode}.
+     * The session will chunk and synthesise them according to `StreamingTTSConfig.chunkingMode`.
      *
      * @param tokens - Text tokens to synthesise (may be partial words).
      */
@@ -776,7 +776,7 @@ export interface StreamingTTSSession extends EventEmitter {
  *
  * Registered via `EXTENSION_KIND_STREAMING_TTS`.
  *
- * @see {@link StreamingTTSSession} for the session interface returned by {@link startSession}.
+ * See {@link StreamingTTSSession} for the session interface returned by `startSession()`.
  */
 export interface IStreamingTTS {
     /**
@@ -793,7 +793,7 @@ export interface IStreamingTTS {
     startSession(config?: StreamingTTSConfig): Promise<StreamingTTSSession>;
 }
 /**
- * Contextual information supplied to {@link IBargeinHandler.handleBargein} so the
+ * Contextual information supplied to `IBargeinHandler.handleBargein()` so the
  * handler can make an informed decision about how to respond to interruption.
  *
  * @see {@link IBargeinHandler} which consumes this context.
@@ -820,20 +820,20 @@ export interface BargeinContext {
     speechDurationMs: number;
     /**
      * The partial TTS text that was interrupted. Used to construct
-     * {@link VoiceTurnMetadata.interruptedRemainder} so the agent knows what
+     * `VoiceTurnMetadata.interruptedRemainder` so the agent knows what
      * information was cut off and can avoid repeating it.
      */
     interruptedText: string;
     /**
      * How many milliseconds of audio had been played at the point of interruption.
-     * Combined with {@link interruptedText}, this allows the agent to estimate
+     * Combined with `interruptedText`, this allows the agent to estimate
      * how much of the response the user actually heard.
      */
     playedDurationMs: number;
 }
 /**
  * Action the pipeline should take in response to a detected barge-in.
- * Returned by {@link IBargeinHandler.handleBargein}.
+ * Returned by `IBargeinHandler.handleBargein()`.
  *
  * @see {@link IBargeinHandler} which returns this type.
  *
@@ -854,7 +854,7 @@ export type BargeinAction = {
      */
     injectMarker?: string;
 } | {
-    /** Fade out TTS audio over {@link fadeMs} milliseconds then pause. */
+    /** Fade out TTS audio over `fadeMs` milliseconds then pause. */
     type: 'pause';
     /** Duration of the fade-out in milliseconds. @defaultValue 150 */
     fadeMs?: number;
@@ -898,7 +898,7 @@ export interface IBargeinHandler {
 /**
  * Adapts any AgentOS agent to the voice pipeline's turn-based protocol.
  *
- * The pipeline calls {@link sendText} with the user's final transcript and
+ * The pipeline calls `sendText()` with the user's final transcript and
  * streams the response back as text tokens for TTS synthesis.
  *
  * @see {@link VoicePipelineOrchestrator} which invokes this during the
@@ -934,7 +934,7 @@ export interface IVoicePipelineAgentSession {
  * Rich metadata attached to each voice turn and passed to the agent session.
  * Enables the agent to tailor its response based on conversation dynamics.
  *
- * @see {@link IVoicePipelineAgentSession.sendText} which receives this metadata.
+ * See `IVoicePipelineAgentSession.sendText()` for the agent-session method that receives this metadata.
  *
  * @example
  * ```typescript
@@ -968,7 +968,7 @@ export interface VoiceTurnMetadata {
      */
     wasInterrupted: boolean;
     /**
-     * When {@link wasInterrupted} is `true`, the text remainder of the agent response
+     * When `wasInterrupted` is `true`, the text remainder of the agent response
      * that was cut off. Useful for the agent to avoid re-stating information
      * the user has already heard.
      */
@@ -1012,7 +1012,7 @@ export interface VoicePipelineConfig {
     tts: string;
     /**
      * Endpoint detection strategy. Defaults to `'heuristic'` when omitted.
-     * @see {@link IEndpointDetector.mode} for the strategy descriptions.
+     * See `IEndpointDetector.mode` for the strategy descriptions.
      */
     endpointing?: 'acoustic' | 'heuristic' | 'semantic';
     /**
@@ -1025,11 +1025,11 @@ export interface VoicePipelineConfig {
      */
     bargeIn?: 'hard-cut' | 'soft-fade' | 'disabled';
     /**
-     * TTS voice identifier. Forwarded to {@link StreamingTTSConfig.voice}.
+     * TTS voice identifier. Forwarded to `StreamingTTSConfig.voice`.
      */
     voice?: string;
     /**
-     * Output audio format for TTS. Forwarded to {@link StreamingTTSConfig.format}.
+     * Output audio format for TTS. Forwarded to `StreamingTTSConfig.format`.
      * @defaultValue 'opus'
      */
     format?: 'pcm' | 'mp3' | 'opus';
@@ -1044,11 +1044,11 @@ export interface VoicePipelineConfig {
      */
     maxTurnDurationMs?: number;
     /**
-     * Provider-level STT options merged into {@link StreamingSTTConfig.providerOptions}.
+     * Provider-level STT options merged into `StreamingSTTConfig.providerOptions`.
      */
     sttOptions?: Record<string, unknown>;
     /**
-     * Provider-level TTS options merged into {@link StreamingTTSConfig.providerOptions}.
+     * Provider-level TTS options merged into `StreamingTTSConfig.providerOptions`.
      */
     ttsOptions?: Record<string, unknown>;
 }
@@ -1063,7 +1063,7 @@ export interface VoicePipelineConfig {
  * any  -> closed
  * ```
  *
- * The state machine is enforced by {@link VoicePipelineOrchestrator._setState}
+ * The state machine is enforced by the internal `VoicePipelineOrchestrator._setState`
  * which emits `'state_changed'` on every transition.
  */
 export type PipelineState = 'idle' | 'listening' | 'processing' | 'speaking' | 'interrupting' | 'closed';
@@ -1083,7 +1083,7 @@ export type PipelineState = 'idle' | 'listening' | 'processing' | 'speaking' | '
  * | `'error'`                | `Error`                   | Unrecoverable pipeline error.           |
  * | `'close'`                | *(none)*                  | Session has been fully torn down.       |
  *
- * @see {@link VoicePipelineOrchestrator.startSession} which creates these sessions.
+ * See `VoicePipelineOrchestrator.startSession()` for the factory method that creates these sessions.
  */
 export interface VoicePipelineSession extends EventEmitter {
     /**
@@ -1143,7 +1143,7 @@ export type ClientTextMessage = {
  * All messages are JSON-serialised.
  *
  * @see {@link ClientTextMessage} for the client-to-server counterpart.
- * @see {@link IStreamTransport.sendControl} which sends these messages.
+ * See `IStreamTransport.sendControl()` for the transport method that sends these messages.
  *
  * @example
  * ```typescript

package/dist/voice-pipeline/types.d.ts.map

@@ -1 +1 @@
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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@framers/agentos",
-  "version": "0.1.112",
+  "version": "0.1.114",
   "description": "Modular AgentOS orchestration library",
   "type": "module",
   "main": "dist/index.js",
@@ -103,6 +103,11 @@
       "default": "./dist/core/tools/*.js",
       "types": "./dist/core/tools/*.d.ts"
     },
+    "./core/vision": {
+      "import": "./dist/core/vision/index.js",
+      "default": "./dist/core/vision/index.js",
+      "types": "./dist/core/vision/index.d.ts"
+    },
     "./discovery": {
       "import": "./dist/discovery/index.js",
       "default": "./dist/discovery/index.js",
@@ -123,6 +128,11 @@
       "default": "./dist/voice-pipeline/index.js",
       "types": "./dist/voice-pipeline/index.d.ts"
     },
+    "./query-router": {
+      "import": "./dist/query-router/index.js",
+      "default": "./dist/query-router/index.js",
+      "types": "./dist/query-router/index.d.ts"
+    },
     "./memory/index": {
       "import": "./dist/memory/index.js",
       "default": "./dist/memory/index.js",
@@ -248,6 +258,8 @@
     "graphology-types": ">=0.24.0",
     "hnswlib-node": ">=3.0.0",
     "neo4j-driver": ">=5.0.0",
+    "ppu-paddle-ocr": ">=3.0.0",
+    "tesseract.js": ">=5.0.0",
     "typescript": ">=5.0.0"
   },
   "peerDependenciesMeta": {
@@ -268,6 +280,12 @@
     },
     "neo4j-driver": {
       "optional": true
+    },
+    "ppu-paddle-ocr": {
+      "optional": true
+    },
+    "tesseract.js": {
+      "optional": true
     }
   }
 }