@runanywhere/web 0.1.0-beta.1

package/dist/Public/Extensions/RunAnywhere+VoiceAgent.js

@@ -0,0 +1,222 @@
+/**
+ * RunAnywhere Web SDK - VoiceAgent Extension
+ *
+ * Orchestrates the complete voice pipeline: VAD -> STT -> LLM -> TTS.
+ * Uses the RACommons rac_voice_agent_* C API for pipeline management.
+ *
+ * Mirrors: sdk/runanywhere-swift/Sources/RunAnywhere/Public/Extensions/VoiceAgent/
+ *
+ * Usage:
+ *   import { VoiceAgent } from '@runanywhere/web';
+ *
+ *   const agent = await VoiceAgent.create();
+ *   await agent.loadModels({ stt: '/models/whisper.bin', llm: '/models/llama.gguf', tts: '/models/piper.onnx' });
+ *   const result = await agent.processVoiceTurn(audioData);
+ *   console.log('Transcription:', result.transcription);
+ *   console.log('Response:', result.response);
+ */
+import { RunAnywhere } from '../RunAnywhere';
+import { WASMBridge } from '../../Foundation/WASMBridge';
+import { SDKError } from '../../Foundation/ErrorTypes';
+import { SDKLogger } from '../../Foundation/SDKLogger';
+import { EventBus } from '../../Foundation/EventBus';
+import { SDKEventType } from '../../types/enums';
+export { PipelineState } from './VoiceAgentTypes';
+const logger = new SDKLogger('VoiceAgent');
+// ---------------------------------------------------------------------------
+// VoiceAgent Instance
+// ---------------------------------------------------------------------------
+export class VoiceAgentSession {
+    handle;
+    eventCallbackPtr = 0;
+    constructor(handle) {
+        this.handle = handle;
+    }
+    /**
+     * Load models for all components.
+     */
+    async loadModels(models) {
+        const bridge = WASMBridge.shared;
+        if (models.stt) {
+            logger.info(`Loading STT model: ${models.stt.id}`);
+            const pathPtr = bridge.allocString(models.stt.path);
+            const idPtr = bridge.allocString(models.stt.id);
+            const namePtr = bridge.allocString(models.stt.name ?? models.stt.id);
+            try {
+                bridge.callWithCheck('rac_voice_agent_load_stt_model', ['number', 'number', 'number', 'number'], [this.handle, pathPtr, idPtr, namePtr]);
+            }
+            finally {
+                bridge.free(pathPtr);
+                bridge.free(idPtr);
+                bridge.free(namePtr);
+            }
+        }
+        if (models.llm) {
+            logger.info(`Loading LLM model: ${models.llm.id}`);
+            const pathPtr = bridge.allocString(models.llm.path);
+            const idPtr = bridge.allocString(models.llm.id);
+            const namePtr = bridge.allocString(models.llm.name ?? models.llm.id);
+            try {
+                bridge.callWithCheck('rac_voice_agent_load_llm_model', ['number', 'number', 'number', 'number'], [this.handle, pathPtr, idPtr, namePtr]);
+            }
+            finally {
+                bridge.free(pathPtr);
+                bridge.free(idPtr);
+                bridge.free(namePtr);
+            }
+        }
+        if (models.tts) {
+            logger.info(`Loading TTS voice: ${models.tts.id}`);
+            const pathPtr = bridge.allocString(models.tts.path);
+            const idPtr = bridge.allocString(models.tts.id);
+            const namePtr = bridge.allocString(models.tts.name ?? models.tts.id);
+            try {
+                bridge.callWithCheck('rac_voice_agent_load_tts_voice', ['number', 'number', 'number', 'number'], [this.handle, pathPtr, idPtr, namePtr]);
+            }
+            finally {
+                bridge.free(pathPtr);
+                bridge.free(idPtr);
+                bridge.free(namePtr);
+            }
+        }
+        // Initialize with loaded models
+        bridge.callWithCheck('rac_voice_agent_initialize_with_loaded_models', ['number'], [this.handle]);
+        logger.info('VoiceAgent initialized with loaded models');
+    }
+    /**
+     * Process a complete voice turn (audio in -> text response + audio out).
+     */
+    async processVoiceTurn(audioData) {
+        const bridge = WASMBridge.shared;
+        const m = bridge.module;
+        const audioPtr = m._malloc(audioData.length);
+        bridge.writeBytes(audioData, audioPtr);
+        // rac_voice_agent_result_t: { speech_detected, transcription, response, synthesized_audio, audio_size }
+        const resultSize = m._rac_wasm_sizeof_voice_agent_result();
+        const resultPtr = m._malloc(resultSize);
+        try {
+            bridge.callWithCheck('rac_voice_agent_process_voice_turn', ['number', 'number', 'number', 'number'], [this.handle, audioPtr, audioData.length, resultPtr]);
+            const speechDetected = m.getValue(resultPtr, 'i32') === 1;
+            const transcriptionPtr = m.getValue(resultPtr + 4, '*');
+            const responsePtr = m.getValue(resultPtr + 8, '*');
+            const audioDataPtr = m.getValue(resultPtr + 12, '*');
+            const audioSize = m.getValue(resultPtr + 16, 'i32');
+            const result = {
+                speechDetected,
+                transcription: transcriptionPtr ? bridge.readString(transcriptionPtr) : undefined,
+                response: responsePtr ? bridge.readString(responsePtr) : undefined,
+            };
+            if (audioDataPtr && audioSize > 0) {
+                const numSamples = audioSize / 4;
+                result.synthesizedAudio = bridge.readFloat32Array(audioDataPtr, numSamples);
+            }
+            // Free C result
+            bridge.callFunction('rac_voice_agent_result_free', null, ['number'], [resultPtr]);
+            EventBus.shared.emit('voice.turnCompleted', SDKEventType.Voice, {
+                speechDetected,
+                transcription: result.transcription ?? '',
+                response: result.response ?? '',
+            });
+            return result;
+        }
+        finally {
+            m._free(audioPtr);
+        }
+    }
+    /** Check if the voice agent is ready. */
+    get isReady() {
+        const bridge = WASMBridge.shared;
+        const m = bridge.module;
+        const outPtr = m._malloc(4);
+        try {
+            bridge.callFunction('rac_voice_agent_is_ready', 'number', ['number', 'number'], [this.handle, outPtr]);
+            return m.getValue(outPtr, 'i32') === 1;
+        }
+        finally {
+            m._free(outPtr);
+        }
+    }
+    /** Transcribe audio without the full pipeline. */
+    async transcribe(audioData) {
+        const bridge = WASMBridge.shared;
+        const m = bridge.module;
+        const audioPtr = m._malloc(audioData.length);
+        bridge.writeBytes(audioData, audioPtr);
+        const outPtr = m._malloc(4);
+        try {
+            bridge.callWithCheck('rac_voice_agent_transcribe', ['number', 'number', 'number', 'number'], [this.handle, audioPtr, audioData.length, outPtr]);
+            const textPtr = m.getValue(outPtr, '*');
+            const text = bridge.readString(textPtr);
+            if (textPtr)
+                m._free(textPtr);
+            return text;
+        }
+        finally {
+            m._free(audioPtr);
+            m._free(outPtr);
+        }
+    }
+    /** Generate LLM response without the full pipeline. */
+    async generateResponse(prompt) {
+        const bridge = WASMBridge.shared;
+        const m = bridge.module;
+        const promptPtr = bridge.allocString(prompt);
+        const outPtr = m._malloc(4);
+        try {
+            bridge.callWithCheck('rac_voice_agent_generate_response', ['number', 'number', 'number'], [this.handle, promptPtr, outPtr]);
+            const textPtr = m.getValue(outPtr, '*');
+            const text = bridge.readString(textPtr);
+            if (textPtr)
+                m._free(textPtr);
+            return text;
+        }
+        finally {
+            bridge.free(promptPtr);
+            m._free(outPtr);
+        }
+    }
+    /** Destroy the voice agent session. */
+    destroy() {
+        if (this.eventCallbackPtr !== 0) {
+            try {
+                WASMBridge.shared.module.removeFunction(this.eventCallbackPtr);
+            }
+            catch { /* ignore */ }
+            this.eventCallbackPtr = 0;
+        }
+        if (this.handle !== 0) {
+            try {
+                WASMBridge.shared.callFunction('rac_voice_agent_destroy', null, ['number'], [this.handle]);
+            }
+            catch { /* ignore */ }
+            this.handle = 0;
+        }
+    }
+}
+// ---------------------------------------------------------------------------
+// VoiceAgent Factory
+// ---------------------------------------------------------------------------
+export const VoiceAgent = {
+    /**
+     * Create a standalone VoiceAgent session.
+     * The agent manages its own STT, LLM, TTS, and VAD components.
+     */
+    async create() {
+        if (!RunAnywhere.isInitialized) {
+            throw SDKError.notInitialized();
+        }
+        const bridge = WASMBridge.shared;
+        const m = bridge.module;
+        const handlePtr = m._malloc(4);
+        const result = bridge.callFunction('rac_voice_agent_create_standalone', 'number', ['number'], [handlePtr]);
+        if (result !== 0) {
+            m._free(handlePtr);
+            throw SDKError.fromRACResult(result, 'rac_voice_agent_create_standalone');
+        }
+        const handle = m.getValue(handlePtr, 'i32');
+        m._free(handlePtr);
+        logger.info('VoiceAgent session created');
+        return new VoiceAgentSession(handle);
+    },
+};
+//# sourceMappingURL=RunAnywhere+VoiceAgent.js.map

package/dist/Public/Extensions/RunAnywhere+VoiceAgent.js.map

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package/dist/Public/Extensions/RunAnywhere+VoicePipeline.d.ts

@@ -0,0 +1,63 @@
+/**
+ * RunAnywhere Web SDK - VoicePipeline Extension
+ *
+ * High-level streaming voice orchestrator: STT -> LLM (streaming) -> TTS.
+ *
+ * Unlike VoiceAgent (which uses the C API for a batch pipeline), VoicePipeline
+ * composes the existing TypeScript SDK extensions (STT, TextGeneration, TTS)
+ * and provides streaming callbacks so the UI can update in real-time as each
+ * stage progresses.
+ *
+ * Usage:
+ *   ```typescript
+ *   import { VoicePipeline } from '@runanywhere/web';
+ *
+ *   const pipeline = new VoicePipeline();
+ *   const result = await pipeline.processTurn(audioData, {
+ *     maxTokens: 150,
+ *     systemPrompt: 'You are a helpful voice assistant.',
+ *   }, {
+ *     onTranscription: (text) => updateUI('You said: ' + text),
+ *     onResponseToken: (_tok, acc) => updateUI('Assistant: ' + acc),
+ *     onSynthesisComplete: (audio, sr) => playAudio(audio, sr),
+ *   });
+ *   ```
+ *
+ * The pipeline does NOT handle audio capture, VAD, or playback — those are
+ * app-level concerns (mic control, VAD thresholds, speaker output).
+ */
+import { PipelineState } from './VoiceAgentTypes';
+import type { VoicePipelineCallbacks, VoicePipelineOptions, VoicePipelineTurnResult } from './VoicePipelineTypes';
+export { PipelineState } from './VoiceAgentTypes';
+export type { VoicePipelineCallbacks, VoicePipelineOptions, VoicePipelineTurnResult, } from './VoicePipelineTypes';
+/**
+ * Streaming voice pipeline that orchestrates STT -> LLM -> TTS.
+ *
+ * Create one instance and reuse it across conversation turns. Each call to
+ * `processTurn()` runs the full pipeline once and returns the result.
+ *
+ * The LLM step uses streaming generation, so the `onResponseToken` callback
+ * fires for each token — enabling real-time UI updates.
+ */
+export declare class VoicePipeline {
+    private _cancelGeneration;
+    private _state;
+    /** Current pipeline state. */
+    get state(): PipelineState;
+    /**
+     * Run a complete voice turn: audio in -> transcription -> LLM response -> TTS audio.
+     *
+     * @param audioData  Float32Array of PCM samples (mono, typically 16 kHz).
+     * @param options    LLM / TTS / STT configuration for this turn.
+     * @param callbacks  Streaming callbacks for real-time UI updates.
+     * @returns          The full turn result once all stages complete.
+     */
+    processTurn(audioData: Float32Array, options?: VoicePipelineOptions, callbacks?: VoicePipelineCallbacks): Promise<VoicePipelineTurnResult>;
+    /**
+     * Cancel in-progress LLM generation.
+     * Safe to call at any time — no-ops if nothing is in progress.
+     */
+    cancel(): void;
+    private transition;
+}
+//# sourceMappingURL=RunAnywhere+VoicePipeline.d.ts.map

package/dist/Public/Extensions/RunAnywhere+VoicePipeline.d.ts.map

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package/dist/Public/Extensions/RunAnywhere+VoicePipeline.js

@@ -0,0 +1,168 @@
+/**
+ * RunAnywhere Web SDK - VoicePipeline Extension
+ *
+ * High-level streaming voice orchestrator: STT -> LLM (streaming) -> TTS.
+ *
+ * Unlike VoiceAgent (which uses the C API for a batch pipeline), VoicePipeline
+ * composes the existing TypeScript SDK extensions (STT, TextGeneration, TTS)
+ * and provides streaming callbacks so the UI can update in real-time as each
+ * stage progresses.
+ *
+ * Usage:
+ *   ```typescript
+ *   import { VoicePipeline } from '@runanywhere/web';
+ *
+ *   const pipeline = new VoicePipeline();
+ *   const result = await pipeline.processTurn(audioData, {
+ *     maxTokens: 150,
+ *     systemPrompt: 'You are a helpful voice assistant.',
+ *   }, {
+ *     onTranscription: (text) => updateUI('You said: ' + text),
+ *     onResponseToken: (_tok, acc) => updateUI('Assistant: ' + acc),
+ *     onSynthesisComplete: (audio, sr) => playAudio(audio, sr),
+ *   });
+ *   ```
+ *
+ * The pipeline does NOT handle audio capture, VAD, or playback — those are
+ * app-level concerns (mic control, VAD thresholds, speaker output).
+ */
+import { STT } from './RunAnywhere+STT';
+import { TextGeneration } from './RunAnywhere+TextGeneration';
+import { TTS } from './RunAnywhere+TTS';
+import { SDKLogger } from '../../Foundation/SDKLogger';
+import { PipelineState } from './VoiceAgentTypes';
+export { PipelineState } from './VoiceAgentTypes';
+const logger = new SDKLogger('VoicePipeline');
+// ---------------------------------------------------------------------------
+// Default options
+// ---------------------------------------------------------------------------
+const DEFAULT_OPTIONS = {
+    maxTokens: 150,
+    temperature: 0.7,
+    systemPrompt: 'You are a helpful voice assistant. Keep responses concise — 1-3 sentences. Be conversational and friendly.',
+    ttsSpeed: 1.0,
+    sampleRate: 16000,
+};
+// ---------------------------------------------------------------------------
+// VoicePipeline
+// ---------------------------------------------------------------------------
+/**
+ * Streaming voice pipeline that orchestrates STT -> LLM -> TTS.
+ *
+ * Create one instance and reuse it across conversation turns. Each call to
+ * `processTurn()` runs the full pipeline once and returns the result.
+ *
+ * The LLM step uses streaming generation, so the `onResponseToken` callback
+ * fires for each token — enabling real-time UI updates.
+ */
+export class VoicePipeline {
+    _cancelGeneration = null;
+    _state = PipelineState.Idle;
+    /** Current pipeline state. */
+    get state() {
+        return this._state;
+    }
+    /**
+     * Run a complete voice turn: audio in -> transcription -> LLM response -> TTS audio.
+     *
+     * @param audioData  Float32Array of PCM samples (mono, typically 16 kHz).
+     * @param options    LLM / TTS / STT configuration for this turn.
+     * @param callbacks  Streaming callbacks for real-time UI updates.
+     * @returns          The full turn result once all stages complete.
+     */
+    async processTurn(audioData, options, callbacks) {
+        const opts = { ...DEFAULT_OPTIONS, ...options };
+        const totalStart = performance.now();
+        // ── Step 1: STT ────────────────────────────────────────────────────
+        this.transition(PipelineState.ProcessingSTT, callbacks);
+        const sttStart = performance.now();
+        logger.info(`STT: ${(audioData.length / opts.sampleRate).toFixed(1)}s of audio`);
+        const sttResult = await STT.transcribe(audioData, {
+            sampleRate: opts.sampleRate,
+        });
+        const sttMs = performance.now() - sttStart;
+        const userText = sttResult.text.trim();
+        logger.info(`STT complete: "${userText}" (${sttMs.toFixed(0)}ms)`);
+        callbacks?.onTranscription?.(userText, sttResult);
+        if (!userText) {
+            this.transition(PipelineState.Idle, callbacks);
+            return {
+                transcription: '',
+                response: '',
+                timing: { sttMs, llmMs: 0, ttsMs: 0, totalMs: performance.now() - totalStart },
+            };
+        }
+        // ── Step 2: LLM (streaming) ────────────────────────────────────────
+        this.transition(PipelineState.GeneratingResponse, callbacks);
+        const llmStart = performance.now();
+        const { stream, result: llmResultPromise, cancel } = await TextGeneration.generateStream(userText, {
+            maxTokens: opts.maxTokens,
+            temperature: opts.temperature,
+            systemPrompt: opts.systemPrompt,
+        });
+        this._cancelGeneration = cancel;
+        let accumulated = '';
+        for await (const token of stream) {
+            accumulated += token;
+            callbacks?.onResponseToken?.(token, accumulated);
+        }
+        this._cancelGeneration = null;
+        const llmResult = await llmResultPromise;
+        const fullResponse = llmResult.text || accumulated;
+        const llmMs = performance.now() - llmStart;
+        logger.info(`LLM complete: ${llmResult.tokensUsed} tokens, ${llmResult.tokensPerSecond.toFixed(1)} tok/s (${llmMs.toFixed(0)}ms)`);
+        callbacks?.onResponseComplete?.(fullResponse, llmResult);
+        if (!fullResponse.trim()) {
+            this.transition(PipelineState.Idle, callbacks);
+            return {
+                transcription: userText,
+                response: '',
+                timing: { sttMs, llmMs, ttsMs: 0, totalMs: performance.now() - totalStart },
+                llmResult,
+            };
+        }
+        // ── Step 3: TTS ────────────────────────────────────────────────────
+        this.transition(PipelineState.PlayingTTS, callbacks);
+        const ttsStart = performance.now();
+        const ttsResult = await TTS.synthesize(fullResponse.trim(), {
+            speed: opts.ttsSpeed,
+        });
+        const ttsMs = performance.now() - ttsStart;
+        logger.info(`TTS complete: ${ttsResult.durationMs}ms audio in ${ttsResult.processingTimeMs}ms`);
+        callbacks?.onSynthesisComplete?.(ttsResult.audioData, ttsResult.sampleRate, ttsResult);
+        // ── Done ───────────────────────────────────────────────────────────
+        this.transition(PipelineState.Idle, callbacks);
+        return {
+            transcription: userText,
+            response: fullResponse,
+            synthesizedAudio: ttsResult.audioData,
+            sampleRate: ttsResult.sampleRate,
+            timing: {
+                sttMs,
+                llmMs,
+                ttsMs,
+                totalMs: performance.now() - totalStart,
+            },
+            llmResult,
+        };
+    }
+    /**
+     * Cancel in-progress LLM generation.
+     * Safe to call at any time — no-ops if nothing is in progress.
+     */
+    cancel() {
+        if (this._cancelGeneration) {
+            this._cancelGeneration();
+            this._cancelGeneration = null;
+            logger.info('Generation cancelled');
+        }
+    }
+    // ---------------------------------------------------------------------------
+    // Internals
+    // ---------------------------------------------------------------------------
+    transition(newState, callbacks) {
+        this._state = newState;
+        callbacks?.onStateChange?.(newState);
+    }
+}
+//# sourceMappingURL=RunAnywhere+VoicePipeline.js.map

package/dist/Public/Extensions/RunAnywhere+VoicePipeline.js.map

@@ -0,0 +1 @@
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package/dist/Public/Extensions/STTTypes.d.ts

@@ -0,0 +1,53 @@
+/**
+ * RunAnywhere Web SDK - Speech-to-Text Types
+ *
+ * Type definitions for STT models and transcription results.
+ * Separated from the main STT extension for clean imports.
+ */
+export declare enum STTModelType {
+    Whisper = "whisper",
+    Zipformer = "zipformer",
+    Paraformer = "paraformer"
+}
+export interface STTModelConfig {
+    modelId: string;
+    type: STTModelType;
+    /**
+     * Model files already written to sherpa-onnx virtual FS.
+     * Paths are FS paths (e.g., '/models/whisper-tiny/encoder.onnx').
+     */
+    modelFiles: STTWhisperFiles | STTZipformerFiles | STTParaformerFiles;
+    /** Sample rate (default: 16000) */
+    sampleRate?: number;
+    /** Language code (e.g., 'en', 'zh') */
+    language?: string;
+}
+export interface STTWhisperFiles {
+    encoder: string;
+    decoder: string;
+    tokens: string;
+}
+export interface STTZipformerFiles {
+    encoder: string;
+    decoder: string;
+    joiner: string;
+    tokens: string;
+}
+export interface STTParaformerFiles {
+    model: string;
+    tokens: string;
+}
+export interface STTTranscriptionResult {
+    text: string;
+    confidence: number;
+    detectedLanguage?: string;
+    processingTimeMs: number;
+    words?: STTWord[];
+}
+export interface STTWord {
+    text: string;
+    startMs: number;
+    endMs: number;
+    confidence: number;
+}
+//# sourceMappingURL=STTTypes.d.ts.map

package/dist/Public/Extensions/STTTypes.d.ts.map

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package/dist/Public/Extensions/STTTypes.js

@@ -0,0 +1,16 @@
+/**
+ * RunAnywhere Web SDK - Speech-to-Text Types
+ *
+ * Type definitions for STT models and transcription results.
+ * Separated from the main STT extension for clean imports.
+ */
+// ---------------------------------------------------------------------------
+// STT Types
+// ---------------------------------------------------------------------------
+export var STTModelType;
+(function (STTModelType) {
+    STTModelType["Whisper"] = "whisper";
+    STTModelType["Zipformer"] = "zipformer";
+    STTModelType["Paraformer"] = "paraformer";
+})(STTModelType || (STTModelType = {}));
+//# sourceMappingURL=STTTypes.js.map

package/dist/Public/Extensions/STTTypes.js.map

@@ -0,0 +1 @@
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package/dist/Public/Extensions/TTSTypes.d.ts

@@ -0,0 +1,31 @@
+/** RunAnywhere Web SDK - TTS Types */
+export interface TTSVoiceConfig {
+    voiceId: string;
+    /** Path to the VITS/Piper model ONNX file in sherpa FS */
+    modelPath: string;
+    /** Path to the tokens.txt file in sherpa FS */
+    tokensPath: string;
+    /** Path to the espeak-ng-data directory in sherpa FS (for Piper models) */
+    dataDir?: string;
+    /** Path to the lexicon file in sherpa FS (optional) */
+    lexicon?: string;
+    /** Number of threads (default: 1) */
+    numThreads?: number;
+}
+export interface TTSSynthesisResult {
+    /** Raw PCM audio data */
+    audioData: Float32Array;
+    /** Audio sample rate */
+    sampleRate: number;
+    /** Duration in milliseconds */
+    durationMs: number;
+    /** Processing time in milliseconds */
+    processingTimeMs: number;
+}
+export interface TTSSynthesizeOptions {
+    /** Speaker ID for multi-speaker models (default: 0) */
+    speakerId?: number;
+    /** Speed factor (default: 1.0, >1 = faster, <1 = slower) */
+    speed?: number;
+}
+//# sourceMappingURL=TTSTypes.d.ts.map

package/dist/Public/Extensions/TTSTypes.d.ts.map

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package/dist/Public/Extensions/TTSTypes.js

@@ -0,0 +1,3 @@
+/** RunAnywhere Web SDK - TTS Types */
+export {};
+//# sourceMappingURL=TTSTypes.js.map

package/dist/Public/Extensions/TTSTypes.js.map

@@ -0,0 +1 @@
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