dvgateway-adapters 1.3.9 → 1.4.1

package/dist/realtime/gemini-live.d.ts

@@ -0,0 +1,196 @@
+/**
+ * Google Gemini Live API — Speech-to-Speech Adapter
+ *
+ * Implements a direct speech-to-speech pipeline using Google's Gemini Live API
+ * (BidiGenerateContent). This adapter is a drop-in alternative to
+ * OpenAIRealtimeAdapter — same RealtimeSpeechAdapter interface, same callbacks,
+ * same pipelineType="s2s" dashboard integration.
+ *
+ * Architecture:
+ *
+ *   DVGateway audio in (16kHz PCM)
+ *       ↓ (Gemini Live native input rate — no resampling needed)
+ *   Gemini Live API (WebSocket)
+ *       ↓ gemini-live-2.5-flash-preview processes speech end-to-end
+ *   DVGateway audio out (24kHz → 16kHz downsample)
+ *       ↓
+ *   TTS injector → call audio
+ *
+ * Supported endpoints (choose via `endpoint` option):
+ * - "ai-studio" (default): AI Studio global endpoint
+ *     wss://generativelanguage.googleapis.com/ws/.../BidiGenerateContent?key=KEY
+ * - "vertex": Vertex AI regional endpoint
+ *     wss://{LOCATION}-aiplatform.googleapis.com/ws/.../BidiGenerateContent
+ *     (requires Bearer access token via `accessToken` option)
+ *
+ * Audio format:
+ *   Input:  PCM16 16kHz mono, base64-encoded, MIME "audio/pcm;rate=16000"
+ *   Output: PCM16 24kHz mono, base64-encoded (downsampled to 16kHz for DVGateway)
+ *
+ * Key differences vs OpenAIRealtimeAdapter:
+ * - Input rate matches DVGateway natively (no upsample)
+ * - Message schema is BidiGenerateContent proto-JSON, not OpenAI events
+ * - Server-side VAD signals are implicit — we infer speech activity from
+ *   `interrupted` / `turnComplete` server messages (best-effort). For precise
+ *   VAD, set `turnDetection.mode = "none"` and call `sendActivityStart()` /
+ *   `sendActivityEnd()` manually.
+ *
+ * API Reference:
+ *   https://ai.google.dev/gemini-api/docs/live-api
+ *   https://ai.google.dev/api/live
+ */
+import type { AudioChunk, TranscriptResult } from 'dvgateway-sdk';
+import type { RealtimeSpeechAdapter, RealtimeSpeechActivityEvent, RealtimeToolCall } from './openai-realtime.js';
+/**
+ * Prebuilt voice names supported by Gemini Live (half-cascade models).
+ * Native-audio models support more voices (Google Cloud TTS full catalog),
+ * but these 8 are the portable defaults available on both paths.
+ * Reference: https://ai.google.dev/gemini-api/docs/live-api/capabilities
+ */
+export type GeminiLiveVoice = 'Puck' | 'Charon' | 'Kore' | 'Fenrir' | 'Aoede' | 'Leda' | 'Orus' | 'Zephyr';
+/** Gemini Live API model identifiers (as of 2026-04). */
+export type GeminiLiveModel = 'gemini-live-2.5-flash-preview' | 'gemini-2.5-flash-native-audio' | 'gemini-2.0-flash-live-001';
+export type GeminiLiveEndpoint = 'ai-studio' | 'vertex';
+export type GeminiLiveTurnDetectionMode = 'server_vad' | 'none';
+export interface GeminiLiveTurnDetectionOptions {
+    /**
+     * Turn detection strategy.
+     * - "server_vad" (default): Gemini's server-side VAD auto-detects speech
+     *   boundaries and barge-in. No manual activityStart/End required.
+     * - "none": client is responsible — set automaticActivityDetection.disabled
+     *   and call sendActivityStart() / sendActivityEnd() yourself.
+     */
+    mode?: GeminiLiveTurnDetectionMode;
+    /** VAD start-of-speech sensitivity (LOW/MEDIUM/HIGH). Default: MEDIUM. */
+    startSensitivity?: 'LOW' | 'MEDIUM' | 'HIGH';
+    /** VAD end-of-speech sensitivity (LOW/MEDIUM/HIGH). Default: MEDIUM. */
+    endSensitivity?: 'LOW' | 'MEDIUM' | 'HIGH';
+    /** Prefix padding in ms applied before detected speech start. Default: 200. */
+    prefixPaddingMs?: number;
+    /** Silence duration in ms before turn is considered ended. Default: 800. */
+    silenceDurationMs?: number;
+}
+/** Function/tool declaration for Gemini Live (same schema as Gemini Generative API). */
+export interface GeminiLiveTool {
+    name: string;
+    description?: string;
+    /** JSON Schema describing the function parameters. */
+    parameters?: Record<string, unknown>;
+}
+export interface GeminiLiveAdapterOptions {
+    /** API key for AI Studio endpoint (required if endpoint="ai-studio"). */
+    apiKey?: string;
+    /** Bearer access token for Vertex endpoint (required if endpoint="vertex"). */
+    accessToken?: string;
+    /** Vertex AI project ID (required if endpoint="vertex"). */
+    project?: string;
+    /** Vertex AI location, e.g. "asia-northeast1" (required if endpoint="vertex"). */
+    location?: string;
+    /**
+     * Which Gemini Live endpoint to use.
+     * - "ai-studio" (default): global AI Studio, API key auth.
+     * - "vertex": regional Vertex AI, Bearer token auth, project/location required.
+     */
+    endpoint?: GeminiLiveEndpoint;
+    /**
+     * Model (default: "gemini-live-2.5-flash-preview").
+     * For Vertex AI this will be normalized to the full resource path internally.
+     */
+    model?: GeminiLiveModel;
+    /** Voice for the AI response (default: "Puck"). */
+    voice?: GeminiLiveVoice;
+    /**
+     * System instructions. Equivalent to OpenAI's `instructions` field.
+     * Default: generic voice assistant prompt.
+     */
+    instructions?: string;
+    /** Turn detection configuration — see GeminiLiveTurnDetectionOptions. */
+    turnDetection?: GeminiLiveTurnDetectionOptions;
+    /**
+     * Enable input (user) audio transcription (default: true).
+     * Emits TranscriptResult events with speaker="customer".
+     */
+    inputTranscription?: boolean;
+    /**
+     * Enable output (assistant) audio transcription (default: true).
+     * Emits TranscriptResult events with speaker="agent".
+     */
+    outputTranscription?: boolean;
+    /**
+     * BCP-47 language hint for transcription (default: empty — auto-detect).
+     * E.g. "ko-KR", "en-US". Gemini is strong on multilingual inference but
+     * a hint improves accuracy for short utterances.
+     */
+    language?: string;
+    /** Response temperature (default: 0.8). Gemini accepts 0.0–2.0. */
+    temperature?: number;
+    /** Maximum response output tokens. Default: no cap. */
+    maxOutputTokens?: number;
+    /** Tools the model can call during the session. */
+    tools?: GeminiLiveTool[];
+}
+export declare class GeminiLiveAdapter implements RealtimeSpeechAdapter {
+    private readonly opts;
+    private readonly tools;
+    private audioOutputHandler;
+    private transcriptHandler;
+    private errorHandler;
+    private toolCallHandler;
+    private speechActivityHandler;
+    private readonly sessions;
+    /** Per-session input transcript accumulator (Gemini streams text incrementally) */
+    private readonly inputTranscriptBuf;
+    /** Per-session output transcript accumulator */
+    private readonly outputTranscriptBuf;
+    /** Per-session "caller speaking" state (for synthetic onSpeechActivity events) */
+    private readonly callerSpeaking;
+    private stopped;
+    constructor(opts: GeminiLiveAdapterOptions);
+    onAudioOutput(handler: (chunk: Buffer, linkedId: string) => void): void;
+    onTranscript(handler: (result: TranscriptResult) => void): void;
+    onError(handler: (err: Error, linkedId?: string) => void): void;
+    onToolCall(handler: (toolCall: RealtimeToolCall) => void): void;
+    /**
+     * Register a handler for speech activity events.
+     *
+     * Gemini Live does NOT emit explicit speech_started/stopped events like
+     * OpenAI Realtime. This adapter approximates speech activity from the
+     * following server signals:
+     *
+     * - `serverContent.interrupted=true` → caller started speaking while AI was
+     *   responding (barge-in). Fires `{ side: "caller", speaking: true }`.
+     * - `serverContent.turnComplete=true` → turn boundary reached. Fires
+     *   `{ side: "caller", speaking: false }` if the caller was marked speaking.
+     *
+     * For precise VAD, set `turnDetection.mode = "none"` and call
+     * `sendActivityStart()` / `sendActivityEnd()` manually from your own VAD
+     * logic — those methods fire this callback synchronously with the server message.
+     */
+    onSpeechActivity(handler: (event: RealtimeSpeechActivityEvent) => void): void;
+    /**
+     * Submit a tool call result back to the model.
+     * Call this after receiving a tool call via onToolCall().
+     */
+    submitToolResult(linkedId: string, callId: string, result: unknown): void;
+    /**
+     * Mark the start of caller speech (manual VAD mode only).
+     * Sends a `realtime_input.activity_start` message to Gemini and fires the
+     * onSpeechActivity callback. No-op if turnDetection.mode !== "none".
+     */
+    sendActivityStart(linkedId: string): void;
+    /**
+     * Mark the end of caller speech (manual VAD mode only).
+     * Sends a `realtime_input.activity_end` message to Gemini and fires the
+     * onSpeechActivity callback. No-op if turnDetection.mode !== "none".
+     */
+    sendActivityEnd(linkedId: string): void;
+    startSession(linkedId: string, audioIn: AsyncIterable<AudioChunk>): Promise<void>;
+    stop(linkedId?: string): Promise<void>;
+    private resolveModelPath;
+    private buildRealtimeInputConfig;
+    private connectLive;
+    private pipeAudioIn;
+    private handleServerMessage;
+    private sendMessage;
+}
+//# sourceMappingURL=gemini-live.d.ts.map

package/dist/realtime/gemini-live.d.ts.map

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package/dist/realtime/gemini-live.js

@@ -0,0 +1,461 @@
+/**
+ * Google Gemini Live API — Speech-to-Speech Adapter
+ *
+ * Implements a direct speech-to-speech pipeline using Google's Gemini Live API
+ * (BidiGenerateContent). This adapter is a drop-in alternative to
+ * OpenAIRealtimeAdapter — same RealtimeSpeechAdapter interface, same callbacks,
+ * same pipelineType="s2s" dashboard integration.
+ *
+ * Architecture:
+ *
+ *   DVGateway audio in (16kHz PCM)
+ *       ↓ (Gemini Live native input rate — no resampling needed)
+ *   Gemini Live API (WebSocket)
+ *       ↓ gemini-live-2.5-flash-preview processes speech end-to-end
+ *   DVGateway audio out (24kHz → 16kHz downsample)
+ *       ↓
+ *   TTS injector → call audio
+ *
+ * Supported endpoints (choose via `endpoint` option):
+ * - "ai-studio" (default): AI Studio global endpoint
+ *     wss://generativelanguage.googleapis.com/ws/.../BidiGenerateContent?key=KEY
+ * - "vertex": Vertex AI regional endpoint
+ *     wss://{LOCATION}-aiplatform.googleapis.com/ws/.../BidiGenerateContent
+ *     (requires Bearer access token via `accessToken` option)
+ *
+ * Audio format:
+ *   Input:  PCM16 16kHz mono, base64-encoded, MIME "audio/pcm;rate=16000"
+ *   Output: PCM16 24kHz mono, base64-encoded (downsampled to 16kHz for DVGateway)
+ *
+ * Key differences vs OpenAIRealtimeAdapter:
+ * - Input rate matches DVGateway natively (no upsample)
+ * - Message schema is BidiGenerateContent proto-JSON, not OpenAI events
+ * - Server-side VAD signals are implicit — we infer speech activity from
+ *   `interrupted` / `turnComplete` server messages (best-effort). For precise
+ *   VAD, set `turnDetection.mode = "none"` and call `sendActivityStart()` /
+ *   `sendActivityEnd()` manually.
+ *
+ * API Reference:
+ *   https://ai.google.dev/gemini-api/docs/live-api
+ *   https://ai.google.dev/api/live
+ */
+import WebSocket from 'ws';
+import { resample, float32ToSlin16, slin16ToFloat32 } from 'dvgateway-sdk';
+// ─── Implementation ──────────────────────────────────────────────────────────
+const AI_STUDIO_URL = 'wss://generativelanguage.googleapis.com/ws/google.ai.generativelanguage.v1beta.GenerativeService.BidiGenerateContent';
+const GEMINI_OUTPUT_SAMPLE_RATE = 24000;
+const DV_SAMPLE_RATE = 16000;
+const INPUT_CHUNK_MS = 20;
+const INPUT_SAMPLES_16K = Math.round(DV_SAMPLE_RATE * INPUT_CHUNK_MS / 1000); // 320 samples
+export class GeminiLiveAdapter {
+    opts;
+    tools;
+    audioOutputHandler = null;
+    transcriptHandler = null;
+    errorHandler = null;
+    toolCallHandler = null;
+    speechActivityHandler = null;
+    sessions = new Map();
+    /** Per-session input transcript accumulator (Gemini streams text incrementally) */
+    inputTranscriptBuf = new Map();
+    /** Per-session output transcript accumulator */
+    outputTranscriptBuf = new Map();
+    /** Per-session "caller speaking" state (for synthetic onSpeechActivity events) */
+    callerSpeaking = new Map();
+    stopped = false;
+    constructor(opts) {
+        const endpoint = opts.endpoint ?? 'ai-studio';
+        if (endpoint === 'ai-studio' && !opts.apiKey) {
+            throw new Error('GeminiLiveAdapter: apiKey is required when endpoint="ai-studio"');
+        }
+        if (endpoint === 'vertex' && (!opts.accessToken || !opts.project || !opts.location)) {
+            throw new Error('GeminiLiveAdapter: accessToken, project, and location are required when endpoint="vertex"');
+        }
+        this.opts = {
+            endpoint,
+            apiKey: opts.apiKey ?? '',
+            accessToken: opts.accessToken ?? '',
+            project: opts.project ?? '',
+            location: opts.location ?? '',
+            model: opts.model ?? 'gemini-live-2.5-flash-preview',
+            voice: opts.voice ?? 'Puck',
+            instructions: opts.instructions ??
+                'You are a helpful voice assistant. Keep answers concise and conversational.',
+            turnDetection: opts.turnDetection ?? {},
+            inputTranscription: opts.inputTranscription ?? true,
+            outputTranscription: opts.outputTranscription ?? true,
+            language: opts.language ?? '',
+            temperature: opts.temperature ?? 0.8,
+            maxOutputTokens: opts.maxOutputTokens ?? null,
+        };
+        this.tools = opts.tools ?? [];
+    }
+    onAudioOutput(handler) {
+        this.audioOutputHandler = handler;
+    }
+    onTranscript(handler) {
+        this.transcriptHandler = handler;
+    }
+    onError(handler) {
+        this.errorHandler = handler;
+    }
+    onToolCall(handler) {
+        this.toolCallHandler = handler;
+    }
+    /**
+     * Register a handler for speech activity events.
+     *
+     * Gemini Live does NOT emit explicit speech_started/stopped events like
+     * OpenAI Realtime. This adapter approximates speech activity from the
+     * following server signals:
+     *
+     * - `serverContent.interrupted=true` → caller started speaking while AI was
+     *   responding (barge-in). Fires `{ side: "caller", speaking: true }`.
+     * - `serverContent.turnComplete=true` → turn boundary reached. Fires
+     *   `{ side: "caller", speaking: false }` if the caller was marked speaking.
+     *
+     * For precise VAD, set `turnDetection.mode = "none"` and call
+     * `sendActivityStart()` / `sendActivityEnd()` manually from your own VAD
+     * logic — those methods fire this callback synchronously with the server message.
+     */
+    onSpeechActivity(handler) {
+        this.speechActivityHandler = handler;
+    }
+    /**
+     * Submit a tool call result back to the model.
+     * Call this after receiving a tool call via onToolCall().
+     */
+    submitToolResult(linkedId, callId, result) {
+        const ws = this.sessions.get(linkedId);
+        if (!ws || ws.readyState !== WebSocket.OPEN)
+            return;
+        // Gemini tool_response schema: functionResponses[].response is a struct
+        const response = typeof result === 'object' && result !== null
+            ? result
+            : { result };
+        this.sendMessage(ws, {
+            tool_response: {
+                function_responses: [{ id: callId, name: '', response }],
+            },
+        });
+    }
+    /**
+     * Mark the start of caller speech (manual VAD mode only).
+     * Sends a `realtime_input.activity_start` message to Gemini and fires the
+     * onSpeechActivity callback. No-op if turnDetection.mode !== "none".
+     */
+    sendActivityStart(linkedId) {
+        const ws = this.sessions.get(linkedId);
+        if (!ws || ws.readyState !== WebSocket.OPEN)
+            return;
+        if (this.opts.turnDetection.mode !== 'none')
+            return;
+        this.sendMessage(ws, { realtime_input: { activity_start: {} } });
+        this.callerSpeaking.set(linkedId, true);
+        this.speechActivityHandler?.({
+            linkedId,
+            side: 'caller',
+            speaking: true,
+            timestampMs: Date.now(),
+        });
+    }
+    /**
+     * Mark the end of caller speech (manual VAD mode only).
+     * Sends a `realtime_input.activity_end` message to Gemini and fires the
+     * onSpeechActivity callback. No-op if turnDetection.mode !== "none".
+     */
+    sendActivityEnd(linkedId) {
+        const ws = this.sessions.get(linkedId);
+        if (!ws || ws.readyState !== WebSocket.OPEN)
+            return;
+        if (this.opts.turnDetection.mode !== 'none')
+            return;
+        this.sendMessage(ws, { realtime_input: { activity_end: {} } });
+        this.callerSpeaking.set(linkedId, false);
+        this.speechActivityHandler?.({
+            linkedId,
+            side: 'caller',
+            speaking: false,
+            timestampMs: Date.now(),
+        });
+    }
+    async startSession(linkedId, audioIn) {
+        this.stopped = false;
+        const ws = await this.connectLive(linkedId);
+        this.sessions.set(linkedId, ws);
+        this.inputTranscriptBuf.set(linkedId, '');
+        this.outputTranscriptBuf.set(linkedId, '');
+        this.callerSpeaking.set(linkedId, false);
+        // ── Send initial setup message ──
+        const setupMsg = {
+            setup: {
+                model: this.resolveModelPath(),
+                generation_config: {
+                    response_modalities: ['AUDIO'],
+                    speech_config: {
+                        voice_config: {
+                            prebuilt_voice_config: { voice_name: this.opts.voice },
+                        },
+                        ...(this.opts.language && { language_code: this.opts.language }),
+                    },
+                    temperature: this.opts.temperature,
+                    ...(this.opts.maxOutputTokens !== null && {
+                        max_output_tokens: this.opts.maxOutputTokens,
+                    }),
+                },
+                system_instruction: {
+                    parts: [{ text: this.opts.instructions }],
+                },
+                realtime_input_config: this.buildRealtimeInputConfig(),
+                ...(this.opts.inputTranscription && { input_audio_transcription: {} }),
+                ...(this.opts.outputTranscription && { output_audio_transcription: {} }),
+                ...(this.tools.length > 0 && {
+                    tools: [
+                        {
+                            function_declarations: this.tools.map((t) => ({
+                                name: t.name,
+                                description: t.description,
+                                ...(t.parameters && { parameters: t.parameters }),
+                            })),
+                        },
+                    ],
+                }),
+            },
+        };
+        this.sendMessage(ws, setupMsg);
+        // Pipe audio in background
+        void this.pipeAudioIn(linkedId, ws, audioIn);
+        // Wait until WS closes or stop() is called
+        await new Promise((resolve) => {
+            ws.once('close', resolve);
+            const check = setInterval(() => {
+                if (this.stopped || !this.sessions.has(linkedId)) {
+                    clearInterval(check);
+                    resolve();
+                }
+            }, 100);
+        });
+    }
+    async stop(linkedId) {
+        if (linkedId) {
+            const ws = this.sessions.get(linkedId);
+            if (ws && ws.readyState === WebSocket.OPEN) {
+                ws.close(1000, 'session ended');
+            }
+            this.sessions.delete(linkedId);
+            this.inputTranscriptBuf.delete(linkedId);
+            this.outputTranscriptBuf.delete(linkedId);
+            this.callerSpeaking.delete(linkedId);
+        }
+        else {
+            this.stopped = true;
+            for (const [id, ws] of this.sessions) {
+                if (ws.readyState === WebSocket.OPEN) {
+                    ws.close(1000, 'adapter stopped');
+                }
+                this.sessions.delete(id);
+                this.inputTranscriptBuf.delete(id);
+                this.outputTranscriptBuf.delete(id);
+                this.callerSpeaking.delete(id);
+            }
+        }
+    }
+    // ── Private helpers ─────────────────────────────────────────────────────
+    resolveModelPath() {
+        if (this.opts.endpoint === 'vertex') {
+            return `projects/${this.opts.project}/locations/${this.opts.location}/publishers/google/models/${this.opts.model}`;
+        }
+        // AI Studio expects "models/MODEL_ID"
+        return `models/${this.opts.model}`;
+    }
+    buildRealtimeInputConfig() {
+        const td = this.opts.turnDetection;
+        if (td.mode === 'none') {
+            return {
+                automatic_activity_detection: { disabled: true },
+            };
+        }
+        const cfg = { automatic_activity_detection: { disabled: false } };
+        const vad = {};
+        if (td.startSensitivity)
+            vad.start_of_speech_sensitivity = `START_SENSITIVITY_${td.startSensitivity}`;
+        if (td.endSensitivity)
+            vad.end_of_speech_sensitivity = `END_SENSITIVITY_${td.endSensitivity}`;
+        if (td.prefixPaddingMs != null)
+            vad.prefix_padding_ms = td.prefixPaddingMs;
+        if (td.silenceDurationMs != null)
+            vad.silence_duration_ms = td.silenceDurationMs;
+        if (Object.keys(vad).length > 0) {
+            cfg.automatic_activity_detection = { disabled: false, ...vad };
+        }
+        return cfg;
+    }
+    async connectLive(linkedId) {
+        let url;
+        const headers = {};
+        if (this.opts.endpoint === 'vertex') {
+            url = `wss://${this.opts.location}-aiplatform.googleapis.com/ws/google.cloud.aiplatform.v1.LlmBidiService/BidiGenerateContent`;
+            headers['Authorization'] = `Bearer ${this.opts.accessToken}`;
+        }
+        else {
+            url = `${AI_STUDIO_URL}?key=${encodeURIComponent(this.opts.apiKey)}`;
+        }
+        const ws = new WebSocket(url, { headers });
+        ws.on('message', (data) => {
+            try {
+                const msg = JSON.parse(data.toString());
+                this.handleServerMessage(msg, linkedId);
+            }
+            catch {
+                // Ignore malformed JSON
+            }
+        });
+        ws.on('error', (err) => {
+            this.errorHandler?.(err, linkedId);
+        });
+        ws.on('close', () => {
+            this.sessions.delete(linkedId);
+        });
+        await new Promise((resolve, reject) => {
+            ws.once('open', resolve);
+            ws.once('error', reject);
+        });
+        return ws;
+    }
+    async pipeAudioIn(linkedId, ws, audioIn) {
+        for await (const chunk of audioIn) {
+            if (this.stopped || ws.readyState !== WebSocket.OPEN)
+                break;
+            // DVGateway is 16kHz; Gemini Live input is 16kHz → no resampling needed.
+            // (Keep the resample call guarded for forward compatibility if DV changes.)
+            const samples16k = chunk.sampleRate === DV_SAMPLE_RATE
+                ? chunk.samples
+                : resample(chunk.samples, chunk.sampleRate, DV_SAMPLE_RATE);
+            // Split into 20ms frames and send
+            let offset = 0;
+            while (offset < samples16k.length) {
+                const end = Math.min(offset + INPUT_SAMPLES_16K, samples16k.length);
+                const frame = samples16k.subarray(offset, end);
+                offset = end;
+                const pcm16 = float32ToSlin16(frame);
+                const b64 = Buffer.from(pcm16.buffer, pcm16.byteOffset, pcm16.byteLength).toString('base64');
+                this.sendMessage(ws, {
+                    realtime_input: {
+                        media_chunks: [
+                            { mime_type: 'audio/pcm;rate=16000', data: b64 },
+                        ],
+                    },
+                });
+            }
+        }
+    }
+    handleServerMessage(msg, linkedId) {
+        // ── setupComplete: ready to stream audio ──
+        if (msg.setupComplete) {
+            return;
+        }
+        // ── serverContent: main response channel ──
+        if (msg.serverContent) {
+            const sc = msg.serverContent;
+            // Audio output (modelTurn.parts[].inlineData)
+            if (sc.modelTurn?.parts) {
+                for (const part of sc.modelTurn.parts) {
+                    if (part.inlineData?.data) {
+                        // Gemini outputs PCM16 at 24kHz — downsample to DVGateway's 16kHz
+                        const pcm24kBuf = Buffer.from(part.inlineData.data, 'base64');
+                        const samples24k = slin16ToFloat32(pcm24kBuf);
+                        const samples16k = resample(samples24k, GEMINI_OUTPUT_SAMPLE_RATE, DV_SAMPLE_RATE);
+                        const pcm16k = float32ToSlin16(samples16k);
+                        this.audioOutputHandler?.(Buffer.from(pcm16k.buffer, pcm16k.byteOffset, pcm16k.byteLength), linkedId);
+                    }
+                    // Text parts are ignored in audio-only mode (still present for tool-call context)
+                }
+            }
+            // Interruption: caller barge-in detected by server-side VAD
+            if (sc.interrupted === true) {
+                if (!this.callerSpeaking.get(linkedId)) {
+                    this.callerSpeaking.set(linkedId, true);
+                    this.speechActivityHandler?.({
+                        linkedId,
+                        side: 'caller',
+                        speaking: true,
+                        timestampMs: Date.now(),
+                    });
+                }
+            }
+            // Turn complete: AI finished generating → if caller was marked speaking, clear it
+            if (sc.turnComplete === true || sc.generationComplete === true) {
+                if (this.callerSpeaking.get(linkedId)) {
+                    this.callerSpeaking.set(linkedId, false);
+                    this.speechActivityHandler?.({
+                        linkedId,
+                        side: 'caller',
+                        speaking: false,
+                        timestampMs: Date.now(),
+                    });
+                }
+            }
+            // Input transcription (caller speech → text, streamed incrementally)
+            if (sc.inputTranscription?.text != null) {
+                const prev = this.inputTranscriptBuf.get(linkedId) ?? '';
+                const next = prev + sc.inputTranscription.text;
+                this.inputTranscriptBuf.set(linkedId, next);
+                // Emit as finalized on turnComplete (below); for now just accumulate
+            }
+            // Output transcription (assistant speech → text, streamed incrementally)
+            if (sc.outputTranscription?.text != null) {
+                const prev = this.outputTranscriptBuf.get(linkedId) ?? '';
+                const next = prev + sc.outputTranscription.text;
+                this.outputTranscriptBuf.set(linkedId, next);
+            }
+            // On turn completion, flush accumulated transcripts as final results
+            if (sc.turnComplete === true) {
+                const inputText = this.inputTranscriptBuf.get(linkedId) ?? '';
+                if (inputText.trim()) {
+                    this.transcriptHandler?.({
+                        linkedId,
+                        text: inputText,
+                        isFinal: true,
+                        speaker: 'customer',
+                        timestampMs: Date.now(),
+                    });
+                }
+                const outputText = this.outputTranscriptBuf.get(linkedId) ?? '';
+                if (outputText.trim()) {
+                    this.transcriptHandler?.({
+                        linkedId,
+                        text: outputText,
+                        isFinal: true,
+                        speaker: 'agent',
+                        timestampMs: Date.now(),
+                    });
+                }
+                this.inputTranscriptBuf.set(linkedId, '');
+                this.outputTranscriptBuf.set(linkedId, '');
+            }
+        }
+        // ── toolCall: model wants to invoke a function ──
+        if (msg.toolCall?.functionCalls) {
+            for (const fc of msg.toolCall.functionCalls) {
+                if (fc.name) {
+                    this.toolCallHandler?.({
+                        callId: fc.id ?? '',
+                        name: fc.name,
+                        args: fc.args ?? {},
+                        linkedId,
+                    });
+                }
+            }
+        }
+        // ── goAway: session will close soon (quota / deployment boundary) ──
+        if (msg.goAway) {
+            this.errorHandler?.(new Error(`Gemini Live goAway: session closing in ${msg.goAway.timeLeft ?? 'unknown'}`), linkedId);
+        }
+    }
+    sendMessage(ws, obj) {
+        if (ws.readyState === WebSocket.OPEN) {
+            ws.send(JSON.stringify(obj));
+        }
+    }
+}
+//# sourceMappingURL=gemini-live.js.map

package/dist/realtime/gemini-live.js.map

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package/dist/realtime/index.d.ts

@@ -1,3 +1,5 @@
 export { OpenAIRealtimeAdapter } from './openai-realtime.js';
 export type { OpenAIRealtimeAdapterOptions, OpenAIRealtimeVoice, OpenAIRealtimeTurnDetectionMode, OpenAIRealtimeTurnDetectionOptions, OpenAIRealtimeTool, RealtimeToolCall, RealtimeSpeechActivityEvent, RealtimeSpeechAdapter, } from './openai-realtime.js';
+export { GeminiLiveAdapter } from './gemini-live.js';
+export type { GeminiLiveAdapterOptions, GeminiLiveVoice, GeminiLiveModel, GeminiLiveEndpoint, GeminiLiveTurnDetectionMode, GeminiLiveTurnDetectionOptions, GeminiLiveTool, } from './gemini-live.js';
 //# sourceMappingURL=index.d.ts.map

package/dist/realtime/index.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../../src/realtime/index.ts"],"names":[],"mappings":"AACA,OAAO,EAAE,qBAAqB,EAAE,MAAM,sBAAsB,CAAC;AAC7D,YAAY,EACV,4BAA4B,EAC5B,mBAAmB,EACnB,+BAA+B,EAC/B,kCAAkC,EAClC,kBAAkB,EAClB,gBAAgB,EAChB,2BAA2B,EAC3B,qBAAqB,GACtB,MAAM,sBAAsB,CAAC"}
+{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../../src/realtime/index.ts"],"names":[],"mappings":"AACA,OAAO,EAAE,qBAAqB,EAAE,MAAM,sBAAsB,CAAC;AAC7D,YAAY,EACV,4BAA4B,EAC5B,mBAAmB,EACnB,+BAA+B,EAC/B,kCAAkC,EAClC,kBAAkB,EAClB,gBAAgB,EAChB,2BAA2B,EAC3B,qBAAqB,GACtB,MAAM,sBAAsB,CAAC;AAE9B,OAAO,EAAE,iBAAiB,EAAE,MAAM,kBAAkB,CAAC;AACrD,YAAY,EACV,wBAAwB,EACxB,eAAe,EACf,eAAe,EACf,kBAAkB,EAClB,2BAA2B,EAC3B,8BAA8B,EAC9B,cAAc,GACf,MAAM,kBAAkB,CAAC"}

package/dist/realtime/index.js

@@ -1,3 +1,4 @@
 // Realtime speech-to-speech adapters
 export { OpenAIRealtimeAdapter } from './openai-realtime.js';
+export { GeminiLiveAdapter } from './gemini-live.js';
 //# sourceMappingURL=index.js.map

package/dist/realtime/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/realtime/index.ts"],"names":[],"mappings":"AAAA,qCAAqC;AACrC,OAAO,EAAE,qBAAqB,EAAE,MAAM,sBAAsB,CAAC"}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../src/realtime/index.ts"],"names":[],"mappings":"AAAA,qCAAqC;AACrC,OAAO,EAAE,qBAAqB,EAAE,MAAM,sBAAsB,CAAC;AAY7D,OAAO,EAAE,iBAAiB,EAAE,MAAM,kBAAkB,CAAC"}

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "dvgateway-adapters",
-  "version": "1.3.9",
+  "version": "1.4.1",
   "description": "AI service adapters for DVGateway SDK (Deepgram, ElevenLabs, Anthropic, OpenAI, OpenAI Realtime)",
   "type": "module",
   "main": "./dist/index.js",