@liforma/client 0.3.0 → 0.4.0

package/dist/openai/index.js

@@ -0,0 +1,282 @@
+import { MAX_PCM_CHUNK_BYTES as e, base64ToPcmBytes as t, createTurnSession as n, floatToPcm16Le as r, pcmBytesToBase64 as i, pcmBytesToBase64 as a, resampleFloat32 as o, shouldCaptureMic as s, startPcmMic as c, writePcmInChunks as l } from "@liforma/client/byo";
+//#region src/openai/connectOpenAiRealtime.ts
+var u = "gpt-realtime-2.1", d = "marin", f = 24e3;
+async function p(e, t) {
+	let { ephemeralKey: r, model: i = u, voice: o = d, instructions: l = "You are a helpful voice assistant. Keep replies short.", autoResponse: p = !0, captureMic: m = !0, mediaStream: h, onLog: g, onDisconnect: _, onError: v } = t, y = r.trim();
+	if (!y) throw Error("Pass ephemeralKey from your server mint route");
+	let b = (e, t = "info") => g?.(e, t), x = f, S = null, C = !1, w = !1, T = null, E = n(e, {
+		sampleRate: x,
+		isEnded: () => C,
+		onWriteError: (e) => v?.(e instanceof Error ? e.message : String(e))
+	});
+	function D(e) {
+		!S || S.readyState !== WebSocket.OPEN || S.send(JSON.stringify(e));
+	}
+	function O(e) {
+		let t;
+		try {
+			t = JSON.parse(e);
+		} catch {
+			return;
+		}
+		switch (t.type) {
+			case "session.created":
+				D({
+					type: "session.update",
+					session: {
+						type: "realtime",
+						model: i,
+						output_modalities: ["audio"],
+						instructions: l,
+						audio: {
+							input: {
+								format: {
+									type: "audio/pcm",
+									rate: x
+								},
+								turn_detection: { type: "semantic_vad" }
+							},
+							output: {
+								format: {
+									type: "audio/pcm",
+									rate: x
+								},
+								voice: o
+							}
+						}
+					}
+				});
+				break;
+			case "session.updated":
+				p && !w && (w = !0, D({ type: "response.create" }), b("Session ready — requesting first assistant turn"));
+				break;
+			case "response.output_audio.delta":
+				typeof t.delta == "string" && t.delta && E.writePcmBase64(t.delta);
+				break;
+			case "response.output_audio_transcript.delta":
+				typeof t.delta == "string" && t.delta && E.setTranscript(t.delta, "append");
+				break;
+			case "response.output_audio_transcript.done":
+				typeof t.transcript == "string" && t.transcript.trim() && E.setTranscript(t.transcript, "replace");
+				break;
+			case "response.done":
+			case "response.output_audio.done":
+				E.closeActiveTurn();
+				break;
+			case "input_audio_buffer.speech_started":
+				E.bargeIn();
+				break;
+			case "error": {
+				let e = String(t.error?.message ?? "OpenAI Realtime error");
+				v?.(e), b(e, "warn");
+				break;
+			}
+			default: break;
+		}
+	}
+	let k = `wss://api.openai.com/v1/realtime?model=${encodeURIComponent(i)}`;
+	if (S = new WebSocket(k, ["realtime", `openai-insecure-api-key.${y}`]), await new Promise((e, t) => {
+		if (!S) {
+			t(/* @__PURE__ */ Error("WebSocket failed to construct"));
+			return;
+		}
+		let n = S, r = () => {
+			a(), e();
+		}, i = () => {
+			a(), t(/* @__PURE__ */ Error("OpenAI Realtime WebSocket failed to open"));
+		}, a = () => {
+			n.removeEventListener("open", r), n.removeEventListener("error", i);
+		};
+		n.addEventListener("open", r), n.addEventListener("error", i);
+	}), S.addEventListener("message", (e) => {
+		typeof e.data == "string" && O(e.data);
+	}), S.addEventListener("close", () => {
+		T?.stop(), T = null, S = null, !C && (E.abandonActiveTurn(), b("OpenAI Realtime disconnected"), _?.());
+	}), s({
+		captureMic: m,
+		mediaStream: h
+	})) try {
+		T = await c({
+			targetSampleRate: x,
+			mediaStream: h,
+			onLog: b,
+			shouldSend: () => !C && S != null && S.readyState === WebSocket.OPEN,
+			onPcm: (e) => {
+				D({
+					type: "input_audio_buffer.append",
+					audio: a(e)
+				});
+			}
+		});
+	} catch (e) {
+		S.close();
+		let t = e instanceof Error ? e.message : String(e);
+		throw Error(`Microphone access failed: ${t}`);
+	}
+	return b("Realtime connected. Speak into your mic — the avatar should lip-sync."), {
+		isConnected: () => {
+			let e = S;
+			return e != null && e.readyState === WebSocket.OPEN && !C;
+		},
+		async end() {
+			if (C) return;
+			C = !0, await E.closeActiveTurn(), T?.stop(), T = null;
+			let e = S;
+			if (S = null, e && e.readyState === WebSocket.OPEN) try {
+				e.close();
+			} catch (e) {
+				console.error(e);
+			}
+		}
+	};
+}
+//#endregion
+//#region src/openai/connectOpenAiRealtimeWebRtc.ts
+async function m(e, t) {
+	let { ephemeralKey: n, model: r = u, voice: i = d, instructions: a = "You are a helpful voice assistant. Keep replies short.", autoResponse: o = !0, captureMic: c = !0, mediaStream: l, sampleRate: p = f, onLog: m, onDisconnect: h, onError: g } = t, _ = n.trim();
+	if (!_) throw Error("Pass ephemeralKey from your server mint route");
+	let v = (e, t = "info") => m?.(e, t), y = !1, b = !1, x = !1, S = null, C = null, w = "", T = null, E = !l, D = new RTCPeerConnection(), O = D.createDataChannel("oai-events");
+	function k(e) {
+		O.readyState === "open" && O.send(JSON.stringify(e));
+	}
+	function A(t) {
+		if (y) return;
+		C = t;
+		let n = w.trim();
+		S = e.speech.createUtterance({
+			track: t,
+			sampleRate: p,
+			queue: "replace-active",
+			...n ? { transcript: n } : {}
+		});
+	}
+	function j(e, t) {
+		let n = t === "replace" ? e.trim() : `${w}${e}`.trim();
+		n && (w = n, S && S.setTranscript(n).catch(() => void 0));
+	}
+	function M() {
+		w = "";
+		let t = S;
+		S = null, t ? t.cancel() : e.speech.interrupt({ scope: "active" }), C && !y && C.readyState === "live" && A(C);
+	}
+	function N(e) {
+		let t;
+		try {
+			t = JSON.parse(e);
+		} catch {
+			return;
+		}
+		switch (t.type) {
+			case "session.created":
+				k({
+					type: "session.update",
+					session: {
+						type: "realtime",
+						model: r,
+						output_modalities: ["audio"],
+						instructions: a,
+						audio: { output: { voice: i } }
+					}
+				});
+				break;
+			case "session.updated":
+				o && !b && (b = !0, k({ type: "response.create" }), v("WebRTC session ready — requesting first assistant turn"));
+				break;
+			case "response.output_audio_transcript.delta":
+				typeof t.delta == "string" && t.delta && j(t.delta, "append");
+				break;
+			case "response.output_audio_transcript.done":
+				typeof t.transcript == "string" && t.transcript.trim() && j(t.transcript, "replace");
+				break;
+			case "input_audio_buffer.speech_started":
+				M();
+				break;
+			case "error": {
+				let e = String(t.error?.message ?? "OpenAI Realtime error");
+				g?.(e), v(e, "warn");
+				break;
+			}
+			default: break;
+		}
+	}
+	if (D.ontrack = (e) => {
+		let t = e.track;
+		!t || t.kind !== "audio" || (v("Remote OpenAI audio track received — bridging to avatar"), A(t), t.addEventListener("ended", () => {
+			C === t && (C = null);
+			let e = S;
+			S = null, e?.close({ history: "none" }).catch(() => void 0);
+		}));
+	}, D.onconnectionstatechange = () => {
+		if (D.connectionState === "connected") {
+			x = !0;
+			return;
+		}
+		if (D.connectionState === "failed" || D.connectionState === "disconnected" || D.connectionState === "closed") {
+			if (x = !1, y) return;
+			let e = S;
+			S = null, e?.cancel().catch(() => void 0), v("OpenAI Realtime WebRTC disconnected"), h?.();
+		}
+	}, O.addEventListener("message", (e) => {
+		typeof e.data == "string" && N(e.data);
+	}), s({
+		captureMic: c,
+		mediaStream: l
+	})) try {
+		T = l ?? await navigator.mediaDevices.getUserMedia({ audio: {
+			channelCount: 1,
+			echoCancellation: !0,
+			noiseSuppression: !0
+		} });
+		for (let e of T.getAudioTracks()) D.addTrack(e, T);
+		v("Mic track added to WebRTC peer connection");
+	} catch (e) {
+		D.close();
+		let t = e instanceof Error ? e.message : String(e);
+		throw Error(`Microphone access failed: ${t}`);
+	}
+	let P = await D.createOffer();
+	await D.setLocalDescription(P);
+	let F = await fetch("https://api.openai.com/v1/realtime/calls", {
+		method: "POST",
+		body: P.sdp ?? "",
+		headers: {
+			Authorization: `Bearer ${_}`,
+			"Content-Type": "application/sdp"
+		}
+	});
+	if (!F.ok) {
+		if (D.close(), E && T) for (let e of T.getTracks()) e.stop();
+		let e = await F.text().catch(() => "");
+		throw Error(`OpenAI Realtime WebRTC SDP exchange failed (${F.status})${e ? `: ${e}` : ""}`);
+	}
+	let I = await F.text();
+	return await D.setRemoteDescription({
+		type: "answer",
+		sdp: I
+	}), v("Realtime WebRTC connected. Speak into your mic — the avatar should lip-sync."), {
+		peerConnection: D,
+		isConnected: () => x && !y && D.connectionState === "connected",
+		async end() {
+			if (y) return;
+			y = !0;
+			let e = S;
+			if (S = null, e) try {
+				await e.cancel();
+			} catch {}
+			try {
+				O.close();
+			} catch {}
+			try {
+				D.close();
+			} catch (e) {
+				console.error(e);
+			}
+			if (E && T) for (let e of T.getTracks()) e.stop();
+			T = null, x = !1;
+		}
+	};
+}
+//#endregion
+export { u as DEFAULT_OPENAI_REALTIME_MODEL, d as DEFAULT_OPENAI_REALTIME_VOICE, e as MAX_PCM_CHUNK_BYTES, f as OPENAI_REALTIME_PCM_RATE, t as base64ToPcmBytes, p as connectOpenAiRealtime, m as connectOpenAiRealtimeWebRtc, r as floatToPcm16Le, i as pcmBytesToBase64, o as resampleFloat32, l as writePcmInChunks };
+
+//# sourceMappingURL=index.js.map

package/dist/openai/index.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"index.js","names":[],"sources":["../../src/openai/connectOpenAiRealtime.ts","../../src/openai/connectOpenAiRealtimeWebRtc.ts"],"sourcesContent":["/**\n * OpenAI Realtime (WebSocket) → Liforma BYO voice helper.\n *\n * Mount a Liforma Experience (typically `mode=\"presenter\"`, `speechInputMode=\"off\"`),\n * wait until the player has started (audio unlocked), mint an ephemeral client secret\n * on your server, then call this helper.\n *\n * Why WebSocket here: per-turn PCM + transcript match the ElevenLabs createUtterance\n * pattern. OpenAI’s preferred browser media path is WebRTC — see docs for that path.\n *\n * Docs: https://docs.liforma.ai/avatar-experiences/bring-your-own-voice/openai\n */\n\nimport {\n\tcreateTurnSession,\n\tpcmBytesToBase64,\n\tshouldCaptureMic,\n\tstartPcmMic,\n\ttype ByoBridge,\n\ttype ByoConnectCallbacks,\n\ttype ByoMicCaptureOptions,\n\ttype ByoPcmExperience,\n\ttype PcmMicSession\n} from '@liforma/client/byo';\n\nexport type OpenAiRealtimeExperience = ByoPcmExperience;\n\n/** Default Realtime model when none is passed. */\nexport const DEFAULT_OPENAI_REALTIME_MODEL = 'gpt-realtime-2.1';\n\n/** Default OpenAI Realtime voice when none is passed. */\nexport const DEFAULT_OPENAI_REALTIME_VOICE = 'marin';\n\n/** OpenAI Realtime PCM rate used by this helper. */\nexport const OPENAI_REALTIME_PCM_RATE = 24_000;\n\nexport type ConnectOpenAiRealtimeOptions = ByoConnectCallbacks &\n\tByoMicCaptureOptions & {\n\t\t/** Ephemeral client secret from your backend (never ship OPENAI_API_KEY to browsers). */\n\t\treadonly ephemeralKey: string;\n\t\treadonly model?: string;\n\t\treadonly voice?: string;\n\t\t/** Session instructions applied via `session.update`. */\n\t\treadonly instructions?: string;\n\t\t/**\n\t\t * When true (default), send `response.create` after `session.updated` so the\n\t\t * assistant speaks first. Set false if the host will drive turns itself.\n\t\t */\n\t\treadonly autoResponse?: boolean;\n\t};\n\nexport type OpenAiRealtimeBridge = ByoBridge;\n\n/**\n * Start OpenAI Realtime over WebSocket and pipe agent PCM + transcript into\n * `experience.speech.createUtterance` so the Liforma avatar talks.\n * Captures the host mic by default (Liforma mic stays off).\n */\nexport async function connectOpenAiRealtime(\n\texperience: OpenAiRealtimeExperience,\n\toptions: ConnectOpenAiRealtimeOptions\n): Promise<OpenAiRealtimeBridge> {\n\tconst {\n\t\tephemeralKey,\n\t\tmodel = DEFAULT_OPENAI_REALTIME_MODEL,\n\t\tvoice = DEFAULT_OPENAI_REALTIME_VOICE,\n\t\tinstructions = 'You are a helpful voice assistant. Keep replies short.',\n\t\tautoResponse = true,\n\t\tcaptureMic = true,\n\t\tmediaStream,\n\t\tonLog,\n\t\tonDisconnect,\n\t\tonError\n\t} = options;\n\n\tconst key = ephemeralKey.trim();\n\tif (!key) throw new Error('Pass ephemeralKey from your server mint route');\n\n\tconst log = (line: string, kind: 'info' | 'warn' = 'info') => onLog?.(line, kind);\n\tconst sampleRate = OPENAI_REALTIME_PCM_RATE;\n\n\tlet ws: WebSocket | null = null;\n\tlet ended = false;\n\tlet greeted = false;\n\tlet mic: PcmMicSession | null = null;\n\n\tconst turns = createTurnSession(experience, {\n\t\tsampleRate,\n\t\tisEnded: () => ended,\n\t\tonWriteError: (err) => onError?.(err instanceof Error ? err.message : String(err))\n\t});\n\n\tfunction send(event: Record<string, unknown>): void {\n\t\tif (!ws || ws.readyState !== WebSocket.OPEN) return;\n\t\tws.send(JSON.stringify(event));\n\t}\n\n\tfunction handleServerEvent(raw: string): void {\n\t\tlet event: {\n\t\t\ttype?: string;\n\t\t\tdelta?: string;\n\t\t\ttranscript?: string;\n\t\t\terror?: { message?: string };\n\t\t};\n\t\ttry {\n\t\t\tevent = JSON.parse(raw) as typeof event;\n\t\t} catch {\n\t\t\treturn;\n\t\t}\n\n\t\tswitch (event.type) {\n\t\t\tcase 'session.created':\n\t\t\t\tsend({\n\t\t\t\t\ttype: 'session.update',\n\t\t\t\t\tsession: {\n\t\t\t\t\t\ttype: 'realtime',\n\t\t\t\t\t\tmodel,\n\t\t\t\t\t\toutput_modalities: ['audio'],\n\t\t\t\t\t\tinstructions,\n\t\t\t\t\t\taudio: {\n\t\t\t\t\t\t\tinput: {\n\t\t\t\t\t\t\t\tformat: { type: 'audio/pcm', rate: sampleRate },\n\t\t\t\t\t\t\t\tturn_detection: { type: 'semantic_vad' }\n\t\t\t\t\t\t\t},\n\t\t\t\t\t\t\toutput: {\n\t\t\t\t\t\t\t\tformat: { type: 'audio/pcm', rate: sampleRate },\n\t\t\t\t\t\t\t\tvoice\n\t\t\t\t\t\t\t}\n\t\t\t\t\t\t}\n\t\t\t\t\t}\n\t\t\t\t});\n\t\t\t\tbreak;\n\n\t\t\tcase 'session.updated':\n\t\t\t\tif (autoResponse && !greeted) {\n\t\t\t\t\tgreeted = true;\n\t\t\t\t\tsend({ type: 'response.create' });\n\t\t\t\t\tlog('Session ready — requesting first assistant turn');\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'response.output_audio.delta':\n\t\t\t\tif (typeof event.delta === 'string' && event.delta) {\n\t\t\t\t\tturns.writePcmBase64(event.delta);\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'response.output_audio_transcript.delta':\n\t\t\t\tif (typeof event.delta === 'string' && event.delta) {\n\t\t\t\t\tturns.setTranscript(event.delta, 'append');\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'response.output_audio_transcript.done':\n\t\t\t\tif (typeof event.transcript === 'string' && event.transcript.trim()) {\n\t\t\t\t\tturns.setTranscript(event.transcript, 'replace');\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'response.done':\n\t\t\tcase 'response.output_audio.done':\n\t\t\t\tvoid turns.closeActiveTurn();\n\t\t\t\tbreak;\n\n\t\t\tcase 'input_audio_buffer.speech_started':\n\t\t\t\tturns.bargeIn();\n\t\t\t\tbreak;\n\n\t\t\tcase 'error': {\n\t\t\t\tconst message = String(event.error?.message ?? 'OpenAI Realtime error');\n\t\t\t\tonError?.(message);\n\t\t\t\tlog(message, 'warn');\n\t\t\t\tbreak;\n\t\t\t}\n\n\t\t\tdefault:\n\t\t\t\tbreak;\n\t\t}\n\t}\n\n\tconst url = `wss://api.openai.com/v1/realtime?model=${encodeURIComponent(model)}`;\n\tws = new WebSocket(url, ['realtime', `openai-insecure-api-key.${key}`]);\n\n\tawait new Promise<void>((resolve, reject) => {\n\t\tif (!ws) {\n\t\t\treject(new Error('WebSocket failed to construct'));\n\t\t\treturn;\n\t\t}\n\t\tconst socket = ws;\n\t\tconst onOpen = () => {\n\t\t\tcleanup();\n\t\t\tresolve();\n\t\t};\n\t\tconst onErrorEvt = () => {\n\t\t\tcleanup();\n\t\t\treject(new Error('OpenAI Realtime WebSocket failed to open'));\n\t\t};\n\t\tconst cleanup = () => {\n\t\t\tsocket.removeEventListener('open', onOpen);\n\t\t\tsocket.removeEventListener('error', onErrorEvt);\n\t\t};\n\t\tsocket.addEventListener('open', onOpen);\n\t\tsocket.addEventListener('error', onErrorEvt);\n\t});\n\n\tws.addEventListener('message', (msg) => {\n\t\tif (typeof msg.data === 'string') handleServerEvent(msg.data);\n\t});\n\n\tws.addEventListener('close', () => {\n\t\tmic?.stop();\n\t\tmic = null;\n\t\tws = null;\n\t\t// Skip after intentional end() — turn was already closed cleanly.\n\t\tif (ended) return;\n\t\tturns.abandonActiveTurn();\n\t\tlog('OpenAI Realtime disconnected');\n\t\tonDisconnect?.();\n\t});\n\n\tif (shouldCaptureMic({ captureMic, mediaStream })) {\n\t\ttry {\n\t\t\tmic = await startPcmMic({\n\t\t\t\ttargetSampleRate: sampleRate,\n\t\t\t\tmediaStream,\n\t\t\t\tonLog: log,\n\t\t\t\tshouldSend: () => !ended && ws != null && ws.readyState === WebSocket.OPEN,\n\t\t\t\tonPcm: (pcm) => {\n\t\t\t\t\tsend({\n\t\t\t\t\t\ttype: 'input_audio_buffer.append',\n\t\t\t\t\t\taudio: pcmBytesToBase64(pcm)\n\t\t\t\t\t});\n\t\t\t\t}\n\t\t\t});\n\t\t} catch (err) {\n\t\t\tws.close();\n\t\t\tconst message = err instanceof Error ? err.message : String(err);\n\t\t\tthrow new Error(`Microphone access failed: ${message}`);\n\t\t}\n\t}\n\n\tlog('Realtime connected. Speak into your mic — the avatar should lip-sync.');\n\n\treturn {\n\t\tisConnected: () => {\n\t\t\tconst socket = ws;\n\t\t\treturn socket != null && socket.readyState === WebSocket.OPEN && !ended;\n\t\t},\n\t\tasync end() {\n\t\t\tif (ended) return;\n\t\t\tended = true;\n\t\t\tawait turns.closeActiveTurn();\n\t\t\tmic?.stop();\n\t\t\tmic = null;\n\t\t\tconst active = ws;\n\t\t\tws = null;\n\t\t\tif (active && active.readyState === WebSocket.OPEN) {\n\t\t\t\ttry {\n\t\t\t\t\tactive.close();\n\t\t\t\t} catch (err) {\n\t\t\t\t\tconsole.error(err);\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t};\n}\n","/**\n * OpenAI Realtime (WebRTC) → Liforma BYO voice helper.\n *\n * Preferred browser media path: remote audio track → `createUtterance({ track })`,\n * transcript side-channel on the `oai-events` data channel for force-align lipsync.\n * Do **not** also attach the remote track to an `<audio>` element.\n *\n * Docs: https://docs.liforma.ai/avatar-experiences/bring-your-own-voice/openai\n */\n\nimport {\n\tshouldCaptureMic,\n\ttype ByoBridge,\n\ttype ByoConnectCallbacks,\n\ttype ByoMicCaptureOptions,\n\ttype ByoPcmExperience,\n\ttype ByoSpeechUtterance\n} from '@liforma/client/byo';\nimport {\n\tDEFAULT_OPENAI_REALTIME_MODEL,\n\tDEFAULT_OPENAI_REALTIME_VOICE,\n\tOPENAI_REALTIME_PCM_RATE,\n\ttype OpenAiRealtimeExperience\n} from './connectOpenAiRealtime';\n\nexport type ConnectOpenAiRealtimeWebRtcOptions = ByoConnectCallbacks &\n\tByoMicCaptureOptions & {\n\t\t/** Ephemeral client secret from your backend (never ship OPENAI_API_KEY to browsers). */\n\t\treadonly ephemeralKey: string;\n\t\treadonly model?: string;\n\t\treadonly voice?: string;\n\t\t/** Session instructions applied via data-channel `session.update`. */\n\t\treadonly instructions?: string;\n\t\t/**\n\t\t * When true (default), send `response.create` after `session.updated` so the\n\t\t * assistant speaks first.\n\t\t */\n\t\treadonly autoResponse?: boolean;\n\t\t/** Sample-rate hint for track capture (OpenAI Realtime audio is typically 24 kHz). */\n\t\treadonly sampleRate?: number;\n\t};\n\nexport type OpenAiRealtimeWebRtcBridge = ByoBridge & {\n\treadonly peerConnection: RTCPeerConnection;\n};\n\n/**\n * Start OpenAI Realtime over WebRTC and pipe the remote audio track + transcript\n * into `experience.speech.createUtterance` so the Liforma avatar talks.\n */\nexport async function connectOpenAiRealtimeWebRtc(\n\texperience: OpenAiRealtimeExperience,\n\toptions: ConnectOpenAiRealtimeWebRtcOptions\n): Promise<OpenAiRealtimeWebRtcBridge> {\n\tconst {\n\t\tephemeralKey,\n\t\tmodel = DEFAULT_OPENAI_REALTIME_MODEL,\n\t\tvoice = DEFAULT_OPENAI_REALTIME_VOICE,\n\t\tinstructions = 'You are a helpful voice assistant. Keep replies short.',\n\t\tautoResponse = true,\n\t\tcaptureMic = true,\n\t\tmediaStream,\n\t\tsampleRate = OPENAI_REALTIME_PCM_RATE,\n\t\tonLog,\n\t\tonDisconnect,\n\t\tonError\n\t} = options;\n\n\tconst key = ephemeralKey.trim();\n\tif (!key) throw new Error('Pass ephemeralKey from your server mint route');\n\n\tconst log = (line: string, kind: 'info' | 'warn' = 'info') => onLog?.(line, kind);\n\n\tlet ended = false;\n\tlet greeted = false;\n\tlet connected = false;\n\tlet utterance: ByoSpeechUtterance | null = null;\n\tlet remoteTrack: MediaStreamTrack | null = null;\n\tlet activeTranscript = '';\n\tlet localStream: MediaStream | null = null;\n\tconst stopLocalTracks = !mediaStream;\n\n\tconst pc = new RTCPeerConnection();\n\tconst dc = pc.createDataChannel('oai-events');\n\n\tfunction send(event: Record<string, unknown>): void {\n\t\tif (dc.readyState !== 'open') return;\n\t\tdc.send(JSON.stringify(event));\n\t}\n\n\tfunction openUtteranceForTrack(track: MediaStreamTrack): void {\n\t\tif (ended) return;\n\t\tremoteTrack = track;\n\t\tconst seed = activeTranscript.trim();\n\t\tutterance = experience.speech.createUtterance({\n\t\t\ttrack,\n\t\t\tsampleRate,\n\t\t\tqueue: 'replace-active',\n\t\t\t...(seed ? { transcript: seed } : {})\n\t\t});\n\t}\n\n\tfunction setTranscript(text: string, mode: 'replace' | 'append'): void {\n\t\tconst next =\n\t\t\tmode === 'replace' ? text.trim() : `${activeTranscript}${text}`.trim();\n\t\tif (!next) return;\n\t\tactiveTranscript = next;\n\t\tif (!utterance) return;\n\t\tvoid utterance.setTranscript(next).catch(() => undefined);\n\t}\n\n\tfunction bargeIn(): void {\n\t\tactiveTranscript = '';\n\t\tconst current = utterance;\n\t\tutterance = null;\n\t\tvoid (current\n\t\t\t? current.cancel()\n\t\t\t: experience.speech.interrupt({ scope: 'active' }));\n\t\t// Continuous remote track — reopen so the next agent audio still lip-syncs.\n\t\tif (remoteTrack && !ended && remoteTrack.readyState === 'live') {\n\t\t\topenUtteranceForTrack(remoteTrack);\n\t\t}\n\t}\n\n\tfunction handleServerEvent(raw: string): void {\n\t\tlet event: {\n\t\t\ttype?: string;\n\t\t\tdelta?: string;\n\t\t\ttranscript?: string;\n\t\t\terror?: { message?: string };\n\t\t};\n\t\ttry {\n\t\t\tevent = JSON.parse(raw) as typeof event;\n\t\t} catch {\n\t\t\treturn;\n\t\t}\n\n\t\tswitch (event.type) {\n\t\t\tcase 'session.created':\n\t\t\t\tsend({\n\t\t\t\t\ttype: 'session.update',\n\t\t\t\t\tsession: {\n\t\t\t\t\t\ttype: 'realtime',\n\t\t\t\t\t\tmodel,\n\t\t\t\t\t\toutput_modalities: ['audio'],\n\t\t\t\t\t\tinstructions,\n\t\t\t\t\t\taudio: {\n\t\t\t\t\t\t\toutput: { voice }\n\t\t\t\t\t\t}\n\t\t\t\t\t}\n\t\t\t\t});\n\t\t\t\tbreak;\n\n\t\t\tcase 'session.updated':\n\t\t\t\tif (autoResponse && !greeted) {\n\t\t\t\t\tgreeted = true;\n\t\t\t\t\tsend({ type: 'response.create' });\n\t\t\t\t\tlog('WebRTC session ready — requesting first assistant turn');\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'response.output_audio_transcript.delta':\n\t\t\t\tif (typeof event.delta === 'string' && event.delta) {\n\t\t\t\t\tsetTranscript(event.delta, 'append');\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'response.output_audio_transcript.done':\n\t\t\t\tif (typeof event.transcript === 'string' && event.transcript.trim()) {\n\t\t\t\t\tsetTranscript(event.transcript, 'replace');\n\t\t\t\t}\n\t\t\t\tbreak;\n\n\t\t\tcase 'input_audio_buffer.speech_started':\n\t\t\t\tbargeIn();\n\t\t\t\tbreak;\n\n\t\t\tcase 'error': {\n\t\t\t\tconst message = String(event.error?.message ?? 'OpenAI Realtime error');\n\t\t\t\tonError?.(message);\n\t\t\t\tlog(message, 'warn');\n\t\t\t\tbreak;\n\t\t\t}\n\n\t\t\tdefault:\n\t\t\t\tbreak;\n\t\t}\n\t}\n\n\tpc.ontrack = (ev) => {\n\t\tconst track = ev.track;\n\t\tif (!track || track.kind !== 'audio') return;\n\t\tlog('Remote OpenAI audio track received — bridging to avatar');\n\t\topenUtteranceForTrack(track);\n\t\ttrack.addEventListener('ended', () => {\n\t\t\tif (remoteTrack === track) remoteTrack = null;\n\t\t\tconst current = utterance;\n\t\t\tutterance = null;\n\t\t\tvoid current?.close({ history: 'none' }).catch(() => undefined);\n\t\t});\n\t};\n\n\tpc.onconnectionstatechange = () => {\n\t\tif (pc.connectionState === 'connected') {\n\t\t\tconnected = true;\n\t\t\treturn;\n\t\t}\n\t\tif (\n\t\t\tpc.connectionState === 'failed' ||\n\t\t\tpc.connectionState === 'disconnected' ||\n\t\t\tpc.connectionState === 'closed'\n\t\t) {\n\t\t\tconnected = false;\n\t\t\tif (ended) return;\n\t\t\tconst current = utterance;\n\t\t\tutterance = null;\n\t\t\tvoid current?.cancel().catch(() => undefined);\n\t\t\tlog('OpenAI Realtime WebRTC disconnected');\n\t\t\tonDisconnect?.();\n\t\t}\n\t};\n\n\tdc.addEventListener('message', (ev) => {\n\t\tif (typeof ev.data === 'string') handleServerEvent(ev.data);\n\t});\n\n\tif (shouldCaptureMic({ captureMic, mediaStream })) {\n\t\ttry {\n\t\t\tlocalStream =\n\t\t\t\tmediaStream ??\n\t\t\t\t(await navigator.mediaDevices.getUserMedia({\n\t\t\t\t\taudio: {\n\t\t\t\t\t\tchannelCount: 1,\n\t\t\t\t\t\techoCancellation: true,\n\t\t\t\t\t\tnoiseSuppression: true\n\t\t\t\t\t}\n\t\t\t\t}));\n\t\t\tfor (const track of localStream.getAudioTracks()) {\n\t\t\t\tpc.addTrack(track, localStream);\n\t\t\t}\n\t\t\tlog('Mic track added to WebRTC peer connection');\n\t\t} catch (err) {\n\t\t\tpc.close();\n\t\t\tconst message = err instanceof Error ? err.message : String(err);\n\t\t\tthrow new Error(`Microphone access failed: ${message}`);\n\t\t}\n\t}\n\n\tconst offer = await pc.createOffer();\n\tawait pc.setLocalDescription(offer);\n\n\tconst sdpResponse = await fetch('https://api.openai.com/v1/realtime/calls', {\n\t\tmethod: 'POST',\n\t\tbody: offer.sdp ?? '',\n\t\theaders: {\n\t\t\tAuthorization: `Bearer ${key}`,\n\t\t\t'Content-Type': 'application/sdp'\n\t\t}\n\t});\n\n\tif (!sdpResponse.ok) {\n\t\tpc.close();\n\t\tif (stopLocalTracks && localStream) {\n\t\t\tfor (const track of localStream.getTracks()) track.stop();\n\t\t}\n\t\tconst detail = await sdpResponse.text().catch(() => '');\n\t\tthrow new Error(\n\t\t\t`OpenAI Realtime WebRTC SDP exchange failed (${sdpResponse.status})${detail ? `: ${detail}` : ''}`\n\t\t);\n\t}\n\n\tconst answerSdp = await sdpResponse.text();\n\tawait pc.setRemoteDescription({ type: 'answer', sdp: answerSdp });\n\n\tlog('Realtime WebRTC connected. Speak into your mic — the avatar should lip-sync.');\n\n\treturn {\n\t\tpeerConnection: pc,\n\t\tisConnected: () => connected && !ended && pc.connectionState === 'connected',\n\t\tasync end() {\n\t\t\tif (ended) return;\n\t\t\tended = true;\n\t\t\tconst current = utterance;\n\t\t\tutterance = null;\n\t\t\tif (current) {\n\t\t\t\ttry {\n\t\t\t\t\tawait current.cancel();\n\t\t\t\t} catch {\n\t\t\t\t\t/* ignore */\n\t\t\t\t}\n\t\t\t}\n\t\t\ttry {\n\t\t\t\tdc.close();\n\t\t\t} catch {\n\t\t\t\t/* ignore */\n\t\t\t}\n\t\t\ttry {\n\t\t\t\tpc.close();\n\t\t\t} catch (err) {\n\t\t\t\tconsole.error(err);\n\t\t\t}\n\t\t\tif (stopLocalTracks && localStream) {\n\t\t\t\tfor (const track of localStream.getTracks()) track.stop();\n\t\t\t}\n\t\t\tlocalStream = null;\n\t\t\tconnected = false;\n\t\t}\n\t};\n}\n\n/** @deprecated Prefer OpenAiRealtimeExperience — same duck type. */\nexport type OpenAiRealtimeWebRtcExperience = 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package/dist/react/byo/audioCapture.d.ts

@@ -0,0 +1,24 @@
+/**
+ * Shared Web Audio capture: AudioWorklet when available, ScriptProcessor fallback.
+ * Used by BYO mic uplink and MediaStreamTrack → PCM piping.
+ */
+export type AudioCaptureSession = {
+    stop(): void;
+    readonly fromSampleRate: number;
+    readonly usingWorklet: boolean;
+};
+export type StartAudioCaptureOptions = {
+    readonly mediaStream: MediaStream;
+    readonly targetSampleRate: number;
+    /** Called with pcm_s16le mono at targetSampleRate. */
+    readonly onPcm: (bytes: Uint8Array) => void;
+    /** When false, frames are dropped. */
+    readonly shouldSend?: () => boolean;
+    readonly onLog?: (line: string, kind?: 'info' | 'warn') => void;
+};
+/**
+ * Capture a MediaStream as pcm_s16le at targetSampleRate.
+ * Prefers AudioWorklet; falls back to ScriptProcessorNode.
+ */
+export declare function startAudioCapture(options: StartAudioCaptureOptions): Promise<AudioCaptureSession>;
+//# sourceMappingURL=audioCapture.d.ts.map

package/dist/react/byo/audioCapture.d.ts.map

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package/dist/react/byo/pcm.d.ts

@@ -0,0 +1,33 @@
+/** Liforma `speech.write` rejects chunks larger than 64 KiB. */
+export declare const MAX_PCM_CHUNK_BYTES: number;
+/**
+ * Playback / chunk guidance for custom BYO bridges (mirrors speech controller limits).
+ * Prefer ~20–200 ms PCM frames; the player starts after an adaptive ~80–250 ms jitter
+ * buffer (default 150 ms) based on chunk arrival variance.
+ */
+export declare const BYO_PLAYBACK_GUIDANCE: {
+    readonly RECOMMENDED_CHUNK_MS_MIN: 20;
+    readonly RECOMMENDED_CHUNK_MS_MAX: 200;
+    readonly PLAYBACK_JITTER_BUFFER_MS: 150;
+    readonly PLAYBACK_JITTER_BUFFER_MS_MIN: 80;
+    readonly PLAYBACK_JITTER_BUFFER_MS_MAX: 250;
+};
+export declare function base64ToPcmBytes(b64: string): Uint8Array;
+export declare function pcmBytesToBase64(bytes: Uint8Array): string;
+export declare function floatToPcm16Le(input: Float32Array): Uint8Array;
+/** Naive linear resample Float32 → target rate. */
+export declare function resampleFloat32(input: Float32Array, fromRate: number, toRate: number): Float32Array;
+/**
+ * Parse vendor format strings like `pcm_16000` / `pcm_24000`.
+ * Returns null when the format is missing or not PCM.
+ */
+export declare function parsePcmSampleRate(format: string | null | undefined): number | null;
+/**
+ * Write even-length PCM, splitting at MAX_PCM_CHUNK_BYTES on even boundaries.
+ * Callers that may receive odd-length frames should run bytes through
+ * {@link takeEvenPcmBytes} first so samples are not dropped across chunks.
+ */
+export declare function writePcmInChunks(utterance: {
+    write: (chunk: ArrayBuffer | ArrayBufferView) => Promise<unknown>;
+}, bytes: Uint8Array): Promise<void>;
+//# sourceMappingURL=pcm.d.ts.map

package/dist/react/byo/pcm.d.ts.map

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package/dist/react/byo/resample.d.ts

@@ -0,0 +1,18 @@
+/**
+ * Stateful cubic (Catmull-Rom) resampler for continuous PCM capture streams.
+ * Better than naive linear for uplink mic / track capture.
+ */
+export type FloatResampler = {
+    /** Push a mono float buffer; returns samples at the target rate (may be empty). */
+    push(input: Float32Array): Float32Array;
+    /** Flush any remaining fractional sample (usually silence). */
+    flush(): Float32Array;
+    readonly fromRate: number;
+    readonly toRate: number;
+};
+/**
+ * Create a continuous resampler. When rates match, push() returns the input as-is
+ * (no copy) when possible.
+ */
+export declare function createCubicResampler(fromRate: number, toRate: number): FloatResampler;
+//# sourceMappingURL=resample.d.ts.map

package/dist/react/byo/resample.d.ts.map

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