@happyvertical/speech 0.97.0 → 0.99.0

package/dist/conversation-server.js

@@ -0,0 +1,174 @@
+import { i as SpeechProviderError, n as SpeechConfigurationError } from "./chunks/errors-CQBX3OPS.js";
+import { t as isBrowserRuntime } from "./chunks/websocket-CR9uXAuS.js";
+//#region src/conversation-server.ts
+/** Server-only OpenAI conversational call lifecycle. Hosts own authorization and admission policy. */
+/** Build the GA speech-to-speech session config. Transcription is an optional side channel. */
+function openAIConversationConfig(options) {
+	if (!options.model?.trim()) throw new SpeechConfigurationError("Conversation model is required");
+	const detection = options.turnDetection ?? {
+		type: "semantic_vad",
+		eagerness: "auto"
+	};
+	let turnDetection;
+	switch (detection.type) {
+		case "manual":
+			turnDetection = null;
+			break;
+		case "semantic_vad":
+			turnDetection = {
+				type: "semantic_vad",
+				eagerness: detection.eagerness ?? "auto",
+				create_response: true,
+				interrupt_response: true
+			};
+			break;
+		case "server_vad":
+			turnDetection = {
+				type: "server_vad",
+				...detection.threshold === void 0 ? {} : { threshold: detection.threshold },
+				...detection.prefixPaddingMs === void 0 ? {} : { prefix_padding_ms: detection.prefixPaddingMs },
+				...detection.silenceDurationMs === void 0 ? {} : { silence_duration_ms: detection.silenceDurationMs },
+				create_response: true,
+				interrupt_response: true
+			};
+			break;
+		default: throw new SpeechConfigurationError("Unknown conversation turn detection");
+	}
+	return {
+		type: "realtime",
+		model: options.model.trim(),
+		output_modalities: ["audio"],
+		...options.instructions ? { instructions: options.instructions } : {},
+		audio: {
+			input: {
+				turn_detection: turnDetection,
+				...options.transcriptionModel ? { transcription: { model: options.transcriptionModel } } : {}
+			},
+			output: { voice: options.voice ?? "marin" }
+		},
+		...options.maxOutputTokens === void 0 ? {} : { max_output_tokens: options.maxOutputTokens },
+		...options.tools ? { tools: options.tools } : {}
+	};
+}
+/** No retries: retrying call creation can create duplicate billable sessions. */
+async function createOpenAIVoiceCall(offer, options) {
+	assertServer(options);
+	if (!validSdp(offer)) throw new SpeechConfigurationError("Expected a bounded SDP offer");
+	const body = new FormData();
+	body.set("sdp", offer);
+	body.set("session", JSON.stringify(openAIConversationConfig(options)));
+	let callId;
+	try {
+		const answer = await (await request(callBase(options.baseUrl), { body }, options, (headers) => {
+			const value = headers.get("location")?.split("/").pop();
+			if (!value || !/^rtc_[A-Za-z0-9_-]+$/.test(value)) throw new SpeechProviderError("openai-conversation", "Call response did not contain a WebRTC call id");
+			callId = value;
+		})).text();
+		if (!validSdp(answer)) throw new SpeechProviderError("openai-conversation", "Call response did not contain a valid SDP answer");
+		options.signal?.throwIfAborted();
+		return {
+			answer,
+			callId
+		};
+	} catch (error) {
+		if (callId) try {
+			await hangupOpenAIVoiceCall(callId, {
+				...options,
+				signal: void 0
+			});
+		} catch {
+			throw new VoiceCallSetupError(callId);
+		}
+		throw error;
+	}
+}
+/** Hosts should call this on close, idle/hard timeout and failed client setup. */
+async function hangupOpenAIVoiceCall(callId, options) {
+	assertServer(options);
+	if (!/^rtc_[A-Za-z0-9_-]+$/.test(callId)) throw new SpeechConfigurationError("Invalid WebRTC call id");
+	await request(`${callBase(options.baseUrl)}/${callId}/hangup`, {}, options);
+}
+function assertServer(options) {
+	if (isBrowserRuntime()) throw new SpeechConfigurationError("Voice call management is server-only; never expose a provider key");
+	if (!options.apiKey?.trim()) throw new SpeechConfigurationError("Voice call management requires a server API key");
+	const timeout = options.timeoutMs ?? 15e3;
+	if (!Number.isFinite(timeout) || timeout <= 0) throw new SpeechConfigurationError("timeoutMs must be positive");
+}
+function callBase(baseUrl = "https://api.openai.com/v1") {
+	const url = new URL(baseUrl);
+	if (!["http:", "https:"].includes(url.protocol) || url.username || url.password || url.search || url.hash) throw new SpeechConfigurationError("Invalid voice call API base URL");
+	url.pathname = `${url.pathname.replace(/\/$/, "")}${/\/v\d+$/.test(url.pathname.replace(/\/$/, "")) ? "" : "/v1"}/realtime/calls`;
+	return url.toString();
+}
+function validSdp(value) {
+	return typeof value === "string" && value.length <= 65536 && value.startsWith("v=0") && value.includes("m=audio");
+}
+async function request(url, init, options, onHeaders) {
+	const controller = new AbortController();
+	const timer = setTimeout(() => controller.abort(), options.timeoutMs ?? 15e3);
+	const signal = AbortSignal.any([controller.signal, ...options.signal ? [options.signal] : []]);
+	try {
+		const response = await (options.fetch ?? fetch)(url, {
+			...init,
+			method: "POST",
+			headers: { authorization: `Bearer ${options.apiKey}` },
+			signal
+		});
+		if (!response.ok) throw new SpeechProviderError("openai-conversation", "OpenAI voice call request failed", { status: response.status });
+		onHeaders?.(response.headers);
+		const body = await readBody(response, signal);
+		return new Response(response.status === 204 ? null : body, {
+			status: response.status,
+			headers: response.headers
+		});
+	} catch (error) {
+		if (error instanceof SpeechProviderError) throw error;
+		throw new SpeechProviderError("openai-conversation", signal.aborted ? "OpenAI voice call request cancelled or timed out" : "OpenAI voice call transport failed");
+	} finally {
+		clearTimeout(timer);
+	}
+}
+/** Failed setup whose cleanup also failed. The host must reserve admission and retry termination. */
+var VoiceCallSetupError = class extends SpeechProviderError {
+	callId;
+	constructor(callId) {
+		super("openai-conversation", "Voice setup failed and termination needs retry");
+		this.callId = callId;
+	}
+};
+async function readBody(response, signal) {
+	const reader = response.body?.getReader();
+	if (!reader) return "";
+	const chunks = [];
+	let size = 0;
+	const cancel = () => {
+		reader.cancel().catch(() => void 0);
+	};
+	signal.addEventListener("abort", cancel, { once: true });
+	try {
+		while (true) {
+			signal.throwIfAborted();
+			const { done, value } = await reader.read();
+			signal.throwIfAborted();
+			if (done) break;
+			size += value.byteLength;
+			if (size > 65536) throw new SpeechProviderError("openai-conversation", "Voice call response exceeded its size limit");
+			chunks.push(value);
+		}
+		const data = new Uint8Array(size);
+		let offset = 0;
+		for (const chunk of chunks) {
+			data.set(chunk, offset);
+			offset += chunk.byteLength;
+		}
+		return new TextDecoder().decode(data);
+	} finally {
+		signal.removeEventListener("abort", cancel);
+		await reader.cancel().catch(() => void 0);
+		reader.releaseLock();
+	}
+}
+//#endregion
+export { VoiceCallSetupError, createOpenAIVoiceCall, hangupOpenAIVoiceCall, openAIConversationConfig };
+
+//# sourceMappingURL=conversation-server.js.map

package/dist/conversation-server.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"conversation-server.js","names":[],"sources":["../src/conversation-server.ts"],"sourcesContent":["/** Server-only OpenAI conversational call lifecycle. Hosts own authorization and admission policy. */\nimport {\n  SpeechConfigurationError,\n  SpeechProviderError,\n} from './shared/errors.js';\nimport type { StreamingTurnDetection } from './shared/streaming-types.js';\nimport type { SpeechFetch } from './shared/types.js';\nimport { isBrowserRuntime } from './shared/websocket.js';\n\nexport interface OpenAIVoiceCallOptions {\n  apiKey: string;\n  /** Fixed by the host, never copied from untrusted browser JSON. */\n  model: string;\n  voice?: string;\n  instructions?: string;\n  transcriptionModel?: string;\n  turnDetection?: StreamingTurnDetection;\n  maxOutputTokens?: number;\n  tools?: Array<{\n    type: 'function';\n    name: string;\n    description?: string;\n    parameters: Record<string, unknown>;\n  }>;\n  baseUrl?: string;\n  fetch?: SpeechFetch;\n  signal?: AbortSignal;\n  timeoutMs?: number;\n}\n\nexport interface OpenAIVoiceCall {\n  answer: string;\n  callId: string;\n}\n\n/** Build the GA speech-to-speech session config. Transcription is an optional side channel. */\nexport function openAIConversationConfig(\n  options: Pick<\n    OpenAIVoiceCallOptions,\n    | 'model'\n    | 'voice'\n    | 'instructions'\n    | 'transcriptionModel'\n    | 'turnDetection'\n    | 'maxOutputTokens'\n    | 'tools'\n  >,\n): Record<string, unknown> {\n  if (!options.model?.trim())\n    throw new SpeechConfigurationError('Conversation model is required');\n  const detection = options.turnDetection ?? {\n    type: 'semantic_vad',\n    eagerness: 'auto',\n  };\n  let turnDetection: Record<string, unknown> | null;\n  switch (detection.type) {\n    case 'manual':\n      turnDetection = null;\n      break;\n    case 'semantic_vad':\n      turnDetection = {\n        type: 'semantic_vad',\n        eagerness: detection.eagerness ?? 'auto',\n        create_response: true,\n        interrupt_response: true,\n      };\n      break;\n    case 'server_vad':\n      turnDetection = {\n        type: 'server_vad',\n        ...(detection.threshold === undefined\n          ? {}\n          : { threshold: detection.threshold }),\n        ...(detection.prefixPaddingMs === undefined\n          ? {}\n          : { prefix_padding_ms: detection.prefixPaddingMs }),\n        ...(detection.silenceDurationMs === undefined\n          ? {}\n          : { silence_duration_ms: detection.silenceDurationMs }),\n        create_response: true,\n        interrupt_response: true,\n      };\n      break;\n    default:\n      throw new SpeechConfigurationError('Unknown conversation turn detection');\n  }\n  return {\n    type: 'realtime',\n    model: options.model.trim(),\n    output_modalities: ['audio'],\n    ...(options.instructions ? { instructions: options.instructions } : {}),\n    audio: {\n      input: {\n        turn_detection: turnDetection,\n        ...(options.transcriptionModel\n          ? { transcription: { model: options.transcriptionModel } }\n          : {}),\n      },\n      output: { voice: options.voice ?? 'marin' },\n    },\n    ...(options.maxOutputTokens === undefined\n      ? {}\n      : { max_output_tokens: options.maxOutputTokens }),\n    ...(options.tools ? { tools: options.tools } : {}),\n  };\n}\n\n/** No retries: retrying call creation can create duplicate billable sessions. */\nexport async function createOpenAIVoiceCall(\n  offer: string,\n  options: OpenAIVoiceCallOptions,\n): Promise<OpenAIVoiceCall> {\n  assertServer(options);\n  if (!validSdp(offer))\n    throw new SpeechConfigurationError('Expected a bounded SDP offer');\n  const body = new FormData();\n  body.set('sdp', offer);\n  body.set('session', JSON.stringify(openAIConversationConfig(options)));\n  let callId: string | undefined;\n  try {\n    const response = await request(\n      callBase(options.baseUrl),\n      { body },\n      options,\n      (headers) => {\n        const value = headers.get('location')?.split('/').pop();\n        if (!value || !/^rtc_[A-Za-z0-9_-]+$/.test(value))\n          throw new SpeechProviderError(\n            'openai-conversation',\n            'Call response did not contain a WebRTC call id',\n          );\n        // Capture identity before consuming a response body that can fail or time out.\n        callId = value;\n      },\n    );\n    const answer = await response.text();\n    if (!validSdp(answer))\n      throw new SpeechProviderError(\n        'openai-conversation',\n        'Call response did not contain a valid SDP answer',\n      );\n    options.signal?.throwIfAborted();\n    return { answer, callId: callId! };\n  } catch (error) {\n    if (callId) {\n      try {\n        await hangupOpenAIVoiceCall(callId, { ...options, signal: undefined });\n      } catch {\n        throw new VoiceCallSetupError(callId);\n      }\n    }\n    throw error;\n  }\n}\n\nexport type OpenAIVoiceCallControlOptions = Pick<\n  OpenAIVoiceCallOptions,\n  'apiKey' | 'baseUrl' | 'fetch' | 'signal' | 'timeoutMs'\n>;\n\n/** Hosts should call this on close, idle/hard timeout and failed client setup. */\nexport async function hangupOpenAIVoiceCall(\n  callId: string,\n  options: OpenAIVoiceCallControlOptions,\n): Promise<void> {\n  assertServer(options);\n  if (!/^rtc_[A-Za-z0-9_-]+$/.test(callId))\n    throw new SpeechConfigurationError('Invalid WebRTC call id');\n  await request(`${callBase(options.baseUrl)}/${callId}/hangup`, {}, options);\n}\n\nfunction assertServer(options: OpenAIVoiceCallControlOptions): void {\n  if (isBrowserRuntime())\n    throw new SpeechConfigurationError(\n      'Voice call management is server-only; never expose a provider key',\n    );\n  if (!options.apiKey?.trim())\n    throw new SpeechConfigurationError(\n      'Voice call management requires a server API key',\n    );\n  const timeout = options.timeoutMs ?? 15_000;\n  if (!Number.isFinite(timeout) || timeout <= 0)\n    throw new SpeechConfigurationError('timeoutMs must be positive');\n}\n\nfunction callBase(baseUrl = 'https://api.openai.com/v1'): string {\n  const url = new URL(baseUrl);\n  if (\n    !['http:', 'https:'].includes(url.protocol) ||\n    url.username ||\n    url.password ||\n    url.search ||\n    url.hash\n  )\n    throw new SpeechConfigurationError('Invalid voice call API base URL');\n  url.pathname = `${url.pathname.replace(/\\/$/, '')}${/\\/v\\d+$/.test(url.pathname.replace(/\\/$/, '')) ? '' : '/v1'}/realtime/calls`;\n  return url.toString();\n}\n\nfunction validSdp(value: string): boolean {\n  return (\n    typeof value === 'string' &&\n    value.length <= 65_536 &&\n    value.startsWith('v=0') &&\n    value.includes('m=audio')\n  );\n}\n\nasync function request(\n  url: string,\n  init: RequestInit,\n  options: OpenAIVoiceCallControlOptions,\n  onHeaders?: (headers: Headers) => void,\n): Promise<Response> {\n  const controller = new AbortController();\n  const timer = setTimeout(\n    () => controller.abort(),\n    options.timeoutMs ?? 15_000,\n  );\n  const signal = AbortSignal.any([\n    controller.signal,\n    ...(options.signal ? [options.signal] : []),\n  ]);\n  try {\n    const response = await (options.fetch ?? fetch)(url, {\n      ...init,\n      method: 'POST',\n      headers: { authorization: `Bearer ${options.apiKey}` },\n      signal,\n    });\n    if (!response.ok)\n      throw new SpeechProviderError(\n        'openai-conversation',\n        'OpenAI voice call request failed',\n        { status: response.status },\n      );\n    onHeaders?.(response.headers);\n    const body = await readBody(response, signal);\n    return new Response(response.status === 204 ? null : body, {\n      status: response.status,\n      headers: response.headers,\n    });\n  } catch (error) {\n    if (error instanceof SpeechProviderError) throw error;\n    // Deliberately omit provider payloads and transport messages, which can contain credentials.\n    throw new SpeechProviderError(\n      'openai-conversation',\n      signal.aborted\n        ? 'OpenAI voice call request cancelled or timed out'\n        : 'OpenAI voice call transport failed',\n    );\n  } finally {\n    clearTimeout(timer);\n  }\n}\n\n/** Failed setup whose cleanup also failed. The host must reserve admission and retry termination. */\nexport class VoiceCallSetupError extends SpeechProviderError {\n  constructor(readonly callId: string) {\n    super(\n      'openai-conversation',\n      'Voice setup failed and termination needs retry',\n    );\n  }\n}\n\nasync function readBody(\n  response: Response,\n  signal: AbortSignal,\n): Promise<string> {\n  const reader = response.body?.getReader();\n  if (!reader) return '';\n  const chunks: Uint8Array[] = [];\n  let size = 0;\n  const cancel = () => {\n    void reader.cancel().catch(() => undefined);\n  };\n  signal.addEventListener('abort', cancel, { once: true });\n  try {\n    while (true) {\n      signal.throwIfAborted();\n      const { done, value } = await reader.read();\n      signal.throwIfAborted();\n      if (done) break;\n      size += value.byteLength;\n      if (size > 65_536)\n        throw new SpeechProviderError(\n          'openai-conversation',\n          'Voice call response exceeded its size limit',\n        );\n      chunks.push(value);\n    }\n    const data = new Uint8Array(size);\n    let offset = 0;\n    for (const chunk of chunks) {\n      data.set(chunk, offset);\n      offset += chunk.byteLength;\n    }\n    return new TextDecoder().decode(data);\n  } finally {\n    signal.removeEventListener('abort', cancel);\n    await reader.cancel().catch(() => undefined);\n    reader.releaseLock();\n  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package/dist/conversation.d.ts

@@ -0,0 +1,4 @@
+/** Browser-safe, optional bidirectional voice conversation entry. */
+export { createOpenAIWebRTCVoiceSession } from './conversation/openai-webrtc.js';
+export type { OpenAIWebRTCVoiceOptions, VoiceSession, VoiceSessionEvents, VoiceSessionState, VoiceTranscript, } from './conversation/types.js';
+//# sourceMappingURL=conversation.d.ts.map

package/dist/conversation.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"conversation.d.ts","sourceRoot":"","sources":["../src/conversation.ts"],"names":[],"mappings":"AAAA,qEAAqE;AACrE,OAAO,EAAE,8BAA8B,EAAE,MAAM,iCAAiC,CAAC;AACjF,YAAY,EACV,wBAAwB,EACxB,YAAY,EACZ,kBAAkB,EAClB,iBAAiB,EACjB,eAAe,GAChB,MAAM,yBAAyB,CAAC"}

package/dist/conversation.js

@@ -0,0 +1,346 @@
+import { i as SpeechProviderError, n as SpeechConfigurationError } from "./chunks/errors-CQBX3OPS.js";
+//#region src/conversation/openai-webrtc.ts
+var ADAPTER = "openai-conversation";
+/** Browser WebRTC transport. No credentials, Node dependencies or globals are read at import. */
+function createOpenAIWebRTCVoiceSession(options) {
+	let state = "idle";
+	let pending;
+	let peer;
+	let channel;
+	let microphone;
+	let audio;
+	let micMuted = options.micMuted ?? false;
+	let outputMuted = options.outputMuted ?? false;
+	let responseId;
+	let responseActive = false;
+	let speaking = false;
+	const abort = new AbortController();
+	const listeners = /* @__PURE__ */ new Map();
+	const turns = /* @__PURE__ */ new Map();
+	const disposers = [];
+	function emit(event, value) {
+		for (const listener of listeners.get(event) ?? []) listener(value);
+	}
+	function change(next) {
+		state = next;
+		emit("state", next);
+	}
+	function playback(value) {
+		if (speaking === value) return;
+		speaking = value;
+		emit("speaking", value);
+	}
+	function cleanup() {
+		abort.abort();
+		for (const dispose of disposers.splice(0)) dispose();
+		microphone?.getTracks().forEach((track) => {
+			track.stop();
+		});
+		microphone = void 0;
+		if (channel) {
+			channel.onopen = channel.onmessage = channel.onclose = channel.onerror = null;
+			channel.close();
+		}
+		if (peer) {
+			peer.ontrack = peer.onconnectionstatechange = null;
+			peer.close();
+		}
+		if (audio) {
+			audio.pause();
+			audio.srcObject = null;
+		}
+		playback(false);
+		responseActive = false;
+		turns.clear();
+	}
+	function fail(message) {
+		if (state === "closed" || state === "failed") return;
+		state = "failed";
+		cleanup();
+		emit("state", state);
+		emit("error", new SpeechProviderError(ADAPTER, message));
+	}
+	function send(event) {
+		if (state !== "connected" || channel?.readyState !== "open") throw new SpeechConfigurationError("Voice session is not connected", ADAPTER);
+		channel.send(JSON.stringify(event));
+	}
+	function transcript(itemId, role, text, final, delta = false) {
+		if (typeof itemId !== "string" || typeof text !== "string") return;
+		const prior = turns.get(itemId);
+		if (prior?.final || prior && prior.role !== role) return;
+		const turn = {
+			itemId,
+			role,
+			text: delta ? (prior?.text ?? "") + text : text,
+			final
+		};
+		turns.set(itemId, turn);
+		emit("transcript", turn);
+	}
+	function receive(data) {
+		let event;
+		try {
+			const parsed = typeof data === "string" ? JSON.parse(data) : data;
+			if (!parsed || typeof parsed !== "object" || Array.isArray(parsed)) return;
+			event = parsed;
+		} catch {
+			return;
+		}
+		switch (event.type) {
+			case "conversation.item.input_audio_transcription.delta":
+				transcript(event.item_id, "user", event.delta, false, true);
+				break;
+			case "conversation.item.input_audio_transcription.completed":
+				transcript(event.item_id, "user", event.transcript, true);
+				break;
+			case "conversation.item.input_audio_transcription.failed":
+				emit("error", new SpeechProviderError(ADAPTER, "Input transcription failed"));
+				break;
+			case "response.output_audio_transcript.delta":
+				transcript(event.item_id, "assistant", event.delta, false, true);
+				break;
+			case "response.output_audio_transcript.done":
+				transcript(event.item_id, "assistant", event.transcript, true);
+				break;
+			case "response.output_text.delta":
+				transcript(event.item_id, "assistant", event.delta, false, true);
+				break;
+			case "response.output_text.done":
+				transcript(event.item_id, "assistant", event.text, true);
+				break;
+			case "input_audio_buffer.speech_started":
+				emit("inputSpeech", true);
+				break;
+			case "input_audio_buffer.speech_stopped":
+				emit("inputSpeech", false);
+				break;
+			case "output_audio_buffer.started":
+				playback(!outputMuted);
+				break;
+			case "output_audio_buffer.stopped":
+			case "output_audio_buffer.cleared":
+				playback(false);
+				break;
+			case "response.created": {
+				const response = record(event.response);
+				responseId = typeof response.id === "string" ? response.id : void 0;
+				responseActive = true;
+				emit("response", {
+					active: true,
+					responseId
+				});
+				break;
+			}
+			case "response.done": {
+				const response = record(event.response);
+				responseActive = false;
+				emit("response", {
+					active: false,
+					responseId
+				});
+				if (response.usage && typeof response.usage === "object") emit("usage", record(response.usage));
+				if (response.status === "failed" || response.status === "incomplete") emit("error", new SpeechProviderError(ADAPTER, response.status === "incomplete" ? "Voice response incomplete" : "Voice response failed"));
+				break;
+			}
+			case "response.function_call_arguments.done":
+				if (typeof event.call_id === "string" && typeof event.name === "string" && typeof event.arguments === "string") emit("toolCall", {
+					callId: event.call_id,
+					name: event.name,
+					arguments: event.arguments
+				});
+				break;
+			case "error":
+				emit("error", new SpeechProviderError(ADAPTER, "Realtime provider rejected a conversation event"));
+				break;
+		}
+	}
+	const session = {
+		get state() {
+			return state;
+		},
+		connect() {
+			if (pending) return pending;
+			if (state === "connected") return Promise.resolve();
+			if (state !== "idle") return Promise.reject(new SpeechConfigurationError("Create a new voice session after close or failure", ADAPTER));
+			const timeoutMs = options.connectTimeoutMs ?? 2e4;
+			if (!Number.isFinite(timeoutMs) || timeoutMs <= 0) return Promise.reject(new SpeechConfigurationError("connectTimeoutMs must be positive", ADAPTER));
+			change("connecting");
+			const signal = AbortSignal.any([abort.signal, ...options.signal ? [options.signal] : []]);
+			pending = (async () => {
+				const timeout = setTimeout(() => fail("Voice connection timed out"), timeoutMs);
+				const onAbort = () => {
+					if (state !== "closed" && state !== "failed") fail("Voice connection cancelled");
+				};
+				signal.addEventListener("abort", onAbort, { once: true });
+				disposers.push(() => signal.removeEventListener("abort", onAbort));
+				try {
+					signal.throwIfAborted();
+					const stream = await abortable(options.getMicrophone().then((stream) => {
+						if (signal.aborted) stream.getTracks().forEach((track) => {
+							track.stop();
+						});
+						return stream;
+					}), signal);
+					signal.throwIfAborted();
+					microphone = stream;
+					stream.getAudioTracks().forEach((track) => {
+						track.enabled = !micMuted;
+						const ended = () => fail("Microphone ended; allow access and start a new session");
+						track.addEventListener("ended", ended);
+						disposers.push(() => track.removeEventListener("ended", ended));
+						if (track.readyState === "ended") throw new SpeechConfigurationError("Microphone stream has ended", ADAPTER);
+					});
+					if (!stream.getAudioTracks().length) throw new SpeechConfigurationError("Microphone stream has no audio tracks", ADAPTER);
+					peer = options.createPeerConnection?.() ?? new RTCPeerConnection();
+					audio = options.audio ?? new Audio();
+					audio.autoplay = true;
+					audio.muted = outputMuted;
+					peer.ontrack = (event) => {
+						if (!audio || signal.aborted) return;
+						audio.srcObject = event.streams[0] ?? new MediaStream([event.track]);
+						audio.play().catch(() => emit("error", new SpeechProviderError(ADAPTER, "Audio playback needs a user gesture")));
+					};
+					peer.onconnectionstatechange = () => {
+						if (peer?.connectionState === "failed" || peer?.connectionState === "disconnected") fail("Voice connection lost; start a new session");
+					};
+					for (const track of stream.getAudioTracks()) peer.addTrack(track, stream);
+					channel = peer.createDataChannel("oai-events");
+					channel.onmessage = (event) => receive(event.data);
+					const opened = new Promise((resolve, reject) => {
+						const cancel = () => reject(new SpeechProviderError(ADAPTER, "Voice connection cancelled"));
+						signal.addEventListener("abort", cancel, { once: true });
+						disposers.push(() => signal.removeEventListener("abort", cancel));
+						if (channel) {
+							channel.onopen = () => {
+								signal.removeEventListener("abort", cancel);
+								resolve();
+							};
+							channel.onclose = () => {
+								fail("Voice channel closed");
+								reject(new SpeechProviderError(ADAPTER, "Voice channel closed"));
+							};
+							channel.onerror = () => {
+								fail("Voice channel failed");
+								reject(new SpeechProviderError(ADAPTER, "Voice channel failed"));
+							};
+						}
+					});
+					opened.catch(() => void 0);
+					const offer = await abortable(peer.createOffer(), signal);
+					await abortable(peer.setLocalDescription(offer), signal);
+					signal.throwIfAborted();
+					if (!offer.sdp) throw new SpeechConfigurationError("WebRTC did not produce an SDP offer", ADAPTER);
+					const answer = await abortable(options.negotiate(offer.sdp, signal), signal);
+					signal.throwIfAborted();
+					await abortable(peer.setRemoteDescription({
+						type: "answer",
+						sdp: answer
+					}), signal);
+					await opened;
+					signal.throwIfAborted();
+					change("connected");
+				} catch (error) {
+					fail("Unable to connect voice session");
+					throw error;
+				} finally {
+					clearTimeout(timeout);
+				}
+			})().finally(() => {
+				pending = void 0;
+			});
+			return pending;
+		},
+		sendText(text) {
+			const trimmed = text.trim();
+			if (!trimmed || trimmed.length > 16e3) throw new SpeechConfigurationError("Text must contain 1–16000 characters", ADAPTER);
+			const id = `hv_${crypto.randomUUID().replaceAll("-", "").slice(0, 24)}`;
+			send({
+				type: "conversation.item.create",
+				item: {
+					id,
+					type: "message",
+					role: "user",
+					content: [{
+						type: "input_text",
+						text: trimmed
+					}]
+				}
+			});
+			transcript(id, "user", trimmed, true);
+			session.respond();
+			return id;
+		},
+		commitInput(respond = true) {
+			send({ type: "input_audio_buffer.commit" });
+			if (respond) session.respond();
+		},
+		respond(instructions) {
+			send({
+				type: "response.create",
+				...instructions ? { response: { instructions } } : {}
+			});
+		},
+		interrupt() {
+			if (state !== "connected") return;
+			if (responseActive) send({
+				type: "response.cancel",
+				...responseId ? { response_id: responseId } : {}
+			});
+			send({ type: "output_audio_buffer.clear" });
+			playback(false);
+		},
+		setMicMuted(muted) {
+			micMuted = muted;
+			microphone?.getAudioTracks().forEach((track) => {
+				track.enabled = !muted;
+			});
+			if (muted && state === "connected") send({ type: "input_audio_buffer.clear" });
+		},
+		setOutputMuted(muted) {
+			outputMuted = muted;
+			if (audio) audio.muted = muted;
+			if (muted) playback(false);
+		},
+		submitToolResult(callId, result) {
+			send({
+				type: "conversation.item.create",
+				item: {
+					type: "function_call_output",
+					call_id: callId,
+					output: JSON.stringify(result)
+				}
+			});
+		},
+		close() {
+			if (state === "closed") return;
+			state = "closed";
+			cleanup();
+			emit("state", state);
+			listeners.clear();
+		},
+		on(event, listener) {
+			const set = listeners.get(event) ?? /* @__PURE__ */ new Set();
+			set.add(listener);
+			listeners.set(event, set);
+			return () => {
+				set.delete(listener);
+			};
+		}
+	};
+	return session;
+}
+function record(value) {
+	return value && typeof value === "object" && !Array.isArray(value) ? value : {};
+}
+function abortable(promise, signal) {
+	if (signal.aborted) return Promise.reject(signal.reason);
+	return new Promise((resolve, reject) => {
+		const cancel = () => reject(signal.reason);
+		signal.addEventListener("abort", cancel, { once: true });
+		promise.then(resolve, reject).finally(() => signal.removeEventListener("abort", cancel));
+	});
+}
+//#endregion
+export { createOpenAIWebRTCVoiceSession };
+
+//# sourceMappingURL=conversation.js.map

package/dist/conversation.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"conversation.js","names":[],"sources":["../src/conversation/openai-webrtc.ts"],"sourcesContent":["import {\n  SpeechConfigurationError,\n  SpeechProviderError,\n} from '../shared/errors.js';\nimport type {\n  OpenAIWebRTCVoiceOptions,\n  VoiceSession,\n  VoiceSessionEvents,\n  VoiceSessionState,\n  VoiceTranscript,\n} from './types.js';\n\nconst ADAPTER = 'openai-conversation';\n\n/** Browser WebRTC transport. No credentials, Node dependencies or globals are read at import. */\nexport function createOpenAIWebRTCVoiceSession(\n  options: OpenAIWebRTCVoiceOptions,\n): VoiceSession {\n  let state: VoiceSessionState = 'idle';\n  let pending: Promise<void> | undefined;\n  let peer: RTCPeerConnection | undefined;\n  let channel: RTCDataChannel | undefined;\n  let microphone: MediaStream | undefined;\n  let audio: HTMLAudioElement | undefined;\n  let micMuted = options.micMuted ?? false;\n  let outputMuted = options.outputMuted ?? false;\n  let responseId: string | undefined;\n  let responseActive = false;\n  let speaking = false;\n  const abort = new AbortController();\n  const listeners = new Map<\n    keyof VoiceSessionEvents,\n    Set<(value: never) => void>\n  >();\n  const turns = new Map<string, VoiceTranscript>();\n  const disposers: (() => void)[] = [];\n\n  function emit<E extends keyof VoiceSessionEvents>(\n    event: E,\n    value: VoiceSessionEvents[E],\n  ): void {\n    for (const listener of listeners.get(event) ?? []) listener(value as never);\n  }\n  function change(next: VoiceSessionState): void {\n    state = next;\n    emit('state', next);\n  }\n  function playback(value: boolean): void {\n    if (speaking === value) return;\n    speaking = value;\n    emit('speaking', value);\n  }\n  function cleanup(): void {\n    abort.abort();\n    for (const dispose of disposers.splice(0)) dispose();\n    microphone?.getTracks().forEach((track) => {\n      track.stop();\n    });\n    microphone = undefined;\n    if (channel) {\n      channel.onopen =\n        channel.onmessage =\n        channel.onclose =\n        channel.onerror =\n          null;\n      channel.close();\n    }\n    if (peer) {\n      peer.ontrack = peer.onconnectionstatechange = null;\n      peer.close();\n    }\n    if (audio) {\n      audio.pause();\n      audio.srcObject = null;\n    }\n    playback(false);\n    responseActive = false;\n    turns.clear();\n  }\n  function fail(message: string): void {\n    if (state === 'closed' || state === 'failed') return;\n    state = 'failed';\n    cleanup();\n    emit('state', state);\n    emit('error', new SpeechProviderError(ADAPTER, message));\n  }\n  function send(event: Record<string, unknown>): void {\n    if (state !== 'connected' || channel?.readyState !== 'open')\n      throw new SpeechConfigurationError(\n        'Voice session is not connected',\n        ADAPTER,\n      );\n    channel.send(JSON.stringify(event));\n  }\n  function transcript(\n    itemId: unknown,\n    role: VoiceTranscript['role'],\n    text: unknown,\n    final: boolean,\n    delta = false,\n  ): void {\n    if (typeof itemId !== 'string' || typeof text !== 'string') return;\n    const prior = turns.get(itemId);\n    // Late duplicates/deltas must not overwrite a final turn or its role.\n    if (prior?.final || (prior && prior.role !== role)) return;\n    const turn = {\n      itemId,\n      role,\n      text: delta ? (prior?.text ?? '') + text : text,\n      final,\n    };\n    turns.set(itemId, turn);\n    emit('transcript', turn);\n  }\n  function receive(data: unknown): void {\n    let event: Record<string, unknown>;\n    try {\n      const parsed: unknown =\n        typeof data === 'string' ? JSON.parse(data) : data;\n      if (!parsed || typeof parsed !== 'object' || Array.isArray(parsed))\n        return;\n      event = parsed as Record<string, unknown>;\n    } catch {\n      return;\n    }\n    switch (event.type) {\n      case 'conversation.item.input_audio_transcription.delta':\n        transcript(event.item_id, 'user', event.delta, false, true);\n        break;\n      case 'conversation.item.input_audio_transcription.completed':\n        transcript(event.item_id, 'user', event.transcript, true);\n        break;\n      case 'conversation.item.input_audio_transcription.failed':\n        emit(\n          'error',\n          new SpeechProviderError(ADAPTER, 'Input transcription failed'),\n        );\n        break;\n      case 'response.output_audio_transcript.delta':\n        transcript(event.item_id, 'assistant', event.delta, false, true);\n        break;\n      case 'response.output_audio_transcript.done':\n        transcript(event.item_id, 'assistant', event.transcript, true);\n        break;\n      case 'response.output_text.delta':\n        transcript(event.item_id, 'assistant', event.delta, false, true);\n        break;\n      case 'response.output_text.done':\n        transcript(event.item_id, 'assistant', event.text, true);\n        break;\n      case 'input_audio_buffer.speech_started':\n        emit('inputSpeech', true);\n        break;\n      case 'input_audio_buffer.speech_stopped':\n        emit('inputSpeech', false);\n        break;\n      case 'output_audio_buffer.started':\n        playback(!outputMuted);\n        break;\n      case 'output_audio_buffer.stopped':\n      case 'output_audio_buffer.cleared':\n        playback(false);\n        break;\n      case 'response.created': {\n        const response = record(event.response);\n        responseId = typeof response.id === 'string' ? response.id : undefined;\n        responseActive = true;\n        emit('response', { active: true, responseId });\n        break;\n      }\n      case 'response.done': {\n        const response = record(event.response);\n        responseActive = false;\n        emit('response', { active: false, responseId });\n        if (response.usage && typeof response.usage === 'object')\n          emit('usage', record(response.usage));\n        if (response.status === 'failed' || response.status === 'incomplete')\n          emit(\n            'error',\n            new SpeechProviderError(\n              ADAPTER,\n              response.status === 'incomplete'\n                ? 'Voice response incomplete'\n                : 'Voice response failed',\n            ),\n          );\n        break;\n      }\n      case 'response.function_call_arguments.done':\n        if (\n          typeof event.call_id === 'string' &&\n          typeof event.name === 'string' &&\n          typeof event.arguments === 'string'\n        )\n          emit('toolCall', {\n            callId: event.call_id,\n            name: event.name,\n            arguments: event.arguments,\n          });\n        break;\n      case 'error':\n        emit(\n          'error',\n          new SpeechProviderError(\n            ADAPTER,\n            'Realtime provider rejected a conversation event',\n          ),\n        );\n        break;\n    }\n  }\n\n  const session: VoiceSession = {\n    get state() {\n      return state;\n    },\n    connect() {\n      if (pending) return pending;\n      if (state === 'connected') return Promise.resolve();\n      if (state !== 'idle')\n        return Promise.reject(\n          new SpeechConfigurationError(\n            'Create a new voice session after close or failure',\n            ADAPTER,\n          ),\n        );\n      const timeoutMs = options.connectTimeoutMs ?? 20_000;\n      if (!Number.isFinite(timeoutMs) || timeoutMs <= 0)\n        return Promise.reject(\n          new SpeechConfigurationError(\n            'connectTimeoutMs must be positive',\n            ADAPTER,\n          ),\n        );\n      change('connecting');\n      const signal = AbortSignal.any([\n        abort.signal,\n        ...(options.signal ? [options.signal] : []),\n      ]);\n      pending = (async () => {\n        const timeout = setTimeout(\n          () => fail('Voice connection timed out'),\n          timeoutMs,\n        );\n        const onAbort = () => {\n          if (state !== 'closed' && state !== 'failed')\n            fail('Voice connection cancelled');\n        };\n        signal.addEventListener('abort', onAbort, { once: true });\n        disposers.push(() => signal.removeEventListener('abort', onAbort));\n        try {\n          signal.throwIfAborted();\n          // Permission acquisition cannot be aborted; dispose even if it resolves after close.\n          const acquisition = options.getMicrophone().then((stream) => {\n            if (signal.aborted)\n              stream.getTracks().forEach((track) => {\n                track.stop();\n              });\n            return stream;\n          });\n          const stream = await abortable(acquisition, signal);\n          signal.throwIfAborted();\n          microphone = stream;\n          stream.getAudioTracks().forEach((track) => {\n            track.enabled = !micMuted;\n            const ended = () =>\n              fail('Microphone ended; allow access and start a new session');\n            track.addEventListener('ended', ended);\n            disposers.push(() => track.removeEventListener('ended', ended));\n            if (track.readyState === 'ended')\n              throw new SpeechConfigurationError(\n                'Microphone stream has ended',\n                ADAPTER,\n              );\n          });\n          if (!stream.getAudioTracks().length)\n            throw new SpeechConfigurationError(\n              'Microphone stream has no audio tracks',\n              ADAPTER,\n            );\n          peer = options.createPeerConnection?.() ?? new RTCPeerConnection();\n          audio = options.audio ?? new Audio();\n          audio.autoplay = true;\n          audio.muted = outputMuted;\n          peer.ontrack = (event) => {\n            if (!audio || signal.aborted) return;\n            audio.srcObject =\n              event.streams[0] ?? new MediaStream([event.track]);\n            void audio\n              .play()\n              .catch(() =>\n                emit(\n                  'error',\n                  new SpeechProviderError(\n                    ADAPTER,\n                    'Audio playback needs a user gesture',\n                  ),\n                ),\n              );\n          };\n          peer.onconnectionstatechange = () => {\n            if (\n              peer?.connectionState === 'failed' ||\n              peer?.connectionState === 'disconnected'\n            )\n              fail('Voice connection lost; start a new session');\n          };\n          for (const track of stream.getAudioTracks())\n            peer.addTrack(track, stream);\n          channel = peer.createDataChannel('oai-events');\n          channel.onmessage = (event) => receive(event.data);\n          // Attach before SDP negotiation; some transports open during setRemoteDescription.\n          const opened = new Promise<void>((resolve, reject) => {\n            const cancel = () =>\n              reject(\n                new SpeechProviderError(ADAPTER, 'Voice connection cancelled'),\n              );\n            signal.addEventListener('abort', cancel, { once: true });\n            disposers.push(() => signal.removeEventListener('abort', cancel));\n            if (channel) {\n              channel.onopen = () => {\n                signal.removeEventListener('abort', cancel);\n                resolve();\n              };\n              channel.onclose = () => {\n                fail('Voice channel closed');\n                reject(\n                  new SpeechProviderError(ADAPTER, 'Voice channel closed'),\n                );\n              };\n              channel.onerror = () => {\n                fail('Voice channel failed');\n                reject(\n                  new SpeechProviderError(ADAPTER, 'Voice channel failed'),\n                );\n              };\n            }\n          });\n          // Keep an early rejection handled while negotiation is pending.\n          void opened.catch(() => undefined);\n          const offer = await abortable(peer.createOffer(), signal);\n          await abortable(peer.setLocalDescription(offer), signal);\n          signal.throwIfAborted();\n          if (!offer.sdp)\n            throw new SpeechConfigurationError(\n              'WebRTC did not produce an SDP offer',\n              ADAPTER,\n            );\n          const answer = await abortable(\n            options.negotiate(offer.sdp, signal),\n            signal,\n          );\n          signal.throwIfAborted();\n          await abortable(\n            peer.setRemoteDescription({ type: 'answer', sdp: answer }),\n            signal,\n          );\n          await opened;\n          signal.throwIfAborted();\n          change('connected');\n        } catch (error) {\n          fail('Unable to connect voice session');\n          throw error;\n        } finally {\n          clearTimeout(timeout);\n        }\n      })().finally(() => {\n        pending = undefined;\n      });\n      return pending;\n    },\n    sendText(text) {\n      const trimmed = text.trim();\n      if (!trimmed || trimmed.length > 16_000)\n        throw new SpeechConfigurationError(\n          'Text must contain 1–16000 characters',\n          ADAPTER,\n        );\n      const id = `hv_${crypto.randomUUID().replaceAll('-', '').slice(0, 24)}`;\n      send({\n        type: 'conversation.item.create',\n        item: {\n          id,\n          type: 'message',\n          role: 'user',\n          content: [{ type: 'input_text', text: trimmed }],\n        },\n      });\n      transcript(id, 'user', trimmed, true);\n      session.respond();\n      return id;\n    },\n    commitInput(respond = true) {\n      send({ type: 'input_audio_buffer.commit' });\n      if (respond) session.respond();\n    },\n    respond(instructions) {\n      send({\n        type: 'response.create',\n        ...(instructions ? 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package/dist/index.d.ts

@@ -21,9 +21,18 @@
  *
  * @packageDocumentation
  */
+export { OPENAI_REALTIME_DEFAULT_MODEL, OPENAI_REALTIME_DEFAULT_URL, openAIRealtimeProtocol, openAIRealtimeTranscriptionSession, parseOpenAIRealtimeEvent, resolveOpenAIRealtimeFormat, resolveRealtimeUrl, } from './adapters/openai-realtime.js';
+export { parseVoxtralRealtimeEvent, resolveVoxtralRealtimeUrl, VOXTRAL_REALTIME_AUDIO_FORMAT, VOXTRAL_REALTIME_DEFAULT_MODEL, voxtralRealtimeProtocol, } from './adapters/voxtral-realtime.js';
 export { DEFAULT_MAX_AUDIO_BYTES, mimeTypeToAudioExtension, } from './shared/audio.js';
 export { type SpeechEnv, TRANSCRIBER_ENV_KEYS, } from './shared/env.js';
 export * from './shared/errors.js';
 export { getAvailableSpeechAdapters, getSpeech, getSpeechSynthesizer, getTranscriber, type SpeechFactoryContext, } from './shared/factory.js';
+export { audioSecondsForBytes, bytesPerSecond, unwrapRawAudio, } from './shared/pcm.js';
+export { type RealtimeConnection, type RealtimeProtocol, type RealtimeProtocolEvent, type RealtimeSessionConfig, type RealtimeSessionInit, RealtimeTranscriptionSession, } from './shared/realtime-session.js';
+export { CLIENT_SECRET_TTL_RANGE, type CreateStreamingClientSecretOptions, createStreamingClientSecret, DEFAULT_CLIENT_SECRET_TTL_SECONDS, resolveClientSecretUrl, type StreamingClientSecretContext, type StreamingClientSecretResult, } from './shared/streaming-client-secret.js';
+export { getStreamingTranscriber, isStreamingTranscriberType, parseMaxTurnSeconds, STREAMING_TRANSCRIBER_ENV_KEYS, STREAMING_TRANSCRIBER_TYPES, type StreamingFactoryContext, type StreamingTranscriberWrapperOptions, wrapStreamingTranscriber, } from './shared/streaming-factory.js';
+export type { GetStreamingTranscriberOptions, OpenAIRealtimeTranscriberOptions, StreamingAudioChunk, StreamingAudioEncoding, StreamingAudioFormat, StreamingClientSecret, StreamingCloseEvent, StreamingFinalEvent, StreamingPartialEvent, StreamingRolloverOptions, StreamingSession, StreamingSessionEventName, StreamingSessionEvents, StreamingSessionListener, StreamingSessionOptions, StreamingSessionSettings, StreamingSessionState, StreamingSpeechEvent, StreamingTranscriber, StreamingTranscriberOptions, StreamingTranscriberType, StreamingTurnDetection, StreamingTurnLimit, VoxtralRealtimeTranscriberOptions, } from './shared/streaming-types.js';
+export { DEFAULT_MAX_TURN_SECONDS, DEFAULT_ROLLOVER_MIN_SILENCE_MS, DEFAULT_ROLLOVER_SILENCE_THRESHOLD, DEFAULT_ROLLOVER_WINDOW_SECONDS, } from './shared/turn-limit.js';
 export type { AudioBytes, AudioInput, AudioSource, GetSpeechOptions, GetSpeechSynthesizerOptions, GetTranscriberOptions, OpenAICompatibleSpeechSynthesizerOptions, OpenAICompatibleTranscriberOptions, Qwen3SpeechSynthesizerOptions, Speech, SpeechAdapterAvailability, SpeechAdapterType, SpeechFetch, SpeechOperation, SpeechRetryOptions, SpeechSynthesizer, SpeechSynthesizerType, SpeechUsage, SpeechUsageCallback, SpeechVoice, SpeechVoiceInput, StudioServerSpeechSynthesizerOptions, StudioServerTranscriberOptions, SynthesisRequest, SynthesizedSpeech, TimestampGranularity, Transcriber, TranscriberType, TranscriptionRequest, TranscriptionResponseFormat, TranscriptResult, TranscriptSegment, WordTiming, } from './shared/types.js';
+export { isBrowserRuntime, type SpeechWebSocket, type SpeechWebSocketConstructor, type SpeechWebSocketFactory, type SpeechWebSocketInit, } from './shared/websocket.js';
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

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