thursday-agent 0.4.0 → 0.6.0

package/lib/realtime/realtime.transport.ts

@@ -1,384 +0,0 @@
-import {
-  createPcmPlayer,
-  decodePcm,
-  encodePcm,
-  MICROPHONE_CONSTRAINTS,
-  type Microphone,
-  openMicrophone,
-  PCM_SAMPLE_RATE,
-  type PcmPlayer,
-} from "./realtime.audio";
-import type {
-  RealtimeClientEvent,
-  RealtimeServerEvent,
-} from "./realtime.protocol";
-import type { RealtimeAudio } from "./realtime.session";
-
-/**
- * Transports own the audio and report events plus whether output audio is
- * playing. WebRTC carries audio as tracks and the server handles barge-in
- * truncation; WebSocket carries PCM16 in JSON and this side plays, stops and
- * truncates by hand.
- */
-
-export type TransportEvents = {
-  event: (event: RealtimeServerEvent) => void;
-  /** Output audio started or stopped playing. */
-  playing: (playing: boolean) => void;
-  /** The wire went away on its own. Fatal, reported once. */
-  dropped: (message: string) => void;
-};
-
-export type RealtimeTransport = {
-  /** Resolves once events can be sent. Rejects if the wire never opens. */
-  connect(): Promise<void>;
-  /** Silently dropped while the wire is not open. */
-  send(event: RealtimeClientEvent): void;
-  close(): void;
-  mute(muted: boolean): void;
-};
-
-type TransportOptions = {
-  /** Provider name for user-facing messages. */
-  label: string;
-  audio: RealtimeAudio;
-  on: TransportEvents;
-};
-
-/**
- * WebRTC: mic as a track, SDP offer POSTed with the ephemeral secret, remote
- * audio to `audio.element` and `audio.listen`, events on the "oai-events" channel.
- */
-export function createWebRtcTransport({
-  url,
-  secret,
-  label,
-  audio,
-  on,
-}: TransportOptions & {
-  /** Where the offer is POSTed. */
-  url: string;
-  secret: string;
-}): RealtimeTransport {
-  let peer: RTCPeerConnection | null = null;
-  let channel: RTCDataChannel | null = null;
-  let microphone: MediaStream | null = null;
-  let closed = false;
-  let opened = false;
-  let dropped = false;
-  let playing = false;
-
-  const setPlaying = (next: boolean) => {
-    if (playing === next) return;
-    playing = next;
-    on.playing(next);
-  };
-
-  const drop = (message: string) => {
-    if (closed || dropped) return;
-    dropped = true;
-    on.dropped(message);
-  };
-
-  const stopTracks = (stream: MediaStream | null) => {
-    for (const track of stream?.getTracks() ?? []) track.stop();
-  };
-
-  const close = () => {
-    closed = true;
-    stopTracks(microphone);
-    microphone = null;
-    channel?.close();
-    channel = null;
-    peer?.close();
-    peer = null;
-    // The element outlives the call; a dead stream keeps the speaker indicator lit.
-    audio.element.srcObject = null;
-  };
-
-  return {
-    async connect() {
-      const connection = new RTCPeerConnection();
-      peer = connection;
-      const events = connection.createDataChannel("oai-events");
-      channel = events;
-
-      // Before the channel opens a failure rejects connect; after, it is a drop.
-      const ready = new Promise<void>((resolve, reject) => {
-        const gone = (message: string) =>
-          opened ? drop(message) : reject(new Error(message));
-        events.onopen = () => {
-          opened = true;
-          resolve();
-        };
-        events.onclose = () => gone(`The connection to ${label} closed`);
-        events.onerror = () => gone(`The connection to ${label} dropped`);
-        connection.onconnectionstatechange = () => {
-          const state = connection.connectionState;
-          if (state === "failed" || state === "closed") {
-            gone(`Could not reach ${label}`);
-          }
-        };
-      });
-
-      events.onmessage = (message) => {
-        if (typeof message.data !== "string") return;
-        let event: RealtimeServerEvent;
-        try {
-          event = JSON.parse(message.data);
-        } catch {
-          // One unreadable frame is not worth ending a call over.
-          return;
-        }
-        // The server plays the audio, so it reports playback state.
-        switch (event.type) {
-          case "output_audio_buffer.started":
-            setPlaying(true);
-            break;
-          case "output_audio_buffer.stopped":
-          case "output_audio_buffer.cleared":
-            setPlaying(false);
-            break;
-        }
-        on.event(event);
-      };
-
-      connection.ontrack = (event) => {
-        const [stream] = event.streams;
-        audio.element.srcObject = stream;
-        audio.listen(stream);
-      };
-
-      const stream = await navigator.mediaDevices.getUserMedia({
-        audio: MICROPHONE_CONSTRAINTS,
-      });
-      if (closed) {
-        // Hung up while the mic permission was pending.
-        stopTracks(stream);
-        throw new Error("Hung up before the call opened.");
-      }
-      microphone = stream;
-      audio.hear?.(stream);
-      connection.addTrack(stream.getAudioTracks()[0], stream);
-
-      const offer = await connection.createOffer();
-      await connection.setLocalDescription(offer);
-      const response = await fetch(url, {
-        method: "POST",
-        body: offer.sdp,
-        headers: {
-          "Content-Type": "application/sdp",
-          Authorization: `Bearer ${secret}`,
-        },
-      });
-      if (!response.ok) throw new Error(await describeRefusal(response, label));
-      await connection.setRemoteDescription({
-        type: "answer",
-        sdp: await response.text(),
-      });
-
-      await ready;
-    },
-
-    send(event) {
-      if (channel?.readyState !== "open") return;
-      channel.send(JSON.stringify(event));
-    },
-
-    close,
-
-    mute(muted) {
-      for (const track of microphone?.getAudioTracks() ?? []) {
-        track.enabled = !muted;
-      }
-    },
-  };
-}
-
-/**
- * WebSocket: audio deltas are decoded and played here. On barge-in the player
- * stops and `conversation.item.truncate` tells the model how much was heard,
- * or it continues as if the whole answer was said.
- */
-export function createSocketTransport({
-  url,
-  protocols,
-  label,
-  audio,
-  on,
-}: TransportOptions & {
-  url: string;
-  /** Carries the secret; a browser cannot set WebSocket headers. */
-  protocols: string[];
-}): RealtimeTransport {
-  let socket: WebSocket | null = null;
-  let microphone: Microphone | null = null;
-  let player: PcmPlayer | null = null;
-  let closed = false;
-  let dropped = false;
-  /** Before `open` a failure rejects connect; after, it is a drop. */
-  let opened = false;
-  let playing = false;
-  /** Item being played and how much has arrived; what a truncate measures against. */
-  let current: {
-    itemId: string;
-    contentIndex: number;
-    startedAt: number;
-    receivedMs: number;
-  } | null = null;
-
-  const setPlaying = (next: boolean) => {
-    if (playing === next) return;
-    playing = next;
-    on.playing(next);
-  };
-
-  const drop = (message: string) => {
-    if (closed || dropped) return;
-    dropped = true;
-    on.dropped(message);
-  };
-
-  const send = (event: RealtimeClientEvent) => {
-    if (socket?.readyState !== WebSocket.OPEN) return;
-    socket.send(JSON.stringify(event));
-  };
-
-  /** Stops playback and tells the model where the user stopped listening. */
-  const bargeIn = () => {
-    if (!player?.speaking || !current) return;
-    const heardMs = Math.min(
-      current.receivedMs,
-      performance.now() - current.startedAt,
-    );
-    send({
-      type: "conversation.item.truncate",
-      item_id: current.itemId,
-      content_index: current.contentIndex,
-      audio_end_ms: Math.max(0, Math.floor(heardMs)),
-    });
-    player.stop();
-    current = null;
-    setPlaying(false);
-  };
-
-  const handle = (event: RealtimeServerEvent) => {
-    switch (event.type) {
-      case "response.output_audio.delta": {
-        if (!player || !event.delta) break;
-        const samples = decodePcm(event.delta);
-        if (current?.itemId !== event.item_id) {
-          current = {
-            itemId: event.item_id,
-            contentIndex: event.content_index ?? 0,
-            startedAt: performance.now(),
-            receivedMs: 0,
-          };
-        }
-        current.receivedMs += (samples.length / PCM_SAMPLE_RATE) * 1000;
-        player.push(samples);
-        setPlaying(true);
-        break;
-      }
-      case "input_audio_buffer.speech_started":
-        bargeIn();
-        break;
-    }
-    on.event(event);
-  };
-
-  const close = () => {
-    closed = true;
-    player?.stop();
-    player = null;
-    microphone?.close();
-    microphone = null;
-    const live = socket;
-    socket = null;
-    live?.close();
-  };
-
-  return {
-    async connect() {
-      player = createPcmPlayer(audio, () => {
-        current = null;
-        setPlaying(false);
-      });
-
-      const opening = new Promise<WebSocket>((resolve, reject) => {
-        const live = new WebSocket(url, protocols);
-        socket = live;
-        live.onopen = () => {
-          opened = true;
-          resolve(live);
-        };
-        // Attached before any await: the socket can open and die while the mic
-        // permission dialog is still up. A refused socket fires onerror then onclose.
-        const gone = (message: string) =>
-          opened ? drop(message) : reject(new Error(message));
-        live.onerror = () => gone(`Could not reach ${label}`);
-        live.onclose = (event) =>
-          gone(
-            event.reason || `The connection to ${label} closed (${event.code})`,
-          );
-      });
-
-      // Requested in parallel with the socket handshake.
-      const granting = openMicrophone((samples) =>
-        send({ type: "input_audio_buffer.append", audio: encodePcm(samples) }),
-      ).then(
-        (granted) => {
-          // Hung up while the mic permission was pending.
-          if (closed) granted.close();
-          else {
-            microphone = granted;
-            audio.hear?.(granted.stream);
-          }
-        },
-        (cause) => {
-          throw new Error(
-            cause instanceof Error
-              ? `Microphone unavailable — ${cause.message}`
-              : "Microphone unavailable",
-          );
-        },
-      );
-
-      const [live] = await Promise.all([opening, granting]);
-      if (closed) throw new Error("Hung up before the call opened.");
-
-      live.onmessage = (message) => {
-        if (typeof message.data !== "string") return;
-        let event: RealtimeServerEvent;
-        try {
-          event = JSON.parse(message.data);
-        } catch {
-          // One unreadable frame is not worth ending a call over.
-          return;
-        }
-        handle(event);
-      };
-      // onerror/onclose stay as set above; with `opened` true they route to `drop`.
-    },
-
-    send,
-
-    close,
-
-    mute(muted) {
-      microphone?.setMuted(muted);
-    },
-  };
-}
-
-/** Error message from the body when there is one. */
-async function describeRefusal(response: Response, label: string) {
-  const text = await response.text().catch(() => "");
-  try {
-    const message = JSON.parse(text)?.error?.message;
-    if (typeof message === "string" && message) return message;
-  } catch {
-    // Not JSON; the status code is all there is.
-  }
-  return `${label} refused the call (${response.status})`;
-}

package/lib/realtime/xai.ts

@@ -1,39 +0,0 @@
-import { PCM_SAMPLE_RATE } from "./realtime.audio";
-import { realtimeSession, toFunctionTools } from "./realtime.driver";
-import type { XaiSessionConfig } from "./realtime.protocol";
-import { createSocketTransport } from "./realtime.transport";
-
-/** xAI Speech to Speech over WebSocket (PCM16 both ways); the protocol on top is OpenAI's. */
-
-const REALTIME_URL = "wss://api.x.ai/v1/realtime";
-
-export const createXaiSession = realtimeSession({
-  transport: ({ credential, setup, audio, on }) =>
-    createSocketTransport({
-      url: `${REALTIME_URL}?model=${encodeURIComponent(setup.model)}`,
-      // A browser cannot set WebSocket headers, so the secret travels as a subprotocol.
-      protocols: [`xai-client-secret.${credential.value}`],
-      label: "Grok",
-      audio,
-      on,
-    }),
-
-  sessionUpdate: (setup) => ({
-    type: "session.update",
-    session: {
-      model: setup.model,
-      instructions: setup.instructions,
-      voice: setup.voice,
-      turn_detection: { type: "server_vad" },
-      audio: {
-        input: {
-          format: { type: "audio/pcm", rate: PCM_SAMPLE_RATE },
-          // Naming the model makes transcripts stream while the user is still speaking.
-          transcription: { model: "grok-transcribe" },
-        },
-        output: { format: { type: "audio/pcm", rate: PCM_SAMPLE_RATE } },
-      },
-      tools: toFunctionTools(setup.tools),
-    } satisfies XaiSessionConfig,
-  }),
-});