@arbidocs/react 0.5.111 → 0.5.113

package/dist/calls/recording-media.js

@@ -0,0 +1,218 @@
+import { continuityPrompt } from "@arbidocs/sdk/browser";
+import { conversationKeyForCall, decryptPart, unwrapTrackKey } from "./recording.js";
+import {
+  EXPORT_SAMPLE_RATE,
+  RECORDING_SEGMENTER,
+  createTrackCutter,
+  encodeWav,
+  floatToWav,
+  extensionFor,
+  fileSafe,
+  splitIntoFiles,
+  standalone,
+  timelineDurationMs,
+  timelineOf,
+  writeSamples
+} from "./recording-playback.js";
+import { storeTranscriptSegment } from "./transcription.js";
+async function openRecording(arbi, queryClient, call, recordingExtId, loaded) {
+  const recording = loaded ?? await arbi.calls.recording.get(call.external_id, recordingExtId);
+  const convKeys = /* @__PURE__ */ new Map();
+  const keys = /* @__PURE__ */ new Map();
+  for (const track of recording.tracks ?? []) {
+    const version = track.key_version ?? null;
+    const id = String(version);
+    if (!convKeys.has(id)) {
+      convKeys.set(
+        id,
+        conversationKeyForCall(arbi, queryClient, call, { keyVersion: version }).then((c) => c.key)
+      );
+    }
+    try {
+      keys.set(track.external_id, await unwrapTrackKey(track.wrapped_key, await convKeys.get(id)));
+    } catch (err) {
+      console.warn("[calls] a recording track key does not open", err);
+    }
+  }
+  return { recording, keys };
+}
+function speakerTracks(recording) {
+  return (recording.tracks ?? []).filter((t) => t.kind === "speaker");
+}
+function compositeTrack(recording) {
+  return (recording.tracks ?? []).find((t) => t.kind === "composite") ?? null;
+}
+function sortedParts(track) {
+  return [...track.parts ?? []].sort((a, b) => a.seq - b.seq);
+}
+async function fetchPart(arbi, part, fileKey) {
+  if (!part.download_url) throw new Error("This recording\u2019s media was deleted");
+  return decryptPart(await arbi.calls.recording.downloadPart(part.download_url), fileKey);
+}
+async function compositeFiles(arbi, track, fileKey) {
+  const parts = [];
+  for (const part of sortedParts(track)) parts.push(await fetchPart(arbi, part, fileKey));
+  const type = track.mime_type.split(";")[0] || "video/webm";
+  return splitIntoFiles(parts).map((file) => new Blob(file.map(standalone), { type }));
+}
+async function decodeMono16k(bytes) {
+  const ctx = new OfflineAudioContext(1, 1, EXPORT_SAMPLE_RATE);
+  const buffer = await ctx.decodeAudioData(standalone(bytes).buffer);
+  return buffer.getChannelData(0);
+}
+async function speakerWav(arbi, track, fileKey, onPart) {
+  const parts = sortedParts(track);
+  const totalMs = timelineDurationMs(timelineOf([track]));
+  const samples = new Int16Array(Math.ceil(totalMs / 1e3 * EXPORT_SAMPLE_RATE) + 1);
+  for (const part of parts) {
+    const bytes = await fetchPart(arbi, part, fileKey);
+    try {
+      writeSamples(
+        samples,
+        await decodeMono16k(bytes),
+        part.offset_ms / 1e3 * EXPORT_SAMPLE_RATE
+      );
+    } catch (err) {
+      console.warn("[calls] a recording part does not decode", err);
+    }
+    onPart?.();
+  }
+  return new Blob([encodeWav(samples, EXPORT_SAMPLE_RATE)], { type: "audio/wav" });
+}
+async function recordingFiles(arbi, opened, title, onProgress) {
+  const { recording, keys } = opened;
+  const base = fileSafe(title || "Meeting");
+  const date = recording.started_at.slice(0, 10);
+  const total = (recording.tracks ?? []).reduce((n, t) => n + (t.parts?.length ?? 0), 0);
+  let done = 0;
+  const tick = () => onProgress?.(++done, total);
+  const files = [];
+  for (const track of speakerTracks(recording)) {
+    const key = keys.get(track.external_id);
+    if (!key || !track.parts?.length) continue;
+    const wav = await speakerWav(arbi, track, key, tick);
+    const who = fileSafe(track.speaker_name || track.speaker_ext_id || "speaker");
+    files.push(new File([wav], `${base}-${date}-${who}.wav`, { type: "audio/wav" }));
+  }
+  const composite = compositeTrack(recording);
+  const compositeKey = composite && keys.get(composite.external_id);
+  if (composite && compositeKey && composite.parts?.length) {
+    const videos = await compositeFiles(arbi, composite, compositeKey);
+    done += composite.parts.length;
+    onProgress?.(done, total);
+    const ext = extensionFor(composite.mime_type);
+    videos.forEach(
+      (blob, i) => files.push(
+        new File([blob], `${base}-${date}-video${videos.length > 1 ? `-${i + 1}` : ""}.${ext}`, {
+          type: blob.type
+        })
+      )
+    );
+  }
+  return files;
+}
+const PROMPT_HISTORY_CHARS = 400;
+async function transcribeRecording(params) {
+  const {
+    arbi,
+    queryClient,
+    callExtId,
+    opened,
+    ref,
+    onProgress,
+    decode = decodeMono16k,
+    language
+  } = params;
+  const { recording, keys } = opened;
+  const tracks = speakerTracks(recording).filter((t) => keys.has(t.external_id));
+  const total = tracks.reduce((n, t) => n + (t.parts?.length ?? 0), 0);
+  const start = Date.parse(recording.started_at);
+  let done = 0;
+  let stored = 0;
+  let failed = 0;
+  for (const track of tracks) {
+    const key = keys.get(track.external_id);
+    const cutter = createTrackCutter(EXPORT_SAMPLE_RATE, RECORDING_SEGMENTER, { mergeShort: true });
+    let previous = "";
+    let nextSegment = 0;
+    const send = async (request) => {
+      try {
+        const result = await arbi.calls.recording.transcribe(callExtId, recording.external_id, {
+          audio: request.audio,
+          trackExtId: track.external_id,
+          seq: request.seq ?? null,
+          segment: request.segment ?? null,
+          segmentOffsetMs: request.seq == null ? request.offsetMs : 0,
+          language,
+          prompt: continuityPrompt(previous)
+        });
+        const text = result.text.trim();
+        if (!text) return;
+        previous = `${previous} ${text}`.slice(-PROMPT_HISTORY_CHARS);
+        const segment = {
+          segmentId: result.segment_id || (request.seq != null ? `rec-${track.external_id}-${request.seq}` : `rec-${track.external_id}-s${request.segment}`),
+          text,
+          speakerIdentity: result.speaker_ext_id ?? track.speaker_ext_id ?? "",
+          speakerName: result.speaker_name ?? track.speaker_name ?? "Speaker",
+          startedAt: Number.isFinite(start) ? new Date(start + (result.offset_ms ?? Math.round(request.offsetMs))).toISOString() : null
+        };
+        await storeTranscriptSegment(arbi, queryClient, ref, segment);
+        stored += 1;
+      } catch (err) {
+        failed += 1;
+        console.warn("[calls] a recording piece could not be transcribed", err);
+      }
+    };
+    const sendPieces = async (pieces) => {
+      for (const piece of pieces) {
+        await send({
+          audio: new Blob([floatToWav(piece.samples, EXPORT_SAMPLE_RATE)], { type: "audio/wav" }),
+          segment: nextSegment++,
+          offsetMs: piece.start / EXPORT_SAMPLE_RATE * 1e3
+        });
+      }
+    };
+    for (const part of sortedParts(track)) {
+      let bytes;
+      try {
+        bytes = await fetchPart(arbi, part, key);
+      } catch (err) {
+        failed += 1;
+        console.warn("[calls] a recording part could not be read", err);
+        onProgress?.(++done, total);
+        continue;
+      }
+      let pcm = null;
+      try {
+        pcm = await decode(bytes);
+      } catch (err) {
+        console.warn("[calls] a recording part does not decode; sending it whole", err);
+      }
+      if (pcm) {
+        await sendPieces(cutter.push(part.offset_ms, pcm));
+      } else {
+        await send({
+          audio: new Blob([standalone(bytes)], { type: track.mime_type }),
+          seq: part.seq,
+          offsetMs: part.offset_ms
+        });
+      }
+      onProgress?.(++done, total);
+    }
+    await sendPieces(cutter.finish());
+  }
+  return { stored, failed };
+}
+export {
+  compositeFiles,
+  compositeTrack,
+  decodeMono16k,
+  fetchPart,
+  openRecording,
+  recordingFiles,
+  sortedParts,
+  speakerTracks,
+  speakerWav,
+  transcribeRecording
+};
+//# sourceMappingURL=recording-media.js.map

package/dist/calls/recording-media.js.map

@@ -0,0 +1,7 @@
+{
+  "version": 3,
+  "sources": ["../../src/calls/recording-media.ts"],
+  "sourcesContent": ["/**\n * A stored call recording, opened in the browser: keys unwrapped, parts downloaded and decrypted\n * on demand, exported as files, and transcribed (ARBI `docs/meeting-chats-plan.md` \u00A76).\n * See `recording-playback.ts` for the playback and export formats.\n */\n\nimport type { QueryClient } from '@tanstack/react-query'\nimport { continuityPrompt, type Arbi, type calls } from '@arbidocs/sdk/browser'\nimport { conversationKeyForCall, decryptPart, unwrapTrackKey } from './recording'\nimport {\n  EXPORT_SAMPLE_RATE,\n  RECORDING_SEGMENTER,\n  createTrackCutter,\n  encodeWav,\n  floatToWav,\n  extensionFor,\n  fileSafe,\n  splitIntoFiles,\n  standalone,\n  timelineDurationMs,\n  timelineOf,\n  writeSamples,\n  type TrackPiece,\n} from './recording-playback'\nimport { storeTranscriptSegment, type TranscriptSegment } from './transcription'\nimport type { MeetingChatRef } from './meeting-chat'\n\nexport interface OpenedRecording {\n  recording: calls.CallRecording\n  /** File key per track ext id (a track whose key does not open is absent). */\n  keys: Map<string, Uint8Array>\n}\n\n/** Read a recording and unwrap its tracks' file keys under the call's conversation key. */\nexport async function openRecording(\n  arbi: Arbi,\n  queryClient: QueryClient,\n  call: calls.CallResponse,\n  recordingExtId: string,\n  loaded?: calls.CallRecording\n): Promise<OpenedRecording> {\n  const recording = loaded ?? (await arbi.calls.recording.get(call.external_id, recordingExtId))\n  const convKeys = new Map<string, Promise<Uint8Array>>()\n  const keys = new Map<string, Uint8Array>()\n  for (const track of recording.tracks ?? []) {\n    const version = track.key_version ?? null\n    const id = String(version)\n    if (!convKeys.has(id)) {\n      convKeys.set(\n        id,\n        conversationKeyForCall(arbi, queryClient, call, { keyVersion: version }).then((c) => c.key)\n      )\n    }\n    try {\n      keys.set(track.external_id, await unwrapTrackKey(track.wrapped_key, await convKeys.get(id)!))\n    } catch (err) {\n      console.warn('[calls] a recording track key does not open', err)\n    }\n  }\n  return { recording, keys }\n}\n\nexport function speakerTracks(recording: calls.CallRecording): calls.RecordingTrack[] {\n  return (recording.tracks ?? []).filter((t) => t.kind === 'speaker')\n}\n\nexport function compositeTrack(recording: calls.CallRecording): calls.RecordingTrack | null {\n  return (recording.tracks ?? []).find((t) => t.kind === 'composite') ?? null\n}\n\nexport function sortedParts(track: calls.RecordingTrack): calls.RecordingPart[] {\n  return [...(track.parts ?? [])].sort((a, b) => a.seq - b.seq)\n}\n\n/** One part's plaintext. */\nexport async function fetchPart(\n  arbi: Arbi,\n  part: calls.RecordingPart,\n  fileKey: Uint8Array\n): Promise<Uint8Array> {\n  if (!part.download_url) throw new Error('This recording\u2019s media was deleted')\n  return decryptPart(await arbi.calls.recording.downloadPart(part.download_url), fileKey)\n}\n\n/** The composite as playable files (normally one). */\nexport async function compositeFiles(\n  arbi: Arbi,\n  track: calls.RecordingTrack,\n  fileKey: Uint8Array\n): Promise<Blob[]> {\n  const parts: Uint8Array[] = []\n  for (const part of sortedParts(track)) parts.push(await fetchPart(arbi, part, fileKey))\n  const type = track.mime_type.split(';')[0] || 'video/webm'\n  return splitIntoFiles(parts).map((file) => new Blob(file.map(standalone), { type }))\n}\n\n/** Decode a standalone audio part to mono samples at 16 kHz (decodeAudioData resamples). */\nexport async function decodeMono16k(bytes: Uint8Array): Promise<Float32Array> {\n  const ctx = new OfflineAudioContext(1, 1, EXPORT_SAMPLE_RATE)\n  const buffer = await ctx.decodeAudioData(standalone(bytes).buffer)\n  return buffer.getChannelData(0)\n}\n\n/** A speaker track as ONE WAV (16 kHz mono), its parts at their offsets from the recording start. */\nexport async function speakerWav(\n  arbi: Arbi,\n  track: calls.RecordingTrack,\n  fileKey: Uint8Array,\n  onPart?: () => void\n): Promise<Blob> {\n  const parts = sortedParts(track)\n  const totalMs = timelineDurationMs(timelineOf([track]))\n  const samples = new Int16Array(Math.ceil((totalMs / 1000) * EXPORT_SAMPLE_RATE) + 1)\n  // decodeAudioData resamples to the context's rate.\n  for (const part of parts) {\n    const bytes = await fetchPart(arbi, part, fileKey)\n    try {\n      writeSamples(\n        samples,\n        await decodeMono16k(bytes),\n        (part.offset_ms / 1000) * EXPORT_SAMPLE_RATE\n      )\n    } catch (err) {\n      console.warn('[calls] a recording part does not decode', err)\n    }\n    onPart?.()\n  }\n  return new Blob([encodeWav(samples, EXPORT_SAMPLE_RATE)], { type: 'audio/wav' })\n}\n\n/** Every decrypted file of a recording: one WAV per speaker, the composite as one video. */\nexport async function recordingFiles(\n  arbi: Arbi,\n  opened: OpenedRecording,\n  title: string | null | undefined,\n  onProgress?: (done: number, total: number) => void\n): Promise<File[]> {\n  const { recording, keys } = opened\n  const base = fileSafe(title || 'Meeting')\n  const date = recording.started_at.slice(0, 10)\n  const total = (recording.tracks ?? []).reduce((n, t) => n + (t.parts?.length ?? 0), 0)\n  let done = 0\n  const tick = () => onProgress?.(++done, total)\n  const files: File[] = []\n  for (const track of speakerTracks(recording)) {\n    const key = keys.get(track.external_id)\n    if (!key || !track.parts?.length) continue\n    const wav = await speakerWav(arbi, track, key, tick)\n    const who = fileSafe(track.speaker_name || track.speaker_ext_id || 'speaker')\n    files.push(new File([wav], `${base}-${date}-${who}.wav`, { type: 'audio/wav' }))\n  }\n  const composite = compositeTrack(recording)\n  const compositeKey = composite && keys.get(composite.external_id)\n  if (composite && compositeKey && composite.parts?.length) {\n    const videos = await compositeFiles(arbi, composite, compositeKey)\n    done += composite.parts.length\n    onProgress?.(done, total)\n    const ext = extensionFor(composite.mime_type)\n    videos.forEach((blob, i) =>\n      files.push(\n        new File([blob], `${base}-${date}-video${videos.length > 1 ? `-${i + 1}` : ''}.${ext}`, {\n          type: blob.type,\n        })\n      )\n    )\n  }\n  return files\n}\n\n/** A track's previous text, kept for the continuity prompt (a little more than it sends). */\nconst PROMPT_HISTORY_CHARS = 400\n\n/**\n * The host's \"Transcribe\": each speaker track is decrypted and decoded here part by part, its parts\n * placed on ONE continuous timeline by their offsets (gaps are silence) and cut at the track's\n * pauses ({@link createTrackCutter} with {@link RECORDING_SEGMENTER}: the live 700 ms pause rule,\n * pieces up to 25 s), so no sentence is split at a part boundary. Each piece (a 16 kHz mono WAV) is\n * sent through the shared transcription helper WITHOUT a part `seq`: its index across the track\n * (`segment`) and its start from the recording start (`segment_offset_ms`) \u2192 segment id\n * `rec-<track>-s<n>`. The user's language and the speaker's previous text (continuity prompt) go\n * with every piece.\n *\n * Each line is stored as an attributed transcript entry exactly like a live one (P5) \u2014 an E2E chat\n * message of kind `transcript` with `dedupe_key` = the segment id, or a thread comment with the\n * transcript marker \u2014 so running it twice stores each line once; the recording's lines then\n * supersede the live ones of the same window. A part this browser cannot decode is sent whole\n * (`rec-<track>-<seq>`) and is silence on the timeline.\n */\nexport async function transcribeRecording(params: {\n  arbi: Arbi\n  queryClient: QueryClient\n  callExtId: string\n  opened: OpenedRecording\n  ref: MeetingChatRef\n  onProgress?(done: number, total: number): void\n  /** Decoder (tests); default WebAudio at 16 kHz. */\n  decode?(bytes: Uint8Array): Promise<Float32Array>\n  /** ISO-639-1 spoken language, when known; otherwise the transcriber detects it. */\n  language?: string\n}): Promise<{ stored: number; failed: number }> {\n  const {\n    arbi,\n    queryClient,\n    callExtId,\n    opened,\n    ref,\n    onProgress,\n    decode = decodeMono16k,\n    language,\n  } = params\n  const { recording, keys } = opened\n  const tracks = speakerTracks(recording).filter((t) => keys.has(t.external_id))\n  const total = tracks.reduce((n, t) => n + (t.parts?.length ?? 0), 0)\n  const start = Date.parse(recording.started_at)\n  let done = 0\n  let stored = 0\n  let failed = 0\n\n  for (const track of tracks) {\n    const key = keys.get(track.external_id)!\n    const cutter = createTrackCutter(EXPORT_SAMPLE_RATE, RECORDING_SEGMENTER, { mergeShort: true })\n    let previous = ''\n    let nextSegment = 0\n\n    const send = async (request: {\n      audio: Blob\n      seq?: number\n      segment?: number\n      /** From the recording start (a piece), or from the part start (a whole part: 0). */\n      offsetMs: number\n    }) => {\n      try {\n        const result = await arbi.calls.recording.transcribe(callExtId, recording.external_id, {\n          audio: request.audio,\n          trackExtId: track.external_id,\n          seq: request.seq ?? null,\n          segment: request.segment ?? null,\n          segmentOffsetMs: request.seq == null ? request.offsetMs : 0,\n          language,\n          prompt: continuityPrompt(previous),\n        })\n        const text = result.text.trim()\n        if (!text) return\n        previous = `${previous} ${text}`.slice(-PROMPT_HISTORY_CHARS)\n        const segment: TranscriptSegment = {\n          segmentId:\n            result.segment_id ||\n            (request.seq != null\n              ? `rec-${track.external_id}-${request.seq}`\n              : `rec-${track.external_id}-s${request.segment}`),\n          text,\n          speakerIdentity: result.speaker_ext_id ?? track.speaker_ext_id ?? '',\n          speakerName: result.speaker_name ?? track.speaker_name ?? 'Speaker',\n          startedAt: Number.isFinite(start)\n            ? new Date(start + (result.offset_ms ?? Math.round(request.offsetMs))).toISOString()\n            : null,\n        }\n        await storeTranscriptSegment(arbi, queryClient, ref, segment)\n        stored += 1\n      } catch (err) {\n        failed += 1\n        console.warn('[calls] a recording piece could not be transcribed', err)\n      }\n    }\n    const sendPieces = async (pieces: TrackPiece[]) => {\n      for (const piece of pieces) {\n        await send({\n          audio: new Blob([floatToWav(piece.samples, EXPORT_SAMPLE_RATE)], { type: 'audio/wav' }),\n          segment: nextSegment++,\n          offsetMs: (piece.start / EXPORT_SAMPLE_RATE) * 1000,\n        })\n      }\n    }\n\n    for (const part of sortedParts(track)) {\n      let bytes: Uint8Array\n      try {\n        bytes = await fetchPart(arbi, part, key)\n      } catch (err) {\n        failed += 1\n        console.warn('[calls] a recording part could not be read', err)\n        onProgress?.(++done, total)\n        continue\n      }\n      let pcm: Float32Array | null = null\n      try {\n        pcm = await decode(bytes)\n      } catch (err) {\n        console.warn('[calls] a recording part does not decode; sending it whole', err)\n      }\n      if (pcm) {\n        await sendPieces(cutter.push(part.offset_ms, pcm))\n      } else {\n        await send({\n          audio: new Blob([standalone(bytes)], { type: track.mime_type }),\n          seq: part.seq,\n          offsetMs: part.offset_ms,\n        })\n      }\n      onProgress?.(++done, total)\n    }\n    // The end of the track: the last piece (a closing line) is sent too.\n    await sendPieces(cutter.finish())\n  }\n  return { stored, failed }\n}\n"],
+  "mappings": 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+  "names": []
+}

package/dist/calls/recording-playback.js

@@ -0,0 +1,260 @@
+import {
+  DEFAULT_SEGMENTER_CONFIG,
+  INITIAL_SEGMENTER_STATE,
+  flushSegmenter,
+  levelFromRms,
+  stepSegmenter
+} from "../documents/audio-segmenter.js";
+const partKey = (p) => `${p.trackExtId}:${p.seq}`;
+function timelineOf(tracks) {
+  return tracks.flatMap(
+    (t) => (t.parts ?? []).map((p) => ({
+      trackExtId: t.external_id,
+      seq: p.seq,
+      offsetMs: p.offset_ms,
+      durationMs: p.duration_ms
+    }))
+  ).sort((a, b) => a.offsetMs - b.offsetMs || a.seq - b.seq);
+}
+function timelineDurationMs(parts) {
+  return parts.reduce((max, p) => Math.max(max, p.offsetMs + p.durationMs), 0);
+}
+const PLAYBACK_LOOKAHEAD_MS = 2e4;
+function partsDue(parts, playheadMs, lookaheadMs, scheduled) {
+  return parts.filter(
+    (p) => !scheduled.has(partKey(p)) && p.offsetMs < playheadMs + lookaheadMs && p.offsetMs + p.durationMs > playheadMs
+  );
+}
+function schedulePart(params) {
+  const { partOffsetMs, bufferDurationS, ctxStart, seekMs, ctxNow } = params;
+  let when = ctxStart + (partOffsetMs - seekMs) / 1e3;
+  let offset = 0;
+  if (when < ctxNow) {
+    offset = ctxNow - when;
+    when = ctxNow;
+  }
+  if (offset >= bufferDurationS) return null;
+  return { when, offset };
+}
+function playheadAt(ctxNow, ctxStart, seekMs) {
+  return seekMs + Math.max(0, ctxNow - ctxStart) * 1e3;
+}
+const EBML_MAGIC = [26, 69, 223, 163];
+const FTYP = [102, 116, 121, 112];
+function startsMediaFile(bytes) {
+  if (EBML_MAGIC.every((b, i) => bytes[i] === b)) return true;
+  return FTYP.every((b, i) => bytes[4 + i] === b);
+}
+function splitIntoFiles(parts) {
+  const files = [];
+  for (const part of parts) {
+    if (!files.length || startsMediaFile(part)) files.push([part]);
+    else files[files.length - 1].push(part);
+  }
+  return files;
+}
+function extensionFor(mimeType) {
+  if (mimeType.includes("mp4")) return mimeType.startsWith("video") ? "mp4" : "m4a";
+  if (mimeType.includes("ogg")) return "ogg";
+  return "webm";
+}
+function fileSafe(name) {
+  return name.replace(/[^\p{L}\p{N} ._-]+/gu, "").trim().replace(/\s+/g, "-").slice(0, 60) || "recording";
+}
+const EXPORT_SAMPLE_RATE = 16e3;
+function writeSamples(target, samples, offsetSamples) {
+  const start = Math.max(0, Math.round(offsetSamples));
+  const n = Math.min(samples.length, target.length - start);
+  for (let i = 0; i < n; i++) {
+    const s = Math.max(-1, Math.min(1, samples[i]));
+    target[start + i] = s < 0 ? s * 32768 : s * 32767;
+  }
+}
+function encodeWav(samples, sampleRate) {
+  const dataBytes = samples.length * 2;
+  const out = new Uint8Array(44 + dataBytes);
+  const view = new DataView(out.buffer);
+  const ascii = (offset, text) => {
+    for (let i = 0; i < text.length; i++) out[offset + i] = text.charCodeAt(i);
+  };
+  ascii(0, "RIFF");
+  view.setUint32(4, 36 + dataBytes, true);
+  ascii(8, "WAVE");
+  ascii(12, "fmt ");
+  view.setUint32(16, 16, true);
+  view.setUint16(20, 1, true);
+  view.setUint16(22, 1, true);
+  view.setUint32(24, sampleRate, true);
+  view.setUint32(28, sampleRate * 2, true);
+  view.setUint16(32, 2, true);
+  view.setUint16(34, 16, true);
+  ascii(36, "data");
+  view.setUint32(40, dataBytes, true);
+  for (let i = 0; i < samples.length; i++) view.setInt16(44 + i * 2, samples[i], true);
+  return out;
+}
+function standalone(bytes) {
+  return new Uint8Array(bytes);
+}
+function floatToWav(samples, sampleRate) {
+  const pcm = new Int16Array(samples.length);
+  writeSamples(pcm, samples, 0);
+  return encodeWav(pcm, sampleRate);
+}
+const SEGMENT_FRAME_MS = 50;
+const SEGMENT_PRE_ROLL_MS = 200;
+const RECORDING_SEGMENTER = {
+  maxMs: 25e3,
+  minVoiceMs: 300,
+  silenceMs: 1500
+};
+const SHORT_PIECE_MERGE_GAP_MS = 1e3;
+function createTrackCutter(sampleRate, config = {}, options = {}) {
+  const cfg = { ...DEFAULT_SEGMENTER_CONFIG, ...config };
+  const frame = Math.max(1, Math.round(sampleRate * SEGMENT_FRAME_MS / 1e3));
+  const preRoll = Math.round(SEGMENT_PRE_ROLL_MS / 1e3 * sampleRate);
+  const toSample = (ms) => Math.round(ms / 1e3 * sampleRate);
+  const toMs = (sample) => sample / sampleRate * 1e3;
+  let buf = new Float32Array(sampleRate * 4);
+  let bufStart = 0;
+  let len = 0;
+  let origin = null;
+  let cursor = 0;
+  let nextFrame = 0;
+  let state = INITIAL_SEGMENTER_STATE;
+  let pending = null;
+  let out = [];
+  const append = (samples) => {
+    if (len + samples.length > buf.length) {
+      const grown = new Float32Array(Math.max(buf.length * 2, len + samples.length));
+      grown.set(buf.subarray(0, len));
+      buf = grown;
+    }
+    buf.set(samples, len);
+    len += samples.length;
+    cursor += samples.length;
+  };
+  const emit = (piece) => {
+    const from = Math.max(piece.start, bufStart);
+    if (piece.end <= from) return;
+    out.push({
+      start: from,
+      end: piece.end,
+      voiceMs: piece.voiceMs,
+      samples: buf.slice(from - bufStart, piece.end - bufStart)
+    });
+  };
+  const onEvents = (events) => {
+    for (const e of events) {
+      if (e.type === "start") {
+        if (pending && e.at - pending.at > SHORT_PIECE_MERGE_GAP_MS) pending = null;
+        continue;
+      }
+      let piece = {
+        start: Math.max(origin ?? 0, toSample(e.startedAt) - preRoll),
+        end: Math.min(cursor, toSample(e.at)),
+        voiceMs: e.voiceMs,
+        at: e.at
+      };
+      if (pending) {
+        if (piece.end - pending.start <= toSample(cfg.maxMs) + preRoll) {
+          piece = { ...piece, start: pending.start, voiceMs: piece.voiceMs + pending.voiceMs };
+        }
+        pending = null;
+      }
+      if (piece.voiceMs >= cfg.minVoiceMs) emit(piece);
+      else if (options.mergeShort && piece.end > piece.start) pending = piece;
+    }
+  };
+  const step = (from, n) => {
+    let sum = 0;
+    const base = from - bufStart;
+    for (let j = 0; j < n; j++) sum += buf[base + j] * buf[base + j];
+    const result = stepSegmenter(state, levelFromRms(Math.sqrt(sum / n)), toMs(from), cfg);
+    state = result.state;
+    onEvents(result.events);
+  };
+  const run = () => {
+    while (nextFrame + frame <= cursor) {
+      step(nextFrame, frame);
+      nextFrame += frame;
+    }
+    let keepFrom = cursor - preRoll;
+    if (state.phase === "voice") keepFrom = Math.min(keepFrom, toSample(state.startedAt) - preRoll);
+    if (pending) keepFrom = Math.min(keepFrom, pending.start);
+    keepFrom = Math.min(Math.max(keepFrom, bufStart), nextFrame);
+    if (keepFrom > bufStart) {
+      buf.copyWithin(0, keepFrom - bufStart, len);
+      len -= keepFrom - bufStart;
+      bufStart = keepFrom;
+    }
+  };
+  const take = () => {
+    const done = out;
+    out = [];
+    return done;
+  };
+  return {
+    push(offsetMs, samples) {
+      const at = toSample(Math.max(0, offsetMs));
+      if (origin === null) {
+        origin = at;
+        bufStart = cursor = nextFrame = at;
+      }
+      if (at > cursor) {
+        const second = new Float32Array(sampleRate);
+        while (cursor < at) {
+          append(second.subarray(0, Math.min(sampleRate, at - cursor)));
+          run();
+        }
+      }
+      const fresh = at < cursor ? samples.subarray(Math.min(samples.length, cursor - at)) : samples;
+      if (fresh.length) {
+        append(fresh);
+        run();
+      }
+      return take();
+    },
+    finish() {
+      if (origin !== null) {
+        if (nextFrame < cursor) {
+          step(nextFrame, cursor - nextFrame);
+          nextFrame = cursor;
+        }
+        onEvents(flushSegmenter(state, toMs(cursor), cfg).events);
+        state = INITIAL_SEGMENTER_STATE;
+      }
+      pending = null;
+      return take();
+    }
+  };
+}
+function cutAtPauses(samples, sampleRate, config = {}) {
+  const cutter = createTrackCutter(sampleRate, config);
+  return [...cutter.push(0, samples), ...cutter.finish()].map(({ start, end }) => ({ start, end }));
+}
+export {
+  EXPORT_SAMPLE_RATE,
+  PLAYBACK_LOOKAHEAD_MS,
+  RECORDING_SEGMENTER,
+  SEGMENT_FRAME_MS,
+  SEGMENT_PRE_ROLL_MS,
+  SHORT_PIECE_MERGE_GAP_MS,
+  createTrackCutter,
+  cutAtPauses,
+  encodeWav,
+  extensionFor,
+  fileSafe,
+  floatToWav,
+  partKey,
+  partsDue,
+  playheadAt,
+  schedulePart,
+  splitIntoFiles,
+  standalone,
+  startsMediaFile,
+  timelineDurationMs,
+  timelineOf,
+  writeSamples
+};
+//# sourceMappingURL=recording-playback.js.map

package/dist/calls/recording-playback.js.map

@@ -0,0 +1,7 @@
+{
+  "version": 3,
+  "sources": ["../../src/calls/recording-playback.ts"],
+  "sourcesContent": ["/**\n * Playing and exporting a call recording in the browser (ARBI `docs/meeting-chats-plan.md` \u00A76).\n * Everything here works on DECRYPTED part bytes; nothing leaves the browser.\n *\n * Speaker audio \u2014 mixed in sync with WebAudio: each speaker part is a standalone file placed at its\n * `offset_ms` on one AudioContext timeline. Parts are fetched, decrypted and decoded only a little\n * ahead of the playhead ({@link partsDue}) and dropped once played, so an hour-long recording never\n * sits decoded in memory.\n *\n * Composite video \u2014 the host's composite is recorded as ONE continuous WebM (MediaRecorder\n * timeslice: only the first part carries the header), so its decrypted parts concatenated in order\n * ARE the file: it plays in a plain <video> and downloads as one file. This was chosen over\n * MediaSource (codec strings and append juggling per browser) and over part-by-part playback\n * (a visible gap every 30 s) as the simplest robust option. A reconnect of the host starts a new\n * file mid-recording; {@link splitIntoFiles} cuts at each new header, so each run plays on its own.\n *\n * Download \u2014 each speaker track becomes one WAV (16 kHz mono) with its parts at their offsets, so\n * every speaker's file starts at the recording's start and they line up in any editor.\n */\n\nimport type { calls } from '@arbidocs/sdk/browser'\nimport {\n  DEFAULT_SEGMENTER_CONFIG,\n  INITIAL_SEGMENTER_STATE,\n  flushSegmenter,\n  levelFromRms,\n  stepSegmenter,\n  type SegmenterConfig,\n  type SegmenterEvent,\n  type SegmenterState,\n} from '../documents/audio-segmenter'\n\n// \u2500\u2500 Timeline \u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\n\nexport interface TimelinePart {\n  trackExtId: string\n  seq: number\n  offsetMs: number\n  durationMs: number\n}\n\nexport const partKey = (p: Pick<TimelinePart, 'trackExtId' | 'seq'>) => `${p.trackExtId}:${p.seq}`\n\n/** Every part of the given tracks, in time order. */\nexport function timelineOf(\n  tracks: Pick<calls.RecordingTrack, 'external_id' | 'parts'>[]\n): TimelinePart[] {\n  return tracks\n    .flatMap((t) =>\n      (t.parts ?? []).map((p: calls.RecordingPart) => ({\n        trackExtId: t.external_id,\n        seq: p.seq,\n        offsetMs: p.offset_ms,\n        durationMs: p.duration_ms,\n      }))\n    )\n    .sort((a, b) => a.offsetMs - b.offsetMs || a.seq - b.seq)\n}\n\n/** The recording's length: the end of its last part. */\nexport function timelineDurationMs(parts: TimelinePart[]): number {\n  return parts.reduce((max, p) => Math.max(max, p.offsetMs + p.durationMs), 0)\n}\n\n/** How far ahead of the playhead parts are fetched and decoded. */\nexport const PLAYBACK_LOOKAHEAD_MS = 20_000\n\n/** Parts not yet scheduled that play within the window `[playhead, playhead + lookahead)`. */\nexport function partsDue(\n  parts: TimelinePart[],\n  playheadMs: number,\n  lookaheadMs: number,\n  scheduled: ReadonlySet<string>\n): TimelinePart[] {\n  return parts.filter(\n    (p) =>\n      !scheduled.has(partKey(p)) &&\n      p.offsetMs < playheadMs + lookaheadMs &&\n      p.offsetMs + p.durationMs > playheadMs\n  )\n}\n\n/**\n * When to start a decoded part on the AudioContext clock. Playback began at context time\n * `ctxStart` from recording position `seekMs`, so position `t` plays at\n * `ctxStart + (t \u2212 seekMs)/1000`. A part whose start already passed (a seek into it, or it decoded\n * late) starts now, `offset` seconds into its buffer. Null: nothing of it is left to play.\n */\nexport function schedulePart(params: {\n  partOffsetMs: number\n  bufferDurationS: number\n  ctxStart: number\n  seekMs: number\n  ctxNow: number\n}): { when: number; offset: number } | null {\n  const { partOffsetMs, bufferDurationS, ctxStart, seekMs, ctxNow } = params\n  let when = ctxStart + (partOffsetMs - seekMs) / 1000\n  let offset = 0\n  if (when < ctxNow) {\n    offset = ctxNow - when\n    when = ctxNow\n  }\n  if (offset >= bufferDurationS) return null\n  return { when, offset }\n}\n\n/** The recording position (ms) the AudioContext clock is at. */\nexport function playheadAt(ctxNow: number, ctxStart: number, seekMs: number): number {\n  return seekMs + Math.max(0, ctxNow - ctxStart) * 1000\n}\n\n// \u2500\u2500 Files \u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\n\nconst EBML_MAGIC = [0x1a, 0x45, 0xdf, 0xa3]\nconst FTYP = [0x66, 0x74, 0x79, 0x70]\n\n/** Whether these bytes start a media file (a WebM/Matroska EBML header or an MP4 `ftyp` box). */\nexport function startsMediaFile(bytes: Uint8Array): boolean {\n  if (EBML_MAGIC.every((b, i) => bytes[i] === b)) return true\n  return FTYP.every((b, i) => bytes[4 + i] === b)\n}\n\n/** Consecutive parts grouped into files: a new file at each part that starts with a header. */\nexport function splitIntoFiles(parts: Uint8Array[]): Uint8Array[][] {\n  const files: Uint8Array[][] = []\n  for (const part of parts) {\n    if (!files.length || startsMediaFile(part)) files.push([part])\n    else files[files.length - 1].push(part)\n  }\n  return files\n}\n\n/** File extension for a recorded MIME type. */\nexport function extensionFor(mimeType: string): string {\n  if (mimeType.includes('mp4')) return mimeType.startsWith('video') ? 'mp4' : 'm4a'\n  if (mimeType.includes('ogg')) return 'ogg'\n  return 'webm'\n}\n\n/** A safe file name piece. */\nexport function fileSafe(name: string): string {\n  return (\n    name\n      .replace(/[^\\p{L}\\p{N} ._-]+/gu, '')\n      .trim()\n      .replace(/\\s+/g, '-')\n      .slice(0, 60) || 'recording'\n  )\n}\n\n// \u2500\u2500 WAV \u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\n\nexport const EXPORT_SAMPLE_RATE = 16_000\n\n/** Copy decoded samples (\u22121\u20261) into a 16-bit timeline at `offsetSamples`, clipped. */\nexport function writeSamples(\n  target: Int16Array,\n  samples: Float32Array,\n  offsetSamples: number\n): void {\n  const start = Math.max(0, Math.round(offsetSamples))\n  const n = Math.min(samples.length, target.length - start)\n  for (let i = 0; i < n; i++) {\n    const s = Math.max(-1, Math.min(1, samples[i]))\n    target[start + i] = s < 0 ? s * 0x8000 : s * 0x7fff\n  }\n}\n\n/** A mono 16-bit PCM WAV file. */\nexport function encodeWav(samples: Int16Array, sampleRate: number): Uint8Array<ArrayBuffer> {\n  const dataBytes = samples.length * 2\n  const out = new Uint8Array(44 + dataBytes)\n  const view = new DataView(out.buffer)\n  const ascii = (offset: number, text: string) => {\n    for (let i = 0; i < text.length; i++) out[offset + i] = text.charCodeAt(i)\n  }\n  ascii(0, 'RIFF')\n  view.setUint32(4, 36 + dataBytes, true)\n  ascii(8, 'WAVE')\n  ascii(12, 'fmt ')\n  view.setUint32(16, 16, true)\n  view.setUint16(20, 1, true) // PCM\n  view.setUint16(22, 1, true) // mono\n  view.setUint32(24, sampleRate, true)\n  view.setUint32(28, sampleRate * 2, true)\n  view.setUint16(32, 2, true)\n  view.setUint16(34, 16, true)\n  ascii(36, 'data')\n  view.setUint32(40, dataBytes, true)\n  for (let i = 0; i < samples.length; i++) view.setInt16(44 + i * 2, samples[i], true)\n  return out\n}\n\n/** Bytes as a standalone ArrayBuffer-backed view (Blob/decodeAudioData want one). */\nexport function standalone(bytes: Uint8Array): Uint8Array<ArrayBuffer> {\n  return new Uint8Array(bytes) as Uint8Array<ArrayBuffer>\n}\n\n/** Decoded mono samples as a 16-bit WAV. */\nexport function floatToWav(samples: Float32Array, sampleRate: number): Uint8Array<ArrayBuffer> {\n  const pcm = new Int16Array(samples.length)\n  writeSamples(pcm, samples, 0)\n  return encodeWav(pcm, sampleRate)\n}\n\n// \u2500\u2500 Cutting at pauses (transcription) \u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\n\n/** One piece of a part, in samples. */\nexport interface AudioPiece {\n  start: number\n  end: number\n}\n\n/** A piece of a whole track: absolute sample positions on the track's timeline, and its audio. */\nexport interface TrackPiece extends AudioPiece {\n  samples: Float32Array\n  /** Voiced time in the piece. */\n  voiceMs: number\n}\n\n/** Meter frame for the segmenter (the live recorder's meter ticks every 50 ms). */\nexport const SEGMENT_FRAME_MS = 50\n/** Kept before a piece's first voiced frame, so a word's onset is not clipped. */\nexport const SEGMENT_PRE_ROLL_MS = 200\n\n/**\n * The segmenter for a RECORDED track (transcribed after the call). Nothing waits on it, so it cuts\n * only at a real turn: 1.5 s without voice (live uses 700 ms for latency). A speaker's own track is\n * silent for seconds while others talk, but a quiet syllable mid-sentence can read as a short\n * \"pause\" \u2014 the 3-person demo split \"\u2026two hundred and fifty / thousand pounds\u2026\" at 700 ms and the\n * cut half came back as \"$250,000\". A piece runs to 25 s before it is cut (a 25 s 16 kHz WAV is\n * ~0.8 MB, well under the 2 MB cap), and a piece needs only 300 ms of voice: a short closing line\n * (\"Perfect. Thank you both.\") is speech, not a click.\n */\nexport const RECORDING_SEGMENTER: Partial<SegmenterConfig> = {\n  maxMs: 25_000,\n  minVoiceMs: 300,\n  silenceMs: 1_500,\n}\n/**\n * A recorded piece with too little voice is held, and joined to the next piece when that starts\n * within this gap (\"Perfect.\" \u2026 \"Thank you both.\"); otherwise it is dropped.\n */\nexport const SHORT_PIECE_MERGE_GAP_MS = 1000\n\nexport interface TrackCutter {\n  /**\n   * Feed the next decoded part, placed at `offsetMs` on the track's timeline (a gap before it is\n   * silence; an overlap with what came before is skipped). Returns the pieces it completed.\n   */\n  push(offsetMs: number, samples: Float32Array): TrackPiece[]\n  /** The end of the track: closes the open piece. Returns the pieces it completed. */\n  finish(): TrackPiece[]\n}\n\n/**\n * Cut a whole track at its pauses, part after part, with the SAME voice-activity segmenter as live\n * transcription (P5): RMS per 50 ms frame \u2192 meter level \u2192 `stepSegmenter`. The parts are placed on\n * ONE continuous timeline by their offsets, so a sentence spoken across a part boundary stays one\n * piece. Only the open piece (\u2264 `maxMs`) is kept in memory, never the whole track.\n *\n * `mergeShort`: a piece with less voice than `minVoiceMs` is held and joined to the next one if it\n * starts within {@link SHORT_PIECE_MERGE_GAP_MS}, instead of being dropped at once.\n */\nexport function createTrackCutter(\n  sampleRate: number,\n  config: Partial<SegmenterConfig> = {},\n  options: { mergeShort?: boolean } = {}\n): TrackCutter {\n  const cfg = { ...DEFAULT_SEGMENTER_CONFIG, ...config }\n  const frame = Math.max(1, Math.round((sampleRate * SEGMENT_FRAME_MS) / 1000))\n  const preRoll = Math.round((SEGMENT_PRE_ROLL_MS / 1000) * sampleRate)\n  const toSample = (ms: number) => Math.round((ms / 1000) * sampleRate)\n  const toMs = (sample: number) => (sample / sampleRate) * 1000\n\n  // Retained audio: `buf[0..len)` holds the absolute samples `[bufStart, bufStart + len)`.\n  let buf = new Float32Array(sampleRate * 4)\n  let bufStart = 0\n  let len = 0\n  /** Absolute index of the first sample of the timeline (null before the first part). */\n  let origin: number | null = null\n  /** Absolute index just past the last sample fed. */\n  let cursor = 0\n  /** Absolute index of the next meter frame. */\n  let nextFrame = 0\n  let state: SegmenterState = INITIAL_SEGMENTER_STATE\n  /** A short piece waiting for the next one (mergeShort). */\n  let pending: { start: number; end: number; voiceMs: number; at: number } | null = null\n  let out: TrackPiece[] = []\n\n  const append = (samples: Float32Array) => {\n    if (len + samples.length > buf.length) {\n      const grown = new Float32Array(Math.max(buf.length * 2, len + samples.length))\n      grown.set(buf.subarray(0, len))\n      buf = grown\n    }\n    buf.set(samples, len)\n    len += samples.length\n    cursor += samples.length\n  }\n\n  const emit = (piece: { start: number; end: number; voiceMs: number }) => {\n    const from = Math.max(piece.start, bufStart)\n    if (piece.end <= from) return\n    out.push({\n      start: from,\n      end: piece.end,\n      voiceMs: piece.voiceMs,\n      samples: buf.slice(from - bufStart, piece.end - bufStart),\n    })\n  }\n\n  const onEvents = (events: SegmenterEvent[]) => {\n    for (const e of events) {\n      if (e.type === 'start') {\n        // Too long a pause after a held short piece: it stands alone, too short to keep.\n        if (pending && e.at - pending.at > SHORT_PIECE_MERGE_GAP_MS) pending = null\n        continue\n      }\n      let piece = {\n        start: Math.max(origin ?? 0, toSample(e.startedAt) - preRoll),\n        end: Math.min(cursor, toSample(e.at)),\n        voiceMs: e.voiceMs,\n        at: e.at,\n      }\n      if (pending) {\n        if (piece.end - pending.start <= toSample(cfg.maxMs) + preRoll) {\n          piece = { ...piece, start: pending.start, voiceMs: piece.voiceMs + pending.voiceMs }\n        }\n        pending = null\n      }\n      if (piece.voiceMs >= cfg.minVoiceMs) emit(piece)\n      else if (options.mergeShort && piece.end > piece.start) pending = piece\n    }\n  }\n\n  const step = (from: number, n: number) => {\n    let sum = 0\n    const base = from - bufStart\n    for (let j = 0; j < n; j++) sum += buf[base + j] * buf[base + j]\n    const result = stepSegmenter(state, levelFromRms(Math.sqrt(sum / n)), toMs(from), cfg)\n    state = result.state\n    onEvents(result.events)\n  }\n\n  /** Run the meter over every whole frame fed so far, then drop audio no piece can reach. */\n  const run = () => {\n    while (nextFrame + frame <= cursor) {\n      step(nextFrame, frame)\n      nextFrame += frame\n    }\n    let keepFrom = cursor - preRoll\n    if (state.phase === 'voice') keepFrom = Math.min(keepFrom, toSample(state.startedAt) - preRoll)\n    if (pending) keepFrom = Math.min(keepFrom, pending.start)\n    keepFrom = Math.min(Math.max(keepFrom, bufStart), nextFrame)\n    if (keepFrom > bufStart) {\n      buf.copyWithin(0, keepFrom - bufStart, len)\n      len -= keepFrom - bufStart\n      bufStart = keepFrom\n    }\n  }\n\n  const take = () => {\n    const done = out\n    out = []\n    return done\n  }\n\n  return {\n    push(offsetMs, samples) {\n      const at = toSample(Math.max(0, offsetMs))\n      if (origin === null) {\n        origin = at\n        bufStart = cursor = nextFrame = at\n      }\n      if (at > cursor) {\n        // A gap between parts (a reconnect): silence, a second at a time.\n        const second = new Float32Array(sampleRate)\n        while (cursor < at) {\n          append(second.subarray(0, Math.min(sampleRate, at - cursor)))\n          run()\n        }\n      }\n      // Overlapping the previous part: skip what is already on the timeline.\n      const fresh = at < cursor ? samples.subarray(Math.min(samples.length, cursor - at)) : samples\n      if (fresh.length) {\n        append(fresh)\n        run()\n      }\n      return take()\n    },\n    finish() {\n      if (origin !== null) {\n        if (nextFrame < cursor) {\n          step(nextFrame, cursor - nextFrame)\n          nextFrame = cursor\n        }\n        onEvents(flushSegmenter(state, toMs(cursor), cfg).events)\n        state = INITIAL_SEGMENTER_STATE\n      }\n      pending = null\n      return take()\n    },\n  }\n}\n\n/**\n * Cut decoded speech at its pauses with the SAME voice-activity segmenter as live transcription\n * (P5): RMS per 50 ms frame \u2192 meter level \u2192 `stepSegmenter` (a piece ends after ~700 ms of\n * silence, is cut at `maxMs`, and is dropped with too little voice). A whole 30 s part sent at\n * once made the transcriber drop repeated sentences; short pieces do not.\n */\nexport function cutAtPauses(\n  samples: Float32Array,\n  sampleRate: number,\n  config: Partial<SegmenterConfig> = {}\n): AudioPiece[] {\n  const cutter = createTrackCutter(sampleRate, config)\n  return [...cutter.push(0, samples), ...cutter.finish()].map(({ start, end }) => ({ start, end }))\n}\n"],
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+}