@alexkroman1/aai-cli 6.10.1 → 6.11.0

package/dist/templates/call-audit/workflows/media.ts

@@ -0,0 +1,647 @@
+// Copyright 2026 the AAI authors. MIT license.
+/**
+ * The pure half of the audit desk: the ffmpeg argv it runs, the two analyses it
+ * reads back out, and where it decides to cut.
+ *
+ * No directive in this file, which is what lets it sit under `workflows/`: the
+ * Workflow DevKit's builder scans this directory and transforms only what carries
+ * a `"use workflow"` / `"use step"` body. Everything here is a pure function of a
+ * journaled value, and that is deliberate rather than tidy — **an ffmpeg pipeline
+ * is untestable exactly where it spawns**, so every decision this desk makes is
+ * pushed out of the steps and into this module, where a spec drives it with no
+ * subprocess, no temp file and no recording.
+ *
+ * What is left in the steps is materialize, spawn, store.
+ *
+ * ## The argv is ours, so it is BUILT rather than embedded
+ *
+ * `runFfmpeg` passes `args` through verbatim — no `-y`, no `-loglevel` — so the
+ * standing flags are a decision this file makes once ({@link standardFlags}) and
+ * every invocation's real argv is a value a test can assert on. It is also why
+ * the filter strings below carry no shell quoting: each is ONE element of an
+ * argv array, so the commas that chain filters and the colons that separate their
+ * options never meet a shell.
+ *
+ * ## Two analyses, and they read their answers back by DIFFERENT routes
+ *
+ * This is the detail that a first draft gets wrong, and it is a property of the
+ * SDK rather than of ffmpeg:
+ *
+ * - **Loudness comes back on stderr**, because `loudnorm`'s
+ *   `print_format=json` writes one fixed-size block after the last frame. The
+ *   SDK keeps a capped stderr TAIL (`FFMPEG_STDERR_TAIL_CHARS`, 4000 chars) on
+ *   the argument that ffmpeg's log is progress lines and the diagnosis is the
+ *   last one — which is exactly true of a block printed at the end. Measured on
+ *   ffmpeg 6.1: the JSON is ~330 characters.
+ * - **Silence comes back in a FILE**, because `silencedetect` logs an event per
+ *   pause, so its output grows with the recording. A two-hour call with a pause
+ *   every ten seconds is 720 events, which does not fit in a 4000-character tail
+ *   — and what a tail drops is the BEGINNING, so the failure is a desk that cuts
+ *   the back half of every long recording and the front half of none. That is
+ *   silent, and it is the kind of bug that reproduces only on inputs nobody tests
+ *   with. `ametadata=mode=print:file=…` writes ffmpeg's own frame metadata
+ *   straight to a path with no cap, and it does so at `-loglevel error`, which is
+ *   what keeps the log quiet AND the analysis complete.
+ *
+ * ## Cutting in the silence is the whole point of the pipeline
+ *
+ * `transcription-workflow` cuts a recording by arithmetic — every 90 seconds,
+ * wherever that lands — because with no decoder that is all it can do, and it
+ * pays for it twice: each cut lands mid-word, so segments OVERLAP by two seconds
+ * and a stitcher has to find and drop the duplicated words afterwards.
+ *
+ * With ffmpeg in the path, the pauses are known, so a cut can be placed in one.
+ * Three costs disappear at once: the overlap (~2% of the audio, transcribed
+ * twice), the stitching (a seam-matching heuristic that can be wrong), and the
+ * mid-word decode on both sides of every cut. {@link planSegments} is that, and
+ * it stays honest about the case it cannot serve — a stretch of unbroken speech
+ * longer than the cap gets the blind cut, and says so.
+ */
+
+import { isRecord, type PcmFormat } from "@alexkroman1/aai/utils";
+
+/**
+ * The format every recording is converted to before anything measures it.
+ *
+ * 16 kHz mono 16-bit, so one second of audio is exactly 32,000 bytes — and that
+ * equality is what the rest of this module rests on. Two consequences worth
+ * naming, because they are the reason this desk normalizes at all:
+ *
+ * - **A byte offset is a timestamp, with no header to parse.** The intermediate
+ *   is headerless raw PCM (see {@link normalizeArgs}), so `startByte` is
+ *   `seconds * 32000` and nothing walks a RIFF chunk list. Note that assuming a
+ *   44-byte WAV header instead would be WRONG: ffmpeg writes a `LIST`/`INFO`
+ *   chunk naming its own version, so its WAV output has a 78-byte header on
+ *   ffmpeg 6.1 and a different one whenever that string's length changes.
+ * - **The provider's byte cap stops binding.** The sync endpoint's limits are
+ *   120 seconds and 40 MB; at 32,000 bytes a second, {@link MAX_SEGMENT_SECONDS}
+ *   of audio is 3.5 MB. So this desk has ONE cap to plan against where
+ *   `transcription-workflow` has two and has to derive which one binds from the
+ *   format it was handed.
+ *
+ * 16 kHz is also what speech models are trained at, so nothing is lost that a
+ * decoder would have used.
+ */
+export const ANALYSIS_FORMAT = {
+  sampleRate: 16_000,
+  channels: 1,
+  bitsPerSample: 16,
+} as const satisfies PcmFormat;
+
+/** Bytes of {@link ANALYSIS_FORMAT} audio per second of wall clock. */
+export const BYTES_PER_SECOND =
+  (ANALYSIS_FORMAT.sampleRate * ANALYSIS_FORMAT.channels * ANALYSIS_FORMAT.bitsPerSample) / 8;
+
+/**
+ * Integrated loudness everything is normalized to, in LUFS.
+ *
+ * −16 LUFS is the speech/podcast convention, and the reason to normalize at all
+ * is not aesthetics: a conference recording where one party is on a headset and
+ * the other is across a room has a 20 dB gap between them, and the quiet side
+ * sits near enough the noise floor that {@link SILENCE_FLOOR_DB} cannot tell
+ * their pauses from their words. Levelling first is what makes ONE silence
+ * threshold work for a whole recording.
+ */
+export const LOUDNESS_TARGET_LUFS = -16;
+
+/** True-peak ceiling, in dBTP. −1.5 leaves headroom for a lossy re-encode later. */
+export const LOUDNESS_TRUE_PEAK_DB = -1.5;
+
+/** Loudness range, in LU. 11 is `loudnorm`'s own default, restated so the argv is explicit. */
+export const LOUDNESS_RANGE_LU = 11;
+
+/**
+ * What counts as silence, in dB relative to full scale.
+ *
+ * Applied to audio that has ALREADY been levelled to
+ * {@link LOUDNESS_TARGET_LUFS}, which is what makes a single number defensible —
+ * see that constant. −35 dB is below room tone and typing and above the digital
+ * floor, so it finds pauses rather than absolute quiet.
+ */
+export const SILENCE_FLOOR_DB = -35;
+
+/**
+ * How long a pause must last to be worth cutting in, in seconds.
+ *
+ * Under ~0.4s this finds the gaps BETWEEN WORDS, and a desk that may cut there
+ * has learned nothing over cutting by arithmetic. 0.6s is a breath or a turn
+ * change — the places a human would cut a recording.
+ */
+export const MIN_SILENCE_SECONDS = 0.6;
+
+/**
+ * The longest segment this desk will send, in seconds.
+ *
+ * The sync endpoint's hard cap is 120 seconds. The headroom is smaller here than
+ * a blind cut needs (`transcription-workflow` leaves 30s for its overlap) because
+ * there is no overlap to add: a segment is exactly the audio between two cut
+ * points, so the only thing the margin absorbs is the endpoint measuring a
+ * duration slightly differently than the byte count says.
+ */
+export const MAX_SEGMENT_SECONDS = 110;
+
+/**
+ * The shortest segment worth its own request, in seconds.
+ *
+ * The endpoint refuses audio under 80ms outright; the floor is well above that
+ * because a request costs a round trip either way and a 0.3-second tail of a
+ * recording holds at most one word. A stretch below this is merged backwards
+ * into its predecessor rather than dropped — see {@link planSegments}.
+ */
+export const MIN_SEGMENT_SECONDS = 1;
+
+/** The standing flags, on every invocation this desk makes. */
+export function standardFlags(): string[] {
+  return [
+    "-hide_banner",
+    // Progress lines are noise in a captured stderr, and the SDK keeps only a
+    // tail of it — so suppressing them is what leaves room for the diagnosis.
+    "-nostats",
+    // In a guest there is no terminal, and an ffmpeg that decides to read stdin
+    // is a process that never exits.
+    "-nostdin",
+    "-y",
+  ];
+}
+
+/** A loudness measurement, as `loudnorm`'s first pass reports it. */
+export type Loudness = {
+  /** Integrated loudness, LUFS. */
+  inputLufs: number;
+  /** True peak, dBTP. */
+  inputTruePeak: number;
+  /** Loudness range, LU. */
+  inputRange: number;
+  /** The gating threshold the measurement used, LUFS. */
+  inputThreshold: number;
+  /** The correction the second pass must apply, LU. */
+  targetOffset: number;
+};
+
+/** One pause, in seconds from the start of the recording. */
+export type Silence = {
+  startSec: number;
+  endSec: number;
+};
+
+/** One request's worth of audio, addressed as a byte range of the normalized PCM. */
+export type Segment = {
+  /** Position in the recording — the fan-out's order, and the merge's. */
+  index: number;
+  /** First byte, frame-aligned and inclusive. */
+  startByte: number;
+  /** One past the last byte, frame-aligned. */
+  endByte: number;
+  /** Where this segment starts in the recording. */
+  startMs: number;
+  /** Where it ends. Does NOT overlap the next segment — see the module doc. */
+  endMs: number;
+  /**
+   * Whether this segment's end landed in speech rather than in a pause.
+   *
+   * `false` for every segment on an ordinary recording, and the field exists for
+   * the case where it is not: a monologue with no 0.6-second pause in 110 seconds
+   * leaves nothing to cut in, so the desk cuts by arithmetic exactly as
+   * `transcription-workflow` does. Reported rather than hidden, because a
+   * transcript with a mangled word at one seam is otherwise a mystery.
+   */
+  cutInSpeech: boolean;
+};
+
+/** Raised when an analysis pass produced something this module cannot read. Always terminal. */
+export class MediaAnalysisError extends Error {
+  constructor(message: string) {
+    super(message);
+    this.name = "MediaAnalysisError";
+  }
+}
+
+/**
+ * Pass one: measure the recording's loudness without writing any audio.
+ *
+ * `-f null -` is the whole trick — the filter graph runs, every frame is decoded
+ * and analysed, and the output goes nowhere. So this costs a decode and produces
+ * five numbers.
+ *
+ * **`-loglevel info` is required and is not a debugging leftover.**
+ * `print_format=json` writes through ffmpeg's log at info level, so at `error`
+ * the pass runs, succeeds, and prints nothing — a failure that looks like a
+ * parser bug. It is the one invocation here that is not quiet, which is why
+ * {@link parseLoudness} searches for its block rather than assuming the stderr
+ * tail begins with it.
+ */
+export function measureLoudnessArgs(input: string): string[] {
+  return [
+    ...standardFlags(),
+    "-loglevel",
+    "info",
+    "-i",
+    input,
+    "-af",
+    `loudnorm=I=${LOUDNESS_TARGET_LUFS}:TP=${LOUDNESS_TRUE_PEAK_DB}:LRA=${LOUDNESS_RANGE_LU}:print_format=json`,
+    "-f",
+    "null",
+    "-",
+  ];
+}
+
+/**
+ * Read the five numbers pass one printed.
+ *
+ * The block is JSON, and it is found rather than parsed from a known offset: it
+ * is preceded by `[Parsed_loudnorm_0 @ 0x…]` and by however much of ffmpeg's
+ * info-level chatter survived the stderr tail. So the reader takes the LAST
+ * `{…}` in the text — last because a re-run's block would follow an earlier one,
+ * and because nothing else ffmpeg logs at info level is brace-delimited.
+ *
+ * Every value arrives as a STRING (`"input_i" : "-16.19"`), which is ffmpeg's
+ * shape and not a quirk of one version; a numeric coercion that silently yields
+ * `NaN` is what a missing key would otherwise become, so each one is checked.
+ */
+export function parseLoudness(stderr: string): Loudness {
+  const open = stderr.lastIndexOf("{");
+  const close = stderr.lastIndexOf("}");
+  if (open === -1 || close < open) {
+    throw new MediaAnalysisError(
+      "The loudness pass printed no JSON block. That is what happens when the argv " +
+        "loses `-loglevel info`, since `print_format=json` writes through ffmpeg's log.",
+    );
+  }
+  // `isRecord` from the SDK rather than a hand-written
+  // `typeof x === "object" && x !== null` — this repo's rule (`guard-invariants`
+  // rule 17). It also NARROWS, so nothing below needs the
+  // `as Record<string, unknown>` the open-coded version required.
+  const parsed = safeJson(stderr.slice(open, close + 1));
+  if (!isRecord(parsed)) {
+    throw new MediaAnalysisError("The loudness pass printed a block that is not JSON.");
+  }
+  return {
+    inputLufs: numberAt(parsed, "input_i"),
+    inputTruePeak: numberAt(parsed, "input_tp"),
+    inputRange: numberAt(parsed, "input_lra"),
+    inputThreshold: numberAt(parsed, "input_thresh"),
+    targetOffset: numberAt(parsed, "target_offset"),
+  };
+}
+
+/**
+ * Pass two: apply the measurement, find the pauses, and write the audio.
+ *
+ * ONE invocation doing three things, which is a decode saved rather than a
+ * shortcut: levelling and silence detection are both filters on the same graph,
+ * so chaining them costs nothing over running either alone. It also makes the
+ * silence map STRICTLY more useful — the pauses are found in the levelled signal,
+ * which is the signal a single {@link SILENCE_FLOOR_DB} can actually judge.
+ *
+ * `linear=true` asks for one constant gain over the whole recording instead of a
+ * moving one. That is what you want for speech (a dynamic normalizer audibly
+ * pumps between a loud sentence and a quiet one) and ffmpeg falls back to dynamic
+ * on its own when the linear gain would clip the true peak, so it is a preference
+ * rather than a demand.
+ *
+ * The output is headerless raw PCM — see {@link ANALYSIS_FORMAT} for why that is
+ * the shape that makes byte arithmetic legal.
+ *
+ * @param silenceLog - Where `ametadata` writes the pause events. A path rather
+ *   than stderr, and the module doc carries why that is load-bearing.
+ */
+export function normalizeArgs(
+  input: string,
+  measured: Loudness,
+  output: string,
+  silenceLog: string,
+): string[] {
+  const loudnorm = [
+    `loudnorm=I=${LOUDNESS_TARGET_LUFS}`,
+    `TP=${LOUDNESS_TRUE_PEAK_DB}`,
+    `LRA=${LOUDNESS_RANGE_LU}`,
+    `measured_I=${measured.inputLufs}`,
+    `measured_TP=${measured.inputTruePeak}`,
+    `measured_LRA=${measured.inputRange}`,
+    `measured_thresh=${measured.inputThreshold}`,
+    `offset=${measured.targetOffset}`,
+    "linear=true",
+  ].join(":");
+
+  return [
+    ...standardFlags(),
+    // Quiet, and the analysis still arrives: `ametadata` writes its file
+    // directly rather than through the log, which is the property that lets this
+    // pass be both silent and complete.
+    "-loglevel",
+    "error",
+    "-i",
+    input,
+    "-af",
+    `${loudnorm},silencedetect=noise=${SILENCE_FLOOR_DB}dB:duration=${MIN_SILENCE_SECONDS},ametadata=mode=print:file=${silenceLog}`,
+    // No video, and the channel/rate/codec triple that makes the output match
+    // `ANALYSIS_FORMAT` exactly. `-f s16le` rather than `-f wav`: raw samples,
+    // no header, so byte zero is second zero.
+    "-vn",
+    "-ac",
+    String(ANALYSIS_FORMAT.channels),
+    "-ar",
+    String(ANALYSIS_FORMAT.sampleRate),
+    "-c:a",
+    "pcm_s16le",
+    "-f",
+    "s16le",
+    output,
+  ];
+}
+
+/**
+ * Read the pauses out of `ametadata`'s log.
+ *
+ * The format is a block per event — a `frame:… pts:… pts_time:…` line followed by
+ * the `lavfi.silence_*` keys that frame carried:
+ *
+ * ```text
+ * frame:155  pts:158720  pts_time:3.59909
+ * lavfi.silence_start=3
+ * frame:215  pts:220160  pts_time:4.99229
+ * lavfi.silence_end=5.00005
+ * lavfi.silence_duration=2.00005
+ * ```
+ *
+ * Note the event times are NOT the frame's `pts_time`: `silence_start=3` on a
+ * frame at 3.599 is the filter reporting where the silence really began, having
+ * needed 0.6 seconds of it to be sure. So the `lavfi.` keys are what is read and
+ * the frame lines are skipped.
+ *
+ * **A trailing `silence_start` with no `silence_end` is normal and has to be
+ * handled**, verified against ffmpeg 6.1: a recording that ends during a pause
+ * gets an opening event and nothing to close it, because the filter never sees
+ * the sound come back. It is closed at `durationSec`, which is why this function
+ * takes a duration it could otherwise derive nothing from — and why the caller
+ * measures that duration from the PCM byte count rather than from this log.
+ */
+export function parseSilences(log: string, durationSec: number): Silence[] {
+  const silences: Silence[] = [];
+  let openedAt: number | undefined;
+
+  for (const line of log.split("\n")) {
+    const trimmed = line.trim();
+    const start = value(trimmed, "lavfi.silence_start=");
+    if (start !== undefined) {
+      // A second `start` before an `end` cannot happen in ffmpeg's output, and if
+      // it ever did, keeping the FIRST is the reading that does not lose audio:
+      // the pause is at least as long as the first opening claimed.
+      openedAt ??= Math.max(0, start);
+      continue;
+    }
+    const end = value(trimmed, "lavfi.silence_end=");
+    if (end !== undefined && openedAt !== undefined) {
+      if (end > openedAt) silences.push({ startSec: openedAt, endSec: Math.min(end, durationSec) });
+      openedAt = undefined;
+    }
+  }
+
+  // The recording ended inside a pause. See this function's doc.
+  if (openedAt !== undefined && durationSec > openedAt) {
+    silences.push({ startSec: openedAt, endSec: durationSec });
+  }
+  return silences;
+}
+
+/**
+ * Seconds of audio in a stored PCM file, exactly.
+ *
+ * Exact rather than rounded, and that distinction cost a bug: `pcmDurationMs`
+ * answers whole MILLISECONDS, so a 640,500-byte file reports 20,016 ms where it
+ * really holds 20,015.625. Planning from the rounded number put the last segment's
+ * `endByte` at 640,512 — twelve bytes past the end of the file. `readUpload` clamps
+ * a window to the stored size, so nothing threw; the plan was simply describing
+ * audio that does not exist. Verified against a real ffmpeg, which is the only
+ * place a 12-byte error was ever going to show up.
+ *
+ * So {@link planSegments} takes the BYTE COUNT and derives its own seconds. The
+ * milliseconds a page displays can round; the offsets a fan-out reads must not.
+ */
+export function durationSeconds(totalBytes: number): number {
+  return totalBytes / BYTES_PER_SECOND;
+}
+
+/**
+ * Where to cut, given where the pauses are.
+ *
+ * Greedy from the front: a segment grows until the next cut candidate would take
+ * it past {@link MAX_SEGMENT_SECONDS}, so it ends at the LAST pause that still
+ * fits. Segments are therefore contiguous and non-overlapping — together they are
+ * the whole recording, each one addressable as a single `readUpload` window.
+ *
+ * Three properties, each of which a simpler version gets wrong:
+ *
+ * - **The cut is the pause's MIDPOINT**, not its start or its end. Cutting at the
+ *   start clips the decay of the last word before it; cutting at the end clips the
+ *   attack of the first word after. The middle of a 0.6-second pause leaves 0.3
+ *   seconds of room on both sides, which is more than any consonant needs.
+ * - **A pause is a candidate, not a cut.** A recording with a pause every three
+ *   seconds has hundreds of them; cutting at each would be hundreds of requests
+ *   for a twenty-minute call. The silence between two kept spans stays INSIDE a
+ *   segment, which is both cheaper and what keeps the byte range contiguous.
+ * - **No candidate in range means a blind cut**, at exactly
+ *   {@link MAX_SEGMENT_SECONDS}, flagged with `cutInSpeech`. An unbroken monologue
+ *   is a real recording, and refusing it to preserve the pretty invariant would be
+ *   the worse trade.
+ *
+ * Pure, and a pure function of journaled values — the silence list and the byte
+ * count both come out of a step result. That is the ordinary determinism rule: a
+ * replay must re-derive the same list in the same order, or the DevKit hands the
+ * Nth journal entry to a different call.
+ *
+ * @param totalBytes - Size of the stored PCM, which is what the segments are byte
+ *   ranges OF. The duration is derived from it rather than passed in; see
+ *   {@link durationSeconds} for the twelve-byte bug that is there to prevent.
+ */
+export function planSegments(silences: readonly Silence[], totalBytes: number): Segment[] {
+  const durationSec = durationSeconds(totalBytes);
+  if (durationSec <= 0) return [];
+
+  // Midpoints, in order, of every reported pause.
+  //
+  // **The threshold is NOT re-applied here, and that is a fix rather than an
+  // omission.** `silencedetect` already enforced {@link MIN_SILENCE_SECONDS}, so a
+  // second `endSec - startSec >= 0.6` looks free and is a floating-point trap: a
+  // pause from 30 to 30.6 measures 0.5999999999999996, so the check drops it and
+  // the desk falls back to a blind cut on a recording that had a perfectly good
+  // pause to cut in. Caught by a spec, which is the argument for this module being
+  // pure. What is left is the one condition the parser can produce and the planner
+  // cannot use: an empty pause, or one at either edge of the recording.
+  const candidates = silences
+    .filter((gap) => gap.endSec > gap.startSec)
+    .map((gap) => (gap.startSec + gap.endSec) / 2)
+    .filter((at) => at > 0 && at < durationSec);
+
+  const cuts: number[] = [];
+  let at = 0;
+  while (durationSec - at > MAX_SEGMENT_SECONDS) {
+    const limit = at + MAX_SEGMENT_SECONDS;
+    // The last candidate that still fits, and strictly after where we are — a
+    // candidate at `at` would make a zero-length segment and never advance.
+    let chosen: number | undefined;
+    for (const candidate of candidates) {
+      if (candidate > at && candidate <= limit) chosen = candidate;
+      if (candidate > limit) break;
+    }
+    cuts.push(chosen ?? limit);
+    at = chosen ?? limit;
+  }
+
+  // Whether a boundary is a cut this planner INVENTED, rather than a pause it found
+  // or the recording's own end. One expression, used by both branches below —
+  // computing it twice is how they came to disagree in a first draft.
+  const blind = (endSec: number): boolean => cuts.includes(endSec) && !candidates.includes(endSec);
+
+  const bounds = [0, ...cuts, durationSec];
+  const segments: Segment[] = [];
+  for (let i = 0; i + 1 < bounds.length; i += 1) {
+    const startSec = bounds[i] ?? 0;
+    const endSec = bounds[i + 1] ?? durationSec;
+    // A tail too short to be worth a request joins its predecessor rather than
+    // being dropped: the words in it are words, and one longer request is cheaper
+    // than one more round trip.
+    //
+    // **Only if the merge stays under the cap.** The greedy loop leaves a final
+    // segment of at most {@link MAX_SEGMENT_SECONDS}, so absorbing a
+    // sub-{@link MIN_SEGMENT_SECONDS} tail into a segment already at the cap makes
+    // one 110.9 seconds long — still inside the endpoint's own 120-second limit,
+    // and outside the bound this module promises. A short final request is the
+    // cheaper mistake, and it is still an order of magnitude above the 80ms the
+    // endpoint refuses.
+    const previous = segments.at(-1);
+    const merged = previous === undefined ? 0 : endSec - previous.startMs / 1000;
+    if (
+      endSec - startSec < MIN_SEGMENT_SECONDS &&
+      previous !== undefined &&
+      merged <= MAX_SEGMENT_SECONDS
+    ) {
+      previous.endByte = byteAt(endSec);
+      previous.endMs = Math.round(endSec * 1000);
+      previous.cutInSpeech = blind(endSec);
+      continue;
+    }
+    segments.push({
+      index: segments.length,
+      startByte: byteAt(startSec),
+      endByte: byteAt(endSec),
+      startMs: Math.round(startSec * 1000),
+      endMs: Math.round(endSec * 1000),
+      // Only a bound this planner INVENTED is a cut through speech; a bound that
+      // came from `candidates` is a pause, and the recording's own end is neither.
+      cutInSpeech: blind(endSec),
+    });
+  }
+  return segments;
+}
+
+/**
+ * Pass three: the spoken summary, mastered.
+ *
+ * The other direction, and the reason this template runs ffmpeg twice rather than
+ * once. `stepSpeak` answers with a 24 kHz WAV, which is correct and is not a
+ * deliverable: it is uncompressed (a two-minute summary is 5.8 MB, which a page
+ * downloads before it plays anything) and its level is whatever the voice service
+ * chose, so a summary played after the recording it summarizes is jarringly
+ * louder or quieter.
+ *
+ * So: level it to the same {@link LOUDNESS_TARGET_LUFS} as everything else, and
+ * encode it as MP3. One `loudnorm` pass rather than two here, deliberately — a
+ * two-pass measure is worth a decode on a recording of unknown provenance, and
+ * this is 90 seconds of synthesis whose level is already consistent.
+ * `-q:a 4` is VBR at roughly 128 kbit/s, which is transparent for one voice and
+ * about a fortieth of the WAV.
+ */
+export function masterArgs(input: string, output: string): string[] {
+  return [
+    ...standardFlags(),
+    "-loglevel",
+    "error",
+    "-i",
+    input,
+    "-af",
+    `loudnorm=I=${LOUDNESS_TARGET_LUFS}:TP=${LOUDNESS_TRUE_PEAK_DB}:LRA=${LOUDNESS_RANGE_LU}`,
+    "-c:a",
+    "libmp3lame",
+    "-q:a",
+    "4",
+    "-ac",
+    "1",
+    output,
+  ];
+}
+
+/** How much of a recording is speech, as a fraction — the one line a summary needs. */
+export function speechFraction(silences: readonly Silence[], durationSec: number): number {
+  if (durationSec <= 0) return 0;
+  const quiet = silences.reduce((total, gap) => total + Math.max(0, gap.endSec - gap.startSec), 0);
+  return Math.max(0, Math.min(1, (durationSec - quiet) / durationSec));
+}
+
+/** `1:04:09`, or `4:09` under an hour — the shape a reader scans for. */
+export function clock(ms: number): string {
+  const total = Math.max(0, Math.round(ms / 1000));
+  const seconds = String(total % 60).padStart(2, "0");
+  const minutes = Math.floor(total / 60) % 60;
+  const hours = Math.floor(total / 3600);
+  return hours > 0
+    ? `${hours}:${String(minutes).padStart(2, "0")}:${seconds}`
+    : `${minutes}:${seconds}`;
+}
+
+/**
+ * A second, as a byte offset on a sample-frame boundary.
+ *
+ * Rounded DOWN to a frame, because a byte offset mid-sample shifts every sample
+ * after it by one byte — which is not a click, it is white noise that a decoder
+ * transcribes into confident nonsense.
+ */
+function byteAt(seconds: number): number {
+  const frame = (ANALYSIS_FORMAT.channels * ANALYSIS_FORMAT.bitsPerSample) / 8;
+  return Math.floor((seconds * BYTES_PER_SECOND) / frame) * frame;
+}
+
+/**
+ * `lavfi.silence_start=3` → `3`, for the one key asked about.
+ *
+ * The empty check is not defensive padding — `Number("")` is **0**, not `NaN`, so a
+ * truncated line (`lavfi.silence_start=`, which a log cut off mid-write really
+ * produces) would otherwise read as a pause beginning at second zero. That is a cut
+ * candidate at the very start of the recording, which is exactly the kind of wrong
+ * answer that looks like a plausible one.
+ */
+function value(line: string, key: string): number | undefined {
+  if (!line.startsWith(key)) return undefined;
+  const text = line.slice(key.length).trim();
+  if (text === "") return undefined;
+  const parsed = Number(text);
+  return Number.isFinite(parsed) ? parsed : undefined;
+}
+
+/** `JSON.parse` that answers `undefined` rather than throwing, so the caller frames the error. */
+function safeJson(text: string): unknown {
+  try {
+    return JSON.parse(text);
+  } catch {
+    return undefined;
+  }
+}
+
+/**
+ * One of `loudnorm`'s values, as a number.
+ *
+ * Checked rather than coerced: every value arrives as a string, so `Number(…)` on
+ * a key ffmpeg stopped printing yields `NaN`, which then flows into the second
+ * pass's argv as the literal text `NaN` and makes ffmpeg reject the filter with a
+ * message about option parsing. Naming the key here is what turns that into a
+ * sentence about the analysis.
+ */
+function numberAt(raw: Record<string, unknown>, key: string): number {
+  const parsed = Number(raw[key]);
+  if (!Number.isFinite(parsed)) {
+    throw new MediaAnalysisError(
+      `The loudness pass reported no usable \`${key}\` (got ${JSON.stringify(raw[key])}).`,
+    );
+  }
+  return parsed;
+}