Audio Loop

Prepare a source that already joins continuously for Web Audio playback: optionally balance stereo, normalize loudness and encode FLAC. The pipeline does not repair source discontinuities; inspect the join before promising a seamless result.

All ffmpeg work happens in scripts/audio-loop.sh — this skill validates the source, orchestrates the pipeline, and turns the script's summary into a report plus a ready-to-paste JS snippet.

Parameters

Where each flag is handled. -t, -o, and -B pass through to scripts/audio-loop.sh. -v is skill-only — the agent reads it from $ARGUMENTS and interpolates it into the TARGET constant of the emitted JS snippet; the script never sees it. -s / -S follow the repo save-mode convention — the agent translates -s into an -o <save_path> passed to the script, where <save_path> is ~/.agents/output/{project}/audio-loop/{slug}/ ({project} = kebab-cased basename of the git toplevel, else cwd; create it $HOME-expanded, report the fully-expanded absolute path — no tilde, no magic).

Deliberately no crossfade flag. If the source WAV has a real sample-level discontinuity at the loop boundary, that's source editing — outside this skill's scope. If the user is hearing a bump with FLAC + Web Audio, see Diagnostic by negative result below.

Workflow

1. Validate tools

command -v ffmpeg ffprobe. Missing → stop and ask the user to install (macOS: brew install ffmpeg, Debian/Ubuntu: sudo apt install ffmpeg). Never auto-install.

2. Probe the source

The script reads duration, sample rate, channel count, and per-channel RMS via astats. Surface these early — the user sees the starting point before any processing.

3. Diagnose stereo imbalance (stereo sources only)

The script uses 1 dB RMS as its chosen correction threshold, applying a pan filter unless -B is set. Use -B to preserve intentional stereo asymmetry:

pan=stereo|c0=FL|c1=<gain>*FR     # if R is louder
pan=stereo|c0=<gain>*FL|c1=FR     # if L is louder

Where gain = 10^(-|delta_dB| / 20). The script computes delta_dB from the astats pass, picks the right direction, and wires the filter accordingly. At or below the chosen 1 dB threshold no filter is applied. This is a workflow default, not a universal perceptual guarantee.

4. Run the pipeline

$SKILL_DIR = this skill's folder — ${CLAUDE_SKILL_DIR} in Claude Code, the directory containing this SKILL.md elsewhere.

bash "$SKILL_DIR"/scripts/audio-loop.sh <input> [flags]

The script chains: probe → (optional pan correction) → loudnorm=I=<target>:TP=-2:LRA=7 → aresample=<source_rate> → asetnsamples=n=4608:p=0 (bounded frames without padding) → encode FLAC (-c:a flac -compression_level 8). It emits RESULT: key=value lines on stdout.

5. Report

Parse RESULT: lines and compose:

• Size comparison (source → FLAC, MB)
• Final integrated loudness + true peak
• Per-channel RMS confirming the balance is centred
• A drop-in Web Audio JS snippet tuned to the file's URL and the -v target

Report template:

| File | Size | Codec | Integrated LUFS | True peak |
|------|------|-------|-----------------|-----------|
| `<stem>.flac` | X MB | FLAC | -28.0 | -2.0 dBFS |

Stereo balance: L -19.70 dB / R -19.71 dB — centred (Δ 0.01 dB)
Duration: 6.50 s · Sample rate: 48 kHz · Channels: 2

Web Audio snippet (paste-in-page):

<script>
(() => {
  const AUDIO_URL = '/audio/<stem>.flac'; // adjust to where you serve the file
  const TARGET = <v-value>;
  const FADE_MS = 700;

  let ctx, gain, bufferPromise, entered = false;

  function preload() {
    if (ctx) return;
    const Ctor = window.AudioContext || window.webkitAudioContext;
    if (!Ctor) return;
    ctx = new Ctor();
    gain = ctx.createGain();
    gain.gain.value = 0;
    gain.connect(ctx.destination);
    bufferPromise = fetch(AUDIO_URL)
      .then(r => r.arrayBuffer())
      .then(ab => ctx.decodeAudioData(ab));
  }

  async function unlock() {
    if (entered || !ctx) return;
    try { await ctx.resume(); } catch { return; }
    if (ctx.state !== 'running') return;
    let buf;
    try { buf = await bufferPromise; } catch { return; }
    if (entered) return; // second guard — a concurrent unlock() may have finished during the await
    const src = ctx.createBufferSource();
    src.buffer = buf;
    src.loop = true;
    src.connect(gain);
    src.start(0);
    entered = true;
    const now = ctx.currentTime;
    gain.gain.setValueAtTime(0, now);
    gain.gain.linearRampToValueAtTime(TARGET, now + FADE_MS / 1000);
  }

  const IGNORED = new Set(['Shift','Control','Alt','Meta','Tab','Escape','CapsLock']);
  const onKeydown = e => {
    if (IGNORED.has(e.key)) return;
    document.removeEventListener('keydown', onKeydown, { capture: true });
    unlock();
  };
  preload();
  document.addEventListener('pointerdown', () => unlock(), { capture: true, once: true });
  document.addEventListener('keydown', onKeydown, { capture: true });
})();
</script>

For scroll-tied volume or any multi-channel control surface on top of the baseline, see references/scroll-tied-pattern.md — it documents the multiplicative factors architecture (gain = TARGET × fadeInFactor × scrollVolumeFactor) so additional control dimensions compose cleanly.

Verify the loop boundary

Web Audio loops a decoded buffer; codec decoding and the original signal both matter. FLAC preserves the processed PCM, not the original WAV after normalization or balance changes. decodeAudioData may resample to the AudioContext rate: see MDN.

Inspect decoded endpoints and listen across repeated joins in the target browser before claiming seamless playback. A failed or worsening crossfade does not prove a codec defect: source discontinuity, fade placement, processing and playback can all contribute. Compare lossless source and decoded output to isolate the cause. Report size from actual files rather than predicting a compression ratio.

Diagnostic by negative result

Treat a worse crossfade as evidence against that attempted edit, not a definitive diagnosis. If the source itself does not join, identify the required source edit and continue only within the user's authorized scope. The bundled script has no crossfade operation.

Browser autoplay constraint

Modern browsers block audible playback without a prior user gesture. The emitted snippet attaches one-time pointerdown and filtered keydown listeners on document — any real interaction anywhere on the page unlocks the audio, no splash screen or dedicated button required. mousemove, scroll, and wheel are not gestures per the spec; don't try to hook into them. pointermove isn't either. On reload the unlock must happen again — this is a per-navigation browser constraint with no workaround short of the user granting the origin autoplay privilege explicitly.

Rules

• NEVER re-encode an already-encoded output — always start from the original WAV (or lossless source). Re-encoding FLAC→FLAC is pointless; re-encoding AAC→FLAC doesn't recover what AAC threw away.
• Preserve the source sample rate after dynamic loudnorm, which can upsample to 192 kHz. Bound encoder frame size without padding; verify duration/sample count when changing that stage.
• FLAC is the only encoded output the skill produces. The opinion is deliberate — the "AAC is fine for short loops" habit is the failure mode this skill prevents.
• Stereo correction uses the chosen Δ > 1 dB default; it does not establish perceptual centering for every source.
• Default loudness target is -28 LUFS for ambient web audio (quiet-enough-to-not-intrude, loud-enough-to-hear over UI sounds). Louder targets (e.g. -18 for hero music) are a user call — pass -t explicitly.
• Report source/output sizes and playback verification; retain already-authorized encoding scope.
• Mono sources skip the balance step (stereo-only concern); everything else (loudnorm, encode) proceeds as normal.
• When the output path would overwrite the input (FLAC reprocessed in its own directory), the script exits with a clear error — pass -o <dir> to write elsewhere.