diff --git a/CLAUDE.md b/CLAUDE.md
index 00ed568..838df8e 100644
--- a/CLAUDE.md
+++ b/CLAUDE.md
@@ -61,7 +61,7 @@ Dependencies: `requests` (streams), `numpy` + `soundfile` (FLAC output and FLAC
 - **Loud-section detection is adaptive — do not regress it to an absolute threshold.** Per-window dB is compared against a rolling noise floor (`NOISE_PERCENTILE`-th percentile per `NOISE_BLOCK_SECONDS` block, min-smoothed over ±2 blocks so events can't raise their own floor; clamped to ≥ `MIN_RMS`). A section needs `margin` dB of prominence and carries a `score` used for ranking: peak dB above floor, **capped by the sharpest rise within `ONSET_SECONDS` (0.5 s)** — so a short (~10 s) swell that outruns the 30 s floor blocks still flags but scores ≈ 0 and sinks in the U/I highlight ranking, while sharp events keep their full prominence. A section starting in the first 0.5 s of a file is scored against the floor instead (events cut off by a file split must not be punished as swells). Do not regress the scoring to raw peak, and do not fight swells with a higher margin. If flagging itself (not just ranking) ever needs improving, the next step is a spectral filter or optional Silero VAD over candidate sections. Sections shorter than `min_duration` (default 0.5 s, after `min_gap` merging) are discarded — without this, isolated 100 ms pops (clicks, single raindrops) produced thousands of zero-length sections per day. The original fixed RMS threshold flagged every ambience change (passing cars, rain) and produced ~600 useless sections/day — that is why it was replaced. Tests in `tests/test_web.py`.
 - **Analysis params are coupled in five places.** CLI `--margin`/`--min-gap`/`--min-duration` → `/api/config` → UI inputs `#margin-input`/`#min-gap-input`/`#min-duration-input` → `/api/analyze` query params → cache JSON head keys. Renaming or adding a param means touching all five plus `cachedParamsMatch()` and the `_cached_analysis_params()` regex (see the threshold→margin change `c84b7d8` and the min_duration addition).
 - **Analysis cache:** results stored as `<analyses-dir>/<file>.analysis.json` keyed by margin+min_gap+min_duration; orphans pruned at web startup. In Docker the recordings mount is **read-only** for the web container, so docker-compose layers a read-write `./recordings/analyses` bind mount over it. The `detector`, `margin`, `min_gap`, and `min_duration` keys MUST stay first in the cache JSON — `_cached_analysis_params()` reads only the first 256 bytes to avoid parsing the large embedded result. `detector` is `DETECTOR_VERSION`: bump it whenever detection/scoring changes make old cached results wrong (e.g. v2 = onset-capped scores); caches with another version (or missing keys) never match and get overwritten on the next analyse.
-- **Analyze responses:** `/api/analyze` returns `rms_display` (~800 points), never the full per-window RMS list (~45x larger). Since the waveform SVG was removed (user is blind, see webui notes) the bundled UI no longer reads `rms_display` at all — it stays in the payload for API stability and because cached results embed it.
+- **Analyze responses:** `/api/analyze` returns only `sections`, `duration`, `window` — no RMS data of any kind. `rms_display` (~800-point waveform preview) and the full per-window list were dropped when the waveform SVGs were removed (user is blind, see webui notes). Old caches that embed `rms`/`rms_display` are still valid; both keys are popped when serving from cache, so no DETECTOR_VERSION bump was needed.
 - **Section playback uses clips, not seeks:** `/api/clip?file&start&end` decodes the slice server-side (wave/soundfile) and returns a standalone 16-bit WAV with exact Content-Length (capped at `CLIP_MAX_SECONDS`), `Cache-Control: private` so re-listening is free. The UI plays chips/J-K through the bottom clip bar (`clipQueue` in webui.html); seeking the full file only happens via "Open in file". Rationale (finding): libsndfile writes FLAC **without a SEEKTABLE**, so a browser seek bisects the whole multi-hundred-MB file with Range requests — seeking big FLACs in `<audio>` is inherently slow and must not be reintroduced as the primary navigation. Server-side `sf.SoundFile.seek()` on local disk is fast and frame-accurate.
 - **HTTP/1.1 keep-alive:** `_Handler.protocol_version = 'HTTP/1.1'`; every response path must set an accurate `Content-Length`. `_copy_to_response()` force-closes the connection if it under-delivers (file truncated mid-serve).
 - **Live playback:** for files listed in status.json, `/stream/` patches the header on the fly so the browser sees the duration recorded so far and can seek; responses get `Cache-Control: no-store`. WAV: `_live_wav_header` derives sizes from the byte count. FLAC: `_live_flac_header` parses the sample count out of the last frame header in the file tail (CRC-8-verified to reject false sync matches) and rewrites STREAMINFO total_samples — duration is NOT derivable from byte size for FLAC.
diff --git a/README.md b/README.md
index c751eac..2ca644b 100644
--- a/README.md
+++ b/README.md
@@ -186,7 +186,7 @@ Everything the UI does goes through these endpoints, so they can also be scripte
 | Endpoint | Description |
 |----------|-------------|
 | `GET /api/files` | File listing with size, mtime, duration, recording state, cached-analysis params |
-| `GET /api/analyze?file=&margin=&min_gap=&min_duration=` | Loud-section analysis: scored `sections`, `duration` (plus `rms_display`, ~800 RMS points, unused by the bundled UI) |
+| `GET /api/analyze?file=&margin=&min_gap=&min_duration=` | Loud-section analysis: scored `sections`, `duration` |
 | `GET /api/clip?file=&start=&end=` | Section of a WAV/FLAC decoded server-side, returned as a standalone WAV (max 600 s) |
 | `GET /api/cut?file=&start=&end=` | ffmpeg-trimmed copy of the file as a download |
 | `GET /stream/<name>` | Inline playback with HTTP Range support; live files get an on-the-fly patched header |
diff --git a/web.py b/web.py
index fc4c400..8edd33b 100644
--- a/web.py
+++ b/web.py
@@ -402,16 +402,9 @@ def _package_result(rms_values: list, framerate: int, n_frames: int,
     window_dur = window_samples / framerate
     duration   = n_frames / framerate
 
-    if len(rms_values) > 800:
-        step = len(rms_values) / 800
-        rms_display = [rms_values[int(i * step)] for i in range(800)]
-    else:
-        rms_display = rms_values
-
-    # Note: the full per-window RMS list is deliberately NOT returned — the UI
-    # only renders rms_display (~800 points), and the full list is ~45x larger.
+    # Note: no RMS data is returned — the UI is screen-reader oriented and
+    # draws no waveform, so sections + duration is all it needs.
     return {
-        'rms_display': rms_display,
         'sections':    _loud_sections(rms_values, window_dur, duration, margin_db, min_gap, min_duration),
         'duration':    round(duration, 2),
         'window':      round(window_dur, 4),
@@ -690,7 +683,9 @@ class _Handler(BaseHTTPRequestHandler):
                     and cached.get('margin') == margin and cached.get('min_gap') == min_gap
                     and cached.get('min_duration') == min_duration):
                 payload = dict(cached['result'])
-                payload.pop('rms', None)  # caches written before the full-RMS field was dropped
+                # strip fields that older cache versions embedded
+                payload.pop('rms', None)          # full per-window RMS list
+                payload.pop('rms_display', None)  # ~800-point waveform preview
                 payload['cached'] = True
                 self._send(200, json.dumps(payload).encode('utf-8'), 'application/json')
                 return