diff --git a/CLAUDE.md b/CLAUDE.md
index 8ee77aa..4af8041 100644
--- a/CLAUDE.md
+++ b/CLAUDE.md
@@ -5,14 +5,16 @@ Guidance for Claude Code when working in this repository.
 ## Rules
 
 - **Always update `README.md`** when user-facing behaviour changes (flags, endpoints, Docker setup, features), and **commit it in the same commit** as the code change. README is the external reference; CLAUDE.md documents internals.
-- Run `python -m pytest tests/` after changing `isr.py` or `web.py` (tests cover the recorder and the loud-section detector).
+- Run `python -m pytest tests/` after changing `isr.py`, `web.py` or `isrweb/` (tests cover the recorder, the detector, the filename clock and the HTTP layer). Test files mirror the modules: `test_isr.py`, `test_loudness.py`, `test_audiofile.py`, `test_web.py`; `tests/conftest.py` owns the sys.path setup and the `write_wav` fixture.
 
 ## Files
 
 | File | Purpose |
 |------|---------|
 | `isr.py` | Recorder: streams (Icecast/HTTP) + ALSA soundcards, time-aligned file splits |
-| `web.py` | Archive browser: HTTP server, file listing, RMS loudness analysis, cut/delete |
+| `web.py` | Archive browser: HTTP server, file listing, analysis cache, cut/delete. Entry point — stays at the repo root so `python web.py` and the `webui.html` lookup work |
+| `isrweb/loudness.py` | The adaptive loud-section detector. Pure computation, imports nothing from the app — future spectral/VAD work lands here and never touches HTTP |
+| `isrweb/audiofile.py` | What an audio file is and what its name means: the filename clock, FLAC/WAV header parsing, live-header patching. Also a leaf |
 | `webui.html` | Single-page UI (HTML/CSS/JS), loaded by `web.py` at startup — must sit next to `web.py` and be copied in the Dockerfile |
 | `config.ini` | Recording sources; copy from `config.example.ini`. `[general]` gives defaults, every other section is a source (`type = stream` or `type = soundcard`) |
 | `asound.conf` | dsnoop device `shared_mic` so ISR and other ALSA apps can share a soundcard |
@@ -30,11 +32,17 @@ Dependencies: `requests` (streams), `numpy` + `soundfile` (FLAC output and FLAC
 
 ## Code map
 
-`web.py`:
+**Import direction is strictly `web.py → {isrweb.loudness, isrweb.audiofile}`.** The two leaves never import each other or `web.py`; keep it that way, it is what lets the detector be tested without an HTTP server. Each module carries its own optional numpy/soundfile try-import — that boilerplate cannot meaningfully diverge, and sharing it would stop `isr.py` being a standalone file.
+
+`isrweb/loudness.py`:
 - Detection: `_compute_rms_windows_wav()` / `analyze_flac()` produce 100 ms RMS windows (at 48 kHz — `WINDOW_SAMPLES` is a fixed sample count, so windows are wider on lower-rate files; `analyze_wav()` rejects anything that is not 16-bit, which the RMS decoder assumes) → `_noise_floor_db()` estimates the rolling floor → `_loud_sections()` emits scored sections → `_package_result()` shapes the `/api/analyze` payload.
-- Clips: `_api_clip()` validates params, `_clip_wav()` / `_clip_flac()` stream the decoded slice, `_wav_header()` builds the 44-byte PCM header.
+
+`isrweb/audiofile.py`:
 - Filenames as a clock: `_recording_start()` parses the start time out of a filename stem; `_cut_filename()` turns a (stem, ext, start, end) into a wall-clock-named cut. Both the listing `date` field and `_api_cut()` use them.
-- Live headers: `_live_wav_header()`, `_live_flac_header()` (+ `_flac_frame_samples()`, CRC-8 verified).
+- Live headers: `_live_wav_header()`, `_live_flac_header()` (+ `_flac_frame_samples()`, CRC-8 verified). Plus `_wav_header()` (44-byte PCM header for clips), `_get_audio_duration()`, `MIME_TYPES`, `AUDIO_EXTENSIONS`.
+
+`web.py`:
+- Clips: `_api_clip()` validates params, `_clip_wav()` / `_clip_flac()` stream the decoded slice.
 - Serving: `_serve_bytes()` is the single Range-capable file sender — `_stream()` (live header patch via its `prefix` argument, `no_store`) and `_download()` (`Content-Disposition`) both go through it, and it is what makes `do_HEAD` a body-less `do_GET`. Plus `_copy_to_response()`, `_safe_path()` (path traversal guard).
 
 `webui.html` (one `<script>` block):
diff --git a/Dockerfile b/Dockerfile
index 7fcc4de..a66f551 100644
--- a/Dockerfile
+++ b/Dockerfile
@@ -11,6 +11,7 @@ COPY requirements.txt .
 RUN pip install --no-cache-dir -r requirements.txt
 
 COPY isr.py web.py webui.html ./
+COPY isrweb/ ./isrweb/
 
 RUN mkdir -p /recordings
 
diff --git a/README.md b/README.md
index 72225cd..532beeb 100644
--- a/README.md
+++ b/README.md
@@ -149,7 +149,7 @@ python web.py --min-duration 1             # discard loud sections shorter than
 python web.py --analyses-dir /path/to/dir  # where to store analysis cache files (default: <recordings>/analyses)
 ```
 
-The browser UI (HTML/CSS/JS) lives in `webui.html`, which `web.py` loads at startup — keep the two files together.
+The browser UI (HTML/CSS/JS) lives in `webui.html`, which `web.py` loads at startup. `web.py` also imports the `isrweb/` package (the loud-section detector and the audio-format helpers) — keep `web.py`, `webui.html` and `isrweb/` together.
 
 Shows recordings grouped by day with collapsible sections. Features:
 
diff --git a/isrweb/__init__.py b/isrweb/__init__.py
new file mode 100644
index 0000000..0bc82e5
--- /dev/null
+++ b/isrweb/__init__.py
@@ -0,0 +1 @@
+"""Supporting modules for web.py (the archive browser)."""
diff --git a/isrweb/audiofile.py b/isrweb/audiofile.py
new file mode 100644
index 0000000..83b0b89
--- /dev/null
+++ b/isrweb/audiofile.py
@@ -0,0 +1,250 @@
+"""What an audio file is, and what its name means.
+
+Two clusters that share a theme — facts about the files on disk — and share
+the property of being touched only when a container format misbehaves:
+
+* the filename clock (recordings are named for their *start* time, and that
+  name is the only reliable time anchor the archive has), and
+* byte-level format handling: the 44-byte PCM header for clips, duration
+  probing, and the live-header patching that lets a browser show a duration
+  for a file still being recorded.
+
+Imports nothing from the rest of the app.
+"""
+
+import struct
+import wave
+from datetime import datetime, timedelta
+from pathlib import Path
+
+try:
+    import soundfile as sf
+    SOUNDFILE_AVAILABLE = True
+except ImportError:
+    SOUNDFILE_AVAILABLE = False
+
+AUDIO_EXTENSIONS = {'.wav', '.mp3', '.ogg', '.flac', '.aac', '.opus'}
+
+# Recording filenames encode the start time as this strftime format (kept in
+# sync with isr.FILENAME_FORMAT). It is the authoritative recording start and
+# the only reliable clock anchor — mtime drifts to the last write, so cut clip
+# names and the displayed date are both derived from this.
+FILENAME_FORMAT = '%Y%m%d_%H%M%S'
+
+MIME_TYPES = {
+    '.wav':  'audio/wav',
+    '.mp3':  'audio/mpeg',
+    '.ogg':  'audio/ogg',
+    '.flac': 'audio/flac',
+    '.aac':  'audio/aac',
+    '.opus': 'audio/ogg',
+}
+
+# ---------------------------------------------------------------------------
+# The filename clock
+# ---------------------------------------------------------------------------
+
+def _recording_start(stem: str):
+    """Parse the recording start time encoded in a filename stem.
+
+    Returns a datetime, or None if the stem is not in FILENAME_FORMAT
+    (e.g. a manually renamed file). strptime ignores any extension because
+    callers pass Path.stem.
+    """
+    try:
+        return datetime.strptime(stem, FILENAME_FORMAT)
+    except ValueError:
+        return None
+
+
+def _cut_filename(stem: str, ext: str, start: float, end: float) -> str:
+    """Name a cut by the real wall-clock span it covers.
+
+    For a recording that started at 22:00:00, a 22:31:30→22:32:30 slice
+    (start=1890, end=1950) becomes ``20260523_22-31-30_22-32-30.flac``.
+    Falls back to the source stem plus second offsets when the filename is
+    not in FILENAME_FORMAT (e.g. a manually renamed recording).
+    """
+    started = _recording_start(stem)
+    if started is None:
+        return f'{stem}_cut_{int(start)}s-{int(end)}s{ext}'
+    cut_start = started + timedelta(seconds=start)
+    cut_end   = started + timedelta(seconds=end)
+    return f'{cut_start:%Y%m%d}_{cut_start:%H-%M-%S}_{cut_end:%H-%M-%S}{ext}'
+
+
+def _live_wav_header(path: Path, size: int):
+    """Return the WAV header (through the 'data' chunk header) with RIFF and
+    data sizes rewritten to match the current file size, or None.
+
+    While a WAV file is still being recorded its header claims ~0 frames, so
+    browsers show no duration and refuse to seek. Serving a header patched to
+    the bytes recorded so far fixes both; the patch is the same length as the
+    original header, so all byte offsets and Range math stay valid.
+    """
+    try:
+        with open(path, 'rb') as fh:
+            hdr = fh.read(512)
+        if len(hdr) < 44 or hdr[:4] != b'RIFF' or hdr[8:12] != b'WAVE':
+            return None
+        pos = 12
+        while pos + 8 <= len(hdr):
+            chunk_id   = hdr[pos:pos + 4]
+            chunk_size = int.from_bytes(hdr[pos + 4:pos + 8], 'little')
+            if chunk_id == b'data':
+                data_off = pos + 8
+                patched = bytearray(hdr[:data_off])
+                patched[4:8] = (size - 8).to_bytes(4, 'little')
+                patched[pos + 4:pos + 8] = (size - data_off).to_bytes(4, 'little')
+                return bytes(patched)
+            pos += 8 + chunk_size + (chunk_size & 1)
+        return None
+    except Exception:
+        return None
+
+
+# CRC-8 (poly 0x07) used by FLAC frame headers
+_CRC8_TABLE = []
+for _i in range(256):
+    _c = _i
+    for _ in range(8):
+        _c = ((_c << 1) ^ 0x07) & 0xFF if _c & 0x80 else (_c << 1) & 0xFF
+    _CRC8_TABLE.append(_c)
+
+_FLAC_BLOCKSIZES = {1: 192, 2: 576, 3: 1152, 4: 2304, 5: 4608, 8: 256, 9: 512,
+                    10: 1024, 11: 2048, 12: 4096, 13: 8192, 14: 16384, 15: 32768}
+
+
+def _crc8(data: bytes) -> int:
+    crc = 0
+    for b in data:
+        crc = _CRC8_TABLE[crc ^ b]
+    return crc
+
+
+def _flac_coded_number(buf: bytes, pos: int):
+    """Decode the UTF-8-style frame/sample number; returns (value, next_pos)."""
+    b0 = buf[pos]
+    if b0 < 0x80:
+        return b0, pos + 1
+    n, mask = 0, 0x40
+    while b0 & mask:
+        n += 1
+        mask >>= 1
+    if n < 1 or n > 6:  # 10xxxxxx is not a valid leading byte
+        return None
+    val = b0 & (mask - 1)
+    for i in range(1, n + 1):
+        c = buf[pos + i]
+        if c & 0xC0 != 0x80:
+            return None
+        val = (val << 6) | (c & 0x3F)
+    return val, pos + 1 + n
+
+
+def _flac_frame_samples(buf: bytes, pos: int, fixed_bs: int):
+    """If a valid FLAC frame header starts at pos, return the stream sample
+    count through the end of that frame, else None. Validity is confirmed by
+    the header's CRC-8, so false sync matches in compressed data are rejected."""
+    try:
+        variable = buf[pos + 1] & 0x01
+        bs_code  = buf[pos + 2] >> 4
+        sr_code  = buf[pos + 2] & 0x0F
+        if bs_code == 0 or sr_code == 15 or buf[pos + 3] & 0x01:
+            return None
+        if (buf[pos + 3] >> 4) > 10:  # reserved channel assignment
+            return None
+        coded = _flac_coded_number(buf, pos + 4)
+        if coded is None:
+            return None
+        val, p = coded
+        bs = _FLAC_BLOCKSIZES.get(bs_code)
+        if bs_code == 6:
+            bs = buf[p] + 1
+            p += 1
+        elif bs_code == 7:
+            bs = int.from_bytes(buf[p:p + 2], 'big') + 1
+            p += 2
+        if sr_code == 12:
+            p += 1
+        elif sr_code in (13, 14):
+            p += 2
+        if _crc8(buf[pos:p]) != buf[p]:
+            return None
+        if variable:           # val is the frame's starting sample number
+            return val + (bs or 0)
+        return val * (fixed_bs or bs or 4096) + (bs or 0)
+    except IndexError:
+        return None
+
+
+def _live_flac_header(path: Path, size: int):
+    """Return the first 26 bytes of a FLAC file with STREAMINFO total_samples
+    patched to the samples recorded so far, or None.
+
+    Like _live_wav_header, but FLAC duration cannot be derived from the byte
+    count (variable compression). Instead the sample count is parsed out of
+    the last frame header in the file tail — each FLAC frame carries its own
+    frame/sample number.
+    """
+    try:
+        with open(path, 'rb') as fh:
+            head = fh.read(42)
+            if len(head) < 42 or head[:4] != b'fLaC':
+                return None
+            # STREAMINFO must be the first metadata block
+            if head[4] & 0x7F != 0 or int.from_bytes(head[5:8], 'big') != 34:
+                return None
+            fixed_bs = int.from_bytes(head[8:10], 'big')
+
+            tail_len = min(size, 65536)
+            fh.seek(size - tail_len)
+            buf = fh.read(tail_len)
+
+        samples = None
+        for i in range(len(buf) - 20, -1, -1):
+            if buf[i] == 0xFF and (buf[i + 1] & 0xFC) == 0xF8:
+                samples = _flac_frame_samples(buf, i, fixed_bs)
+                if samples:
+                    break
+        if not samples:
+            return None
+
+        # Bytes 18-25 hold: sample rate (20 bits) | channels-1 (3) |
+        # bps-1 (5) | total_samples (36). Replace only the low 36 bits.
+        field = int.from_bytes(head[18:26], 'big')
+        field = (field & ~((1 << 36) - 1)) | min(samples, (1 << 36) - 1)
+        patched = bytearray(head[:26])
+        patched[18:26] = field.to_bytes(8, 'big')
+        return bytes(patched)
+    except Exception:
+        return None
+
+
+def _wav_header(n_frames: int, channels: int, framerate: int, sampwidth: int) -> bytes:
+    """Standard 44-byte PCM WAV header for a clip of known length."""
+    data_len  = n_frames * channels * sampwidth
+    byte_rate = framerate * channels * sampwidth
+    return (b'RIFF' + (36 + data_len).to_bytes(4, 'little') + b'WAVE'
+            + b'fmt ' + (16).to_bytes(4, 'little')
+            + struct.pack('<HHIIHH', 1, channels, framerate, byte_rate,
+                          channels * sampwidth, sampwidth * 8)
+            + b'data' + data_len.to_bytes(4, 'little'))
+
+
+def _get_audio_duration(path: Path):
+    """Return duration in seconds for any supported audio file, or None."""
+    ext = path.suffix.lower()
+    if ext == '.wav':
+        try:
+            with wave.open(str(path), 'rb') as wf:
+                return round(wf.getnframes() / wf.getframerate(), 2)
+        except Exception:
+            return None
+    if SOUNDFILE_AVAILABLE and ext in ('.flac', '.ogg', '.opus'):
+        try:
+            with sf.SoundFile(path) as f:
+                return round(len(f) / f.samplerate, 2)
+        except Exception:
+            return None
+    return None
diff --git a/isrweb/loudness.py b/isrweb/loudness.py
new file mode 100644
index 0000000..2c68ae4
--- /dev/null
+++ b/isrweb/loudness.py
@@ -0,0 +1,229 @@
+"""Adaptive loud-section detection.
+
+Pure computation: RMS windows -> a rolling noise floor -> scored sections.
+Nothing here knows about HTTP, caching or the archive on disk, and nothing
+here imports the rest of the app — so the detector can be worked on, and
+tested, on its own. That matters because this is the part most likely to
+change: if flagging (rather than ranking) ever needs improving, the next
+step is a spectral filter or an optional Silero VAD over candidate sections,
+and it all lands in this file.
+
+**Detection is adaptive — do not regress it to an absolute threshold.** A
+fixed RMS threshold flagged every ambience change (passing cars, rain) and
+produced ~600 useless sections a day; that is why it was replaced.
+"""
+
+import math
+import struct
+import wave
+
+try:
+    import numpy as np
+    NUMPY_AVAILABLE = True
+except ImportError:
+    NUMPY_AVAILABLE = False
+
+try:
+    import soundfile as sf
+    SOUNDFILE_AVAILABLE = True
+except ImportError:
+    SOUNDFILE_AVAILABLE = False
+
+# ---------------------------------------------------------------------------
+# Tuning
+# ---------------------------------------------------------------------------
+
+WINDOW_SAMPLES   = 4800    # 100 ms at 48 kHz
+MARGIN_DB        = 12.0    # sections must rise this many dB above the noise floor
+MIN_GAP_SECONDS  = 2.0     # merge loud sections separated by less than this
+MIN_DURATION_SECONDS = 0.5 # discard loud sections shorter than this
+
+NOISE_BLOCK_SECONDS = 30.0 # noise floor is estimated per block of this length
+NOISE_PERCENTILE    = 20   # percentile of windowed dB levels taken as the floor
+MIN_RMS             = 0.002  # ≈ −54 dBFS; the floor never drops below this, so
+                             # digital silence does not make every tiny sound loud
+ONSET_SECONDS       = 0.5  # a section's score is capped by its sharpest dB rise
+                           # within this span, so slow swells rank low
+
+# Bumped whenever section detection/scoring changes in a way that makes old
+# cached results wrong (not just differently parameterised). Caches written
+# with another version never match and are recomputed on the next analyse.
+DETECTOR_VERSION = 3   # v3: a section open at EOF ends at its last loud
+                       # window, not at the file duration
+
+def _compute_rms_windows_wav(wf, channels: int, sampwidth: int, framerate: int,
+                              window_samples: int):
+    """Yield rms_0_to_1 for every window in the open wave file."""
+    while True:
+        raw = wf.readframes(window_samples)
+        if not raw:
+            break
+        n_samp = len(raw) // (sampwidth * channels)
+        if n_samp == 0:
+            break
+
+        if NUMPY_AVAILABLE:
+            arr = np.frombuffer(raw[:n_samp * sampwidth * channels], dtype='<i2')
+            if channels > 1:
+                arr = arr.reshape(-1, channels).mean(axis=1)
+            rms = float(np.sqrt(np.mean(arr.astype(np.float64) ** 2))) / 32768.0
+        else:
+            fmt = f'<{n_samp * channels}h'
+            samples = struct.unpack(fmt, raw[:n_samp * sampwidth * channels])
+            mono = samples[::channels] if channels > 1 else samples
+            rms = math.sqrt(sum(s * s for s in mono) / len(mono)) / 32768.0
+
+        yield round(rms, 5)
+
+
+def _noise_floor_db(db_values: list, window_dur: float) -> list:
+    """Per-window background noise floor in dBFS.
+
+    The floor is the NOISE_PERCENTILE-th percentile of the windowed dB levels
+    in each NOISE_BLOCK_SECONDS block, then min-smoothed over ±2 neighbouring
+    blocks so an event spanning a whole block cannot raise its own floor.
+    Tracks slow ambience changes (day/night, rain, traffic hum) so detection
+    is relative to "how loud it normally is right now"."""
+    n = len(db_values)
+    block = max(1, int(round(NOISE_BLOCK_SECONDS / window_dur)))
+    floors = []
+    for i in range(0, n, block):
+        chunk = sorted(db_values[i:i + block])
+        floors.append(chunk[int(len(chunk) * NOISE_PERCENTILE / 100)])
+    smoothed = [min(floors[max(0, b - 2):b + 3]) for b in range(len(floors))]
+    return [smoothed[min(i // block, len(smoothed) - 1)] for i in range(n)]
+
+
+def _loud_sections(rms_values: list, window_dur: float, duration: float,
+                   margin_db: float, min_gap: float = MIN_GAP_SECONDS,
+                   min_duration: float = MIN_DURATION_SECONDS) -> list:
+    """Sections whose level rises at least margin_db above the local noise
+    floor. Each section carries a 'score': its peak dB above the floor,
+    capped by the sharpest rise observed within ONSET_SECONDS. Real events
+    (voices, impacts, barks) have steep onsets, so their cap equals their
+    peak; a swell that drifts up slower than the noise-floor blocks can track
+    (wind, a distant approaching car) still flags but scores near zero, so
+    score-ranked review (chips, U/I highlights) surfaces events first.
+
+    Sections shorter than min_duration (after min_gap merging) are discarded:
+    without this, every isolated 100 ms window that pops above the floor — a
+    click, a single raindrop — becomes its own section and a day can drown in
+    thousands of sub-second clips."""
+    db = [20 * math.log10(max(r, 1e-6)) for r in rms_values]
+    floor = _noise_floor_db(db, window_dur)
+    min_db = 20 * math.log10(MIN_RMS)
+    onset_win = max(1, int(round(ONSET_SECONDS / window_dur)))
+
+    sections = []
+    start_t = None
+    last_loud_t = None
+    peak = 0.0
+    onset = 0.0
+
+    def _score():
+        return round(max(0.0, min(peak, onset)), 1)
+
+    for i, d in enumerate(db):
+        t = i * window_dur
+        floor_eff = max(floor[i], min_db)
+        if d >= floor_eff + margin_db:
+            if start_t is None:
+                start_t = t
+                peak = 0.0
+                onset = 0.0
+            last_loud_t = t
+            peak = max(peak, d - floor_eff)
+            # Rise within the onset span; a section starting before the file
+            # has history is measured against the floor instead (an event cut
+            # off by a file split must not be punished as a swell).
+            rise = d - db[i - onset_win] if i >= onset_win else d - floor_eff
+            onset = max(onset, rise)
+        else:
+            if start_t is not None and (t - last_loud_t) > min_gap:
+                end_t = last_loud_t + window_dur
+                if end_t - start_t >= min_duration - 1e-9:
+                    sections.append({'start': round(start_t, 1),
+                                     'end':   round(end_t, 1),
+                                     'score': _score()})
+                start_t = None
+                last_loud_t = None
+
+    # A section still open at EOF ends at its last loud window, exactly like
+    # every section closed inside the loop — not at the file duration, which
+    # used to stretch it by up to min_gap of trailing silence and put that
+    # silence into the clip served for it.
+    if start_t is not None and (last_loud_t + window_dur - start_t) >= min_duration - 1e-9:
+        sections.append({'start': round(start_t, 1),
+                         'end':   round(min(last_loud_t + window_dur, duration), 1),
+                         'score': _score()})
+
+    return sections
+
+
+def _package_result(rms_values: list, framerate: int, n_frames: int,
+                    window_samples: int, margin_db: float,
+                    min_gap: float = MIN_GAP_SECONDS,
+                    min_duration: float = MIN_DURATION_SECONDS) -> dict:
+    window_dur = window_samples / framerate
+    duration   = n_frames / framerate
+
+    # Note: no RMS data is returned — the UI is screen-reader oriented and
+    # draws no waveform, so sections + duration is all it needs. The window
+    # length went the same way: it only ever existed to place RMS bars.
+    return {
+        'sections':    _loud_sections(rms_values, window_dur, duration, margin_db, min_gap, min_duration),
+        'duration':    round(duration, 2),
+    }
+
+
+def analyze_wav(path: Path, window_samples: int = WINDOW_SAMPLES,
+                margin_db: float = MARGIN_DB,
+                min_gap: float = MIN_GAP_SECONDS,
+                min_duration: float = MIN_DURATION_SECONDS) -> dict:
+    try:
+        with wave.open(str(path), 'rb') as wf:
+            channels  = wf.getnchannels()
+            sampwidth = wf.getsampwidth()
+            framerate = wf.getframerate()
+            n_frames  = wf.getnframes()
+            # _compute_rms_windows_wav decodes as int16 unconditionally, so an
+            # 8- or 24-bit file would yield garbage RMS and therefore garbage
+            # sections — silently. The recorder only writes 16-bit, but --dir
+            # can point at any directory.
+            if sampwidth != 2:
+                return {'error': f'only 16-bit WAV files can be analysed '
+                                 f'(this one is {sampwidth * 8}-bit)'}
+            rms_values = list(_compute_rms_windows_wav(
+                wf, channels, sampwidth, framerate, window_samples))
+    except Exception as e:
+        return {'error': str(e)}
+
+    return _package_result(rms_values, framerate, n_frames, window_samples, margin_db, min_gap, min_duration)
+
+
+def analyze_flac(path: Path, window_samples: int = WINDOW_SAMPLES,
+                 margin_db: float = MARGIN_DB,
+                 min_gap: float = MIN_GAP_SECONDS,
+                 min_duration: float = MIN_DURATION_SECONDS) -> dict:
+    """Analyse a FLAC file for loudness. Requires numpy and soundfile."""
+    if not NUMPY_AVAILABLE or not SOUNDFILE_AVAILABLE:
+        return {'error': 'FLAC analysis requires: pip install numpy soundfile'}
+
+    try:
+        with sf.SoundFile(path) as f:
+            framerate = f.samplerate
+            channels  = f.channels
+            n_frames  = len(f)
+
+            rms_values = []
+            while True:
+                frames = f.read(window_samples, dtype='float32', always_2d=True)
+                if len(frames) == 0:
+                    break
+                mono = frames.mean(axis=1) if channels > 1 else frames[:, 0]
+                rms  = float(np.sqrt(np.mean(mono.astype(np.float64) ** 2)))
+                rms_values.append(round(rms, 5))
+    except Exception as e:
+        return {'error': str(e)}
+
+    return _package_result(rms_values, framerate, n_frames, window_samples, margin_db, min_gap, min_duration)
diff --git a/tests/conftest.py b/tests/conftest.py
new file mode 100644
index 0000000..f485c9e
--- /dev/null
+++ b/tests/conftest.py
@@ -0,0 +1,30 @@
+"""Shared test setup.
+
+Puts the repo root on sys.path so `import isr`, `import web` and
+`import isrweb.*` work regardless of where pytest is invoked from. This used
+to be an inline sys.path.insert at the top of test_isr.py, which only
+happened to work for the other test modules because that file imported first.
+"""
+
+import os
+import sys
+import wave
+
+import pytest
+
+sys.path.insert(0, os.path.join(os.path.dirname(__file__), ".."))
+
+
+@pytest.fixture
+def write_wav():
+    """Factory for small 16-bit WAV files. Returns the path written."""
+    def _write(path, seconds=2, framerate=8000, channels=1, sampwidth=2,
+               frames=None):
+        with wave.open(str(path), 'wb') as wf:
+            wf.setnchannels(channels)
+            wf.setsampwidth(sampwidth)
+            wf.setframerate(framerate)
+            wf.writeframes(frames if frames is not None
+                           else b'\x00\x00' * (seconds * framerate * channels))
+        return path
+    return _write
diff --git a/tests/test_audiofile.py b/tests/test_audiofile.py
new file mode 100644
index 0000000..93cec2f
--- /dev/null
+++ b/tests/test_audiofile.py
@@ -0,0 +1,35 @@
+"""Tests for the filename clock in isrweb/audiofile.py.
+
+Recordings are named for their *start* time, and that name is the only
+reliable time anchor the archive has — mtime drifts to the last write.
+"""
+
+from isrweb.audiofile import _cut_filename, _recording_start
+
+
+def test_recording_start_parses_standard_name():
+    from datetime import datetime
+    assert _recording_start("20260523_220000") == datetime(2026, 5, 23, 22, 0, 0)
+
+
+def test_recording_start_rejects_nonstandard_name():
+    assert _recording_start("radio1_20260523") is None
+    assert _recording_start("notes") is None
+
+
+def test_cut_filename_uses_wall_clock_span():
+    # Recording started 22:00:00; cut covers 22:31:30 → 22:32:30.
+    name = _cut_filename("20260523_220000", ".flac", 1890, 1950)
+    assert name == "20260523_22-31-30_22-32-30.flac"
+
+
+def test_cut_filename_rolls_over_the_hour():
+    name = _cut_filename("20260523_220000", ".wav", 3590, 3661)
+    assert name == "20260523_22-59-50_23-01-01.wav"
+
+
+def test_cut_filename_falls_back_for_nonstandard_name():
+    name = _cut_filename("mixtape", ".mp3", 740, 750.4)
+    assert name == "mixtape_cut_740s-750s.mp3"
+
+
diff --git a/tests/test_isr.py b/tests/test_isr.py
index d7004ba..dcd2562 100644
--- a/tests/test_isr.py
+++ b/tests/test_isr.py
@@ -31,14 +31,10 @@ def fixed_clock(dt: datetime):
 
 
 # ---------------------------------------------------------------------------
-# Import the module under test
+# Import the module under test (sys.path is set up in conftest.py)
 # ---------------------------------------------------------------------------
 
-import sys
-import os
-sys.path.insert(0, os.path.join(os.path.dirname(__file__), ".."))
-
-import isr  # noqa: E402 (must come after sys.path manipulation)
+import isr
 
 
 # ===========================================================================
diff --git a/tests/test_loudness.py b/tests/test_loudness.py
new file mode 100644
index 0000000..c264c92
--- /dev/null
+++ b/tests/test_loudness.py
@@ -0,0 +1,176 @@
+"""Tests for the adaptive loud-section detector in isrweb/loudness.py."""
+
+import math
+import wave
+
+from isrweb.loudness import _loud_sections, _noise_floor_db, analyze_wav
+
+WINDOW_DUR = 0.1  # 100 ms windows, as produced by WINDOW_SAMPLES at 48 kHz
+
+
+def _run(rms, margin_db=12.0, min_gap=2.0, min_duration=0.5):
+    duration = len(rms) * WINDOW_DUR
+    return _loud_sections(rms, WINDOW_DUR, duration, margin_db, min_gap, min_duration)
+
+
+def test_burst_above_quiet_floor_is_detected():
+    rms = [0.002] * 1200          # 2 min of quiet ambience (−54 dBFS)
+    rms[600:610] = [0.05] * 10    # 1 s burst at −26 dBFS (+28 dB)
+    sections = _run(rms)
+    assert len(sections) == 1
+    s = sections[0]
+    assert s['start'] == 60.0
+    assert 60.5 <= s['end'] <= 62.0
+    assert 26.0 <= s['score'] <= 30.0
+
+
+def test_slow_ambience_swell_is_not_detected():
+    # Level rises 20 dB over 10 minutes and back down — e.g. rain setting in.
+    # The old fixed threshold would have flagged the entire loud half.
+    up   = [0.002 * 10 ** (i / 6000 * 1.0) for i in range(6000)]   # −54 → −34 dB
+    down = list(reversed(up))
+    assert _run(up + down) == []
+
+
+def test_burst_still_detected_on_loud_ambience():
+    # Same +28 dB prominence as the quiet test, but on a 20 dB louder floor.
+    rms = [0.02] * 1200
+    rms[600:610] = [0.5] * 10
+    sections = _run(rms)
+    assert len(sections) == 1
+    assert sections[0]['start'] == 60.0
+
+
+def test_min_rms_floor_suppresses_blips_in_digital_silence():
+    rms = [0.000001] * 1200
+    rms[600:605] = [0.004] * 5    # −48 dBFS: audible blip, but below MIN_RMS+12dB
+    assert _run(rms) == []
+    rms[600:605] = [0.05] * 5     # −26 dBFS clearly clears the clamped floor
+    assert len(_run(rms)) == 1
+
+
+def test_min_gap_merges_nearby_bursts():
+    rms = [0.002] * 1200
+    rms[600:605] = [0.05] * 5
+    rms[615:620] = [0.05] * 5     # 1 s gap < 2 s min_gap → one section
+    rms[900:905] = [0.05] * 5     # 28 s away → separate section
+    sections = _run(rms)
+    assert len(sections) == 2
+    assert sections[0]['start'] == 60.0
+    assert sections[0]['end'] >= 61.5
+    assert sections[1]['start'] == 90.0
+
+
+def test_min_duration_drops_subsecond_blips():
+    # Isolated single-window pops (clicks, single raindrops) spaced wider than
+    # min_gap must not each become their own section — this is what used to
+    # produce thousands of zero-length sections per day.
+    rms = [0.002] * 1200
+    for i in range(600, 660, 30):   # 0.1 s blips, 3 s apart (> min_gap)
+        rms[i] = 0.05
+    assert _run(rms) == []
+    # With the filter disabled they are all reported
+    assert len(_run(rms, min_duration=0.0)) == 2
+
+
+def test_min_duration_keeps_sections_at_or_above_it():
+    rms = [0.002] * 1200
+    rms[600:605] = [0.05] * 5     # exactly 0.5 s
+    sections = _run(rms, min_duration=0.5)
+    assert len(sections) == 1
+    assert sections[0]['start'] == 60.0
+
+
+def test_min_duration_applies_after_gap_merging():
+    # Two sub-min_duration blips within min_gap merge into one section whose
+    # loud span exceeds min_duration — the merged section must survive.
+    rms = [0.002] * 1200
+    rms[600] = 0.05
+    rms[610] = 0.05               # 1 s apart < 2 s min_gap → merged, 1.1 s span
+    sections = _run(rms, min_duration=1.0)
+    assert len(sections) == 1
+    assert sections[0]['start'] == 60.0
+    assert sections[0]['end'] >= 61.0
+
+
+def test_slow_swell_scores_near_zero_sharp_burst_scores_high():
+    # A ~10 s swell rises faster than the 30 s floor blocks can track, so it
+    # still flags — but its score must collapse (onset cap) so it ranks at the
+    # bottom, while a sharp burst with the same peak keeps its full score.
+    rms = [0.002] * 1200
+    for k in range(80):                       # rise −54 → −28 dB over 8 s
+        rms[400 + k] = 0.002 * 10 ** (26 * (k / 80) / 20)
+    rms[480:500] = [0.04] * 20                # hold 2 s at −28 dB
+    for k in range(80):                       # fall back over 8 s
+        rms[500 + k] = 0.04 * 10 ** (-26 * (k / 80) / 20)
+    rms[900:910] = [0.04] * 10                # sharp 1 s burst, same peak
+    sections = _run(rms)
+    assert len(sections) == 2
+    swell, burst = sections
+    assert swell['start'] < 50.0 < 90.0 <= burst['start']
+    assert burst['score'] >= 24.0             # ≈ full 26 dB prominence
+    assert swell['score'] <= 5.0              # capped by its slow onset
+    assert swell['score'] < burst['score']
+
+
+def test_burst_at_file_start_keeps_full_score():
+    # No pre-event history to measure a rise against: the onset cap falls back
+    # to prominence above the floor, so events cut off by a file split are not
+    # punished as swells.
+    rms = [0.05] * 10 + [0.002] * 1190
+    sections = _run(rms)
+    assert len(sections) == 1
+    assert sections[0]['start'] == 0.0
+    assert sections[0]['score'] >= 24.0
+
+
+def test_section_open_at_eof_ends_at_last_loud_window():
+    # Burst ends 1 s before EOF — less than min_gap, so the section is still
+    # open when the windows run out. It must end where the sound ended, not at
+    # the file duration; otherwise the clip carries a second of trailing
+    # silence and disagrees with every section closed mid-file.
+    rms = [0.002] * 1200
+    rms[1180:1190] = [0.05] * 10          # loud 118.0 → 119.0, file is 120.0 s
+    sections = _run(rms)
+    assert len(sections) == 1
+    assert sections[0]['start'] == 118.0
+    assert sections[0]['end'] == 119.0
+
+
+def test_section_running_to_the_very_end_is_capped_at_duration():
+    rms = [0.002] * 1200
+    rms[1190:1200] = [0.05] * 10          # loud right up to EOF
+    sections = _run(rms)
+    assert len(sections) == 1
+    assert sections[0]['end'] == 120.0
+
+
+def test_noise_floor_tracks_blocks_and_ignores_short_events():
+    quiet_db = 20 * math.log10(0.002)
+    db = [quiet_db] * 1200
+    db[600:650] = [-20.0] * 50    # 5 s event must not raise its own floor
+    floor = _noise_floor_db(db, WINDOW_DUR)
+    assert len(floor) == len(db)
+    assert all(abs(f - quiet_db) < 1.0 for f in floor)
+
+
+def test_non_16_bit_wav_is_rejected_rather_than_misread(tmp_path, write_wav):
+    # RMS windowing decodes as int16 regardless of the declared sample width,
+    # so an 8-bit file used to produce meaningless sections with no warning.
+    path = tmp_path / 'eight_bit.wav'
+    with wave.open(str(path), 'wb') as wf:
+        wf.setnchannels(1)
+        wf.setsampwidth(1)
+        wf.setframerate(8000)
+        wf.writeframes(b'\x80' * 8000)
+    result = analyze_wav(path)
+    assert 'error' in result
+    assert '16-bit' in result['error']
+
+
+def test_16_bit_wav_still_analyses(tmp_path, write_wav):
+    path = tmp_path / 'ok.wav'
+    write_wav(path)
+    assert 'error' not in analyze_wav(path)
+
+
diff --git a/tests/test_web.py b/tests/test_web.py
index 92f5b13..7f891f4 100644
--- a/tests/test_web.py
+++ b/tests/test_web.py
@@ -1,189 +1,21 @@
-"""Tests for the adaptive loud-section detector in web.py."""
+"""Tests for the HTTP layer and file listing in web.py."""
 
 import io
-import math
-import wave
 
 import web
-from web import _cut_filename, _loud_sections, _noise_floor_db, _recording_start
-
-WINDOW_DUR = 0.1  # 100 ms windows, as produced by WINDOW_SAMPLES at 48 kHz
 
 
-def _run(rms, margin_db=12.0, min_gap=2.0, min_duration=0.5):
-    duration = len(rms) * WINDOW_DUR
-    return _loud_sections(rms, WINDOW_DUR, duration, margin_db, min_gap, min_duration)
+def test_listing_is_ordered_by_the_filename_clock_not_mtime(tmp_path, write_wav):
+    # mtime is the last write (≈ the end of the recording) and drifts from the
+    # name on any copy or round-trip. Here the older recording is touched last.
+    import os
+    early, late = tmp_path / '20260523_080000.wav', tmp_path / '20260523_090000.wav'
+    write_wav(early)
+    write_wav(late)
+    os.utime(late, (1000, 1000))      # newest recording, oldest mtime
 
-
-def test_burst_above_quiet_floor_is_detected():
-    rms = [0.002] * 1200          # 2 min of quiet ambience (−54 dBFS)
-    rms[600:610] = [0.05] * 10    # 1 s burst at −26 dBFS (+28 dB)
-    sections = _run(rms)
-    assert len(sections) == 1
-    s = sections[0]
-    assert s['start'] == 60.0
-    assert 60.5 <= s['end'] <= 62.0
-    assert 26.0 <= s['score'] <= 30.0
-
-
-def test_slow_ambience_swell_is_not_detected():
-    # Level rises 20 dB over 10 minutes and back down — e.g. rain setting in.
-    # The old fixed threshold would have flagged the entire loud half.
-    up   = [0.002 * 10 ** (i / 6000 * 1.0) for i in range(6000)]   # −54 → −34 dB
-    down = list(reversed(up))
-    assert _run(up + down) == []
-
-
-def test_burst_still_detected_on_loud_ambience():
-    # Same +28 dB prominence as the quiet test, but on a 20 dB louder floor.
-    rms = [0.02] * 1200
-    rms[600:610] = [0.5] * 10
-    sections = _run(rms)
-    assert len(sections) == 1
-    assert sections[0]['start'] == 60.0
-
-
-def test_min_rms_floor_suppresses_blips_in_digital_silence():
-    rms = [0.000001] * 1200
-    rms[600:605] = [0.004] * 5    # −48 dBFS: audible blip, but below MIN_RMS+12dB
-    assert _run(rms) == []
-    rms[600:605] = [0.05] * 5     # −26 dBFS clearly clears the clamped floor
-    assert len(_run(rms)) == 1
-
-
-def test_min_gap_merges_nearby_bursts():
-    rms = [0.002] * 1200
-    rms[600:605] = [0.05] * 5
-    rms[615:620] = [0.05] * 5     # 1 s gap < 2 s min_gap → one section
-    rms[900:905] = [0.05] * 5     # 28 s away → separate section
-    sections = _run(rms)
-    assert len(sections) == 2
-    assert sections[0]['start'] == 60.0
-    assert sections[0]['end'] >= 61.5
-    assert sections[1]['start'] == 90.0
-
-
-def test_min_duration_drops_subsecond_blips():
-    # Isolated single-window pops (clicks, single raindrops) spaced wider than
-    # min_gap must not each become their own section — this is what used to
-    # produce thousands of zero-length sections per day.
-    rms = [0.002] * 1200
-    for i in range(600, 660, 30):   # 0.1 s blips, 3 s apart (> min_gap)
-        rms[i] = 0.05
-    assert _run(rms) == []
-    # With the filter disabled they are all reported
-    assert len(_run(rms, min_duration=0.0)) == 2
-
-
-def test_min_duration_keeps_sections_at_or_above_it():
-    rms = [0.002] * 1200
-    rms[600:605] = [0.05] * 5     # exactly 0.5 s
-    sections = _run(rms, min_duration=0.5)
-    assert len(sections) == 1
-    assert sections[0]['start'] == 60.0
-
-
-def test_min_duration_applies_after_gap_merging():
-    # Two sub-min_duration blips within min_gap merge into one section whose
-    # loud span exceeds min_duration — the merged section must survive.
-    rms = [0.002] * 1200
-    rms[600] = 0.05
-    rms[610] = 0.05               # 1 s apart < 2 s min_gap → merged, 1.1 s span
-    sections = _run(rms, min_duration=1.0)
-    assert len(sections) == 1
-    assert sections[0]['start'] == 60.0
-    assert sections[0]['end'] >= 61.0
-
-
-def test_slow_swell_scores_near_zero_sharp_burst_scores_high():
-    # A ~10 s swell rises faster than the 30 s floor blocks can track, so it
-    # still flags — but its score must collapse (onset cap) so it ranks at the
-    # bottom, while a sharp burst with the same peak keeps its full score.
-    rms = [0.002] * 1200
-    for k in range(80):                       # rise −54 → −28 dB over 8 s
-        rms[400 + k] = 0.002 * 10 ** (26 * (k / 80) / 20)
-    rms[480:500] = [0.04] * 20                # hold 2 s at −28 dB
-    for k in range(80):                       # fall back over 8 s
-        rms[500 + k] = 0.04 * 10 ** (-26 * (k / 80) / 20)
-    rms[900:910] = [0.04] * 10                # sharp 1 s burst, same peak
-    sections = _run(rms)
-    assert len(sections) == 2
-    swell, burst = sections
-    assert swell['start'] < 50.0 < 90.0 <= burst['start']
-    assert burst['score'] >= 24.0             # ≈ full 26 dB prominence
-    assert swell['score'] <= 5.0              # capped by its slow onset
-    assert swell['score'] < burst['score']
-
-
-def test_burst_at_file_start_keeps_full_score():
-    # No pre-event history to measure a rise against: the onset cap falls back
-    # to prominence above the floor, so events cut off by a file split are not
-    # punished as swells.
-    rms = [0.05] * 10 + [0.002] * 1190
-    sections = _run(rms)
-    assert len(sections) == 1
-    assert sections[0]['start'] == 0.0
-    assert sections[0]['score'] >= 24.0
-
-
-def test_section_open_at_eof_ends_at_last_loud_window():
-    # Burst ends 1 s before EOF — less than min_gap, so the section is still
-    # open when the windows run out. It must end where the sound ended, not at
-    # the file duration; otherwise the clip carries a second of trailing
-    # silence and disagrees with every section closed mid-file.
-    rms = [0.002] * 1200
-    rms[1180:1190] = [0.05] * 10          # loud 118.0 → 119.0, file is 120.0 s
-    sections = _run(rms)
-    assert len(sections) == 1
-    assert sections[0]['start'] == 118.0
-    assert sections[0]['end'] == 119.0
-
-
-def test_section_running_to_the_very_end_is_capped_at_duration():
-    rms = [0.002] * 1200
-    rms[1190:1200] = [0.05] * 10          # loud right up to EOF
-    sections = _run(rms)
-    assert len(sections) == 1
-    assert sections[0]['end'] == 120.0
-
-
-def test_noise_floor_tracks_blocks_and_ignores_short_events():
-    quiet_db = 20 * math.log10(0.002)
-    db = [quiet_db] * 1200
-    db[600:650] = [-20.0] * 50    # 5 s event must not raise its own floor
-    floor = _noise_floor_db(db, WINDOW_DUR)
-    assert len(floor) == len(db)
-    assert all(abs(f - quiet_db) < 1.0 for f in floor)
-
-
-# ---------------------------------------------------------------------------
-# Filename parsing / cut naming
-# ---------------------------------------------------------------------------
-
-def test_recording_start_parses_standard_name():
-    from datetime import datetime
-    assert _recording_start("20260523_220000") == datetime(2026, 5, 23, 22, 0, 0)
-
-
-def test_recording_start_rejects_nonstandard_name():
-    assert _recording_start("radio1_20260523") is None
-    assert _recording_start("notes") is None
-
-
-def test_cut_filename_uses_wall_clock_span():
-    # Recording started 22:00:00; cut covers 22:31:30 → 22:32:30.
-    name = _cut_filename("20260523_220000", ".flac", 1890, 1950)
-    assert name == "20260523_22-31-30_22-32-30.flac"
-
-
-def test_cut_filename_rolls_over_the_hour():
-    name = _cut_filename("20260523_220000", ".wav", 3590, 3661)
-    assert name == "20260523_22-59-50_23-01-01.wav"
-
-
-def test_cut_filename_falls_back_for_nonstandard_name():
-    name = _cut_filename("mixtape", ".mp3", 740, 750.4)
-    assert name == "mixtape_cut_740s-750s.mp3"
+    names = [f['name'] for f in web.list_files(str(tmp_path))]
+    assert names == ['20260523_090000.wav', '20260523_080000.wav']
 
 
 # ---------------------------------------------------------------------------
@@ -211,52 +43,11 @@ class _OfflineHandler(web._Handler):
         pass
 
 
-def _write_wav(path, seconds=2, framerate=8000):
-    with wave.open(str(path), 'wb') as wf:
-        wf.setnchannels(1)
-        wf.setsampwidth(2)
-        wf.setframerate(framerate)
-        wf.writeframes(b'\x00\x00' * (seconds * framerate))
-
-
-def test_listing_is_ordered_by_the_filename_clock_not_mtime(tmp_path):
-    # mtime is the last write (≈ the end of the recording) and drifts from the
-    # name on any copy or round-trip. Here the older recording is touched last.
-    import os
-    early, late = tmp_path / '20260523_080000.wav', tmp_path / '20260523_090000.wav'
-    _write_wav(early)
-    _write_wav(late)
-    os.utime(late, (1000, 1000))      # newest recording, oldest mtime
-
-    names = [f['name'] for f in web.list_files(str(tmp_path))]
-    assert names == ['20260523_090000.wav', '20260523_080000.wav']
-
-
-def test_non_16_bit_wav_is_rejected_rather_than_misread(tmp_path):
-    # RMS windowing decodes as int16 regardless of the declared sample width,
-    # so an 8-bit file used to produce meaningless sections with no warning.
-    path = tmp_path / 'eight_bit.wav'
-    with wave.open(str(path), 'wb') as wf:
-        wf.setnchannels(1)
-        wf.setsampwidth(1)
-        wf.setframerate(8000)
-        wf.writeframes(b'\x80' * 8000)
-    result = web.analyze_wav(path)
-    assert 'error' in result
-    assert '16-bit' in result['error']
-
-
-def test_16_bit_wav_still_analyses(tmp_path):
-    path = tmp_path / 'ok.wav'
-    _write_wav(path)
-    assert 'error' not in web.analyze_wav(path)
-
-
-def test_clip_failure_after_headers_closes_connection_instead_of_writing_json(tmp_path):
+def test_clip_failure_after_headers_closes_connection_instead_of_writing_json(tmp_path, write_wav):
     # Once the audio headers are out, an error body would be a second response
     # on a keep-alive connection that already promised audio/wav plus an exact
     # Content-Length — desyncing every later request on that connection.
-    _write_wav(tmp_path / 'rec.wav')
+    write_wav(tmp_path / 'rec.wav')
     h = _OfflineHandler(tmp_path)
 
     def boom(path, start, end):
@@ -271,8 +62,8 @@ def test_clip_failure_after_headers_closes_connection_instead_of_writing_json(tm
     assert b'error' not in h.wfile.getvalue()  # no JSON body
 
 
-def test_clip_failure_before_headers_still_reports_json(tmp_path):
-    _write_wav(tmp_path / 'rec.wav')
+def test_clip_failure_before_headers_still_reports_json(tmp_path, write_wav):
+    write_wav(tmp_path / 'rec.wav')
     h = _OfflineHandler(tmp_path)
 
     def boom(path, start, end):
diff --git a/web.py b/web.py
index 557f40e..d20d90b 100644
--- a/web.py
+++ b/web.py
@@ -15,24 +15,22 @@ Usage:
 
 import argparse
 import json
-import math
 import os
 import re
 import shutil
-import struct
 import subprocess
 import sys
 import tempfile
 import threading
 import wave
-from datetime import datetime, timedelta
+from datetime import datetime
 from http.server import BaseHTTPRequestHandler, ThreadingHTTPServer
 from pathlib import Path
 from urllib.parse import parse_qs, unquote, urlparse
 
 try:
-    import numpy as np
-    NUMPY_AVAILABLE = True
+    import numpy       # noqa: F401 — only the availability flag is used here,
+    NUMPY_AVAILABLE = True   # to refuse FLAC work rather than fail halfway
 except ImportError:
     NUMPY_AVAILABLE = False
 
@@ -42,434 +40,22 @@ try:
 except ImportError:
     SOUNDFILE_AVAILABLE = False
 
+# The detector and the byte-level format helpers live beside this file. Both
+# are leaves: they import nothing from here, so `web.py -> {loudness,
+# audiofile}` is the only import edge and there are no cycles.
+from isrweb.audiofile import (AUDIO_EXTENSIONS, MIME_TYPES, _cut_filename,
+                              _get_audio_duration, _live_flac_header,
+                              _live_wav_header, _recording_start, _wav_header)
+from isrweb.loudness import (DETECTOR_VERSION, MARGIN_DB, MIN_DURATION_SECONDS,
+                             MIN_GAP_SECONDS, analyze_flac, analyze_wav)
+
+
 # ---------------------------------------------------------------------------
 # Constants
 # ---------------------------------------------------------------------------
 
-AUDIO_EXTENSIONS = {'.wav', '.mp3', '.ogg', '.flac', '.aac', '.opus'}
-WINDOW_SAMPLES   = 4800    # 100 ms at 48 kHz
-MARGIN_DB        = 12.0    # sections must rise this many dB above the noise floor
-MIN_GAP_SECONDS  = 2.0     # merge loud sections separated by less than this
-MIN_DURATION_SECONDS = 0.5 # discard loud sections shorter than this
-
-NOISE_BLOCK_SECONDS = 30.0 # noise floor is estimated per block of this length
-NOISE_PERCENTILE    = 20   # percentile of windowed dB levels taken as the floor
-MIN_RMS             = 0.002  # ≈ −54 dBFS; the floor never drops below this, so
-                             # digital silence does not make every tiny sound loud
-ONSET_SECONDS       = 0.5  # a section's score is capped by its sharpest dB rise
-                           # within this span, so slow swells rank low
-
-# Bumped whenever section detection/scoring changes in a way that makes old
-# cached results wrong (not just differently parameterised). Caches written
-# with another version never match and are recomputed on the next analyse.
-DETECTOR_VERSION = 3   # v3: a section open at EOF ends at its last loud
-                       # window, not at the file duration
-
 CLIP_MAX_SECONDS = 600     # upper bound on /api/clip length
 
-# Recording filenames encode the start time as this strftime format (kept in
-# sync with isr.FILENAME_FORMAT). It is the authoritative recording start and
-# the only reliable clock anchor — mtime drifts to the last write, so cut clip
-# names and the displayed date are both derived from this.
-FILENAME_FORMAT = '%Y%m%d_%H%M%S'
-
-MIME_TYPES = {
-    '.wav':  'audio/wav',
-    '.mp3':  'audio/mpeg',
-    '.ogg':  'audio/ogg',
-    '.flac': 'audio/flac',
-    '.aac':  'audio/aac',
-    '.opus': 'audio/ogg',
-}
-
-
-# ---------------------------------------------------------------------------
-# Audio analysis helpers
-# ---------------------------------------------------------------------------
-
-def _recording_start(stem: str):
-    """Parse the recording start time encoded in a filename stem.
-
-    Returns a datetime, or None if the stem is not in FILENAME_FORMAT
-    (e.g. a manually renamed file). strptime ignores any extension because
-    callers pass Path.stem.
-    """
-    try:
-        return datetime.strptime(stem, FILENAME_FORMAT)
-    except ValueError:
-        return None
-
-
-def _cut_filename(stem: str, ext: str, start: float, end: float) -> str:
-    """Name a cut by the real wall-clock span it covers.
-
-    For a recording that started at 22:00:00, a 22:31:30→22:32:30 slice
-    (start=1890, end=1950) becomes ``20260523_22-31-30_22-32-30.flac``.
-    Falls back to the source stem plus second offsets when the filename is
-    not in FILENAME_FORMAT (e.g. a manually renamed recording).
-    """
-    started = _recording_start(stem)
-    if started is None:
-        return f'{stem}_cut_{int(start)}s-{int(end)}s{ext}'
-    cut_start = started + timedelta(seconds=start)
-    cut_end   = started + timedelta(seconds=end)
-    return f'{cut_start:%Y%m%d}_{cut_start:%H-%M-%S}_{cut_end:%H-%M-%S}{ext}'
-
-
-def _live_wav_header(path: Path, size: int):
-    """Return the WAV header (through the 'data' chunk header) with RIFF and
-    data sizes rewritten to match the current file size, or None.
-
-    While a WAV file is still being recorded its header claims ~0 frames, so
-    browsers show no duration and refuse to seek. Serving a header patched to
-    the bytes recorded so far fixes both; the patch is the same length as the
-    original header, so all byte offsets and Range math stay valid.
-    """
-    try:
-        with open(path, 'rb') as fh:
-            hdr = fh.read(512)
-        if len(hdr) < 44 or hdr[:4] != b'RIFF' or hdr[8:12] != b'WAVE':
-            return None
-        pos = 12
-        while pos + 8 <= len(hdr):
-            chunk_id   = hdr[pos:pos + 4]
-            chunk_size = int.from_bytes(hdr[pos + 4:pos + 8], 'little')
-            if chunk_id == b'data':
-                data_off = pos + 8
-                patched = bytearray(hdr[:data_off])
-                patched[4:8] = (size - 8).to_bytes(4, 'little')
-                patched[pos + 4:pos + 8] = (size - data_off).to_bytes(4, 'little')
-                return bytes(patched)
-            pos += 8 + chunk_size + (chunk_size & 1)
-        return None
-    except Exception:
-        return None
-
-
-# CRC-8 (poly 0x07) used by FLAC frame headers
-_CRC8_TABLE = []
-for _i in range(256):
-    _c = _i
-    for _ in range(8):
-        _c = ((_c << 1) ^ 0x07) & 0xFF if _c & 0x80 else (_c << 1) & 0xFF
-    _CRC8_TABLE.append(_c)
-
-_FLAC_BLOCKSIZES = {1: 192, 2: 576, 3: 1152, 4: 2304, 5: 4608, 8: 256, 9: 512,
-                    10: 1024, 11: 2048, 12: 4096, 13: 8192, 14: 16384, 15: 32768}
-
-
-def _crc8(data: bytes) -> int:
-    crc = 0
-    for b in data:
-        crc = _CRC8_TABLE[crc ^ b]
-    return crc
-
-
-def _flac_coded_number(buf: bytes, pos: int):
-    """Decode the UTF-8-style frame/sample number; returns (value, next_pos)."""
-    b0 = buf[pos]
-    if b0 < 0x80:
-        return b0, pos + 1
-    n, mask = 0, 0x40
-    while b0 & mask:
-        n += 1
-        mask >>= 1
-    if n < 1 or n > 6:  # 10xxxxxx is not a valid leading byte
-        return None
-    val = b0 & (mask - 1)
-    for i in range(1, n + 1):
-        c = buf[pos + i]
-        if c & 0xC0 != 0x80:
-            return None
-        val = (val << 6) | (c & 0x3F)
-    return val, pos + 1 + n
-
-
-def _flac_frame_samples(buf: bytes, pos: int, fixed_bs: int):
-    """If a valid FLAC frame header starts at pos, return the stream sample
-    count through the end of that frame, else None. Validity is confirmed by
-    the header's CRC-8, so false sync matches in compressed data are rejected."""
-    try:
-        variable = buf[pos + 1] & 0x01
-        bs_code  = buf[pos + 2] >> 4
-        sr_code  = buf[pos + 2] & 0x0F
-        if bs_code == 0 or sr_code == 15 or buf[pos + 3] & 0x01:
-            return None
-        if (buf[pos + 3] >> 4) > 10:  # reserved channel assignment
-            return None
-        coded = _flac_coded_number(buf, pos + 4)
-        if coded is None:
-            return None
-        val, p = coded
-        bs = _FLAC_BLOCKSIZES.get(bs_code)
-        if bs_code == 6:
-            bs = buf[p] + 1
-            p += 1
-        elif bs_code == 7:
-            bs = int.from_bytes(buf[p:p + 2], 'big') + 1
-            p += 2
-        if sr_code == 12:
-            p += 1
-        elif sr_code in (13, 14):
-            p += 2
-        if _crc8(buf[pos:p]) != buf[p]:
-            return None
-        if variable:           # val is the frame's starting sample number
-            return val + (bs or 0)
-        return val * (fixed_bs or bs or 4096) + (bs or 0)
-    except IndexError:
-        return None
-
-
-def _live_flac_header(path: Path, size: int):
-    """Return the first 26 bytes of a FLAC file with STREAMINFO total_samples
-    patched to the samples recorded so far, or None.
-
-    Like _live_wav_header, but FLAC duration cannot be derived from the byte
-    count (variable compression). Instead the sample count is parsed out of
-    the last frame header in the file tail — each FLAC frame carries its own
-    frame/sample number.
-    """
-    try:
-        with open(path, 'rb') as fh:
-            head = fh.read(42)
-            if len(head) < 42 or head[:4] != b'fLaC':
-                return None
-            # STREAMINFO must be the first metadata block
-            if head[4] & 0x7F != 0 or int.from_bytes(head[5:8], 'big') != 34:
-                return None
-            fixed_bs = int.from_bytes(head[8:10], 'big')
-
-            tail_len = min(size, 65536)
-            fh.seek(size - tail_len)
-            buf = fh.read(tail_len)
-
-        samples = None
-        for i in range(len(buf) - 20, -1, -1):
-            if buf[i] == 0xFF and (buf[i + 1] & 0xFC) == 0xF8:
-                samples = _flac_frame_samples(buf, i, fixed_bs)
-                if samples:
-                    break
-        if not samples:
-            return None
-
-        # Bytes 18-25 hold: sample rate (20 bits) | channels-1 (3) |
-        # bps-1 (5) | total_samples (36). Replace only the low 36 bits.
-        field = int.from_bytes(head[18:26], 'big')
-        field = (field & ~((1 << 36) - 1)) | min(samples, (1 << 36) - 1)
-        patched = bytearray(head[:26])
-        patched[18:26] = field.to_bytes(8, 'big')
-        return bytes(patched)
-    except Exception:
-        return None
-
-
-def _wav_header(n_frames: int, channels: int, framerate: int, sampwidth: int) -> bytes:
-    """Standard 44-byte PCM WAV header for a clip of known length."""
-    data_len  = n_frames * channels * sampwidth
-    byte_rate = framerate * channels * sampwidth
-    return (b'RIFF' + (36 + data_len).to_bytes(4, 'little') + b'WAVE'
-            + b'fmt ' + (16).to_bytes(4, 'little')
-            + struct.pack('<HHIIHH', 1, channels, framerate, byte_rate,
-                          channels * sampwidth, sampwidth * 8)
-            + b'data' + data_len.to_bytes(4, 'little'))
-
-
-def _get_audio_duration(path: Path):
-    """Return duration in seconds for any supported audio file, or None."""
-    ext = path.suffix.lower()
-    if ext == '.wav':
-        try:
-            with wave.open(str(path), 'rb') as wf:
-                return round(wf.getnframes() / wf.getframerate(), 2)
-        except Exception:
-            return None
-    if SOUNDFILE_AVAILABLE and ext in ('.flac', '.ogg', '.opus'):
-        try:
-            with sf.SoundFile(path) as f:
-                return round(len(f) / f.samplerate, 2)
-        except Exception:
-            return None
-    return None
-
-
-def _compute_rms_windows_wav(wf, channels: int, sampwidth: int, framerate: int,
-                              window_samples: int):
-    """Yield rms_0_to_1 for every window in the open wave file."""
-    while True:
-        raw = wf.readframes(window_samples)
-        if not raw:
-            break
-        n_samp = len(raw) // (sampwidth * channels)
-        if n_samp == 0:
-            break
-
-        if NUMPY_AVAILABLE:
-            arr = np.frombuffer(raw[:n_samp * sampwidth * channels], dtype='<i2')
-            if channels > 1:
-                arr = arr.reshape(-1, channels).mean(axis=1)
-            rms = float(np.sqrt(np.mean(arr.astype(np.float64) ** 2))) / 32768.0
-        else:
-            fmt = f'<{n_samp * channels}h'
-            samples = struct.unpack(fmt, raw[:n_samp * sampwidth * channels])
-            mono = samples[::channels] if channels > 1 else samples
-            rms = math.sqrt(sum(s * s for s in mono) / len(mono)) / 32768.0
-
-        yield round(rms, 5)
-
-
-def _noise_floor_db(db_values: list, window_dur: float) -> list:
-    """Per-window background noise floor in dBFS.
-
-    The floor is the NOISE_PERCENTILE-th percentile of the windowed dB levels
-    in each NOISE_BLOCK_SECONDS block, then min-smoothed over ±2 neighbouring
-    blocks so an event spanning a whole block cannot raise its own floor.
-    Tracks slow ambience changes (day/night, rain, traffic hum) so detection
-    is relative to "how loud it normally is right now"."""
-    n = len(db_values)
-    block = max(1, int(round(NOISE_BLOCK_SECONDS / window_dur)))
-    floors = []
-    for i in range(0, n, block):
-        chunk = sorted(db_values[i:i + block])
-        floors.append(chunk[int(len(chunk) * NOISE_PERCENTILE / 100)])
-    smoothed = [min(floors[max(0, b - 2):b + 3]) for b in range(len(floors))]
-    return [smoothed[min(i // block, len(smoothed) - 1)] for i in range(n)]
-
-
-def _loud_sections(rms_values: list, window_dur: float, duration: float,
-                   margin_db: float, min_gap: float = MIN_GAP_SECONDS,
-                   min_duration: float = MIN_DURATION_SECONDS) -> list:
-    """Sections whose level rises at least margin_db above the local noise
-    floor. Each section carries a 'score': its peak dB above the floor,
-    capped by the sharpest rise observed within ONSET_SECONDS. Real events
-    (voices, impacts, barks) have steep onsets, so their cap equals their
-    peak; a swell that drifts up slower than the noise-floor blocks can track
-    (wind, a distant approaching car) still flags but scores near zero, so
-    score-ranked review (chips, U/I highlights) surfaces events first.
-
-    Sections shorter than min_duration (after min_gap merging) are discarded:
-    without this, every isolated 100 ms window that pops above the floor — a
-    click, a single raindrop — becomes its own section and a day can drown in
-    thousands of sub-second clips."""
-    db = [20 * math.log10(max(r, 1e-6)) for r in rms_values]
-    floor = _noise_floor_db(db, window_dur)
-    min_db = 20 * math.log10(MIN_RMS)
-    onset_win = max(1, int(round(ONSET_SECONDS / window_dur)))
-
-    sections = []
-    start_t = None
-    last_loud_t = None
-    peak = 0.0
-    onset = 0.0
-
-    def _score():
-        return round(max(0.0, min(peak, onset)), 1)
-
-    for i, d in enumerate(db):
-        t = i * window_dur
-        floor_eff = max(floor[i], min_db)
-        if d >= floor_eff + margin_db:
-            if start_t is None:
-                start_t = t
-                peak = 0.0
-                onset = 0.0
-            last_loud_t = t
-            peak = max(peak, d - floor_eff)
-            # Rise within the onset span; a section starting before the file
-            # has history is measured against the floor instead (an event cut
-            # off by a file split must not be punished as a swell).
-            rise = d - db[i - onset_win] if i >= onset_win else d - floor_eff
-            onset = max(onset, rise)
-        else:
-            if start_t is not None and (t - last_loud_t) > min_gap:
-                end_t = last_loud_t + window_dur
-                if end_t - start_t >= min_duration - 1e-9:
-                    sections.append({'start': round(start_t, 1),
-                                     'end':   round(end_t, 1),
-                                     'score': _score()})
-                start_t = None
-                last_loud_t = None
-
-    # A section still open at EOF ends at its last loud window, exactly like
-    # every section closed inside the loop — not at the file duration, which
-    # used to stretch it by up to min_gap of trailing silence and put that
-    # silence into the clip served for it.
-    if start_t is not None and (last_loud_t + window_dur - start_t) >= min_duration - 1e-9:
-        sections.append({'start': round(start_t, 1),
-                         'end':   round(min(last_loud_t + window_dur, duration), 1),
-                         'score': _score()})
-
-    return sections
-
-
-def _package_result(rms_values: list, framerate: int, n_frames: int,
-                    window_samples: int, margin_db: float,
-                    min_gap: float = MIN_GAP_SECONDS,
-                    min_duration: float = MIN_DURATION_SECONDS) -> dict:
-    window_dur = window_samples / framerate
-    duration   = n_frames / framerate
-
-    # Note: no RMS data is returned — the UI is screen-reader oriented and
-    # draws no waveform, so sections + duration is all it needs. The window
-    # length went the same way: it only ever existed to place RMS bars.
-    return {
-        'sections':    _loud_sections(rms_values, window_dur, duration, margin_db, min_gap, min_duration),
-        'duration':    round(duration, 2),
-    }
-
-
-def analyze_wav(path: Path, window_samples: int = WINDOW_SAMPLES,
-                margin_db: float = MARGIN_DB,
-                min_gap: float = MIN_GAP_SECONDS,
-                min_duration: float = MIN_DURATION_SECONDS) -> dict:
-    try:
-        with wave.open(str(path), 'rb') as wf:
-            channels  = wf.getnchannels()
-            sampwidth = wf.getsampwidth()
-            framerate = wf.getframerate()
-            n_frames  = wf.getnframes()
-            # _compute_rms_windows_wav decodes as int16 unconditionally, so an
-            # 8- or 24-bit file would yield garbage RMS and therefore garbage
-            # sections — silently. The recorder only writes 16-bit, but --dir
-            # can point at any directory.
-            if sampwidth != 2:
-                return {'error': f'only 16-bit WAV files can be analysed '
-                                 f'(this one is {sampwidth * 8}-bit)'}
-            rms_values = list(_compute_rms_windows_wav(
-                wf, channels, sampwidth, framerate, window_samples))
-    except Exception as e:
-        return {'error': str(e)}
-
-    return _package_result(rms_values, framerate, n_frames, window_samples, margin_db, min_gap, min_duration)
-
-
-def analyze_flac(path: Path, window_samples: int = WINDOW_SAMPLES,
-                 margin_db: float = MARGIN_DB,
-                 min_gap: float = MIN_GAP_SECONDS,
-                 min_duration: float = MIN_DURATION_SECONDS) -> dict:
-    """Analyse a FLAC file for loudness. Requires numpy and soundfile."""
-    if not NUMPY_AVAILABLE or not SOUNDFILE_AVAILABLE:
-        return {'error': 'FLAC analysis requires: pip install numpy soundfile'}
-
-    try:
-        with sf.SoundFile(path) as f:
-            framerate = f.samplerate
-            channels  = f.channels
-            n_frames  = len(f)
-
-            rms_values = []
-            while True:
-                frames = f.read(window_samples, dtype='float32', always_2d=True)
-                if len(frames) == 0:
-                    break
-                mono = frames.mean(axis=1) if channels > 1 else frames[:, 0]
-                rms  = float(np.sqrt(np.mean(mono.astype(np.float64) ** 2)))
-                rms_values.append(round(rms, 5))
-    except Exception as e:
-        return {'error': str(e)}
-
-    return _package_result(rms_values, framerate, n_frames, window_samples, margin_db, min_gap, min_duration)
-
 
 # ---------------------------------------------------------------------------
 # Analysis cache helpers