Files
adminandClaude Opus 5 c91264c253 refactor: split web.py — detector and audio-format helpers into isrweb/
Pure move, no behaviour change. web.py 1221 -> 807 lines.

The reason is not file size but isolation of change. CLAUDE.md records
that the next step for detection, should flagging ever need improving, is
a spectral filter or an optional Silero VAD over candidate sections —
that work now lands entirely in isrweb/loudness.py and never has to open
the HTTP layer. The byte-level format code (FLAC frame parsing, CRC-8,
live header patching) is a second self-contained cluster, touched only
when a container format misbehaves.

  isrweb/loudness.py   229  detector + its tuning constants
  isrweb/audiofile.py  250  filename clock, WAV/FLAC headers, MIME types
  web.py               807  cache, listing, handler, main()

Import direction is strictly web.py -> {loudness, audiofile}; the two
leaves import neither each other nor web.py, which is what lets the
detector be imported and tested without an HTTP server.

A package rather than flat modules because `audiofile` is a real PyPI
name and `detect` is generic enough to collide; a local file silently
shadows an installed one, which fails confusingly rather than loudly.
web.py stays at the repo root so `python web.py`, the docker-compose
command and the webui.html lookup are all unchanged.

Each module keeps its own optional numpy/soundfile try-import rather than
sharing one. That block cannot meaningfully diverge — if numpy is
installed every copy says True — and sharing it would stop isr.py being a
standalone file you can drop on a Pi and run.

Tests mirror the modules: test_loudness.py, test_audiofile.py, test_web.py
(+ unchanged test_isr.py). New tests/conftest.py owns the sys.path setup
that was previously an inline hack in test_isr.py — which only worked for
the other modules because that file happened to import first — and
provides the write_wav fixture the split would otherwise have duplicated.

Verified:
* 78 passed, same count as before the split; each test file also passes
  standalone.
* All 21 moved functions diffed against HEAD:web.py — byte-identical.
* Import graph asserted acyclic; both leaves import with `web` absent
  from sys.modules.
* An analysis cache written *before* the split still hits afterwards,
  with identical sections — the one way a "pure" move could quietly
  invalidate every user's cache.
* Endpoint smoke (analyze bounds, cache hit, cached_analysis in the
  listing, clip Content-Length, 8-bit rejection) and the byte-serving
  harness (ranges, download Range, HEAD, 405, live-header patching whole
  and straddling) both re-run green.
* Dockerfile COPY set verified by building the image's file tree in a
  temp dir and importing web.py with nothing else present.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
2026-07-30 23:14:54 +02:00

230 lines
10 KiB
Python
Raw Permalink Blame History

This file contains ambiguous Unicode characters
This file contains Unicode characters that might be confused with other characters. If you think that this is intentional, you can safely ignore this warning. Use the Escape button to reveal them.
"""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)