Raise tail pad to 3s so short transmissions are not clipped

The recording window is anchored to OP25 control-channel timestamps, but
the buffered audio lags those by roughly 1.5s. Trim logs across seven
calls measured the offset at 0.84-1.62s, consistently present.

At a 1.0s pad a short call closed its window before the voice arrived:
a 0.97s control-channel call closed at T+1.97 while voice started around
T+1.5, capturing ~0.4s of speech and cutting mid-word. Confirmed by a
0.57s file whose final 0.10s measured -12.2dB against its own -18.2dB
average - clipped speech, not a tail - and by two short calls that
logged no trim at all because no trailing silence remained.

Being generous is free here: trim_silence already strips trailing
silence back to the guard margin before upload, so long calls are
unaffected while short ones gain the window they need. Over-capture
costs nothing; under-capture loses words permanently.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>

.env.example

@@ -38,9 +38,13 @@ CALL_IDLE_TIMEOUT=3
 
 # Seconds of audio kept after the last transmission ends. This is the only
 # headroom protecting the final word of a transmission — usually the disposition
-# or the address. Measured at 0.5s it left ~0.3s of real margin and one recording
-# ended mid-word, hence 1.0.
-CALL_TAIL_PAD_SECONDS=1.0
+# or the address. Raised 1.0 -> 3.0 after field measurement showed the
+# grant-to-speech offset is ~0.84-1.62s (typically ~1.5s): at 1.0s pad, short
+# calls had their recording window close before the voice even started,
+# clipping speech mid-word. Safe to be generous — trim_silence already strips
+# the extra back off long calls before upload, so only short transmissions
+# actually benefit from the larger window.
+CALL_TAIL_PAD_SECONDS=3.0
 
 # Strip leading/trailing dead air before upload. ~63% of an untrimmed recording
 # is silence, which costs Whisper spend and makes it hallucinate text that was

drb-edge-node/app/config.py

@@ -47,10 +47,29 @@ class Settings(BaseSettings):
     # Audio kept after the observed end of the last transmission. The srcaddr
     # 1→0 edge can be up to one poll (0.5 s) late and the encoder adds its own
     # latency, so this is the only headroom protecting the last word of a
-    # transmission — which is usually the disposition or the address. Field
-    # measurement at 0.5 s left only 0.29–0.37 s of real trailing margin and one
-    # recording ended mid-word, hence 1.0 s.
-    call_tail_pad_seconds: float = 1.0
+    # transmission — which is usually the disposition or the address.
+    #
+    # Raised 1.0 -> 3.0 after field measurement showed the recording WINDOW
+    # (anchored to OP25 control-channel timestamps) closing well before the
+    # actual voice audio arrives: grant->speech offset measured 0.84-1.62s
+    # across 7 calls (~1.5s typical). At the old 1.0s pad, a short
+    # transmission (e.g. a 0.97s control-channel call) had its window close
+    # at T+1.97 while voice didn't start until ~T+1.5 — leaving ~0.4s of
+    # captured speech, clipped mid-word. Confirmed by a 0.57s output file
+    # whose final 0.10s measured -12.2dB, louder than its own -18.2dB
+    # average (i.e. clipped speech, not trailing silence), and by two short
+    # calls that produced no "Trimmed" log line at all because there was no
+    # trailing silence left to trim.
+    #
+    # Safe to be generous here: trim_silence already strips trailing silence
+    # back to trim_silence_guard_seconds before upload, so a larger pad costs
+    # long calls nothing (the extra is trimmed away) while giving short
+    # transmissions enough window to actually capture the voice. Over-capture
+    # is free; under-capture loses words permanently. Do not tune this back
+    # down without new field data showing the grant->speech offset has
+    # shrunk — see DEFERRED.md for the call_idle_timeout coupling this value
+    # now sits at.
+    call_tail_pad_seconds: float = 3.0
 
     # Strip leading/trailing dead air before upload. ~63% of a typical recording
     # is silence (the grant→speech delay plus the tail pad), which inflates

drb-edge-node/app/internal/metadata_watcher.py

@@ -58,8 +58,13 @@ def _tail_pad() -> float:
     edge (up to one poll late) plus encoder latency never clips the tail.
 
     Read live from settings (env CALL_TAIL_PAD_SECONDS) rather than frozen into a
-    module constant, so it is tunable per node. See the setting for why the
-    default moved 0.5 → 1.0.
+    module constant, so it is tunable per node. See the setting in config.py for
+    why the default moved 1.0 → 3.0 (short calls' recording window was closing
+    before the ~1.5s grant→speech offset let voice audio even start).
+
+    Only the idle-timeout close path below uses this pad — the tgid_change and
+    tgid_change_unlogged paths close at an exact, already-known boundary (the
+    new grant's timestamp, or the same poll tick) and intentionally add none.
     """
     return settings.call_tail_pad_seconds
 

drb-edge-node/tests/test_metadata_watcher.py

@@ -233,15 +233,17 @@ async def test_srcaddr_edge_then_idle_timeout_ends_call(watcher, clock):
 
 
 @pytest.mark.asyncio
-async def test_tail_pad_is_configurable_and_defaults_to_one_second(watcher, clock, monkeypatch):
+async def test_tail_pad_is_configurable_and_defaults_to_three_seconds(watcher, clock, monkeypatch):
     """
-    0.5s left only ~0.3s of real trailing margin in field measurement and one
-    recording ended mid-word, so the default moved to 1.0 — and it has to be a
-    setting, not a magic number, so it can be tuned per node.
+    Field measurement showed the grant->speech offset runs ~0.84-1.62s, so a
+    1.0s pad let short calls' windows close before voice audio even started
+    (clipping mid-word). The default moved to 3.0 — and it has to be a
+    setting, not a magic number, so it can be tuned per node without a code
+    change (and so tests can prove it isn't hardcoded anywhere downstream).
     """
-    assert settings.call_tail_pad_seconds == 1.0
+    assert settings.call_tail_pad_seconds == 3.0
 
-    monkeypatch.setattr(settings, "call_tail_pad_seconds", 2.5)
+    monkeypatch.setattr(settings, "call_tail_pad_seconds", 5.0)
 
     await tick(watcher, update(
         call_log=[grant(1234, clock.now)],
@@ -251,11 +253,51 @@ async def test_tail_pad_is_configurable_and_defaults_to_one_second(watcher, cloc
     edge_time = clock.now
     await tick(watcher, update(channels=[channel(tgid=1234, srcaddr=0, hold_tgid=1234)]))
 
-    clock.advance(settings.call_idle_timeout + 1.0)
+    # Advance well past both the idle timeout AND the monkeypatched 5.0s pad so
+    # the "now" cap in _handle_channels never masks the pad value under test.
+    clock.advance(settings.call_idle_timeout + 6.0)
     await tick(watcher, update(channels=[channel()]))
 
     payload = watcher.on_call_end.call_args[0][0]
-    assert payload["ended_at_epoch"] == pytest.approx(edge_time + 2.5)
+    assert payload["ended_at_epoch"] == pytest.approx(edge_time + 5.0)
+
+
+@pytest.mark.asyncio
+async def test_short_call_window_now_covers_delayed_voice_arrival(watcher, clock):
+    """
+    Regression test for the truncation bug: a ~0.97s control-channel call
+    (grant to srcaddr-drop) previously closed its window at
+    last_tx_end + 1.0s pad, i.e. ~1.97s after the grant — but field
+    measurement shows voice audio doesn't start until ~1.5s after the grant
+    (0.84-1.62s measured), so the old window left as little as ~0.4s of
+    captured speech and clipped it mid-word.
+
+    With the 3.0s default pad, the same short call's window must extend well
+    past the ~1.5s point where voice actually starts.
+    """
+    call_start = clock.now
+    await tick(watcher, update(
+        call_log=[grant(1234, call_start)],
+        channels=[channel(tgid=1234, srcaddr=555)],
+    ))
+
+    # The control-channel call itself is short — under 1 second.
+    clock.advance(0.97)
+    edge_time = clock.now
+    await tick(watcher, update(channels=[channel(tgid=1234, srcaddr=0, hold_tgid=1234)]))
+
+    clock.advance(settings.call_idle_timeout + 0.5)
+    await tick(watcher, update(channels=[channel()]))
+
+    payload = watcher.on_call_end.call_args[0][0]
+    assert payload["end_reason"] == "idle_timeout"
+
+    voice_arrival = call_start + 1.5  # measured grant->speech offset, typical case
+    assert payload["ended_at_epoch"] == pytest.approx(edge_time + settings.call_tail_pad_seconds)
+    assert payload["ended_at_epoch"] > voice_arrival, (
+        "recording window must extend past the point voice audio actually arrives, "
+        "not just past the control-channel call_log timestamps"
+    )
 
 
 @pytest.mark.asyncio