diff --git a/README.md b/README.md
index cfa0fee..4ebc132 100644
--- a/README.md
+++ b/README.md
@@ -74,13 +74,20 @@ re-run `--write` unless you intend to remap slots.
 C920 mics are 16 kHz **stereo**. Each hop (default 200 ms) is kept
 stereo through a SpeechBrain preprocessor, then published as mono:
 
-1. `AudioNormalizer(mix="keep")` at 16 kHz
-2. Delay-and-Sum beamform (GCC-PHAT TDOA) — the live path from the
-   SpeechBrain [multi-microphone beamforming tutorial](https://speechbrain.readthedocs.io/en/latest/tutorials/preprocessing/multi-microphone-beamforming.html)
-   (`speechbrain.lobes.beamform_multimic.DelaySum_Beamformer`)
-3. `speechbrain/metricgan-plus-voicebank` neural enhancement on a 1 s
+1. Delay-and-Sum beamform (GCC-PHAT TDOA, frozen for
+   `BEAMFORM_TDOA_EVERY=5` hops) — live path from the SpeechBrain
+   [multi-microphone beamforming tutorial](https://speechbrain.readthedocs.io/en/latest/tutorials/preprocessing/multi-microphone-beamforming.html)
+2. `speechbrain/metricgan-plus-voicebank` neural enhancement on a 1 s
    rolling window (`ENHANCE_CHUNK_S=1.0`, `ENHANCE_HOP_S=0.2`)
 
+The 1 s window is **past context**, not +1 s delay: only the latest hop
+is published. Algorithmic delay ≈ hop + compute.
+
+Each worker caps Torch/BLAS to `TORCH_NUM_THREADS=1` (default) so 14–20
+processes do not oversubscribe the i7-14700. This host has an Intel Arc
+B580; the venv Torch is NVIDIA CUDA and **cannot** use it. Do not set
+`run_opts={"device":"cuda"}`.
+
 One worker process per camera loads the modules once and reuses them.
 `enhance_batch` must be called with `lengths=torch.tensor([1.0])`
 (relative, not sample count). `ENHANCE_MODE=off` (or a failed model
@@ -89,7 +96,8 @@ instead of Delay-and-Sum. DSP high-pass + soft-clip is the last-resort
 fallback so streams still work.
 
 Until the 1 s window is full, hops use the DSP fallback so LiveKit is
-not silent on startup.
+not silent on startup. Workers log `process_ms` / `capture_ms` every 50
+hops.
 
 ## Appliance boot / power button
 
diff --git a/scripts/bench_enhance.py b/scripts/bench_enhance.py
index 1598c7b..6422da8 100644
--- a/scripts/bench_enhance.py
+++ b/scripts/bench_enhance.py
@@ -1,5 +1,12 @@
 #!/usr/bin/env python3
-"""RTF bench for hop/window speechbrain enhance."""
+"""RTF bench for hop/window speechbrain enhance.
+
+Stop livekit-cameras.service first or the numbers fight the fleet.
+
+  .venv/bin/python scripts/bench_enhance.py
+  .venv/bin/python scripts/bench_enhance.py --stereo --hop 0.1
+"""
+import argparse
 import sys
 import time
 from pathlib import Path
@@ -10,18 +17,49 @@ import numpy as np
 
 from audio_cleanup import AudioCleaner
 
-c = AudioCleaner(mode="force", chunk_s=1.0, hop_s=0.2)
-hop = 3200
-pcm = (np.random.randn(hop) * 800).astype(np.int16)
-for _ in range(8):
-    c.process(pcm)
-n = 50
-t0 = time.perf_counter()
-for _ in range(n):
-    c.process(pcm)
-dt = time.perf_counter() - t0
-audio_s = n * 0.2
-rtf = dt / audio_s
-print(f"hops={n} wall={dt:.3f}s audio={audio_s:.1f}s rtf={rtf:.3f}")
-if rtf > 0.8:
-    raise SystemExit("RTF too high for 20-way realtime")
+
+def main() -> int:
+    ap = argparse.ArgumentParser()
+    ap.add_argument("--stereo", action="store_true")
+    ap.add_argument("--hop", type=float, default=0.2)
+    ap.add_argument("--chunk", type=float, default=1.0)
+    ap.add_argument("--n", type=int, default=50)
+    args = ap.parse_args()
+
+    c = AudioCleaner(
+        mode="force",
+        beamform="force" if args.stereo else "off",
+        chunk_s=args.chunk,
+        hop_s=args.hop,
+        torch_num_threads=1,
+    )
+    hop_n = int(args.hop * 16000)
+    if args.stereo:
+        t = np.arange(hop_n)
+        ch0 = (np.sin(2 * np.pi * 440 * t / 16000) * 800).astype(np.int16)
+        pcm = np.stack([ch0, np.roll(ch0, 2)], axis=1)
+    else:
+        pcm = (np.random.randn(hop_n) * 800).astype(np.int16)
+    warmup = max(8, int(args.chunk / args.hop) + 2)
+    for _ in range(warmup):
+        c.process(pcm)
+    n = args.n
+    t0 = time.perf_counter()
+    for _ in range(n):
+        c.process(pcm)
+    dt = time.perf_counter() - t0
+    audio_s = n * args.hop
+    rtf = dt / audio_s
+    print(
+        f"backend={c.backend} stereo={args.stereo} hops={n} "
+        f"wall={dt:.3f}s audio={audio_s:.1f}s rtf={rtf:.3f} "
+        f"last_ms={c.last_dt_ms:.1f}"
+    )
+    limit = 0.5 if args.stereo else 0.8
+    if rtf > limit:
+        raise SystemExit(f"RTF too high for 20-way realtime (limit {limit})")
+    return 0
+
+
+if __name__ == "__main__":
+    sys.exit(main())