diff --git a/README.md b/README.md
index 37ae9e9..65bc1c9 100644
--- a/README.md
+++ b/README.md
@@ -140,6 +140,44 @@ whisper = LightningWhisperMLX(model="distil-large-v3", quant="4bit")
 
 Higher batch sizes improve throughput but require more memory. Start with the recommended values and adjust based on your hardware.
 
+### Measuring the speed-up
+
+The speed-up from batching depends on the model, batch size, chip and audio. To measure it on your Mac, time sequential against batched decoding on one of your own files:
+
+```bash
+python scripts/benchmark.py audio.mp3 --model distil-large-v3 --batch-size 12
+```
+
+The script loads the model and warms it up first, then reports the best of `--runs` timings for `batch_size=1` and for the batch size you chose.
+
+## Batched vs Sequential Decoding
+
+With `batch_size=1`, Vayu decodes like OpenAI's Whisper. Each 30-second window starts where the previous segment ended and is conditioned on the text so far.
+
+With `batch_size > 1`, fixed 30-second windows are decoded together. This is much faster, with some trade-offs:
+
+- Every window in a batch is conditioned on the text from *before* the batch, so `condition_on_previous_text` applies between batches, not between windows.
+- Windows don't move to follow the speech. Text cut off at a window boundary is kept as a segment that ends at the boundary, so a word spanning two windows can be split.
+- `hallucination_silence_threshold` needs window-by-window seeking and is ignored (with a warning).
+- `best_of` is not used. Temperature fallback re-decodes only the windows that fail the quality checks.
+
+Word-level timestamps work in both modes.
+
+## Loading Local Models
+
+`load_model` and `--model` accept a local directory with MLX weights (`config.json` plus `weights.safetensors`, `model.safetensors` or `weights.npz`). For safety, local directories are only loaded from:
+
+- the HuggingFace cache (`~/.cache/huggingface/hub`)
+- `/usr/local/share/whisper-mlx`
+- directories listed in `WHISPER_MLX_MODEL_DIRS` (separated by `:`)
+
+```bash
+export WHISPER_MLX_MODEL_DIRS=~/models:/Volumes/External/whisper
+vayu audio.mp3 --model ~/models/whisper-large-v3-mlx
+```
+
+Anything else fails with `Model path '...' is not within allowed directories`. HuggingFace repo names (`mlx-community/whisper-turbo`) are downloaded to the cache and are not affected.
+
 ## API Reference
 
 ### transcribe()
diff --git a/scripts/benchmark.py b/scripts/benchmark.py
new file mode 100644
index 0000000..89d100b
--- /dev/null
+++ b/scripts/benchmark.py
@@ -0,0 +1,80 @@
+#!/usr/bin/env python3
+"""
+Time sequential (batch_size=1) against batched decoding on one audio file.
+
+Usage:
+    python scripts/benchmark.py audio.mp3 --model distil-large-v3 --batch-size 12
+
+The model is loaded and warmed up before timing, and the audio is decoded once
+up front, so the numbers cover transcription only. Use a few minutes of real
+speech: short clips fit in one window and can't benefit from batching.
+"""
+
+import argparse
+import platform
+import time
+
+import mlx.core as mx
+
+from whisper_mlx import SAMPLE_RATE, load_audio, transcribe
+from whisper_mlx.utils import resolve_model_path
+
+
+def time_transcription(audio: mx.array, repo: str, batch_size: int, language):
+    start = time.perf_counter()
+    result = transcribe(
+        audio,
+        path_or_hf_repo=repo,
+        batch_size=batch_size,
+        language=language,
+        verbose=None,
+    )
+    return time.perf_counter() - start, result
+
+
+def main():
+    parser = argparse.ArgumentParser(description=__doc__.split("\n\n")[1])
+    parser.add_argument("audio", help="Audio file to transcribe")
+    parser.add_argument(
+        "--model", default="distil-large-v3", help="Model name or HuggingFace repo"
+    )
+    parser.add_argument("--quant", default=None, choices=["4bit", "8bit"])
+    parser.add_argument("--batch-size", type=int, default=12)
+    parser.add_argument(
+        "--language",
+        default=None,
+        help="Language code; set it to leave language detection out of the timings",
+    )
+    parser.add_argument("--runs", type=int, default=3, help="Timed runs per setting")
+    args = parser.parse_args()
+
+    repo = resolve_model_path(args.model, args.quant)
+    audio = load_audio(args.audio)
+    duration = audio.shape[0] / SAMPLE_RATE
+
+    print(f"Model:  {repo}")
+    print(f"Audio:  {args.audio} ({duration:.1f}s)")
+    print(f"System: {platform.platform()}, MLX {mx.__version__}")
+    print()
+
+    # Load the model and compile kernels outside the timed runs
+    time_transcription(audio[: 30 * SAMPLE_RATE], repo, 1, args.language)
+
+    best = {}
+    for batch_size in dict.fromkeys([1, args.batch_size]):
+        times = [
+            time_transcription(audio, repo, batch_size, args.language)[0]
+            for _ in range(args.runs)
+        ]
+        best[batch_size] = min(times)
+        print(
+            f"batch_size={batch_size:<3} best {best[batch_size]:7.2f}s  "
+            f"({duration / best[batch_size]:5.1f}x real time)"
+        )
+
+    if args.batch_size != 1:
+        print(f"\nSpeed-up: {best[1] / best[args.batch_size]:.2f}x")
+
+
+if __name__ == "__main__":
+    main()