Metro-ASR Small

Non-autoregressive CTC speech recognition for Egyptian Arabic and Arabic–English
code-switching, with a detachable n-gram language head you can retrain on text alone.

What this is

Metro-ASR separates the two things a speech recogniser has to know — what the audio sounds like and what the words are likely to be — into two artefacts trained and shipped separately. This repository holds the first one: Metro-Small, a 61.6M-parameter Conformer acoustic model trained with CTC. It is non-autoregressive — one forward pass turns an utterance into a matrix of per-frame log-probabilities, with no decoder loop and no dependence on previously emitted tokens — which is why it runs at 40–55× real time on a laptop CPU with no GPU involved anywhere in this card's examples.

The second artefact, the language model, is an n-gram over text. It never sees audio, trains in minutes on a laptop, and plugs into the decoder at run time — including domain-specific variants you build yourself from nothing but text (see The language head below). The general-purpose one is included in this repository as lm_5gram.bin.

For the full architecture writeup, training-from-scratch instructions, and how to build your own domain language head, see the GitHub repository. For real transcripts, audio players, and a measured comparison of greedy vs. beam+LM decoding across three interchangeable language heads, see the interactive evaluation report.

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Architecture

Twelve identical Conformer-style blocks, each a Macaron sandwich of two half-weighted feed-forward networks around an attention module and a convolution module:

Frame rate. 16 kHz audio → 80-bin log-Mel (100 fps) → Conv2D ×4 subsampling → 25 fps through the encoder and the CTC head. Subsampling by 4 before the first block cuts attention's quadratic cost 16× before a single block runs; one output token covers 40 ms of audio.

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The language head

CTC's per-frame independence produces a specific, recognisable error pattern: doubled syllables, dropped affixes, malformed English fragments — the model heard correctly and wrote something that isn't a word. A word-level n-gram model fixes this during beam search, because it knows which sequences are plausible, without ever having heard a single second of audio:

The language head is a separate file with no learned interaction with the acoustic weights. Swap it, and nothing about model.pt changes:

engine = MetroASREngine.from_pretrained("small")

engine.load_lm("lm/medical_head.bin")      # swap language heads at run time,
print(engine.transcribe("call.wav", beam_search=True).text)   # same acoustic weights throughout

This is measured, not asserted. The GitHub repo ships two extra language heads — technical and medical — built from real text (Egyptian medical chat/QA, Egyptian Arabic Wikipedia's technical articles, real Arabic-English code-switching text) plus synthesised domain-term carrier phrases, and decodes 11 real clips with all three heads against human references. Headline results from the full interactive report:

Two things worth reading out of that table. The domain heads win in their own domain despite being 4-grams built from far less text than the general 5-gram — domain fit beats scale for this component. And the technical head actively hurts on general speech (36.9% vs. greedy's 25.6%) — a language head is a strong prior, and matching it to your traffic matters. (These numbers are from a small, hard 11-clip demo set — unscripted, overlapping speech, dense code-switching — and are not the same evaluation as the Performance section below; see the report for methodology.)

Building your own head takes text and minutes, no GPU:

python scripts/train_lm.py --corpus my_domain.txt --out lm/my_domain_4gram.arpa --order 4

Full walkthrough — including a from-scratch corpus-building example for a technical and a medical domain — in the GitHub README's Domain-specialised heads section.

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Model variants

Three sizes share one block definition and one training recipe — only width, depth and vocabulary change. Small (this repository) is the only one currently trained. Medium and Large exist as configs in the GitHub repo with exact parameter counts, but no weights — see Scaling to Medium and Large for what training them actually requires (data volume most of all).

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Performance

Held-out test-set WER/CER (the numbers in this card's metadata):

Speed — measured, Intel Core Ultra 7 155H, 4 CPU threads, PyTorch 2.13 CPU build, fp32, minimum of 15 runs after warm-up:

Beam search with the 5-gram head adds roughly 10–250 ms per utterance depending on length (RTF ≈ 0.024 overall). Loading the 5.9 GB binary itself takes about 3.4 s, once, at startup.

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Usage

Install

pip install metro-asr          # greedy decoding only
pip install "metro-asr[lm]"    # + KenLM beam search
pip install -U "numpy>=2.0"    # see note below

[!WARNING] pyctcdecode's only PyPI release pins numpy<2.0.0 in its own metadata, even though it runs fine under numpy 2.x. Installing the [lm] extra will downgrade numpy to satisfy that — on an environment that already had numpy 2.x with other packages built against it (Colab, most fresh installs today), that breaks those packages with numpy.dtype size changed. The third line above fixes it. If [lm] isn't installed at all, lm_path="auto" now degrades to greedy with a warning rather than crashing engine construction.

Quick start

from metro_asr import MetroASREngine

engine = MetroASREngine.from_pretrained("small")   # auto-downloads weights + tokenizer, caches locally
result = engine.transcribe("audio.wav")
print(result.text)

The 5.9 GB language model is not downloaded by this call.

With the language head (beam search)

engine = MetroASREngine.from_pretrained("small", lm_path="auto")   # also fetches lm_5gram.bin
result = engine.transcribe("audio.wav", beam_search=True)
print(result.text)

From a manual download

If you've already run snapshot_download (or git cloned this repo) into a local directory, point from_pretrained at that directory instead — nothing gets re-downloaded, and it works fully offline:

from huggingface_hub import snapshot_download
from metro_asr import MetroASREngine

snapshot_download(repo_id="MohammedAly22/metro-asr-small", local_dir="checkpoints")

engine = MetroASREngine.from_pretrained("checkpoints", lm_path="auto")

[!TIP] This is the fix for a common mistake: calling from_pretrained("checkpoints") used to be interpreted as a HuggingFace repo id named literally "checkpoints" and fail with Repository Not Found. Current versions check for an existing local directory first — update if you hit that error.

Without the package — loading the raw PyTorch model

import torch
from metro_asr.utils.config import load_config
from metro_asr.model.metro import MetroASR
from metro_asr.model.tokenizer import build_tokenizer

config = load_config("checkpoints/config.yaml")
tokenizer = build_tokenizer(config, "checkpoints")   # must run before MetroASR.from_config —
model = MetroASR.from_config(config)                 # it fixes the CTC head's vocab size
ckpt = torch.load("checkpoints/model.pt", map_location="cpu", weights_only=False)
model.load_state_dict(ckpt["model_state_dict"])
model.eval()

Batch transcription

results = engine.transcribe_batch(["audio1.wav", "audio2.wav", "audio3.wav"])
for r in results:
    print(f"{r.text}  (RTF={r.rtf:.4f})")

Streaming and serving

engine.transcribe_stream(chunk_generator) yields incremental transcriptions from any audio generator, and scripts/serve.py in the GitHub repo wraps the same engine in a Flask REST API (/transcribe, /transcribe/batch, /health, /info). See the README's Streaming and Serving sections, or the runnable streaming_server.ipynb notebook.

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Files in this repository

Weights alone are 235 MB; model.pt is larger because it also carries optimizer state so training can be resumed from it. Strip that for deployment:

import torch
ckpt = torch.load("model.pt", map_location="cpu", weights_only=False)
torch.save({"model_state_dict": ckpt["model_state_dict"], "config": ckpt["config"]},
           "model_inference.pt")

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Training

• Audio data: 130K+ clips from the 8 audio datasets listed in this card's metadata, plus additional Egyptian Arabic content, covering Arabic-only and Arabic-English code-switching speech.
• Text data (for the shipped language head): ~1.9M Egyptian Arabic sentences plus Arabic-English code-switching text, upsampled to balance against the larger Arabic-only portion.
• Recipe: CTC loss + 0.3-weighted auxiliary CTC at layer 6, AdamW (β = 0.9, 0.98), linear warmup into cosine decay, SpecAugment, speed perturbation (0.9×/1.0×/1.1×), 443K steps, batch size 32 with 4× gradient accumulation, bf16.
• Hardware: single GPU.

Full step-by-step instructions to reproduce this from scratch — tokenizer training, language model training, data preparation, the acoustic training loop, and how to scale the recipe to Medium/Large — are in the GitHub README's Training from scratch section.

Fine-tuning this checkpoint

Adapting to a new domain or accent starts from these weights, not from scratch — 10-20× lower learning rate, encoder frozen for the first few thousand steps so the CTC head adapts first:

python scripts/finetune.py \
    --checkpoint checkpoints/model.pt \
    --tokenizer-dir checkpoints \
    --dataset your/dataset-id \
    --lr 5e-5 --max-steps 30000 --freeze-steps 3000

Runnable end-to-end in fine_tuning.ipynb (Colab, needs a GPU), or see Fine-tuning in the README.

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Limitations

• Dialect. Trained on Egyptian Arabic; Modern Standard Arabic and other dialects — Gulf, Levantine, Maghrebi — degrade, Maghrebi most of all.
• Code-switching is Arabic-English only, and it's the weakest part of the system:
  • Technical/domain vocabulary degrades under plain greedy decoding — see The language head above for the fix and the measured improvement.
  • A lone English word surrounded by Arabic is harder than a full English clause, which tends to survive intact.
  • Acronyms and short initialisms (CNN, MRI, AIC) are acoustically ambiguous and unreliable without a matching language head.
  • Roughly 12K code-switching training utterances were available against ~130K Arabic-only ones — the imbalance is a data problem, not an architectural one.
• Clip length. Trained and evaluated on 0.5–30 s; segment longer recordings before transcribing.
• Output is lowercase and unpunctuated, matching the training transcripts.
• Not a streaming model in the strict sense — the encoder is bidirectional, so a chunk must be complete before it can be decoded. transcribe_stream is chunked offline decoding, with a floor latency of one chunk.
• The language head is large. The shipped 5-gram is 5.9 GB resident in RAM; a smaller 4-gram (like the domain heads above) trades some accuracy for a much smaller footprint.
• The language head asserts priors. It corrects toward what its training text considers likely — which is exactly what makes it useful, and exactly how it gets unfamiliar proper nouns wrong. Train a head on your own text if this matters for your use case.
• Medium and Large are unreleased. Their configs exist; no weights do.

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Citation

@software{metro_asr_2025,
  title  = {Metro-ASR: Non-Autoregressive Speech Recognition for Egyptian Arabic
            and Code-Switching with a Detachable N-gram Language Head},
  author = {Mohammed Aly},
  year   = {2025},
  url    = {https://github.com/MohammedAly22/metro-asr}
}

License

MIT — see LICENSE.