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Qwen/Qwen3-ASR-1.7B-hf for Japanese noisy-recording/device-domain speech
recognition. The repository contains full merged model weights; it does not
require a separate LoRA adapter at inference time.lowrank_loraplus8_coverage recipe:Qwen/Qwen3-ASR-1.7B-hf8e-6, B-matrix ratio 8xsample_id list with identical normalized ground truth; each
side produced exactly 17,619 predictions with no duplicate IDs.max_new_tokens=256. CER is character-level Levenshtein distance after the
same whitespace normalization for both models. Macro-CER is the unweighted
mean of per-segment CER.| Model | Evaluated segments | Micro-CER | Macro-CER | Character errors | Exact-match rate |
|---|---|---|---|---|---|
| Base Qwen3-ASR-1.7B | 17,619 | 0.34295 | 0.41445 | 102,563 | 14.995% |
| Bro-ASR-1.7B | 17,619 | 0.33763 | 0.39566 | 100,970 | 18.208% |
sample_id values with identical normalized ground truth and exact fractional
audio time boundaries. This fixed-set audit is separate from the 17,619-segment
strict-unseen evaluation above.| Fixed audit subset | Model | Segments | Micro-CER | Macro-CER | Character errors | Exact-match rate | Decoding |
|---|---|---|---|---|---|---|---|
| Delivery_JP frozen 1,000 | Bro-ASR-1.7B | 1,000 | 0.47119 | 0.49852 | 7,581 | 14.4% | Japanese, beam 5 |
| Delivery_JP frozen 1,000 | Qwen3-ASR-1.7B baseline | 1,000 | 0.48574 | 0.54712 | 7,815 | 10.3% | Japanese, beam 5 |
| Delivery_JP frozen 1,000 | Cohere Transcribe 03-2026 | 1,000 | 0.56871 | 0.63912 | 9,150 | 7.1% | Official model.transcribe, ja, punctuation, greedy |
| Delivery_JP frozen 1,000 | Whisper large-v3-turbo (untrained) | 1,000 | 1.53969 | 3.86799 | 24,772 | 0.7% | Historical frozen TTA: speed factors 1.0/0.95/1.05, Japanese, beam 5 |
best_solution_snapshot.py TTA decoder and an asserted zero LoRA B-matrix.
Its result is substantially weaker on this audit; the decode policy is
reported separately because it uses three TTA speed factors while the Qwen
runs use beam-5 decoding.1pip install "git+https://github.com/huggingface/transformers"
2pip install torch soundfile1import torch
2from transformers import AutoModelForMultimodalLM, AutoProcessor
3
4model_id = "Junlaii/Bro-ASR-1.7B"
5processor = AutoProcessor.from_pretrained(model_id)
6model = AutoModelForMultimodalLM.from_pretrained(
7 model_id, torch_dtype=torch.bfloat16, device_map="auto"
8).eval()
9
10inputs = processor.apply_transcription_request(
11 audio="sample.wav", language="ja"
12).to(model.device, model.dtype)
13with torch.inference_mode():
14 output = model.generate(**inputs, num_beams=5, max_new_tokens=256)
15text = processor.decode(
16 output[:, inputs["input_ids"].shape[1]:], return_format="transcription_only"
17)[0]
18print(text)max_new_tokens=256.