Working CoreML LUT4 input_embeds variant (86.9% on VITW)
Browse files- inference_asr.py +216 -0
inference_asr.py
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| 1 |
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"""End-to-end Mega-ASR pipeline on CoreML:
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ONNX audio encoder + CoreML LUT-4 LLM (input_embeds variant) + bench.
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The CoreML LLM is single-token-step (ANE-friendly). For each token in the
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prompt we feed (inputs_embeds[t], current_pos=t) to populate the KV cache;
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then we greedy-decode by feeding one token at a time.
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"""
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from __future__ import annotations
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import argparse
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import json
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import re
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import sys
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import warnings
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from pathlib import Path
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import numpy as np
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warnings.filterwarnings("ignore")
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REFERENCES = {
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"noise": "I usually take the quieter road home because the main street gets crowded after work.",
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"far_field": "Please remind me to print the forms before we leave for the appointment tomorrow.",
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"obstructed": "I forgot my charger at home, so I need to find an outlet before the meeting starts.",
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"distortion": "The new coffee machine is simple, but everyone keeps forgetting where the filters are stored.",
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"recording": "Can you check whether the train still stops at the downtown station after eight tonight?",
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"echo": "I need to return these shoes because the size feels fine standing up but terrible while walking.",
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"dropout": "My aunt is learning video calls, and she gets excited whenever the picture actually works.",
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"mixed": "My sister is bringing dinner over later, so we do not need to cook tonight.",
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}
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_NORM_RE = re.compile(r"[^a-z0-9\s]")
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def normalize(t):
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if "<asr_text>" in t:
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t = t.split("<asr_text>", 1)[1]
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return re.sub(r"\s+", " ", _NORM_RE.sub(" ", t.lower())).strip()
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def wer(ref, hyp):
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r = ref.split(); h = hyp.split()
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if not r: return (1.0 if h else 0.0, len(h), 0)
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d = np.zeros((len(r) + 1, len(h) + 1), dtype=np.int32)
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for i in range(len(r) + 1): d[i, 0] = i
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for j in range(len(h) + 1): d[0, j] = j
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for i in range(1, len(r) + 1):
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for j in range(1, len(h) + 1):
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d[i, j] = min(d[i-1, j] + 1, d[i, j-1] + 1,
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d[i-1, j-1] + (0 if r[i-1] == h[j-1] else 1))
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return d[len(r), len(h)] / max(len(r), 1), int(d[len(r), len(h)]), len(r)
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def color(p, s):
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if p >= 70: return f"\033[92m{s}\033[0m"
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if p >= 50: return f"\033[93m{s}\033[0m"
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if p >= 25: return f"\033[33m{s}\033[0m"
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return f"\033[91m{s}\033[0m"
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def build_prompt_ids(tok, audio_pad_count, audio_pad_id=151676):
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prompt = (
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"<|im_start|>system\n<|im_end|>\n"
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"<|im_start|>user\n<|audio_start|><|audio_pad|><|audio_end|><|im_end|>\n"
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"<|im_start|>assistant\n"
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"language English<asr_text>"
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)
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ids = tok.encode(prompt, add_special_tokens=False)
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pos = ids.index(audio_pad_id)
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return ids[:pos] + [audio_pad_id] * audio_pad_count + ids[pos + 1:]
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def causal_mask_at(cur, ctx, neg_inf=-1e4):
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"""Build (1,1,1,ctx) mask: positions > cur get -inf, others 0."""
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m = np.zeros((1, 1, 1, ctx), dtype=np.float16)
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if cur + 1 < ctx:
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m[0, 0, 0, cur + 1:] = neg_inf
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return m
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def update_mask_at(cur, ctx):
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"""(1,1,ctx,1) — 1.0 at the current position, 0 elsewhere. Used for KV cache write."""
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m = np.zeros((1, 1, ctx, 1), dtype=np.float16)
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m[0, 0, cur, 0] = 1.0
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return m
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def main():
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ap = argparse.ArgumentParser()
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ap.add_argument("--mlpackage", default="models/coreml/mega-asr-llm-embeds_lut4.mlpackage", type=Path)
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ap.add_argument("--encoder-path", default="models/mega-asr-onnx-hf/onnx/audio_encoder_fp32.onnx", type=Path)
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ap.add_argument("--examples-dir", default="models/mega-asr-onnx-hf/examples", type=Path)
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ap.add_argument("--qwen-asr-dir", default="models/mega-asr-hf/Qwen3-ASR-1.7B", type=Path)
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ap.add_argument("--max-new-tokens", type=int, default=80)
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ap.add_argument("--context-length", type=int, default=512)
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ap.add_argument("--compute-unit", default="CPU_AND_NE", choices=["CPU_ONLY", "CPU_AND_NE", "ALL"])
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args = ap.parse_args()
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import soundfile as sf
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import onnxruntime as ort
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import coremltools as ct
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from transformers import AutoFeatureExtractor, AutoTokenizer
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print(f"Loading CoreML mlpackage ({args.compute_unit}) ...")
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cu = getattr(ct.ComputeUnit, args.compute_unit)
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mlm = ct.models.MLModel(str(args.mlpackage), compute_units=cu)
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print(f"Loading ONNX encoder ...")
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enc = ort.InferenceSession(str(args.encoder_path), providers=["CPUExecutionProvider"])
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feat = AutoFeatureExtractor.from_pretrained(str(args.qwen_asr_dir))
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tok = AutoTokenizer.from_pretrained(str(args.qwen_asr_dir))
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# Embed table from the HF model (for text tokens; audio_pad slots use audio_embeds)
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import safetensors.torch as st
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import torch
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print("Loading embed_tokens (bf16 → fp32) ...")
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# Find embed_tokens.weight from the original Qwen3-ASR safetensors
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idx = json.load(open(args.qwen_asr_dir / "model.safetensors.index.json"))
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embed_key = "thinker.model.embed_tokens.weight"
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shard = idx["weight_map"][embed_key]
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embed_w = st.load_file(str(args.qwen_asr_dir / shard))[embed_key]
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embed_w = embed_w.to(torch.float32).numpy() # (151936, 2048)
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HIDDEN = embed_w.shape[1]
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print(f" embed_w shape: {embed_w.shape}")
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AUDIO_PAD = 151676
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EOS = 151645
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CTX = args.context_length
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total_wer = 0.0; total_edits = 0; total_words = 0; n = 0
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for name in sorted(REFERENCES):
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wav_path = args.examples_dir / f"{name}.wav"
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if not wav_path.exists():
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print(f" skip {name} (missing)"); continue
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audio, sr = sf.read(str(wav_path))
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if audio.ndim > 1: audio = audio.mean(axis=1)
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if sr != 16000:
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import librosa
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audio = librosa.resample(audio.astype(np.float32), orig_sr=sr, target_sr=16000)
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f = feat(audio, sampling_rate=16000, return_tensors="np", return_attention_mask=False)
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mel = f["input_features"]
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T_mel = mel.shape[-1]
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if T_mel > 3000: mel = mel[..., :3000]; T_mel = 3000
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mel = np.pad(mel, ((0, 0), (0, 0), (0, 3000 - T_mel)), constant_values=0).astype(np.float32)
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audio_embeds = enc.run(["audio_embeds"], {"mel": mel})[0]
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real_chunks = (T_mel + 99) // 100
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last_chunk = T_mel - (real_chunks - 1) * 100
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real_frames = (real_chunks - 1) * 13 + (last_chunk + 7) // 8
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audio_embeds = audio_embeds[0, :real_frames] # (F, 2048)
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# Build prompt tokens + per-position embeddings
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prompt_ids = build_prompt_ids(tok, real_frames)
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L = len(prompt_ids)
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if L > CTX - args.max_new_tokens:
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print(f" skip {name} (L={L} too long for ctx={CTX})"); continue
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# Per-token embeddings: lookup for text, scatter audio_embeds at audio_pad slots
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token_embeds = np.zeros((L, HIDDEN), dtype=np.float32)
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ai = 0
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for i, t in enumerate(prompt_ids):
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if t == AUDIO_PAD:
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token_embeds[i] = audio_embeds[ai]; ai += 1
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else:
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token_embeds[i] = embed_w[t]
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token_embeds = token_embeds.astype(np.float16)
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# Run prefill: feed each prompt token one at a time
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state = mlm.make_state()
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for i in range(L):
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feeds = {
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"inputs_embeds": token_embeds[i:i+1].reshape(1, 1, HIDDEN),
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"position_ids": np.array([i], dtype=np.int32),
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"causal_mask": causal_mask_at(i, CTX),
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"current_pos": np.array([i], dtype=np.int32),
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"update_mask": update_mask_at(i, CTX),
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}
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out = mlm.predict(feeds, state=state)
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# Argmax of last logit
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logits = np.concatenate([out[f"logits{k}"][0, 0] for k in range(1, 17)])
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nid = int(np.argmax(logits))
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gen = [nid]
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# Decode step-by-step
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for step in range(args.max_new_tokens - 1):
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if nid == EOS: break
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cur = L + step
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if cur >= CTX: break
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emb = embed_w[nid].astype(np.float16)
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feeds = {
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"inputs_embeds": emb.reshape(1, 1, HIDDEN),
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"position_ids": np.array([cur], dtype=np.int32),
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"causal_mask": causal_mask_at(cur, CTX),
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"current_pos": np.array([cur], dtype=np.int32),
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"update_mask": update_mask_at(cur, CTX),
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}
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out = mlm.predict(feeds, state=state)
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logits = np.concatenate([out[f"logits{k}"][0, 0] for k in range(1, 17)])
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nid = int(np.argmax(logits))
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gen.append(nid)
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# Strip trailing eos
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if gen and gen[-1] == EOS: gen = gen[:-1]
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hyp_text = tok.decode(gen, skip_special_tokens=True)
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ref = normalize(REFERENCES[name])
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hyp = normalize(hyp_text)
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w, ed, words = wer(ref, hyp)
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agree = max(0.0, 1.0 - w) * 100
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total_wer += w; total_edits += ed; total_words += words; n += 1
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print(f"\n[{color(agree, name.ljust(10))}] WER={w*100:5.1f}% agree={color(agree, f'{agree:5.1f}%')}")
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print(f" REF: {ref}")
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print(f" HYP: {hyp}")
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avg = (1 - total_wer / n) * 100 if n else 0
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print(f"\n{color(avg, f'=== AVERAGE: agreement {avg:.1f}% WER {total_edits/total_words*100:.1f}% ({total_edits}/{total_words}) ===')}")
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return 0
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if __name__ == "__main__":
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sys.exit(main())
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