Spaces:
Running
Running
- analyzer/ASR_en_us.py +56 -132
analyzer/ASR_en_us.py
CHANGED
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@@ -1,89 +1,50 @@
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import torch
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import soundfile as sf
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import librosa
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import os
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import json
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import epitran
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from phonemizer import phonemize
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import numpy as np
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from datetime import datetime, timezone
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import re
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# ---
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# 移除了全域的 processor 和 model
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DEVICE = "cuda" if torch.cuda.is_available() else "cpu"
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print(f"INFO:
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#
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try:
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if os.path.exists(LEXICON_PATH):
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with open(LEXICON_PATH, "r", encoding="utf-8") as f:
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LEXICON = json.load(f)
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else:
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LEXICON = {}
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except Exception:
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LEXICON = {}
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json.dump(LEXICON, f, ensure_ascii=False, indent=2)
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except Exception:
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pass
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# 初始化 Epitran(記憶體 lexicon,不寫 JSON)
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try:
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epi = epitran.Epitran("eng-Latn")
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print("INFO: Epitran initialized for English (eng-Latn)")
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except Exception as e:
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print(f"WARN: Epitran init failed for en_us: {e}")
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epi = None
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def _get_word_ipa(word: str, cache: dict) -> str:
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"""
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不寫檔案,僅記在記憶體 cache 中。
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回傳 IPA 字串(可能包含多字元 token),一個 word -> 一個 IPA 字串保證。
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"""
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# 2) 若 epitran 無效或回傳空字串,使用 phonemizer/espeak 單字呼叫作為備援
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if not ipa:
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try:
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ipa = phonemize(word, language='en-us', backend='espeak', with_stress=True, strip=True)
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ipa = ipa.strip()
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except Exception:
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ipa = ""
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# 3) 最後 fallback:直接使用 word characters(保證回傳非 None)
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if ipa is None or ipa == "":
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ipa = "".join(list(word))
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lex[key] = ipa
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return ipa
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# --- 2. 智能 IPA 切分函數 (
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MULTI_CHAR_PHONEMES = {
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'tʃ', 'dʒ',
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'eɪ', 'aɪ', 'oʊ', 'aʊ', 'ɔɪ',
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'ɪə', 'eə', 'ʊə', 'ər'
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}
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def _tokenize_ipa(ipa_string: str) -> list:
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@@ -102,23 +63,20 @@ def _tokenize_ipa(ipa_string: str) -> list:
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i += 1
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return phonemes
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# --- 3. 核心分析函數 (主入口) (
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def analyze(audio_file_path: str, target_sentence: str, cache: dict = None) -> dict:
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"""
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接收音訊檔案路徑和目標句子,回傳詳細的發音分析字典。
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模型會被載入並儲存在此函數獨立的 'cache' 中,實現狀態隔離。
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"""
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if cache is None:
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cache = {}
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# 檢查快取中是否已有模型,如果沒有則載入
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if "model" not in cache:
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print(f"快取未命中 (
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try:
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# 載入模型並存入此函數的快取字典
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cache["processor"] =
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cache["model"] =
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cache["model"].to(DEVICE)
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print(f"模型 '{MODEL_NAME}' 已載入並快取。")
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except Exception as e:
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@@ -130,15 +88,13 @@ def analyze(audio_file_path: str, target_sentence: str, cache: dict = None) -> d
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model = cache["model"]
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# --- 以下為原始分析邏輯,保持不變 ---
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target_ipa_by_word = [
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target_words_original = words
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try:
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speech, sample_rate = sf.read(audio_file_path)
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@@ -152,26 +108,21 @@ def analyze(audio_file_path: str, target_sentence: str, cache: dict = None) -> d
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with torch.no_grad():
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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#
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for w in asr_words:
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ipa_str = _get_word_ipa(w, cache)
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cleaned = ipa_str.replace('ˌ', '').replace('ˈ', '').replace('ː', '')
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user_ipa_word_tokens.append(_tokenize_ipa(cleaned))
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# 4) 合併成供對齊使用的單一 IPA 字串(不含空格)
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user_ipa_full = "".join("".join(toks) for toks in user_ipa_word_tokens)
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word_alignments = _get_phoneme_alignments_by_word(user_ipa_full, target_ipa_by_word)
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return _format_to_json_structure(word_alignments, target_sentence, target_words_original)
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# --- 4. 對齊函數 (
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def _get_phoneme_alignments_by_word(user_phoneme_str, target_words_ipa_tokenized):
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"""
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(已修改) 使用新的切分邏輯執行音素對齊。
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@@ -226,7 +177,8 @@ def _get_phoneme_alignments_by_word(user_phoneme_str, target_words_ipa_tokenized
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return alignments_by_word
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# --- 5. 格式化函數 (
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def _format_to_json_structure(alignments, sentence, original_words) -> dict:
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total_phonemes = 0
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total_errors = 0
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}
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return final_result
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# 將原本的 _get_target_phonemes_by_word (或相等功能) 改為使用 lexicon 優先 + epitran 備援
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def _get_target_phonemes_by_word(text: str) -> tuple[list[str], list[list[str]]]:
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"""
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針對 English (en_us) 的詞到音素處理:字典優先、Epitran 備援、快取至 lexicon_en_us.json。
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回傳 (原始詞列表, 每個詞的音素列表)
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"""
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if not text or not text.strip():
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return [], []
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# 簡單以空白分詞;若輸入無空白則逐字
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words = text.split() if ' ' in text.strip() else list(text.strip())
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target_words_original = []
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target_ipa_by_word = []
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for w in words:
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w_stripped = w.strip()
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if not w_stripped:
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continue
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try:
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phonemes = _get_phonemes_for_word_en(w_stripped)
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except Exception:
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phonemes = list(w_stripped)
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target_words_original.append(w_stripped)
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target_ipa_by_word.append(phonemes)
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return target_words_original, target_ipa_by_word
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import torch
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import soundfile as sf
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import librosa
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# 【【【【【 修改 #1:從 transformers 匯入 AutoProcessor 和 AutoModelForCTC 】】】】】
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from transformers import AutoProcessor, AutoModelForCTC
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import os
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from phonemizer import phonemize
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import numpy as np
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from datetime import datetime, timezone
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# --- 全域設定 (已修改) ---
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# 移除了全域的 processor 和 model 變數。
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# 刪除了舊的 load_model() 函數。
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DEVICE = "cuda" if torch.cuda.is_available() else "cpu"
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print(f"INFO: ASR_en_us_v2.py is configured to use device: {DEVICE}")
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# 【【【【【 修改 #2:更新為最終選定的 KoelLabs 模型名稱 】】】】】
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MODEL_NAME = "KoelLabs/xlsr-english-01"
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# 【【【【【 新增程式碼 #1:為 KoelLabs 模型設計的 IPA 正規化器 】】】】】
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# 【保持不變】
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def normalize_koel_ipa(raw_phonemes: list) -> list:
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"""
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將 KoelLabs 模型輸出的高級 IPA 序列,正規化為與 eSpeak 輸出可比的基礎 IPA 序列。
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"""
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normalized_phonemes = []
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for phoneme in raw_phonemes:
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if not phoneme:
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continue
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base_phoneme = phoneme.replace('ʰ', '').replace('̃', '').replace('̥', '')
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if base_phoneme == 'β':
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base_phoneme = 'v'
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elif base_phoneme in ['x', 'ɣ', 'ɦ']:
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base_phoneme = 'h'
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normalized_phonemes.append(base_phoneme)
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return normalized_phonemes
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# --- 2. 智能 IPA 切分函數 (與您的原版邏輯完全相同) ---
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# 【保持不變】
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MULTI_CHAR_PHONEMES = {
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'tʃ', 'dʒ',
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'eɪ', 'aɪ', 'oʊ', 'aʊ', 'ɔɪ',
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'ɪə', 'eə', 'ʊə', 'ər'
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}
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def _tokenize_ipa(ipa_string: str) -> list:
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i += 1
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return phonemes
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# --- 3. 核心分析函數 (主入口) (已修改以整合正規化器和快取邏輯) ---
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def analyze(audio_file_path: str, target_sentence: str, cache: dict = {}) -> dict:
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"""
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接收音訊檔案路徑和目標句子,回傳詳細的發音分析字典。
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模型會被載入並儲存在此函數獨立的 'cache' 中,實現狀態隔離。
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"""
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# 檢查快取中是否已有模型,如果沒有則載入
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if "model" not in cache:
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print(f"快取未命中 (ASR_en_us_v2)。正在載入模型 '{MODEL_NAME}'...")
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try:
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# 【【【【【 修改 #3:使用 AutoProcessor 和 AutoModelForCTC 載入模型 】】】】】
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# 載入模型並存入此函數的快取字典
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cache["processor"] = AutoProcessor.from_pretrained(MODEL_NAME)
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cache["model"] = AutoModelForCTC.from_pretrained(MODEL_NAME)
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cache["model"].to(DEVICE)
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print(f"模型 '{MODEL_NAME}' 已載入並快取。")
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except Exception as e:
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model = cache["model"]
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# --- 以下為原始分析邏輯,保持不變 ---
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target_ipa_by_word_str = phonemize(target_sentence, language='en-us', backend='espeak', with_stress=True, strip=True).split()
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target_ipa_by_word = [
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_tokenize_ipa(word.replace('ˌ', '').replace('ˈ', '').replace('ː', ''))
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for word in target_ipa_by_word_str
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]
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target_words_original = target_sentence.split()
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try:
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speech, sample_rate = sf.read(audio_file_path)
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with torch.no_grad():
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logits = model(input_values).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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# 【【【【【 修改 #4:在此處插入正規化步驟 】】】】】
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# 【保持不變】
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raw_user_ipa_str = processor.decode(predicted_ids[0])
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raw_user_phonemes = raw_user_ipa_str.split(' ')
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normalized_user_phonemes = normalize_koel_ipa(raw_user_phonemes)
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user_ipa_full = "".join(normalized_user_phonemes)
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word_alignments = _get_phoneme_alignments_by_word(user_ipa_full, target_ipa_by_word)
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return _format_to_json_structure(word_alignments, target_sentence, target_words_original)
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# --- 4. 對齊函數 (與您的原版邏輯完全相同) ---
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# 【保持不變】
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def _get_phoneme_alignments_by_word(user_phoneme_str, target_words_ipa_tokenized):
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"""
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(已修改) 使用新的切分邏輯執行音素對齊。
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return alignments_by_word
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# --- 5. 格式化函數 (與您的原版邏輯完全相同) ---
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# 【保持不變】
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def _format_to_json_structure(alignments, sentence, original_words) -> dict:
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total_phonemes = 0
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total_errors = 0
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}
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return final_result
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