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# =======================================================================
# analyzer/ASR_pt_br.py
# 巴西葡萄牙語發音分析器
# 版本:v2.0 (與 en_us.py 邏輯對齊)
# 描述:此版本完全遵循 en_us.py 的程式碼結構和算法實現,
# 僅在語言特定配置(模型名稱、G2P語言)上有所不同,
# 並採用了更健壯的、基於 Unicode 的 IPA 切分方法以適應葡萄牙語。
# =======================================================================
# --- 1. 匯入區 (與 en_us.py 保持一致) ---
import torch
import soundfile as sf
import librosa
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
import os
from phonemizer import phonemize
import numpy as np
from datetime import datetime, timezone
import unicodedata # 【保留】這是處理葡萄牙語鼻音等音素的更優方案
import re # 【保留】用於更準確地切分單詞
# --- 2. 全域設定與模型載入 ---
# 【已修改】移除了全域的 processor 和 model 變數。
# 【已修改】刪除了舊的 load_model() 函數。
DEVICE = "cuda" if torch.cuda.is_available() else "cpu"
print(f"INFO: ASR_pt_br.py is configured to use device: {DEVICE}")
# 【關鍵修改 1:設定為葡萄牙語 ASR 模型】
MODEL_NAME = "caiocrocha/wav2vec2-large-xlsr-53-phoneme-portuguese"
# --- 3. 智能 IPA 切分函數 ---
# 【關鍵修改 2:保留更優越的通用切分邏輯】
# 【保持不變】
def _tokenize_ipa(ipa_string: str) -> list:
"""
將 IPA 字串智能地切分為音素列表,能正確處理帶有附加符號的組合字符。
"""
phonemes = []
s = ipa_string.replace(' ', '')
i = 0
while i < len(s):
if i + 1 < len(s) and s[i:i+2] in {'dʒ', 'tʃ'}:
phonemes.append(s[i:i+2])
i += 2
continue
current_char = s[i]
i += 1
while i < len(s) and unicodedata.category(s[i]) == 'Mn':
current_char += s[i]
i += 1
phonemes.append(current_char)
return phonemes
# --- 4. 核心分析函數 (主入口) ---
# 【已修改】將模型載入和快取邏輯合併至此。
def analyze(audio_file_path: str, target_sentence: str, cache: dict = {}) -> dict:
"""
接收音訊檔案路徑和目標葡萄牙語句子,回傳詳細的發音分析字典。
模型會被載入並儲存在此函數獨立的 'cache' 中,實現狀態隔離。
"""
# 檢查快取中是否已有模型,如果沒有則載入
if "model" not in cache:
print(f"快取未命中 (ASR_pt_br)。正在載入模型 '{MODEL_NAME}'...")
try:
# 載入模型並存入此函數的快取字典
cache["processor"] = Wav2Vec2Processor.from_pretrained(MODEL_NAME)
cache["model"] = Wav2Vec2ForCTC.from_pretrained(MODEL_NAME)
cache["model"].to(DEVICE)
print(f"模型 '{MODEL_NAME}' 已載入並快取。")
except Exception as e:
print(f"處理或載入模型 '{MODEL_NAME}' 時發生錯誤: {e}")
raise RuntimeError(f"Failed to load model '{MODEL_NAME}': {e}")
# 從此函數的獨立快取中獲取模型和處理器
processor = cache["processor"]
model = cache["model"]
# --- 以下為原始分析邏輯,保持不變 ---
# 1. 準備目標音素 (G2P)
target_words_original = re.findall(r"[\w'-]+", target_sentence)
cleaned_sentence = " ".join(target_words_original)
# 【關鍵修改 3:設定 G2P 語言為 'pt-br'】
target_ipa_by_word_str = phonemize(
cleaned_sentence,
language='pt-br',
backend='espeak',
with_stress=True,
strip=True
).split()
if len(target_words_original) != len(target_ipa_by_word_str):
print(f"警告: G2P 後單詞數量 ({len(target_ipa_by_word_str)}) 與原始單詞數量 ({len(target_words_original)}) 不匹配。將進行截斷。")
min_len = min(len(target_words_original), len(target_ipa_by_word_str))
target_words_original = target_words_original[:min_len]
target_ipa_by_word_str = target_ipa_by_word_str[:min_len]
# 【關鍵修改 4:與 en_us.py 對齊,在準備目標音素時就清除所有不比較的符號】
target_ipa_by_word = [
_tokenize_ipa(word.replace('ˈ', '').replace('ˌ', '').replace('ː', ''))
for word in target_ipa_by_word_str
]
# 2. 處理音訊並進行語音辨識 (ASR)
try:
speech, sample_rate = sf.read(audio_file_path)
if sample_rate != 16000:
speech = librosa.resample(y=speech, orig_sr=sample_rate, target_sr=16000)
except Exception as e:
raise IOError(f"讀取或處理音訊時發生錯誤: {e}")
input_values = processor(speech, sampling_rate=16000, return_tensors="pt").input_values
input_values = input_values.to(DEVICE)
with torch.no_grad():
logits = model(input_values).logits
predicted_ids = torch.argmax(logits, dim=-1)
# 【關鍵修改 5:與 en_us.py 對齊,清理模型輸出以匹配目標音素的處理方式】
user_ipa_full = processor.decode(predicted_ids[0]).replace('|', '').replace('ː', '')
# 3. 執行對齊並格式化輸出
word_alignments = _get_phoneme_alignments_by_word(user_ipa_full, target_ipa_by_word)
return _format_to_json_structure(word_alignments, target_sentence, target_words_original)
# --- 5. 對齊函數 (與 en_us.py 的實現邏輯完全對齊) ---
# 【保持不變】
def _get_phoneme_alignments_by_word(user_phoneme_str, target_words_ipa_tokenized):
"""
使用動態規劃執行音素對齊。
(此函數實現與 en_us.py 完全相同)
"""
user_phonemes = _tokenize_ipa(user_phoneme_str)
target_phonemes_flat = []
word_boundaries_indices = []
current_idx = 0
for word_ipa_tokens in target_words_ipa_tokenized:
target_phonemes_flat.extend(word_ipa_tokens)
current_idx += len(word_ipa_tokens)
word_boundaries_indices.append(current_idx - 1)
dp = np.zeros((len(user_phonemes) + 1, len(target_phonemes_flat) + 1))
for i in range(1, len(user_phonemes) + 1): dp[i][0] = i
for j in range(1, len(target_phonemes_flat) + 1): dp[0][j] = j
for i in range(1, len(user_phonemes) + 1):
for j in range(1, len(target_phonemes_flat) + 1):
cost = 0 if user_phonemes[i-1] == target_phonemes_flat[j-1] else 1
dp[i][j] = min(dp[i-1][j] + 1, dp[i][j-1] + 1, dp[i-1][j-1] + cost)
i, j = len(user_phonemes), len(target_phonemes_flat)
user_path, target_path = [], []
while i > 0 or j > 0:
cost = float('inf') if i == 0 or j == 0 else (0 if user_phonemes[i-1] == target_phonemes_flat[j-1] else 1)
if i > 0 and j > 0 and dp[i][j] == dp[i-1][j-1] + cost:
user_path.insert(0, user_phonemes[i-1]); target_path.insert(0, target_phonemes_flat[j-1]); i -= 1; j -= 1
elif i > 0 and dp[i][j] == dp[i-1][j] + 1:
user_path.insert(0, user_phonemes[i-1]); target_path.insert(0, '-'); i -= 1
else:
user_path.insert(0, '-'); target_path.insert(0, target_phonemes_flat[j-1]); j -= 1
alignments_by_word = []
word_start_idx_in_path = 0
target_phoneme_counter_in_path = 0
for path_idx, p in enumerate(target_path):
if p != '-':
if target_phoneme_counter_in_path in word_boundaries_indices:
alignments_by_word.append({
"target": target_path[word_start_idx_in_path : path_idx + 1],
"user": user_path[word_start_idx_in_path : path_idx + 1]
})
word_start_idx_in_path = path_idx + 1
target_phoneme_counter_in_path += 1
return alignments_by_word
# --- 6. 格式化函數 (與 en_us.py 的實現邏輯完全對齊) ---
# 【保持不變】
def _format_to_json_structure(alignments, sentence, original_words) -> dict:
"""
將對齊結果格式化為最終的 JSON 結構。
(此函數實現與 en_us.py 完全相同,僅 G2P 語言設定不同)
"""
total_phonemes, total_errors, correct_words_count = 0, 0, 0
words_data = []
num_words_to_process = min(len(alignments), len(original_words))
for i in range(num_words_to_process):
alignment = alignments[i]
word_is_correct = True
phonemes_data = []
for j in range(len(alignment['target'])):
target_phoneme = alignment['target'][j]
user_phoneme = alignment['user'][j]
is_match = (user_phoneme == target_phoneme)
phonemes_data.append({"target": target_phoneme, "user": user_phoneme, "isMatch": is_match})
if not is_match:
word_is_correct = False
if not (user_phoneme == '-' and target_phoneme == '-'): total_errors += 1
if word_is_correct:
correct_words_count += 1
words_data.append({"word": original_words[i], "isCorrect": word_is_correct, "phonemes": phonemes_data})
total_phonemes += sum(1 for p in alignment['target'] if p != '-')
if len(alignments) < len(original_words):
for i in range(len(alignments), len(original_words)):
# 【關鍵修改 6:確保此處的 G2P 語言和符號清理也保持一致】
missed_word_ipa_str = phonemize(original_words[i], language='pt-br', backend='espeak', strip=True).replace('ː', '')
missed_word_ipa = _tokenize_ipa(missed_word_ipa_str)
phonemes_data = []
for p_ipa in missed_word_ipa:
phonemes_data.append({"target": p_ipa, "user": "-", "isMatch": False})
total_errors += 1
total_phonemes += 1
words_data.append({"word": original_words[i], "isCorrect": False, "phonemes": phonemes_data})
total_words = len(original_words)
overall_score = (correct_words_count / total_words) * 100 if total_words > 0 else 0
phoneme_error_rate = (total_errors / total_phonemes) * 100 if total_phonemes > 0 else 0
return {
"sentence": sentence,
"analysisTimestampUTC": datetime.now(timezone.utc).strftime('%Y-%m-%d %H:%M:%S (UTC)'),
"summary": {
"overallScore": round(overall_score, 1),
"totalWords": total_words,
"correctWords": correct_words_count,
"phonemeErrorRate": round(phoneme_error_rate, 2),
"total_errors": total_errors,
"total_target_phonemes": total_phonemes
},
"words": words_data
}
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