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1from huggingsound import SpeechRecognitionModel
2
3model = SpeechRecognitionModel("jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn")
4audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
5
6transcriptions = model.transcribe(audio_paths)1import torch
2import librosa
3from datasets import load_dataset
4from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
5
6LANG_ID = "zh-CN"
7MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn"
8SAMPLES = 10
9
10test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
11
12processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
13model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
14
15# Preprocessing the datasets.
16# We need to read the audio files as arrays
17def speech_file_to_array_fn(batch):
18 speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
19 batch["speech"] = speech_array
20 batch["sentence"] = batch["sentence"].upper()
21 return batch
22
23test_dataset = test_dataset.map(speech_file_to_array_fn)
24inputs = processor(test_dataset["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
25
26with torch.no_grad():
27 logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
28
29predicted_ids = torch.argmax(logits, dim=-1)
30predicted_sentences = processor.batch_decode(predicted_ids)
31
32for i, predicted_sentence in enumerate(predicted_sentences):
33 print("-" * 100)
34 print("Reference:", test_dataset[i]["sentence"])
35 print("Prediction:", predicted_sentence)| Reference | Prediction |
|---|---|
| 宋朝末年年间定居粉岭围。 | 宋朝末年年间定居分定为 |
| 渐渐行动不便 | 建境行动不片 |
| 二十一年去世。 | 二十一年去世 |
| 他们自称恰哈拉。 | 他们自称家哈 |
| 局部干涩的例子包括有口干、眼睛干燥、及阴道干燥。 | 菊物干寺的例子包括有口肝眼睛干照以及阴到干 |
| 嘉靖三十八年,登进士第三甲第二名。 | 嘉靖三十八年登进士第三甲第二名 |
| 这一名称一直沿用至今。 | 这一名称一直沿用是心 |
| 同时乔凡尼还得到包税合同和许多明矾矿的经营权。 | 同时桥凡妮还得到包税合同和许多民繁矿的经营权 |
| 为了惩罚西扎城和塞尔柱的结盟,盟军在抵达后将外城烧毁。 | 为了曾罚西扎城和塞尔素的节盟盟军在抵达后将外曾烧毁 |
| 河内盛产黄色无鱼鳞的鳍射鱼。 | 合类生场环色无鱼林的骑射鱼 |
1import torch
2import re
3import librosa
4from datasets import load_dataset, load_metric
5from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
6
7LANG_ID = "zh-CN"
8MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn"
9DEVICE = "cuda"
10
11CHARS_TO_IGNORE = [",", "?", "¿", ".", "!", "¡", ";", ";", ":", '""', "%", '"', "�", "ʿ", "·", "჻", "~", "՞",
12 "؟", "،", "।", "॥", "«", "»", "„", "“", "”", "「", "」", "‘", "’", "《", "》", "(", ")", "[", "]",
13 "{", "}", "=", "`", "_", "+", "<", ">", "…", "–", "°", "´", "ʾ", "‹", "›", "©", "®", "—", "→", "。",
14 "、", "﹂", "﹁", "‧", "~", "﹏", ",", "{", "}", "(", ")", "[", "]", "【", "】", "‥", "〽",
15 "『", "』", "〝", "〟", "⟨", "⟩", "〜", ":", "!", "?", "♪", "؛", "/", "\\", "º", "−", "^", "'", "ʻ", "ˆ"]
16
17test_dataset = load_dataset("common_voice", LANG_ID, split="test")
18
19wer = load_metric("wer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/wer.py
20cer = load_metric("cer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/cer.py
21
22chars_to_ignore_regex = f"[{re.escape(''.join(CHARS_TO_IGNORE))}]"
23
24processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
25model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
26model.to(DEVICE)
27
28# Preprocessing the datasets.
29# We need to read the audio files as arrays
30def speech_file_to_array_fn(batch):
31 with warnings.catch_warnings():
32 warnings.simplefilter("ignore")
33 speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
34 batch["speech"] = speech_array
35 batch["sentence"] = re.sub(chars_to_ignore_regex, "", batch["sentence"]).upper()
36 return batch
37
38test_dataset = test_dataset.map(speech_file_to_array_fn)
39
40# Preprocessing the datasets.
41# We need to read the audio files as arrays
42def evaluate(batch):
43 inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
44
45 with torch.no_grad():
46 logits = model(inputs.input_values.to(DEVICE), attention_mask=inputs.attention_mask.to(DEVICE)).logits
47
48 pred_ids = torch.argmax(logits, dim=-1)
49 batch["pred_strings"] = processor.batch_decode(pred_ids)
50 return batch
51
52result = test_dataset.map(evaluate, batched=True, batch_size=8)
53
54predictions = [x.upper() for x in result["pred_strings"]]
55references = [x.upper() for x in result["sentence"]]
56
57print(f"WER: {wer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}")
58print(f"CER: {cer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}")| Model | WER | CER |
|---|---|---|
| jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn | 82.37% | 19.03% |
| ydshieh/wav2vec2-large-xlsr-53-chinese-zh-cn-gpt | 84.01% | 20.95% |
1@misc{grosman2021xlsr53-large-chinese,
2 title={Fine-tuned {XLSR}-53 large model for speech recognition in {C}hinese},
3 author={Grosman, Jonatas},
4 howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn}},
5 year={2021}
6}